March 2019 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] opening rawmidi on user ports
by Clemens Ladisch 13 Mar '19

13 Mar '19

AlGrenadine wrote: > it seems ALSA seq looses sysex messages. Is there a buffer size which can be set ? Messages from where to where? Can you check with aseqdump or "amidi --dump"? Messages that are too large for the internal buffer are split into multiple parts. Regards, Clemens

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[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 13 Mar '19

13 Mar '19

The patch ASoC: rsnd: src: fix compiler warnings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 825b22e5961a47ba7387d3ee6729956d6bb3b495 Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 11:34:53 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index db81e066b92e..79716725f67b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -135,7 +135,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -144,7 +144,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -153,7 +153,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -162,7 +162,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -171,7 +171,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -180,7 +180,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

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[alsa-devel] [PATCH] ASoC: nau8810: fix the issue of widget with prefixed name
by John Hsu 13 Mar '19

13 Mar '19

The driver changes the stream name of DAC and ADC to avoid the issue of widget with prefixed name. When the machine adds prefixed name for codec, the stream name of DAI may not find the widgets. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 39512a78a556..832657720a5f 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -411,9 +411,9 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], ARRAY_SIZE(nau8810_mono_mixer_controls)), - SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + SND_SOC_DAPM_DAC("DAC", "Playback", NAU8810_REG_POWER3, NAU8810_DAC_EN_SFT, 0), - SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + SND_SOC_DAPM_ADC("ADC", "Capture", NAU8810_REG_POWER2, NAU8810_ADC_EN_SFT, 0), SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, NAU8810_NSPK_EN_SFT, 0, NULL, 0), -- 2.18.0

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[alsa-devel] [PATCH] ASoC: samsung: odroid: Fix clock configuration for 44100 sample rate
by Sylwester Nawrocki 13 Mar '19

13 Mar '19

After commit fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") the audio root clock frequency is configured improperly for 44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead of 22579000 Hz. This results in a too low value of the PSR clock divider in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g. 1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2). Fix this by increasing the correction passed to clk_set_rate() to take into account inaccuracy of the EPLL frequency properly. Fixes: fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") Reported-by: JaeChul Lee <jcsing.lee(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 694512f980fd..1dc54c4206f0 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -91,11 +91,11 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, return ret; /* - * We add 1 to the rclk_freq value in order to avoid too low clock + * We add 2 to the rclk_freq value in order to avoid too low clock * frequency values due to the EPLL output frequency not being exact * multiple of the audio sampling rate. */ - rclk_freq = params_rate(params) * rfs + 1; + rclk_freq = params_rate(params) * rfs + 2; ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) -- 2.17.1

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[alsa-devel] Applied "ASoC: nau8810: use 64-bit arithmetic instead of 32-bit" to the asoc tree
by Mark Brown 13 Mar '19

13 Mar '19

The patch ASoC: nau8810: use 64-bit arithmetic instead of 32-bit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b517229ca2f7836125be58997808f2803f9ebc86 Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Wed, 13 Mar 2019 14:47:18 +0800 Subject: [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index bfd74b86c9d2..e7fd0b2645ea 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,7 @@ static int nau88l0_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- 2.20.1

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Re: [alsa-devel] [PATCH 2/7] drm/edid: Allow to ignore the audio EDID data
by Ville Syrjälä 13 Mar '19

13 Mar '19

On Tue, Mar 05, 2019 at 10:12:40AM +0100, Maxime Ripard wrote: > On Mon, Mar 04, 2019 at 03:05:31PM -0500, Alex Deucher wrote: > > On Mon, Mar 4, 2019 at 2:53 PM Eric Anholt <eric(a)anholt.net> wrote: > > > > > > Maxime Ripard <maxime.ripard(a)bootlin.com> writes: > > > > > > > In some cases, in order to accomodate with displays with poor EDIDs, we > > > > need to ignore that the monitor alledgedly supports audio output and > > > > disable the audio output. > > > > > > > > Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> > > > > --- > > > > drivers/gpu/drm/drm_edid.c | 8 ++++++++ > > > > 1 file changed, 8 insertions(+) > > > > > > > > diff --git a/drivers/gpu/drm/drm_edid.c b/drivers/gpu/drm/drm_edid.c > > > > index 990b1909f9d7..c0258b011bb2 100644 > > > > --- a/drivers/gpu/drm/drm_edid.c > > > > +++ b/drivers/gpu/drm/drm_edid.c > > > > @@ -4190,6 +4190,11 @@ bool drm_detect_hdmi_monitor(struct edid *edid) > > > > } > > > > EXPORT_SYMBOL(drm_detect_hdmi_monitor); > > > > > > > > +static bool ignore_edid_audio = false; > > > > +module_param(ignore_edid_audio, bool, 0644); > > > > +MODULE_PARM_DESC(ignore_edid_audio, > > > > + "Ignore the EDID and always consider that a monitor doesn't have audio capabilities"); > > > > + > > > > /** > > > > * drm_detect_monitor_audio - check monitor audio capability > > > > * @edid: EDID block to scan > > > > @@ -4209,6 +4214,9 @@ bool drm_detect_monitor_audio(struct edid *edid) > > > > bool has_audio = false; > > > > int start_offset, end_offset; > > > > > > > > + if (ignore_edid_audio) > > > > + goto end; > > > > + > > > > edid_ext = drm_find_cea_extension(edid); > > > > if (!edid_ext) > > > > goto end; > > > > > > It looks like the motivation for the original flag on Raspberry Pi was > > > "I've got a non-audio monitor, but the system comes up trying to play > > > audio to HDMI instead of the analog jack". Do we have some way for DRM > > > to communicate to ALSA that this is not the right place to try to play > > > audio by default? > > > > Apparently not. We have users using debug knobs in our drivers to > > disable display audio because ALSA defaults to that rather than other > > audio. > > I guess one way to do this would be to register the card only when an > audio-capable monitor is connected instead of doing this at probe > time. I'm not sure how convenient it is for userspace though. We already provide the ELD to alsa. I'm pretty sure pulseaudio uses that stuff somehow to figure out whether to play audio over HDMI. But since I don't use pulseaudio myself I can't be 100% sure. Cc:ing Takashi/alsa folks for confirmation. -- Ville Syrjälä Intel

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[alsa-devel] [PATCH] ALSA: hda - add more quirks for HP Z2 G4 and HP Z240
by Jaroslav Kysela 13 Mar '19

13 Mar '19

From: Jaroslav Kysela <jkysela(a)redhat.com> Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and HP Z240 models. Reported-by: Jeff Burrell <jeff.burrell(a)hp.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_conexant.c | 3 +++ sound/pci/hda/patch_realtek.c | 6 ++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a4ee7656d9ee..fb65ad31e86c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -936,6 +936,9 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8458, "HP Z2 G4 mini premium", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c8413d44973c..476c3ee31dc8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6764,11 +6764,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), - SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.20.1

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[alsa-devel] opening rawmidi on user ports
by AlGrenadine 13 Mar '19

13 Mar '19

Hi, I use btmidi-server from the bluez stack, which creates user ports to send and receive midi via bluetooth. I can't manager which parameter string i must provide to snd_rawmidi_open function, can you help ? (of course the user ports don't appear in amidi -l) I only have a client and port ids (128:0) when using aconnect -l Thanks

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[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headset MIC of Acer TravelMate X514-51T with ALC255
by Jian-Hong Pan 13 Mar '19

13 Mar '19

The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the internal DMIC uses another module - snd_soc_skl as the driver. We still need the NID 0x1a in the quirk to enable the headset MIC. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c8413d44973c..8debd661e6f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5673,6 +5673,7 @@ enum { ALC225_FIXUP_HEADSET_JACK, ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, + ALC255_FIXUP_ACER_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6639,6 +6640,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_LENOVO_HEADPHONE_NOISE }, + [ALC255_FIXUP_ACER_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11130 }, + { 0x1a, 0x90a60140 }, /* use as internal mic */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6658,6 +6669,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- 2.20.1

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Re: [alsa-devel] [PATCH] Fix speakers on Acer Predator Helios 500 Ryzen laptops
by Takashi Iwai 13 Mar '19

13 Mar '19

On Tue, 05 Mar 2019 00:38:19 +0100, Bernhard Rosenkraenzer wrote: > > On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers > don't work out of the box. > > The problem can be worked around with hdajackretask, remapping the > "Black Headphone, Right side" pin (0x21) to the Internal speaker. > > This patch adds a quirk to change this mapping by default. > > Signed-off-by: Bernhard Rosenkraenzer <bero(a)lindev.ch> Applied, thanks. Takashi

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