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

[alsa-devel] BUG: soft lockup (2)
by syzbot 10 Jan '18

10 Jan '18

syzkaller has found reproducer for the following crash on e1915c8195b38393005be9b74bfa6a3a367c83b3 git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. C reproducer is attached syzkaller reproducer is attached. See https://goo.gl/kgGztJ for information about syzkaller reproducers IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f76f3c62dfadce022fd1c1deff15a61e09ac708a(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. watchdog: BUG: soft lockup - CPU#0 stuck for 135s! [syzkaller670324:3527] Modules linked in: irq event stamp: 2531226 hardirqs last enabled at (2531225): [<00000000f1ec093f>] snd_pcm_stream_unlock_irq+0x78/0xe0 sound/core/pcm_native.c:166 hardirqs last disabled at (2531226): [<000000003c6ef1cd>] apic_timer_interrupt+0xa4/0xb0 arch/x86/entry/entry_64.S:920 softirqs last enabled at (41848): [<0000000081bd5f03>] __do_softirq+0x7a0/0xb85 kernel/softirq.c:311 softirqs last disabled at (41829): [<00000000d02c6d52>] invoke_softirq kernel/softirq.c:365 [inline] softirqs last disabled at (41829): [<00000000d02c6d52>] irq_exit+0x1cc/0x200 kernel/softirq.c:405 CPU: 0 PID: 3527 Comm: syzkaller670324 Not tainted 4.15.0-rc6+ #158 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:memcpy+0x45/0x50 mm/kasan/kasan.c:305 RSP: 0018:ffff8801bf6676f0 EFLAGS: 00000246 ORIG_RAX: ffffffffffffff11 RAX: ffffc9000137ba06 RBX: 0000000000000002 RCX: 0000000000000000 RDX: 0000000000000002 RSI: ffff8801bf6677da RDI: ffffc9000137ba08 RBP: ffff8801bf667708 R08: fffff5200026f741 R09: fffff5200026f741 R10: 0000000000000001 R11: fffff5200026f740 R12: ffffc9000137ba06 R13: ffff8801bf6677d8 R14: dffffc0000000000 R15: ffffc9000137ba06 FS: 0000000000000000(0000) GS:ffff8801db200000(0063) knlGS:00000000f7ec6b40 CS: 0010 DS: 002b ES: 002b CR0: 0000000080050033 CR2: 0000000020735ee0 CR3: 00000001bfba8002 CR4: 00000000001606f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: memcpy include/linux/string.h:344 [inline] cvt_s16_to_native sound/core/oss/mulaw.c:164 [inline] mulaw_decode+0x52f/0x770 sound/core/oss/mulaw.c:195 mulaw_transfer+0x222/0x270 sound/core/oss/mulaw.c:273 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1372 [inline] snd_pcm_oss_write+0x5fe/0x830 sound/core/oss/pcm_oss.c:2646 __vfs_write+0xef/0x970 fs/read_write.c:480 vfs_write+0x189/0x510 fs/read_write.c:544 SYSC_write fs/read_write.c:589 [inline] SyS_write+0xef/0x220 fs/read_write.c:581 do_syscall_32_irqs_on arch/x86/entry/common.c:327 [inline] do_fast_syscall_32+0x3ee/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7f0cc79 RSP: 002b:00000000f7ec61fc EFLAGS: 00000246 ORIG_RAX: 0000000000000004 RAX: ffffffffffffffda RBX: 0000000000000005 RCX: 0000000020735ee0 RDX: 00000000fffffee4 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 00000000003d0f00 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Code: 89 de 31 d2 e8 9d f9 ff ff 48 8b 4d 08 48 89 de 4c 89 e7 ba 01 00 00 00 e8 89 f9 ff ff 48 89 da 4c 89 ee 4c 89 e7 e8 7b eb c0 03 <5b> 41 5c 41 5d 5d c3 0f 1f 40 00 89 f1 b8 00 10 00 00 55 48 d3

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[alsa-devel] snd_hda_intel: coil whine noises after resume
by Francisco G. 10 Jan '18

10 Jan '18

Hello. Some users included myself have noticed coil-whine-like noises coming through the headphones after the computer resumes from sleep. This noise is clearly associated to the laptop activity, e.g. when scrolling through a webpage or compiling something, the processor can be heard doing the work. Interestingly, the coil whine that can be heard from the computer with naked ears (which is a common problem with this laptop) is also increased after resume. The problem has been observed on a Dell XPS 13 9360 with an i7-8550U CPU by myself, as well as with an i5-8250U by a user in the Arch Linux forums <https://bbs.archlinux.org/viewtopic.php?pid=1759588#p1759588>. This user has also noticed that reloading the snd_hda_intel module fixes the issue. Please let me know of any inormation that would be needed to debug this problem.

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[alsa-devel] [PATCH v5][RESEND] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 10 Jan '18

10 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- 2week passed. resend sound/soc/codecs/ak4613.c | 78 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..d85c61e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,83 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * + * Note + * + * To avoid extra delay, we want to avoid preemption here, + * but we can't. Because it uses I2C access which is using IRQ + * and sleep. Thus, delay might be more than 5 LR clocks + * see also + * ak4613_dai_trigger() + */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for writing). + * Thus, use schedule_work() to switching to normal context + * immediately. + * + * Note + * + * Calling ak4613_dummy_write() function might be delayed. + * In such case, ak4613 volume might be temporarily 0dB when + * beggining of playback. + * see also + * ak4613_dummy_write() + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +668,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: codecs: dmic: Make number of channels configurable
by Matthias Kaehlcke 09 Jan '18

09 Jan '18

The DMIC DAI driver specifies a number of 1 to 8 channels for each DAI. The actual number of mics can currently not be configured in the device tree or audio glue, but is derived from the min/max channels of the CPU and codec DAI. A typical CPU DAI has two or more channels, in consequence a single mic is treated as a stereo/multi channel device, even though only one channel carries audio data. This change adds the option to specify the number of used DMIC channels in the device tree. When specified this value overwrites the default channels_max value of 8 in the snd_soc_dai_driver struct of the codec. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Documentation/devicetree/bindings/sound/dmic.txt | 2 ++ sound/soc/codecs/dmic.c | 24 +++++++++++++++++++++++- 2 files changed, 25 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/dmic.txt b/Documentation/devicetree/bindings/sound/dmic.txt index 54c8ef6498a8..f7bf65611453 100644 --- a/Documentation/devicetree/bindings/sound/dmic.txt +++ b/Documentation/devicetree/bindings/sound/dmic.txt @@ -7,10 +7,12 @@ Required properties: Optional properties: - dmicen-gpios: GPIO specifier for dmic to control start and stop + - num-channels: Number of microphones on this DAI Example node: dmic_codec: dmic@0 { compatible = "dmic-codec"; dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; + num-channels = <1>; }; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index b88a1ee66f80..c88f974ebe3e 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -107,8 +107,30 @@ static const struct snd_soc_codec_driver soc_dmic = { static int dmic_dev_probe(struct platform_device *pdev) { + int err; + u32 chans; + struct snd_soc_dai_driver *dai_drv = &dmic_dai; + + if (pdev->dev.of_node) { + err = of_property_read_u32(pdev->dev.of_node, "num-channels", &chans); + if (err && (err != -ENOENT)) + return err; + + if (!err) { + if (chans < 1 || chans > 8) + return -EINVAL; + + dai_drv = devm_kzalloc(&pdev->dev, sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + memcpy(dai_drv, &dmic_dai, sizeof(*dai_drv)); + dai_drv->capture.channels_max = chans; + } + } + return snd_soc_register_codec(&pdev->dev, - &soc_dmic, &dmic_dai, 1); + &soc_dmic, dai_drv, 1); } static int dmic_dev_remove(struct platform_device *pdev) -- 2.16.0.rc0.223.g4a4ac83678-goog

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[alsa-devel] [PATCH v1] ASoC: TSCS42xx: Fix control names
by Steven Eckhoff 09 Jan '18

09 Jan '18

The tscs42xx CODEC driver can confuse userspace with non-standard control names. Remove "Switch" from enum control type names. Add "Switch" to on/off control type names. Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 42 +++++++++++++++++++++--------------------- 1 file changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index eedd600875e5..4a5b32a717f2 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -631,7 +631,7 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { 0, mic_boost_scale), /* Input Channel Map */ - SOC_ENUM("Input Channel Map Switch", ch_map_select_enum), + SOC_ENUM("Input Channel Map", ch_map_select_enum), /* Coefficient Ram */ COEFF_RAM_CTL("Cascade1L BiQuad1", BIQUAD_SIZE, 0x00), @@ -708,13 +708,13 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { /* EQ */ SOC_SINGLE("EQ1 Switch", R_CONFIG1, FB_CONFIG1_EQ1_EN, 1, 0), SOC_SINGLE("EQ2 Switch", R_CONFIG1, FB_CONFIG1_EQ2_EN, 1, 0), - SOC_ENUM("EQ1 Band Enable Switch", eq1_band_enable_enum), - SOC_ENUM("EQ2 Band Enable Switch", eq2_band_enable_enum), + SOC_ENUM("EQ1 Band Enable", eq1_band_enable_enum), + SOC_ENUM("EQ2 Band Enable", eq2_band_enable_enum), /* CLE */ - SOC_ENUM("CLE Level Detect Switch", + SOC_ENUM("CLE Level Detect", cle_level_detection_enum), - SOC_ENUM("CLE Level Detect Win Switch", + SOC_ENUM("CLE Level Detect Win", cle_level_detection_window_enum), SOC_SINGLE("Expander Switch", R_CLECTL, FB_CLECTL_EXP_EN, 1, 0), @@ -726,7 +726,7 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { R_MUGAIN, FB_MUGAIN_CLEMUG, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("Comp Thresh Playback Volume", R_COMPTH, FB_COMPTH, 0xff, 0, compth_scale), - SOC_ENUM("Comp Ratio Switch", compressor_ratio_enum), + SOC_ENUM("Comp Ratio", compressor_ratio_enum), SND_SOC_BYTES("Comp Atk Time", R_CATKTCL, 2), /* Effects */ @@ -740,50 +740,50 @@ static const struct snd_kcontrol_new tscs42xx_snd_controls[] = { SOC_SINGLE("MBC Band1 Switch", R_DACMBCEN, FB_DACMBCEN_MBCEN1, 1, 0), SOC_SINGLE("MBC Band2 Switch", R_DACMBCEN, FB_DACMBCEN_MBCEN2, 1, 0), SOC_SINGLE("MBC Band3 Switch", R_DACMBCEN, FB_DACMBCEN_MBCEN3, 1, 0), - SOC_ENUM("MBC Band1 Level Detect Switch", + SOC_ENUM("MBC Band1 Level Detect", mbc_level_detection_enums[0]), - SOC_ENUM("MBC Band2 Level Detect Switch", + SOC_ENUM("MBC Band2 Level Detect", mbc_level_detection_enums[1]), - SOC_ENUM("MBC Band3 Level Detect Switch", + SOC_ENUM("MBC Band3 Level Detect", mbc_level_detection_enums[2]), - SOC_ENUM("MBC Band1 Level Detect Win Switch", + SOC_ENUM("MBC Band1 Level Detect Win", mbc_level_detection_window_enums[0]), - SOC_ENUM("MBC Band2 Level Detect Win Switch", + SOC_ENUM("MBC Band2 Level Detect Win", mbc_level_detection_window_enums[1]), - SOC_ENUM("MBC Band3 Level Detect Win Switch", + SOC_ENUM("MBC Band3 Level Detect Win", mbc_level_detection_window_enums[2]), - SOC_SINGLE("MBC1 Phase Invert", R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, - 1, 0), + SOC_SINGLE("MBC1 Phase Invert Switch", + R_DACMBCMUG1, FB_DACMBCMUG1_PHASE, 1, 0), SOC_SINGLE_TLV("DAC MBC1 Make-Up Gain Playback Volume", R_DACMBCMUG1, FB_DACMBCMUG1_MUGAIN, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("DAC MBC1 Comp Thresh Playback Volume", R_DACMBCTHR1, FB_DACMBCTHR1_THRESH, 0xff, 0, compth_scale), - SOC_ENUM("DAC MBC1 Comp Ratio Switch", + SOC_ENUM("DAC MBC1 Comp Ratio", dac_mbc1_compressor_ratio_enum), SND_SOC_BYTES("DAC MBC1 Comp Atk Time", R_DACMBCATK1L, 2), SND_SOC_BYTES("DAC MBC1 Comp Rel Time Const", R_DACMBCREL1L, 2), - SOC_SINGLE("MBC2 Phase Invert", R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, - 1, 0), + SOC_SINGLE("MBC2 Phase Invert Switch", + R_DACMBCMUG2, FB_DACMBCMUG2_PHASE, 1, 0), SOC_SINGLE_TLV("DAC MBC2 Make-Up Gain Playback Volume", R_DACMBCMUG2, FB_DACMBCMUG2_MUGAIN, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("DAC MBC2 Comp Thresh Playback Volume", R_DACMBCTHR2, FB_DACMBCTHR2_THRESH, 0xff, 0, compth_scale), - SOC_ENUM("DAC MBC2 Comp Ratio Switch", + SOC_ENUM("DAC MBC2 Comp Ratio", dac_mbc2_compressor_ratio_enum), SND_SOC_BYTES("DAC MBC2 Comp Atk Time", R_DACMBCATK2L, 2), SND_SOC_BYTES("DAC MBC2 Comp Rel Time Const", R_DACMBCREL2L, 2), - SOC_SINGLE("MBC3 Phase Invert", R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, - 1, 0), + SOC_SINGLE("MBC3 Phase Invert Switch", + R_DACMBCMUG3, FB_DACMBCMUG3_PHASE, 1, 0), SOC_SINGLE_TLV("DAC MBC3 Make-Up Gain Playback Volume", R_DACMBCMUG3, FB_DACMBCMUG3_MUGAIN, 0x1f, 0, mugain_scale), SOC_SINGLE_TLV("DAC MBC3 Comp Thresh Playback Volume", R_DACMBCTHR3, FB_DACMBCTHR3_THRESH, 0xff, 0, compth_scale), - SOC_ENUM("DAC MBC3 Comp Ratio Switch", + SOC_ENUM("DAC MBC3 Comp Ratio", dac_mbc3_compressor_ratio_enum), SND_SOC_BYTES("DAC MBC3 Comp Atk Time", R_DACMBCATK3L, 2), SND_SOC_BYTES("DAC MBC3 Comp Rel Time Const", -- 2.11.0

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[alsa-devel] [PATCH][next] ASoC: TSCS42xx: make functions pll_event and dac_event static
by Colin King 09 Jan '18

09 Jan '18

From: Colin Ian King <colin.king(a)canonical.com> The functions pll_event and dac_event are local to the source and do not need to be in global scope, so make them static. Cleans up sparse warnings: symbol 'pll_event' was not declared. Should it be static? symbol 'dac_event' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tscs42xx.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index eedd600875e5..9e8155766ea9 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -355,8 +355,8 @@ static int dapm_micb_event(struct snd_soc_dapm_widget *w, return 0; } -int pll_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); int ret; @@ -369,8 +369,8 @@ int pll_event(struct snd_soc_dapm_widget *w, return ret; } -int dac_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct tscs42xx *tscs42xx = snd_soc_codec_get_drvdata(codec); -- 2.15.1

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[alsa-devel] 4.16 merge window: please send earlier
by Takashi Iwai 09 Jan '18

09 Jan '18

Hi, currently 4.16 merge window is planned to be opened in two weeks, around Jan 21. At this time, there will be a conflict with my private schedule: I'll be traveling from Jan 20 for a couple of weeks. I'll be with a laptop and the net should work, so I likely can still keep merging but it'll be quite much reduced. So, at this time, I'd like to set the merge deadline earlier. If anyone has some material to be merged for 4.16, please submit ASAP, preferably in this week, or in the beginning of the next week. Luckily Mark already sent a pull request for the major changes, so the big changes have been already there. If any, Mark, send the further fixes earlier than usual, please. thanks, Takashi

1 0

[alsa-devel] Set multiple mixer channel values at once
by Petr Kulhavy 09 Jan '18

09 Jan '18

Hi, does the ALSA library provide a method to set multiple channel values of a multi-channel mixer element at once? There are the snd_mixer_selem_set_XXXXX_all() methods, which set all channels to the same value. I would like to write a different value to each channel, is there such a method? My motivation is the following. I have a HW with DSP/equaliser function, which uses a 5-channel mixer element (volume type element) to store biquad coefficients. To avoid invalid values in the transfer function the coefficients have to be sent to the HW over I2C all together in one transfer. Though the low level driver does that, the snd_mixer_selem_set_playback_volume() API method I'm using sets only one coef at a time. This results in partial updates of the biquad coefficients and temporary errors in the transfer function. Thanks Petr

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[alsa-devel] [PATCH] ASoC: rockchip: i2s: Support mono capture
by Matthias Kaehlcke 09 Jan '18

09 Jan '18

The Rockchip I2S controller only allows to configure even numbers of capture channels. It is still possible to capture monophonic audio by using dual-channel mode and ignoring the 'data' from the second channel. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/rockchip/rockchip_i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 908211e1d6fc..cc22ab3d10dd 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -328,6 +328,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, val |= I2S_CHN_4; break; case 2: + case 1: val |= I2S_CHN_2; break; default: @@ -460,7 +461,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = (SNDRV_PCM_FMTBIT_S8 | @@ -654,7 +655,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) { - if (val >= 2 && val <= 8) + if (val >= 1 && val <= 8) soc_dai->capture.channels_max = val; } -- 2.16.0.rc0.223.g4a4ac83678-goog

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Re: [alsa-devel] Headphone issues iMac 14.1 (A1418)
by Takashi Iwai 09 Jan '18

09 Jan '18

On Tue, 09 Jan 2018 16:40:09 +0100, Freaky wrote: > > Hi, > > the speaker is always working on this Mac, whilst the headphone is > unplugged that is, haven't gotten it to not function. That said, the > unplug/plug events switch data I/O flags and unplugging the headphones > correctly enables the speakers voiding anything I had done with hda-verb. > > I'm not sure what the parameters do btw :). I understand the last parameter > is some value being written and that it's probably a bunch of binary flags, > but have little clue on what they exactly do. > > >From flipping flags and looking at the alsa-info.sh it seems that each bit > corresponds to a data I/O channel. 0x1 being IO[0], 0x2 IO[1], 0x4 IO[2], > 0x8 IO[3]. > > When IO[3]'s data=1 there's audio on the speakers (0x8) > So we wrote > 0x0 and verified alsa-info, all data IO's are disabled. Audio is gone. > Wrote 0x8, enabling bit 4, which flipped IO[3]'s data back to 1 and we have > audio on speaker. I presume that makes it safe to assume IO[3] is the Amp > for speaker. > > Plugged the headphone and looked at alsa-info. > Now IO[1]'s data is enabled and that seems to be where it goes wrong. wrote > 0x0 ensuring all IO's data are disabled (checked headphones - no sound > obviously) and then 0x4 which enabled only IO[2]'s data and there's audio > again. I presume IO[2] is the Amp for the headphones thus. > > Using only 0x4 works as well thus. I have no clue if there are any Amps > behind IO[0] and IO[1] and if so where they would be heading I'm afraid. > The driver enables IO[1]'s data upon plugging the headphone, which should > be IO[2]'s data on this system and unplugging enabled IO[3]'s data. That > might also be why I never have seen it as broken as I need to unplug the > headphone first, which switches flags correctly for the speaker output. > > Plugging / unplugging doesn't seem to have any effect on the other flags > required (direction and enabled). However, they do need to be set initially > and in an earlier test, when booted with the headphone already plugged and > then enabling them before any audio was played audio still didn't work. > Needed to enable them again. Perhaps pulse did something with the flags > then though (restoring the state at first audio playback or something). I > can test this if desired. Plugging/unplugging the headphones doesn't seem > to affect any of the IO's direction nor enabled flags, they remain the same > as they were. OK, so this looks like the gpio 2/3 case, and it corresponds to imac27_122 model. Could you try to add the model option like below? Create a module config file, e.g. /etc/modprobe.d/50-hda.conf containing: options snd-hda-intel model=,imac27_122 (Note the comma before imac27_122, it's no typo, but indicates to specify the second device -- where Haswell gives the HDMI/DP controller for the first PCI device while the second one is the analog I/O). If this works, the patch like below should work without the model option above. Takashi --- diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 80bbadc83721..17660d03a314 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -408,6 +408,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ /* codec SSID */ + SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),

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