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

[alsa-devel] [PATCH 1/2] ASoC: wm9712: Use empty struct initializer
by Fabio Estevam 14 Feb '18

14 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> { 0 } only clears the first member of the structure. The first member of the snd_soc_dapm_update struct is pointer, so wrting 0 to a pointer results in the following sparse warning: sound/soc/codecs/wm9712.c:229:47: warning: Using plain integer as NULL pointer Use the empty struct initializer that clears all the struct members and fixes the sparse warning. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 7aeeeec..953d94d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -226,7 +226,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int mixer, mask, shift, old; - struct snd_soc_dapm_update update = { 0 }; + struct snd_soc_dapm_update update = {}; bool change; mixer = mc->shift >> 8; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: uniphier: fix an error code in uniphier_aio_comprdma_new()
by Dan Carpenter 14 Feb '18

14 Feb '18

The dma_mapping_error() returns true or false, but we want to return -ENOMEM if there was an error. Fixes: e98131222ff9 ("ASoC: uniphier: add support for UniPhier AIO compress audio") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c index 7f7abe3ae99d..4751c8bdabec 100644 --- a/sound/soc/uniphier/aio-compress.c +++ b/sound/soc/uniphier/aio-compress.c @@ -54,12 +54,11 @@ static int uniphier_aio_comprdma_new(struct snd_soc_pcm_runtime *rtd) dma_dir = DMA_TO_DEVICE; sub->compr_addr = dma_map_single(dev, sub->compr_area, size, dma_dir); - ret = dma_mapping_error(dev, sub->compr_addr); - if (ret) { + if (dma_mapping_error(dev, sub->compr_addr)) { kfree(sub->compr_area); sub->compr_area = NULL; - return ret; + return -ENOMEM; } sub->compr_bytes = size;

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[alsa-devel] [PATCH][RFC] ASoC: soc-core: soc_probe_dai() code simplification
by Kuninori Morimoto 14 Feb '18

14 Feb '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current soc_probe_dai() is using deep nested condition. Thus, it is difficult to read/understand. This patch simplification it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9c65477..7687515 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1619,22 +1619,21 @@ static int soc_probe_link_components(struct snd_soc_card *card, static int soc_probe_dai(struct snd_soc_dai *dai, int order) { - int ret; + if (dai->probed || + dai->driver->probe_order != order) + return 0; - if (!dai->probed && dai->driver->probe_order == order) { - if (dai->driver->probe) { - ret = dai->driver->probe(dai); - if (ret < 0) { - dev_err(dai->dev, - "ASoC: failed to probe DAI %s: %d\n", - dai->name, ret); - return ret; - } + if (dai->driver->probe) { + int ret = dai->driver->probe(dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", + dai->name, ret); + return ret; } - - dai->probed = 1; } + dai->probed = 1; + return 0; } -- 1.9.1

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[alsa-devel] [PATCH v2 0/7] ASoC: samsung: Add HDMI sound support for tm2 board
by Sylwester Nawrocki 14 Feb '18

14 Feb '18

This series adds support for HDMI audio on exynos5433 tm2 board, it includes a few fixes for the I2S driver and an extension of the existing card driver to support the I2S1<->HDMI transmitter path. In v2 only support for 48k/96k/192k samples rate is left, as adjusting the AUD PLL output frequency might cause regressions. Sylwester Nawrocki (7): ASoC: samsung: i2s: Update clock-output-names property documentation ASoC: samsung: i2s: Ensure names of supplied clocks are unique ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access ASoC: samsung: i2s: Define the parameters list for SAMSUNG_I2S_OPCLK ASoC: samsung,tm2-audio DT binding documentation update ASoC: samsung: Add support for HDMI audio on TM2 board ASoC: samsung: Add missing #sound-dai-cells property documentation .../bindings/sound/samsung,tm2-audio.txt | 14 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 +-- sound/soc/samsung/i2s-regs.h | 11 +- sound/soc/samsung/i2s.c | 39 ++++-- sound/soc/samsung/i2s.h | 11 +- sound/soc/samsung/tm2_wm5110.c | 152 +++++++++++++++++---- 6 files changed, 191 insertions(+), 58 deletions(-) -- 2.14.2

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[alsa-devel] [PATCH 0/8] ASoC: samsung: Add HDMI sound support for tm2 board
by Sylwester Nawrocki 14 Feb '18

14 Feb '18

This series adds support for HDMI audio on exynos5433 tm2 board, it includes a few fixes for the I2S driver and an extension of the existing card driver to support the I2S1<->HDMI transmitter path. Sylwester Nawrocki (8): ASoC: samsung: i2s: Ensure the RCLK rate is properly determined ASoC: samsung: i2s: Ensure names of supplied clocks are unique ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access ASoC: samsung: i2s: Define the parameters list for SAMSUNG_I2S_OPCLK ASoC: samsung: Add support for HDMI audio on TM2 board ASoC: samsung: i2s: Update clock-output-names property documentation ASoC: samsung: Add missing #sound-dai-cells property documentation ASoC: samsung,tm2-audio DT binding documentation update .../bindings/sound/samsung,tm2-audio.txt | 14 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 +-- sound/soc/samsung/i2s-regs.h | 11 +- sound/soc/samsung/i2s.c | 50 ++++-- sound/soc/samsung/i2s.h | 11 +- sound/soc/samsung/tm2_wm5110.c | 180 ++++++++++++++++++--- 6 files changed, 229 insertions(+), 59 deletions(-) -- 2.14.2

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[alsa-devel] [PATCH v5] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 14 Feb '18

14 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes will be only added if the codec is clock master. In case the role changes runtime, the applied routes are removed. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reviewed-by: Jyri Sarha <jsarha(a)ti.com> --- Hi, CHanges since v4: - Rebased on top of CODEC to component conversion - Added Reviewed-by from Jyri Changes since v3: - install or remove the master mode DAPM routes if needed - move the clock master DAPM route 'management' to a separate function Changes since v2: - Leftover debug prints removed. Changes since v1: - Only apply the master mode DAPM routes when the codec is clock master - comments added to explain the need. Regards, Peter sound/soc/codecs/tlv320aic31xx.c | 73 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 72 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3cd924dc300..7090342e8285 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_component *component) { int ret = 0; @@ -912,6 +936,53 @@ static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int aic31xx_clock_master_routes(struct snd_soc_component *component, + unsigned int fmt) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + int ret; + + fmt &= SND_SOC_DAIFMT_MASTER_MASK; + if (fmt == SND_SOC_DAIFMT_CBS_CFS && + aic31xx->master_dapm_route_applied) { + /* + * Remove the DAPM route(s) for codec clock master modes, + * if applied + */ + ret = snd_soc_dapm_del_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_del_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = false; + } else if (fmt != SND_SOC_DAIFMT_CBS_CFS && + !aic31xx->master_dapm_route_applied) { + /* + * Add the needed DAPM route(s) for codec clock master modes, + * if it is not done already + */ + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + + return 0; +} + static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -992,7 +1063,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); - return 0; + return aic31xx_clock_master_routes(component, fmt); } static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [RfC][PATCH] ALSA: hda/realtek Pin quirk for Fujitsu U7[25]7
by Jan-Marek Glogowski 14 Feb '18

14 Feb '18

Hi everybody, I got new HW from our supplier, a Fujitsu U727 and a U757 laptop, which have a headset jack. Attaching a headset, the microphone doesn't work. After some reading and trying various tools to understand the problem, I found hdajacksensetest, which showed a sense change for a seemingly wrong unconnected pin. Overriding this pins' config has fixed the headsets microphone, so now I have the attached kernel patch with a SND_HDA_PIN_QUIRK. After compiling the patched module (for 4.4), everything works as expected, but hdajackretask still shows the pin as unconnected, so I'm wondering, if I need to do something in addition? Is this normal behaviour? This is just a port to master, as we (Munich) are on Ubuntu Trusty with Xenial HWE (4.4 kernel). Thanks for the review. Jan-Marek Glogowski P.S. checkpatch.pl warned about too long lines for the SND_HDA_PIN_QUIRK. Should I fix that? >From 69f1c296b22ab6dce93ba50de9cf05e279d52088 Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski <glogow(a)fbihome.de> Date: Wed, 24 Jan 2018 18:57:38 +0100 Subject: [PATCH] ALSA: hda/realtek Pin quirk for Fujitsu U7[25]7 These laptops have a combined jack to attach headsets, the U727 on the left, the U757 on the right, but a headsets microphone doesn't work. Using hdajacksensetest I found that pin 0x19 changed the present state when plugging the headset, in addition to 0x21, but didn't have the correct configuration (shown as "Not connected"). So this sets the configuration to the same values as the headphone pin 0x21 except for the device type microphone, which makes it work correctly. With the patch the configured pins for U727 are Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x14 (Internal Speaker): present = No Pin 0x19 (Black Mic, Left side): present = No Pin 0x1d (Internal Aux): present = No Pin 0x21 (Black Headphone, Left side): present = No Signed-off-by: Jan-Marek Glogowski <glogow(a)fbihome.de> --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9aafc6c..52c6beb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5275,6 +5275,8 @@ enum { ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC255_FIXUP_FUJITSU_U727_MIC, + ALC255_FIXUP_FUJITSU_U757_MIC, ALC255_FIXUP_HEADSET_MODE, ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -5786,6 +5788,20 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC255_FIXUP_FUJITSU_U727_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* detect headset mic, left */ + { } + }, + }, + [ALC255_FIXUP_FUJITSU_U757_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11020 }, /* detect headset mic, right */ + { } + }, + }, [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, @@ -6589,6 +6605,18 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x1b, 0x01011020}, {0x21, 0x02211010}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x10cf, "Fujitsu Lifebook U727", ALC255_FIXUP_FUJITSU_U727_MIC, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x19, 0x411111f0}, + {0x1d, 0x909701f0}, + {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x10cf, "Fujitsu Livebook U757", ALC255_FIXUP_FUJITSU_U757_MIC, + {0x12, 0xb7a60130}, + {0x14, 0x90170110}, + {0x19, 0x411111f0}, + {0x1d, 0x909701f0}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x14, 0x90170110}, -- 2.1.4

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[alsa-devel] kernel crash in snd_ctl_notify
by Sven Schnelle 14 Feb '18

14 Feb '18

Hi List, i just got external screens for my Thinkpad T470p. They are connected via Displayport to the docking station of the thinkpad. However, when i put the Thinkpad in the docking station, i see the following kernel crash: Feb 5 08:39:21 t470p kernel: [ 106.616102] BUG: unable to handle kernel NULL pointer dereference at 0000000000000 Feb 5 08:39:21 t470p kernel: [ 106.616119] IP: snd_ctl_notify.part.9+0xb3/0x190 Feb 5 08:39:21 t470p kernel: [ 106.616123] PGD 0 P4D 0 Feb 5 08:39:21 t470p kernel: [ 106.616129] Oops: 0000 [#1] SMP PTI Feb 5 08:39:21 t470p kernel: [ 106.616133] Modules linked in: e1000e Feb 5 08:39:21 t470p kernel: [ 106.616140] CPU: 6 PID: 1298 Comm: Xorg Not tainted 4.15.0+ #148 Feb 5 08:39:21 t470p kernel: [ 106.616143] Hardware name: LENOVO 20J6003DGE/20J6003DGE, BIOS R0FET39W (1.19 ) 12/ Feb 5 08:39:21 t470p kernel: [ 106.616149] RIP: 0010:snd_ctl_notify.part.9+0xb3/0x190 Feb 5 08:39:21 t470p kernel: [ 106.616152] RSP: 0018:ffffb7d2c068fa88 EFLAGS: 00010086 Feb 5 08:39:21 t470p kernel: [ 106.616156] RAX: ffff949ab97edf60 RBX: ffff949abb4ca4c8 RCX: 0000000000000000 Feb 5 08:39:21 t470p kernel: [ 106.616159] RDX: 0000000000000060 RSI: 0000000000000000 RDI: ffff949ab97edfc0 Feb 5 08:39:21 t470p kernel: [ 106.616162] RBP: ffff949abb4ca000 R08: ffff949abaac3a10 R09: ffff949ab97edf60 Feb 5 08:39:21 t470p kernel: [ 106.616164] R10: ffff949abaac3a10 R11: 00000000000003d8 R12: ffff949aaea132c0 Feb 5 08:39:21 t470p kernel: [ 106.616167] R13: 0000000000000202 R14: 0000000000000010 R15: ffff949aaea13280 Feb 5 08:39:21 t470p kernel: [ 106.616171] FS: 00007f57ded37500(0000) GS:ffff949ad1580000(0000) knlGS:0000000000 Feb 5 08:39:21 t470p kernel: [ 106.616174] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Feb 5 08:39:21 t470p kernel: [ 106.616176] CR2: 0000000000000010 CR3: 000000043cf06005 CR4: 00000000003606e0 Feb 5 08:39:21 t470p kernel: [ 106.616179] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 Feb 5 08:39:21 t470p kernel: [ 106.616182] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Feb 5 08:39:21 t470p kernel: [ 106.616184] Call Trace: Feb 5 08:39:21 t470p kernel: [ 106.616194] hdmi_present_sense+0xdc/0x370 Feb 5 08:39:21 t470p kernel: [ 106.616200] check_presence_and_report+0x5e/0x80 Feb 5 08:39:21 t470p kernel: [ 106.616206] intel_audio_codec_disable+0xab/0xd0 Feb 5 08:39:21 t470p kernel: [ 106.616213] intel_encoders_disable.isra.97+0x70/0x90 Feb 5 08:39:21 t470p kernel: [ 106.616219] haswell_crtc_disable+0x47/0x140 Feb 5 08:39:21 t470p kernel: [ 106.616225] intel_atomic_commit_tail+0x70e/0xcb0 Feb 5 08:39:21 t470p kernel: [ 106.616231] ? intel_atomic_commit_ready+0x44/0x4c Feb 5 08:39:21 t470p kernel: [ 106.616236] intel_atomic_commit+0x21f/0x2d0 Feb 5 08:39:21 t470p kernel: [ 106.616241] drm_atomic_connector_commit_dpms+0xe5/0xf0 Feb 5 08:39:21 t470p kernel: [ 106.616248] drm_mode_obj_set_property_ioctl+0x153/0x260 Feb 5 08:39:21 t470p kernel: [ 106.616253] ? drm_mode_connector_set_obj_prop+0x70/0x70 Feb 5 08:39:21 t470p kernel: [ 106.616258] drm_mode_connector_property_set_ioctl+0x2e/0x40 Feb 5 08:39:21 t470p kernel: [ 106.616264] drm_ioctl_kernel+0x59/0xb0 Feb 5 08:39:21 t470p kernel: [ 106.616270] drm_ioctl+0x29f/0x340 Feb 5 08:39:21 t470p kernel: [ 106.616275] ? drm_mode_connector_set_obj_prop+0x70/0x70 Feb 5 08:39:21 t470p kernel: [ 106.616282] ? signal_setup_done+0x57/0x90 Feb 5 08:39:21 t470p kernel: [ 106.616288] ? __fpu__restore_sig+0x80/0x400 Feb 5 08:39:21 t470p kernel: [ 106.616295] do_vfs_ioctl+0x8d/0x5d0 Feb 5 08:39:21 t470p kernel: [ 106.616301] SyS_ioctl+0x3b/0x70 Feb 5 08:39:21 t470p kernel: [ 106.616307] entry_SYSCALL_64_fastpath+0x22/0x8a Feb 5 08:39:21 t470p kernel: [ 106.616312] RIP: 0033:0x7f57dc1b7307 Feb 5 08:39:21 t470p kernel: [ 106.616314] RSP: 002b:00007ffe3459f138 EFLAGS: 00003246 Feb 5 08:39:21 t470p kernel: [ 106.616317] Code: 8b 00 48 39 c8 75 ef ba 58 00 00 00 be 20 80 08 01 48 89 4c 24 0 Feb 5 08:39:21 t470p kernel: [ 106.616384] RIP: snd_ctl_notify.part.9+0xb3/0x190 RSP: ffffb7d2c068fa88 Feb 5 08:39:21 t470p kernel: [ 106.616387] CR2: 0000000000000010 Feb 5 08:39:21 t470p kernel: [ 106.616391] ---[ end trace 0e8742968e349514 ]--- RIP is at snd_ctl_notify, because the id parameter is NULL. Looking a bit further, i see the following in the kernel log: Feb 6 08:14:20 t470p kernel: [ 6.266169] snd_hda_intel 0000:00:1f.3: Too many HDMI devices Feb 6 08:14:20 t470p kernel: [ 6.266170] snd_hda_intel 0000:00:1f.3: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y Feb 6 08:14:20 t470p kernel: [ 6.266171] snd_hda_intel 0000:00:1f.3: Too many HDMI devices Feb 6 08:14:20 t470p kernel: [ 6.266172] snd_hda_intel 0000:00:1f.3: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y Feb 6 08:14:20 t470p kernel: [ 6.266172] snd_hda_intel 0000:00:1f.3: Too many HDMI devices Feb 6 08:14:20 t470p kernel: [ 6.266173] snd_hda_intel 0000:00:1f.3: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y So it seems like commit 1f7f51a63114bab3a05920f4b1343154e95e2cb6 ("Fix regression of hdmi eld control created based on invalid pcm") might be causing this issue. While it makes the probe work, it doesn't assign the pcm pointer in the hdmi data structure. Later it get's referenced and the kernel crashes. Enabling CONFIG_SND_DYNAMIC_MINORS makes the issue go away, but the kernel shouldn't crash if it's disabled.I made the attached patch to fix this. However, i'm not sure whether this is the right way to fix it and whether it needs to be patched in other places. Regards Sven

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[alsa-devel] [PATCH v3] ASoC: tlv320aic31xx: Fix master mode clock I2S bus clocks
by Peter Ujfalusi 14 Feb '18

14 Feb '18

In the reset state of the codec we do not have complete playback or capture routes. The audio playback/capture will not work due to missing clock signals on the I2S bus if PLL, MDAC/NDAC/DAC MADC/NADC/ADC is powered down. To make sure that even if all output/input is disconnected the codec is generating clocks, we need to have valid DAPM route in every case to power up the must needed parts of the codec. I have verified that switching DAC (during playback) or ADC (during capture) will stop the I2S clocks, so the only solution is to connect the 'Off' routes as well to output/input. The routes only going to be added if the codec is configured as clock master. Tested on am43x-epos-evm with aic3111 codec in master mode. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v2: - Leftover debug prints removed. Changes since v1: - Only apply the master mode DAPM routes when the codec is clock master - comments added to explain the need. Regards, Peter sound/soc/codecs/tlv320aic31xx.c | 44 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 858cb8be445f..088648791111 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -166,6 +166,7 @@ struct aic31xx_priv { unsigned int sysclk; u8 p_div; int rate_div_line; + bool master_dapm_route_applied; }; struct aic31xx_rate_divs { @@ -670,6 +671,29 @@ aic310x_audio_map[] = { {"SPK", NULL, "SPK ClassD"}, }; +/* + * Always connected DAPM routes for codec clock master modes. + * If the codec is the master on the I2S bus, we need to power on components + * to have valid DAC_CLK and also the DACs and ADC for playback/capture. + * Otherwise the codec will not generate clocks on the bus. + */ +static const struct snd_soc_dapm_route +common31xx_cm_audio_map[] = { + {"DAC Left Input", "Off", "DAC IN"}, + {"DAC Right Input", "Off", "DAC IN"}, + + {"HPL", NULL, "DAC Left"}, + {"HPR", NULL, "DAC Right"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_cm_audio_map[] = { + {"MIC1LP P-Terminal", "Off", "MIC1LP"}, + {"MIC1RP P-Terminal", "Off", "MIC1RP"}, + {"MIC1LM P-Terminal", "Off", "MIC1LM"}, + {"MIC1LM M-Terminal", "Off", "MIC1LM"}, +}; + static int aic31xx_add_controls(struct snd_soc_codec *codec) { int ret = 0; @@ -916,6 +940,7 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); u8 iface_reg1 = 0; u8 iface_reg2 = 0; u8 dsp_a_val = 0; @@ -992,6 +1017,25 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, AIC31XX_BCLKINV_MASK, iface_reg2); + /* Add the needed DAPM route(s) for codec clock master modes, once */ + if (iface_reg1 & (AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER) && + !aic31xx->master_dapm_route_applied) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int ret; + + ret = snd_soc_dapm_add_routes(dapm, common31xx_cm_audio_map, + ARRAY_SIZE(common31xx_cm_audio_map)); + if (!ret && !(aic31xx->codec_type & DAC31XX_BIT)) + ret = snd_soc_dapm_add_routes(dapm, + aic31xx_cm_audio_map, + ARRAY_SIZE(aic31xx_cm_audio_map)); + + if (ret) + return ret; + + aic31xx->master_dapm_route_applied = true; + } + return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ALSA: seq: Fix racy pool initializations
by Takashi Iwai 14 Feb '18

14 Feb '18

ALSA sequencer core initializes the event pool on demand by invoking snd_seq_pool_init() when the first write happens and the pool is empty. Meanwhile user can reset the pool size manually via ioctl concurrently, and this may lead to UAF or out-of-bound accesses since the function tries to vmalloc / vfree the buffer. A simple fix is to just wrap the snd_seq_pool_init() call with the recently introduced client->ioctl_mutex; as the calls for snd_seq_pool_init() from other side are always protected with this mutex, we can avoid the race. Reported-by: 范龙飞 <long7573(a)126.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 60db32785f62..04d4db44fae5 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1003,7 +1003,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, { struct snd_seq_client *client = file->private_data; int written = 0, len; - int err = -EINVAL; + int err; struct snd_seq_event event; if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT)) @@ -1018,11 +1018,15 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, /* allocate the pool now if the pool is not allocated yet */ if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) { - if (snd_seq_pool_init(client->pool) < 0) + mutex_lock(&client->ioctl_mutex); + err = snd_seq_pool_init(client->pool); + mutex_unlock(&client->ioctl_mutex); + if (err < 0) return -ENOMEM; } /* only process whole events */ + err = -EINVAL; while (count >= sizeof(struct snd_seq_event)) { /* Read in the event header from the user */ len = sizeof(event); -- 2.15.1

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