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

[alsa-devel] [PATCH 00/10] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 13 Oct '18

13 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (9): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec: add support for speaker amp GPIO ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: sunxi: Add bindings for A64 codec's analog path controls ASoC: sunxi: Add support for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 14 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 30 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 33 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 30 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 451 ++++++++++++++++++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +-- sound/soc/sunxi/sun8i-codec.c | 61 ++- sound/soc/sunxi/sunxi-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sunxi-adda-pr-regmap.h | 7 + 14 files changed, 824 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.h -- 2.19.0

2 15

[alsa-devel] [PATCH v2 0/5] MAX98088/9 of features
by Marco Felsch 12 Oct '18

12 Oct '18

Hi, this the v2 with the integrated review from the v1 [1]. I picked up the max98088 patches from Andreas series [2] and ported them to the new kernel. Regards, Marco [1] http://mailman.alsa-project.org/pipermail/alsa-devel/ \ 2018-September/140501.html [2] http://lists.infradead.org/pipermail/linux-arm-kernel/ \ 2015-February/324440.html Changes: v2: - make master clock optional - be more verbose in commit messages Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable .../bindings/sound/maxim,max98088.txt | 23 ++++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98088.c | 36 +++++++++++++++++++ 3 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt -- 2.19.0

3 7

[alsa-devel] [PATCH] ASoC: sta32x: set ->component pointer in private struct
by Daniel Mack 12 Oct '18

12 Oct '18

The ESD watchdog code in sta32x_watchdog() dereferences the pointer which is never assigned. This is a regression from a1be4cead9b950 ("ASoC: sta32x: Convert to direct regmap API usage.") which went unnoticed since nobody seems to use that ESD workaround. Fixes: a1be4cead9b950 ("ASoC: sta32x: Convert to direct regmap API usage.") Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/sta32x.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..a878c8d92e68 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -879,6 +879,9 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { -- 2.17.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: pcm3168a: add TDM support
by Kuninori Morimoto 12 Oct '18

12 Oct '18

Hi Mark, Jaroslav These patches adds missing TDM support, and necessary patches for it. Kuninori Morimoto (3): ASoC: pcm3168a: add HW constraint for non RIGHT_J ASoC: pcm3168a: add hw constraint for channel ASoC: pcm3168a: add I2S/Left_J TDM support sound/soc/codecs/pcm3168a.c | 75 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 75 insertions(+) -- 2.7.4 Best regards --- Kuninori Morimoto

2 6

[alsa-devel] [bug report] ALSA: usb: initial USB Audio Device Class 3.0 support
by Dan Carpenter 12 Oct '18

12 Oct '18

Hello Ruslan Bilovol, The patch 9a2fe9b801f5: "ALSA: usb: initial USB Audio Device Class 3.0 support" from Mar 21, 2018, leads to the following static checker warning: sound/usb/stream.c:971 snd_usb_get_audioformat_uac3() warn: potentially allocating too little. 7 vs 1 sound/usb/stream.c 943 /* 944 * Get number of channels and channel map through 945 * High Capability Cluster Descriptor 946 * 947 * First step: get High Capability header and 948 * read size of Cluster Descriptor 949 */ 950 err = snd_usb_ctl_msg(chip->dev, 951 usb_rcvctrlpipe(chip->dev, 0), 952 UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, 953 USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 954 cluster_id, 955 snd_usb_ctrl_intf(chip), 956 &hc_header, sizeof(hc_header)); ^^^^^^^^^^ It looks like this comes from the USB (untrusted). 957 if (err < 0) 958 return ERR_PTR(err); 959 else if (err != sizeof(hc_header)) { 960 dev_err(&dev->dev, 961 "%u:%d : can't get High Capability descriptor\n", 962 iface_no, altno); 963 return ERR_PTR(-EIO); 964 } 965 966 /* 967 * Second step: allocate needed amount of memory 968 * and request Cluster Descriptor 969 */ 970 wLength = le16_to_cpu(hc_header.wLength); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ My private build of Smatch complains that all le16_to_cpu() data probably comes from untrusted sources. 971 cluster = kzalloc(wLength, GFP_KERNEL); ^^^^^^^ Maybe we're not allocating enough bytes for the cluster struct (8 bytes). 972 if (!cluster) 973 return ERR_PTR(-ENOMEM); 974 err = snd_usb_ctl_msg(chip->dev, 975 usb_rcvctrlpipe(chip->dev, 0), 976 UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, 977 USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 978 cluster_id, 979 snd_usb_ctrl_intf(chip), 980 cluster, wLength); 981 if (err < 0) { 982 kfree(cluster); 983 return ERR_PTR(err); 984 } else if (err != wLength) { 985 dev_err(&dev->dev, 986 "%u:%d : can't get Cluster Descriptor\n", 987 iface_no, altno); 988 kfree(cluster); 989 return ERR_PTR(-EIO); 990 } 991 992 num_channels = cluster->bNrChannels; ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 993 chmap = convert_chmap_v3(cluster); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ This code assumes that cluster is large enough without checking. 994 kfree(cluster); 995 regards, dan carpenter

1 1

[alsa-devel] [PATCH v2] pinctrl: madera: Fix uninitialized variable bug in madera_mux_set_mux
by Gustavo A. R. Silva 12 Oct '18

12 Oct '18

There is a potential execution path in which variable *ret* is checked in an IF statement, and then its value is used to report an error at line 659 without being properly initialized previously: 659 if (ret) 660 dev_err(priv->dev, "Failed to write to 0x%x (%d)\n", reg, ret); Fix this by initializing variable *ret* to 0 in order to avoid unpredictable or unintended results. Addresses-Coverity-ID: 1471969 ("Uninitialized scalar variable") Fixes: 218d72a77b0b ("pinctrl: madera: Add driver for Cirrus Logic Madera codecs") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- Changes in v2: - Initialize variable ret to 0 instead of -1. Thanks to Charles Keepax for pointing this out. - Update subject and commit log. drivers/pinctrl/cirrus/pinctrl-madera-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/pinctrl/cirrus/pinctrl-madera-core.c b/drivers/pinctrl/cirrus/pinctrl-madera-core.c index 0266302..a5dda83 100644 --- a/drivers/pinctrl/cirrus/pinctrl-madera-core.c +++ b/drivers/pinctrl/cirrus/pinctrl-madera-core.c @@ -608,7 +608,7 @@ static int madera_mux_set_mux(struct pinctrl_dev *pctldev, unsigned int n_chip_groups = priv->chip->n_pin_groups; const char *func_name = madera_mux_funcs[selector].name; unsigned int reg; - int i, ret; + int i, ret = 0; dev_dbg(priv->dev, "%s selecting %u (%s) for group %u (%s)\n", __func__, selector, func_name, group, -- 2.7.4

3 3

[alsa-devel] [PATCH] ALSA: firewire-motu: add missing entries to Kconfig
by Takashi Sakamoto 12 Oct '18

12 Oct '18

MOTU Traveler and Audio Express are supported as well. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 529d9f405fa9..8a146b039276 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -147,7 +147,9 @@ config SND_FIREWIRE_MOTU help Say Y here to enable support for FireWire devices which MOTU produced: * 828mk2 + * Traveler * 828mk3 + * Audio Express To compile this driver as a module, choose M here: the module will be called snd-firewire-motu. -- 2.19.1

2 3

[alsa-devel] [PATCH] ALSA: isight: fix leak of reference to firewire unit in error path of .probe callback
by Takashi Sakamoto 12 Oct '18

12 Oct '18

In some error paths, reference count of firewire unit is not decreased. This commit fixes the bug. Fixes: 5b14ec25a79b('ALSA: firewire: release reference count of firewire unit in .remove callback of bus driver') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/isight.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index de4decfb74d5..9ebe510ea26b 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -638,7 +638,7 @@ static int isight_probe(struct fw_unit *unit, if (!isight->audio_base) { dev_err(&unit->device, "audio unit base not found\n"); err = -ENXIO; - goto err_unit; + goto error; } fw_iso_resources_init(&isight->resources, unit); @@ -667,12 +667,12 @@ static int isight_probe(struct fw_unit *unit, dev_set_drvdata(&unit->device, isight); return 0; - -err_unit: - fw_unit_put(isight->unit); - mutex_destroy(&isight->mutex); error: snd_card_free(card); + + mutex_destroy(&isight->mutex); + fw_unit_put(isight->unit); + return err; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add mic quirk for the Lenovo G50-30 (17aa:3905)
by Jeremy Cline 12 Oct '18

12 Oct '18

The Lenovo G50-30, like other G50 models, has a Conexant codec that requires a quirk for its inverted stereo dmic. Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1249364 Reported-by: Alexander Ploumistos <alex.ploumistos(a)gmail.com> Tested-by: Alexander Ploumistos <alex.ploumistos(a)gmail.com> Cc: stable(a)vger.kernel.org Signed-off-by: Jeremy Cline <jcline(a)redhat.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5592557fe50e..950e02e71766 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -943,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), + SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), -- 2.19.1

2 1

[alsa-devel] [PATCH linux-next v3 1/1] ASoC: rsnd: ssi: Request dedicated dma channels for busif0 to 7
by jiada_wang＠mentor.com 12 Oct '18

12 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Currently ssi driver only requests dma channel for SSI_0, with naming of 'rxu/txu', which is used to transfer data to/from busif0. But for GEN2/GEN3, there are also busif1 ~ busif7, which need to be used when SSI works in TDM Split/Ex-Split mode. This patch adds support to firstly try to request busif indexed dma channel with naming "rxu[busif]/txu[busif]", if these dma channel don't exist, then try to request 'rxu/txu' dma channel to keep compatibility with platform hasn't had device-tree updated. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- v3: - Drop device-tree changes - Drop automatic busif selection v2: - Instead of configure BUSIF by kctl interface "SSI Out/In BUSIF", use rsnd_ssi_select_busif() to automatically select BUSIF. - Keep 'rxu/txu' dma request, to keep compatibility with GEN2 series - Re-order patch-set to satisfy git bisect v1: initial version --- sound/soc/sh/rcar/ssi.c | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..7e21b29249d7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -76,6 +76,8 @@ #define SSI_NAME "ssi" +#define SSI_DMA_NAME_SIZE 10 + struct rsnd_ssi { struct rsnd_mod mod; @@ -938,12 +940,37 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); int is_play = rsnd_io_is_play(io); - char *name; + char name[SSI_DMA_NAME_SIZE]; - if (rsnd_ssi_use_busif(io)) - name = is_play ? "rxu" : "txu"; - else - name = is_play ? "rx" : "tx"; + if (rsnd_ssi_use_busif(io)) { + int busif = rsnd_ssi_get_busif(io); + struct dma_chan *chan = NULL; + + /* try to request rxu/txu channel with busif index first */ + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rxu%d", busif); + else + snprintf(name, SSI_DMA_NAME_SIZE, "txu%d", busif); + + chan = rsnd_dma_request_channel(rsnd_ssi_of_node(priv), + mod, name); + if (!IS_ERR(chan)) + return chan; + + /* + * If dma channel with busif index doesn't exist + * request rxu/txu dma channel + */ + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rxu"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "txu"); + } else { + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rx"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "tx"); + } return rsnd_dma_request_channel(rsnd_ssi_of_node(priv), mod, name); -- 2.17.0

2 1

[alsa-devel] [PATCH] ALSA: cs4236: fix return check from call to pnp_port_start
by Colin King 11 Oct '18

11 Oct '18

From: Colin Ian King <colin.king(a)canonical.com> Currently the return from pnp_port_start is being checked for validity by checking if it is greater or equal to zero, however, the return is unsigned and hence this check is always true. An 0 return from pnp_port_start is actually an error, so fix the check by checking for validity when it is non-zero. Detected by CoverityScan, CID#1468334 ("Unsigned compared to 0") Fixes: a9824c868a2c ("[ALSA] Add CS4232 PnP BIOS support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/cs423x/cs4236.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 70559e59d18f..2211c39d2c18 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -292,7 +292,7 @@ static int snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) mpu_irq[dev] = SNDRV_AUTO_IRQ; } else { mpu_port[dev] = pnp_port_start(pdev, 0); - if (mpu_irq[dev] >= 0 && + if (!mpu_irq[dev] && pnp_irq_valid(pdev, 0) && pnp_irq(pdev, 0) >= 0) { mpu_irq[dev] = pnp_irq(pdev, 0); } else { -- 2.17.1

3 2

[alsa-devel] [PATCH 1/4] ASoC: max98373: Sort Digital Volume in reverse order
by Ryan Lee 11 Oct '18

11 Oct '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Sort 'max98373_digital_tlv' in reverse order. Fixed wrong use case of 'TLV_DB_SCALE_ITEM' - TLV_DB_SCALE_ITEM(min, step, mute) Max value was set in min field and negative value was set in step field before. sound/soc/codecs/max98373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1093f76..bb5ef52 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -454,7 +454,7 @@ SND_SOC_DAPM_SIGGEN("IMON"), SND_SOC_DAPM_SIGGEN("FBMON"), }; -static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0); +static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1); static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv, 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), @@ -604,7 +604,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG, SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0), SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL, - 0, 0x7F, 0, max98373_digital_tlv), + 0, 0x7F, 1, max98373_digital_tlv), SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN, MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv), SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN, -- 2.7.4

2 7

[alsa-devel] Oops when unloading SOF driver modules
by Guennadi Liakhovetski 11 Oct '18

11 Oct '18

Hi, I'm tracking down a kernel Oops, occurring when unloading driver modules. A complete Oops is below. It triggers because cpu_dai->driver is NULL after remove_dai() in soc-topology.c freed it. On my Debian system this only happens when I unload modules directly after loading. If some time passes after loading the drivers, pulseaudio completes its probing and releases the hardware, then remove_dai() isn't called before snd_pcm_do_stop() but after, then everything is running correctly. I'm not sure what the best way to fix this would be. Any ideas? Thanks Guennadi Oct 1 10:23:32 UP-APL01 kernel: [ 142.978059] BUG: unable to handle kernel NULL pointer dereference at 0000000000000088 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978066] PGD 0 P4D 0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978075] Oops: 0000 [#1] SMP NOPTI Oct 1 10:23:32 UP-APL01 kernel: [ 142.978083] CPU: 3 PID: 1240 Comm: alsa-sink-Port0 Tainted: G W E 4.18.0-upboard+ #1 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978087] Hardware name: AAEON UP-APL01/UP-APL01, BIOS UPA1AM36 04/10/2018 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978114] RIP: 0010:soc_pcm_trigger+0x16d/0x1d0 [snd_soc_core] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978117] Code: 31 c0 48 85 c9 74 07 4c 8b 81 88 00 00 00 ba 6c 04 00 00 48 c7 c6 20 14 84 c0 48 c7 c7 72 34 84 c0 e8 55 a3 4a df 49 8b 45 18 <48> 8b 80 88 00 00 00 48 8b 80 88 00 00 00 48 85 c0 74 16 4c 89 ea Oct 1 10:23:32 UP-APL01 kernel: [ 142.978204] RSP: 0018:ffffb6b781243da8 EFLAGS: 00010286 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978207] RAX: 0000000000000000 RBX: ffff9146ae2a0800 RCX: 0000000000000000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978210] RDX: 0000000000000000 RSI: ffff9146bfd96738 RDI: ffff9146bfd96738 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978212] RBP: ffff9146afdbbc00 R08: 00000000000006fd R09: 0000000000000004 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978215] R10: 00000000000001c3 R11: 0000000000000001 R12: 0000000000000000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978217] R13: ffff9146b1b92f80 R14: ffff9146ae2a0eb0 R15: ffff9146ae2a0ea8 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978220] FS: 00007f3db6b24700(0000) GS:ffff9146bfd80000(0000) knlGS:0000000000000000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978223] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978225] CR2: 0000000000000088 CR3: 0000000177ce6000 CR4: 00000000003406e0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978228] Call Trace: Oct 1 10:23:32 UP-APL01 kernel: [ 142.978247] dpcm_fe_dai_do_trigger+0xee/0x220 [snd_soc_core] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978258] snd_pcm_do_stop+0x65/0x80 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978267] snd_pcm_action_single+0x36/0x80 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978275] snd_pcm_drop+0x6f/0xa0 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978283] snd_pcm_release_substream+0x39/0xd0 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978291] snd_pcm_release+0x5d/0xb0 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978298] __fput+0xd0/0x1e0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978305] task_work_run+0x8a/0xb0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978311] exit_to_usermode_loop+0xe6/0xf0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978315] do_syscall_64+0xf3/0x110 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978321] entry_SYSCALL_64_after_hwframe+0x44/0xa9 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978325] RIP: 0033:0x7f3dc0988ab7 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978327] Code: 10 e9 67 ff ff ff 0f 1f 44 00 00 48 8b 15 c9 f3 2c 00 f7 d8 64 89 02 48 c7 c0 ff ff ff ff e9 6b ff ff ff b8 0b 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d a1 f3 2c 00 f7 d8 64 89 01 48 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978374] RSP: 002b:00007f3db6b23c38 EFLAGS: 00000202 ORIG_RAX: 000000000000000b Oct 1 10:23:32 UP-APL01 kernel: [ 142.978377] RAX: 0000000000000000 RBX: 0000000000000000 RCX: 00007f3dc0988ab7 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978380] RDX: 0000000000000010 RSI: 0000000000001000 RDI: 00007f3dc284a000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978382] RBP: 000056431cb73ac0 R08: 00007f3db763a578 R09: 00007f3db6b23b50 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978384] R10: 00000000fffffff8 R11: 0000000000000202 R12: 000056431cb73ac0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978387] R13: 000056431cb73ac0 R14: 0000000000000000 R15: 000056431caad4b0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978390] Modules linked in: sof_pci_dev(E-) snd_sof_intel_hda_common(E) sof_acpi_dev(E) snd_sof_intel_bdw(E) snd_sof_intel_byt(E) snd_sof_nocodec(E) snd_sof_intel_hsw(E) snd_sof(E) snd_sof_xtensa_dsp(E) snd_soc_acpi_intel_match(E) snd_soc_acpi(E) snd_soc_da7213(E) snd_soc_rt5640(E) snd_soc_rt5651(E) snd_soc_rt5645(E) snd_soc_rt5670(E) snd_soc_rl6231(E) nls_ascii(E) nls_cp437(E) vfat(E) fat(E) squashfs(E) loop(E) spi_pxa2xx_platform(E) intel_rapl(E) i2c_designware_platform(E) i2c_designware_core(E) x86_pkg_temp_thermal(E) coretemp(E) kvm_intel(E) kvm(E) irqbypass(E) crct10dif_pclmul(E) crc32_pclmul(E) ghash_clmulni_intel(E) snd_hda_core(E) efi_pstore(E) pcbc(E) snd_soc_core(E) snd_compress(E) snd_pcm(E) snd_seq_midi(E) snd_seq_midi_event(E) snd_rawmidi(E) aesni_intel(E) aes_x86_64(E) crypto_simd(E) Oct 1 10:23:32 UP-APL01 kernel: [ 142.978445] cryptd(E) glue_helper(E) intel_cstate(E) intel_rapl_perf(E) pwm_lpss_pci(E) efivars(E) lpc_ich(E) pwm_lpss(E) snd_seq(E) idma64(E) virt_dma(E) intel_lpss_pci(E) intel_lpss(E) mfd_core(E) snd_seq_device(E) snd_timer(E) tpm_crb(E) snd(E) tpm_tis(E) tpm_tis_core(E) tpm(E) evdev(E) rng_core(E) mei_me(E) mei(E) soundcore(E) pcc_cpufreq(E) sch_fq_codel(E) parport_pc(E) ppdev(E) lp(E) parport(E) efivarfs(E) ip_tables(E) x_tables(E) autofs4(E) ext4(E) crc32c_generic(E) crc16(E) mbcache(E) jbd2(E) fscrypto(E) hid_generic(E) usbhid(E) hid(E) mmc_block(E) dwc3(E) sdhci_pci(E) xhci_pci(E) cqhci(E) sdhci(E) crc32c_intel(E) i915(E) mmc_core(E) ahci(E) i2c_algo_bit(E) xhci_hcd(E) libahci(E) drm_kms_helper(E) dwc3_pci(E) usbcore(E) libata(E) r8169(E) mii(E) drm(E) scsi_mod(E) video(E) button(E) [last unloaded: snd_sof] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978528] CR2: 0000000000000088 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978532] ---[ end trace f60338bc9477e8ed ]---

1 0

[alsa-devel] Trying to write control utility for YAMAHA Cavit DP-U50/AP-U70/RP-U200.
by Tahiro Hashizume 11 Oct '18

11 Oct '18

Hi The devices of my interest is in fact 18yrs old, but just can't stop using them for being too useful. (I own DP-U50 and RP-U200). For these devices: A.)Source selecting as well as alteration of the labels (for each input) to be displayed on the VDF are possible. B.)Internal DSP is highly configurable c.)Tuner presets are editable (RP-U200 only) through the Win32 utility (DP-U50.exe/AP-U70.exe/RP-U200.exe). Most of sound-related details should be at http://www.alsa-project.org/db/?f=9aad4f38c69efd1c6ae90de239063efeb5a61dce. Basically I am stuck with almost no experience in this, but am trying to write some code to implement the Linux-alternative of these utilities to at least enable the source selecting from Linux via USB (mostly intended to be used with shairplay-sync). Thought examples of device-specific utilities as found in alsa-tools may help but I just don't see connections between USB-dump and control signals that may be accepted by those listed in amixer. Side-notes(Findings so far): The device appears as a composite with several Audio interfaces and a HID-Class interface. Knowing this from early days, I first decided to see hexdump outputs on "/dev/hidraw0" and it indicated what looked like the representation of current status of the device while turning knobs and switching inputs. My guess is, the device takes special commands via the control and echos the current status via HID which is visible as "/dev/hidraw0". ("/dev/input/event0" is also present but doesn't really spit anything.) At this point, thought this was a bi-directional communication via HID-Class (by which I was wrong, as I found out later). Thinking collecting several packet captures under controlled conditions would help understand the mechanism, moved onto capturing WindowMessages (of RP-U200.exe) with Spy++ from which I was able to implement the automated operation of RP-U200.exe via Python. See: https://www.youtube.com/watch?v=-Il9vKtnr1A Figured out 1.)messages of "WM_USER + 1" are from "hid_io.dll" to RP-U200.exe 2.)messages of "WM_USER + 201" are the actual inputs to RP-U200.exe About files attached: a.)RP-U200.hexdump.txt shows outputs observed on "/dev/hidraw0" as I turned the volume knob from min to max and down to min. b.)"util_launch.pcapng.json" shows the interactions between host (Win10 x64) and RP-U200 as RP-U200.exe was being launched (Power:Off->Off). c.)"util_power_on.pcapng.json" shows the interactions between host (Win10 x64) and RP-U200 starting shortly before I pressed the Power button on RP-U200.exe, ending with the output relays closed for speaker outputs (Power:Off->On). Any comments should keep me encouraged. Thanks in advance. Tahiro.H

1 0

[alsa-devel] [asoc:for-4.20 333/490] sound/pci//hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 11 Oct '18

11 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 8036dbc490d16dc3d998246e14c9507ec8272ae2 commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/490] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (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 6bae5ea9498926440ffc883f3dbceb0adc65e492 # 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 'patch_ca0132': >> sound/pci//hda/patch_ca0132.c:7650:20: 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:7650:18: 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 +7650 sound/pci//hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Fixing EAPD for STAC9202 on an embedded platform.
by Andy Davison 11 Oct '18

11 Oct '18

Hi everyone, I’m hoping that somebody here might be able to spot where I’m going wrong; I'm working with a picture-frame-like device manufactured by the late OpenPeak Inc. known as the OpenFrame (or O2 Joggler in the UK). It has an internal speaker and a 3.5mm line out socket; none of the other connections to its STAC9202 codec chip are exposed to the end user. I've been porting Ubuntu Bionic to the device and everything is done, except for the audio driver. Back in kernel 3.18 it was possible to use the following kernel patch and userland ALSA patch to have these two outputs work correctly, with the insertion of a plug into the line out properly muting the internal speaker. https://github.com/andydvsn/OpenFrame-Kernel/blob/master/patches/03-stac920… https://github.com/andydvsn/OpenFrame-Ubuntu/blob/master/overlay-trusty/lib… In kernel 4.14 the same trick no longer works, so I've rewritten the patch to more properly add support for this variant of STAC9202 to ALSA. It now looks like this: https://gist.github.com/andydvsn/7bffa8de0ed691b14508485d75501ffc This seems to line up the pin configurations appropriately, but when a plug is inserted to the line out, although the audio to the internal speaker is muted, the speaker emits a crackling sound whenever the device is processing. It seems that it is not being properly powered down and is amplifying general circuit noise. Disconnecting the plug returns sound to the internal speaker and the crackling ceases. This is reliable across multiple devices. Here is the alsa-info output from both the crackling kernel 4.14 and the working-just-fine kernel 3.18. https://gist.github.com/andydvsn/ccf55e6f7961dd405fc39d41a6ce9ef6 I don't have sufficient knowledge of ALSA drivers and patches and have been working purely from the configurations used by others over the years, so I would really appreciate somebody looking at the alsa-info output and the new patch to determine where I should go from here. I believe this is all to do with EAPD control, but what options I have for controlling that aren't known to me right now. Ideally, I want to end up with the audio muting and switching correctly when the jack is inserted. A bonus would be to lose the unused (or at least inaccessible) audio input and output options in alsamixer and gain individual control of the internal speaker and line out levels (which appear to be paired together at the moment). All the best, A. -- Andrew Davison

2 11

[alsa-devel] [PATCH] pinctrl: madera: Fix uninitialized variable issue in madera_mux_set_mux
by Gustavo A. R. Silva 10 Oct '18

10 Oct '18

There is a potential execution path in which variable *ret* is checked in an IF statement, and then its value is used to report an error at line 659 without being properly initialized previously: 659 if (ret) 660 dev_err(priv->dev, "Failed to write to 0x%x (%d)\n", reg, ret); Fix this by initializing variable *ret* to -1. Addresses-Coverity-ID: 1471969 ("Uninitialized scalar variable") Fixes: 218d72a77b0b ("pinctrl: madera: Add driver for Cirrus Logic Madera codecs") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- drivers/pinctrl/cirrus/pinctrl-madera-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/pinctrl/cirrus/pinctrl-madera-core.c b/drivers/pinctrl/cirrus/pinctrl-madera-core.c index 0266302..a99ace8 100644 --- a/drivers/pinctrl/cirrus/pinctrl-madera-core.c +++ b/drivers/pinctrl/cirrus/pinctrl-madera-core.c @@ -608,7 +608,7 @@ static int madera_mux_set_mux(struct pinctrl_dev *pctldev, unsigned int n_chip_groups = priv->chip->n_pin_groups; const char *func_name = madera_mux_funcs[selector].name; unsigned int reg; - int i, ret; + int i, ret = -1; dev_dbg(priv->dev, "%s selecting %u (%s) for group %u (%s)\n", __func__, selector, func_name, group, -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: max98988: add I2C dependency
by Arnd Bergmann 10 Oct '18

10 Oct '18

max98988 only builds with I2C support enabled, otherwise we get a build error: sound/soc/codecs/max98088.c:1789:1: error: data definition has no type or storage class [-Werror] module_i2c_driver(max98088_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/max98088.c:1789:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/max98088.c:1789:1: error: parameter names (without types) in function declaration [-Werror] sound/soc/codecs/max98088.c:1780:26: error: 'max98088_i2c_driver' defined but not used [-Werror=unused-variable] Fixes: 24ae67c58250 ("ASoC: max98988: make it selectable") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3c6bd6019b92..774d38310875 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -641,6 +641,7 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" + depends on I2C config SND_SOC_MAX98090 tristate -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: use 32bit TDM width as default
by Kuninori Morimoto 10 Oct '18

10 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit fb2815f44a9e ("ASoC: rsnd: add support for 16/24 bit slot widths") added TDM width check, and return error if it was not 16/24/32 bit. But it is too strict. This patch uses 32bit same as default. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4..f930f51 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -744,8 +744,8 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, case 32: break; default: - dev_err(dev, "unsupported slot width value: %d\n", slot_width); - return -EINVAL; + /* use default */ + slot_width = 32; } switch (slots) { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: enable mclk-fs on codec node
by Kuninori Morimoto 10 Oct '18

10 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current audio-graph-card is supporting mclk-fs on CPU node side only. But having Codec node also is good idea. It will be just ignored if not defined. "rcpu_ep" is same as "cpu_ep", This patch tidyup it, too. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index fb6635f..25c819e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -182,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, if (ret < 0) goto dai_link_of_err; - of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); if (ret < 0) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ALSA: firewire: block .remove callback of bus
by Takashi Sakamoto 10 Oct '18

10 Oct '18

Hi, In a discussion for devres support[1], I realize difference of unbind behaviour of drivers in ALSA firewire stack and in the others. For consistency behaviour inner the same subsystem for users, it's better to imitate the behaviour. Additionally, blocking .remove function simplifies codes to releasing device. This commit uses 'snd_card_free()' instead of 'snd_card_free_when_closed()' in .remove function and performs refactoring for release codes. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140431.… Takashi Sakamoto (4): ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released ALSA: firewire: release reference count of firewire unit in .remove callback of bus driver ALSA: bebob/fireworks: simplify handling of local device entry table ALSA: firewire: simplify cleanup process when failing to register sound card sound/firewire/bebob/bebob.c | 43 ++++++--------------- sound/firewire/dice/dice.c | 35 ++++------------- sound/firewire/digi00x/digi00x.c | 28 +++++--------- sound/firewire/fireface/ff.c | 28 +++++--------- sound/firewire/fireworks/fireworks.c | 56 ++++++++-------------------- sound/firewire/isight.c | 8 ++-- sound/firewire/motu/motu.c | 39 +++++-------------- sound/firewire/oxfw/oxfw.c | 39 +++++-------------- sound/firewire/tascam/tascam.c | 32 +++++----------- 9 files changed, 90 insertions(+), 218 deletions(-) -- 2.19.0

2 8

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the problem of the front MIC on the Lenovo M715
by Hui Wang 10 Oct '18

10 Oct '18

The front MIC on the Lenovo M715 can't record sound, after applying the ALC294_FIXUP_LENOVO_MIC_LOCATION, the problem is fixed. So add the pin configuration of this machine to the pin quirk table. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f3c8e888c2a..2871c54f62ab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6840,6 +6840,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x02a11040}, {0x1b, 0x01014020}, {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, + {0x14, 0x90170110}, + {0x19, 0x02a11030}, + {0x1a, 0x02a11040}, + {0x1b, 0x01011020}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, {0x14, 0x90170110}, {0x19, 0x02a11020}, -- 2.17.1

2 1

[alsa-devel] [PATCH linux-next v2 9/9] ASoC: rsnd: add busif property to dai stream
by jiada_wang＠mentor.com 10 Oct '18

10 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> in GEN3 SSI may use different BUSIF for data transfer, this patch adds busif property to each dai stream, to indicate the BUSIF used by playback/capture stream. Also adds rsnd_ssi_select_busif() to automatically select BUSIF (currently only BUSIF0 is selected) Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 3 +++ sound/soc/sh/rcar/dma.c | 31 +++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 30 +++++++++++++++++++++++++++++- 4 files changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..5e3e6e65bcdf 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1158,6 +1158,9 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); int ret; + int chan = rsnd_runtime_channel_for_ssi_with_params(io, hw_params); + + rsnd_ssi_select_busif(io, chan); ret = rsnd_dai_call(hw_params, io, substream, hw_params); if (ret) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 6d1947515dc8..bf87939f79bb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -792,6 +792,37 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } +void rsnd_dma_addr_update(struct rsnd_dai_stream *io) +{ + struct rsnd_mod *ssi_mod = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *mod_from = NULL; + struct rsnd_mod *mod_to = NULL; + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int is_play = rsnd_io_is_play(io); + struct rsnd_dma *dma = rsnd_mod_to_dma(io->dma); + + if (!dmac) + return; + + rsnd_dma_of_path(ssi_mod, io, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + + if (mod_from && mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + + dmapp->chcr = rsnd_dmapp_get_chcr(io, mod_from, mod_to) | + PDMACHCR_DE; + } + + dev_dbg(dev, "%s[%d] %pad -> %pad\n", + rsnd_mod_name(ssi_mod), rsnd_mod_id(ssi_mod), + &dma->src_addr, &dma->dst_addr); +} + int rsnd_dma_probe(struct rsnd_priv *priv) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 4464d1d0a042..7ce10267c87b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -227,12 +227,14 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); +void rsnd_ssi_select_busif(struct rsnd_dai_stream *io, int chan); /* * R-Car DMA */ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, struct rsnd_mod **dma_mod); +void rsnd_dma_addr_update(struct rsnd_dai_stream *io); int rsnd_dma_probe(struct rsnd_priv *priv); struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); @@ -457,6 +459,7 @@ struct rsnd_dai_stream { struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ u32 parent_ssi_status; + u32 busif; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 992aeac09e76..d515ee192276 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -154,7 +154,35 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) { - return 0; /* BUSIF0 only for now */ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + + if (!rsnd_is_gen3(priv)) + return 0; + + if (!rsnd_ssi_use_busif(io)) + return 0; + + return io->busif; +} + +void rsnd_ssi_select_busif(struct rsnd_dai_stream *io, int chan) +{ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + u32 busif = 0; /* BUSIF0 only for now */ + + if (!rsnd_is_gen3(priv)) + return; + + if (!rsnd_ssi_use_busif(io)) + return; + + /* FIXME: calcuate BUSIF based on SSI mode in future */ + if (io->busif == busif) + return; + + io->busif = busif; + + rsnd_dma_addr_update(io); } static void rsnd_ssi_status_clear(struct rsnd_mod *mod) -- 2.17.0

3 9

[alsa-devel] [PATCH] ALSA: intel8x0: fix a redundant check bug
by Wenwen Wang 09 Oct '18

09 Oct '18

In snd_intel8x0_codec_semaphore(), the parameter 'codec' is firstly checked to see whether it is greater than 2. If yes, an error code EIO is returned. Otherwise, 'chip->in_sdin_init' is further checked. If 'chip->in_sdin_init' is not zero, 'codec' is updated immediately with 'chip->codec_isr_bits'. That means, the parameter 'codec' is not used in this branch. Actually, it is only used in the else branch when 'chip->in_sdin_init' is zero. Thus, the check on the parameter 'codec' is redundant if 'chip->in_sdin_init' is not zero. This can cause potential incorrect execution in this function. Suppose the parameter 'codec' is 3, which is greater than 2, and 'chip->in_sdin_init' is not zero. The current version of this function will return EIO after the first check because 'codec' is greater than 2. However, since 'codec' will be updated in the following execution when 'chip->in_sdin_init' is not zero, this check will be meaningless and the execution should continue, instead of returning the error code EIO. This patch avoids the above issue by moving the check on the parameter 'codec' to the else branch of the if statement that checks 'chip->in_sdin_init'. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/pci/intel8x0.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 5ee468d..f619e58 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -515,13 +515,13 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code { int time; - if (codec > 2) - return -EIO; if (chip->in_sdin_init) { /* we don't know the ready bit assignment at the moment */ /* so we check any */ codec = chip->codec_isr_bits; } else { + if (codec > 2) + return -EIO; codec = chip->codec_bit[chip->ac97_sdin[codec]]; } -- 2.7.4

2 1

[alsa-devel] Remap pin 0x16 of ASUS ROG G751 to Headphone
by Håvard 09 Oct '18

09 Oct '18

Hi! On ASUS ROG G751 laptops the heaphone port is mapped to nothing, causing it to not function at all. I propose that we add a remap of pin 0x16 to Headphone in order to prevent this issue, and fix compability with the new mode=asus-g751. I don't know how to write patches, but this is what needs adding in sound/pci/hda/patch_realtek.c: [ALC668_FIXUP_ASUS_G751_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x16, 0x0421101f }, /* HP */ {} }, }, [ALC668_FIXUP_ASUS_G751] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { { 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x4000 }, {} }, .chained = true, .chain_id = ALC668_FIXUP_ASUS_G751_PINS }, And declaring the function earlier. -Håvard

2 2

[alsa-devel] [PATCH 0/4] Various cleanup + Mic Fix
by Connor McAdams 09 Oct '18

09 Oct '18

This patch set fixes the microphone inconsistency issue, which means the microphone now works all the time on all of the cards I've tested (ZxR, Z, AE-5), along with the input effects. It also includes changes suggested by Takashi Sakamoto, I believe I did what he asked properly, but if I messed it up I'm sure you guys will let me know. This should finish up most of the ca0132 work, with all inputs and outputs working on the desktop cards. Connor McAdams (4): ALSA: hda/ca0132 - Fix microphone inconsistency issues ALSA: hda/ca0132 - Clean up patch_ca0132() ALSA: hda/ca0132 - Add error checking in ca0132_build_controls() ALSA: hda/ca0132 - Fix input effect controls for desktop cards sound/pci/hda/patch_ca0132.c | 75 +++++++++++++++++++++++++++++++------------- 1 file changed, 54 insertions(+), 21 deletions(-) -- 2.7.4

3 6

[alsa-devel] [asoc:for-4.20 333/487] sound/pci//hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 09 Oct '18

09 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 24ae67c5825004bcbce90e7c89fed63f25d96260 commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/487] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (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 6bae5ea9498926440ffc883f3dbceb0adc65e492 # 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 'patch_ca0132': >> sound/pci//hda/patch_ca0132.c:7650:20: 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:7650:18: 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 +7650 sound/pci//hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH][RFC][alsa-utils 0/9] alsactl: monitor mode friendly for
by Takashi Sakamoto 09 Oct '18

09 Oct '18

Hi, A command of 'alsactl' has 'monitor' mode to print events emitted by observed control nodes. This mode is not friendly for plug-and-play devices. A bug is reported that the mode consumes 100% of CPU time when detecting disconnection of observed sound card[1]. A process of alsactl doesn't release ALSA control character device at disconnection. This is quite bad in a view of system. Furthermore, this mode cannot handle control nodes for newly-detected sound card. This patchset is my attempt to improve the mode to handle disconnection of observed sound card and new detection of sound card. A rough summary of this patchset: - maintains observed control nodes by link-list - uses epoll(7) instead of poll(2) to avoid following the list every time of poll loop, - control node for disconnected sound card is not observed anymore and dropped from the list. - the list is rescan and observation is updated when detecting new sound card. Linux-specific inotify(7) is used for this purpose. I'm happy if receiving test reports and indication of new bugs. [1] [alsa-devel] alsactl monitor spins at 100% cpu after unplugging USB headphones http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140580.… Takashi Sakamoto (9): alsactl: install signal handler alsactl: split event loop code to a function alsactl: add an iterator of registered instances of sound card alsactl: use epoll(7) instead of poll(2) alsactl: use link list to maintain source of events alsactl: use a list of source for event dispatcher instead of an array of source alsactl: obsolete array for maintenance of handlers alsactl: handle disconnection of sound card alsactl: handle detection of new sound card alsactl/monitor.c | 403 ++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 356 insertions(+), 47 deletions(-) -- 2.19.0

3 15

[alsa-devel] Applied "ASoC: max98988: make it selectable" to the asoc tree
by Mark Brown 08 Oct '18

08 Oct '18

The patch ASoC: max98988: make it selectable 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 24ae67c5825004bcbce90e7c89fed63f25d96260 Mon Sep 17 00:00:00 2001 From: Marco Felsch <m.felsch(a)pengutronix.de> Date: Fri, 5 Oct 2018 09:58:12 +0200 Subject: [PATCH] ASoC: max98988: make it selectable Currently the driver will build only if SND_SOC_ALL_CODECS is set. Adding a Kconfig menu description to build the driver standalone. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9989d35e0fc6..3c6bd6019b92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -640,7 +640,7 @@ config SND_SOC_LM49453 tristate config SND_SOC_MAX98088 - tristate + tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" config SND_SOC_MAX98090 tristate -- 2.19.0

1 0

[alsa-devel] [PATCH 00/11] Add ZxR support + bugfixes
by Connor McAdams 08 Oct '18

08 Oct '18

This patch series adds support for the Sound Blaster ZxR, as well as a few bug fixes. This should be the last ca0132 based Creative card that needed support to be added. Also, I did check to make sure each patch compiles properly this time, but you can check yourself just to be sure. :) Connor McAdams (11): ALSA: hda/ca0132 - Fix AE-5 control type ALSA: hda/ca0132 - Fix surround sound with output effects ALSA: hda/ca0132 - Add ZxR quirks + new quirk check function ALSA: hda/ca0132 - Add ZxR pincfg ALSA: hda/ca0132 - Add DBpro hda_codec_ops ALSA: hda/ca0132 - Add ZxR init commands ALSA: hda/ca0132 - Add ZxR DSP post-download commands ALSA: hda/ca0132 - Add ZxR input/output select commands ALSA: hda/ca0132 - Remove input select enum for ZxR ALSA: hda/ca0132 - Add ZxR 600 ohm gain control ALSA: hda/ca0132 - Add ZxR exit commands sound/pci/hda/patch_ca0132.c | 369 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 345 insertions(+), 24 deletions(-) -- 2.7.4

3 17

[alsa-devel] [PATCH 1/2] ASoC: Add device tree documentation file for wm8782 stereo DAC
by Daniel Mack 08 Oct '18

08 Oct '18

Add a device tree documentation file for the wm8782 stereo DAC to describe the regulator handles. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../devicetree/bindings/sound/wm8782.txt | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8782.txt diff --git a/Documentation/devicetree/bindings/sound/wm8782.txt b/Documentation/devicetree/bindings/sound/wm8782.txt new file mode 100644 index 000000000000..256cdec6ec4d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8782.txt @@ -0,0 +1,17 @@ +WM8782 stereo ADC + +This device does not have any control interface or reset pins. + +Required properties: + + - compatible : "wlf,wm8782" + - Vdda-supply : phandle to a regulator for the analog power supply (2.7V - 5.5V) + - Vdd-supply : phandle to a regulator for the digital power supply (2.7V - 3.6V) + +Example: + +wm8782: stereo-adc { + compatible = "wlf,wm8782"; + Vdda-supply = <&vdda_supply>; + Vdd-supply = <&vdd_supply>; +}; -- 2.17.1

3 6

Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Connor McAdams 08 Oct '18

08 Oct '18

I might not get around to writing the fix myself, but, here's the COEF's from within Windows. I've marked the ones missing inside the main init function. So, if anyone else wants to add these in, feel free. On Wed, Sep 12, 2018 at 1:16 PM, Håvard <hovardslill(a)gmail.com> wrote: > That was without running the program (sorry) > > This is output when running program: > > Before there was no "Current verb...." > They came when I opened up windows microphone settings. (something I didn't > do before) > > -Håvard > > Den ons. 12. sep. 2018 kl. 19:11 skrev Håvard <hovardslill(a)gmail.com>: >> >> I think this is right! >> >> I first booted up, then tested speakers (That I could not hear) and then >> tested microphone (Which worked perfectly) in windows settings. >> >> (My first email was dissaproved as it was too large) >> >> Den ons. 12. sep. 2018 kl. 19:05 skrev Håvard <hovardslill(a)gmail.com>: >>> >>> I think this is right! >>> >>> I first booted up, then tested speakers (That I could not hear) and then >>> tested microphone (Which worked perfectly) in windows settings. >>> >>> -Håvard >>> >>> Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams >>> <conmanx360(a)gmail.com>: >>>> >>>> Also, sorry to keep sending more replies, make sure the only trace you >>>> have on is vfio_region_write, and not the read one. The read spams up >>>> the console. >>>> >>>> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >>>> wrote: >>>> > Reason being, it may not have identified the CORB buffer properly or >>>> > something. That's very odd to have no data. Never seen that before. >>>> > >>>> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams >>>> > <conmanx360(a)gmail.com> wrote: >>>> >> Could you try again and make a copy of your terminal output? >>>> >> >>>> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> >>>> >> wrote: >>>> >>> Hi again. >>>> >>> Sorry for not noticing all files were empty, did it not run for long >>>> >>> enough? >>>> >>> >>>> >>> -Håvard >>>> >>> >>>> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard >>>> >>> <hovardslill(a)gmail.com>: >>>> >>>> >>>> >>>> Sorry my bad again it was working perfectly, just missed a step. >>>> >>>> >>>> >>>> I'll add the dumps as an attatchment. I had both external headset >>>> >>>> and >>>> >>>> microphone plugged in and let the VM run in the background for >>>> >>>> ~30min while >>>> >>>> doing other stuff. I did not play music or test my microphone in >>>> >>>> the VM >>>> >>>> either, If i did anything wrong, please tell me! >>>> >>>> >>>> >>>> It stopped at 0x104f4 >>>> >>>> >>>> >>>> -Håvard >>>> >>>> >>>> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >>>> >>>> <conmanx360(a)gmail.com>: >>>> >>>>> >>>> >>>>> Hm.... that's odd. They should show up in the folder you ran the >>>> >>>>> command from. Does your console show any "DumpMem entered..." or >>>> >>>>> something like that? You may have a permissions error. I've had >>>> >>>>> some >>>> >>>>> people report that as an issue before. >>>> >>>>> >>>> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >>>> >>>>> wrote: >>>> >>>>> > Hi! >>>> >>>>> > >>>> >>>>> > Took a while, but I think I got it working. Did not see any >>>> >>>>> > "frame[xx]" >>>> >>>>> > files though. What dumps do you want? >>>> >>>>> > >>>> >>>>> > -Håvard >>>> >>>>> > >>>> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard >>>> >>>>> > <hovardslill(a)gmail.com>: >>>> >>>>> >> >>>> >>>>> >> Sorry about that, I forgot to enable a kernel config... >>>> >>>>> >> >>>> >>>>> >> -Håvard >>>> >>>>> >> >>>> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >>>> >>>>> >> <conmanx360(a)gmail.com>: >>>> >>>>> >>> >>>> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >>>> >>>>> >>> controls. >>>> >>>>> >>> You can check what module is loaded by doing lspci -v . It >>>> >>>>> >>> should say >>>> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the >>>> >>>>> >>> vfio-bind >>>> >>>>> >>> script >>>> >>>>> >>> before trying it? >>>> >>>>> >>> >>>> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard >>>> >>>>> >>> <hovardslill(a)gmail.com> >>>> >>>>> >>> wrote: >>>> >>>>> >>> > Thank you for the offer! >>>> >>>>> >>> > >>>> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >>>> >>>>> >>> > seem to >>>> >>>>> >>> > bind my >>>> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >>>> >>>>> >>> > to >>>> >>>>> >>> > function as >>>> >>>>> >>> > normal when I (think I) have bound it to a stud. >>>> >>>>> >>> > >>>> >>>>> >>> > -Håvard >>>> >>>>> >>> > >>>> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >>>> >>>>> >>> > <conmanx360(a)gmail.com>: >>>> >>>>> >>> >> >>>> >>>>> >>> >> One thing you could try, is using the program I used to >>>> >>>>> >>> >> reverse >>>> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. >>>> >>>>> >>> >> If your >>>> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >>>> >>>>> >>> >> you >>>> >>>>> >>> >> could >>>> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >>>> >>>>> >>> >> capture >>>> >>>>> >>> >> the commands. I'd be willing to look through the dumps to >>>> >>>>> >>> >> see if >>>> >>>>> >>> >> there >>>> >>>>> >>> >> are any special verbs or anything. >>>> >>>>> >>> >> >>>> >>>>> >>> >> You can find the program here: >>>> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >>>> >>>>> >>> >> >>>> >>>>> >>> >> Let me know. >>>> >>>>> >>> >> >>>> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard >>>> >>>>> >>> >> <hovardslill(a)gmail.com> >>>> >>>>> >>> >> wrote: >>>> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response >>>> >>>>> >>> >> > before I >>>> >>>>> >>> >> > read >>>> >>>>> >>> >> > your >>>> >>>>> >>> >> > last one. >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >>>> >>>>> >>> >> > headset >>>> >>>>> >>> >> > multijacks. >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the >>>> >>>>> >>> >> > tip! >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > -Håvard >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >>>> >>>>> >>> >> > <hovardslill(a)gmail.com>: >>>> >>>>> >>> >> > >>>> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> Do you know where I can ask around for more help on the >>>> >>>>> >>> >> >> issue? >>>> >>>>> >>> >> >> I >>>> >>>>> >>> >> >> don't >>>> >>>>> >>> >> >> want to give up yet. >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> -Håvard >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >>>> >>>>> >>> >> >> <tiwai(a)suse.de>: >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >>>> >>>>> >>> >> >>> Håvard wrote: >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >>>> >>>>> >>> >> >>> > automatically >>>> >>>>> >>> >> >>> > changes >>>> >>>>> >>> >> >>> > to >>>> >>>>> >>> >> >>> it >>>> >>>>> >>> >> >>> > in pavucontrol. >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> Then it likely requires some additional initialization >>>> >>>>> >>> >> >>> outside >>>> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >>>> >>>>> >>> >> >>> vendor-specific. >>>> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >>>> >>>>> >>> >> >>> Windows >>>> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> Takashi >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >>>> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >>>> >>>>> >>> >> >>> > > Håvard wrote: >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Thank you for replying! >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >>>> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic >>>> >>>>> >>> >> >>> > > > Playback >>>> >>>>> >>> >> >>> > > > Volume" >>>> >>>>> >>> >> >>> > > > in >>>> >>>>> >>> >> >>> > > hdaanalyzer: >>>> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >>>> >>>>> >>> >> >>> > > > made white noise come from the headset. However >>>> >>>>> >>> >> >>> > > > it did >>>> >>>>> >>> >> >>> > > > not >>>> >>>>> >>> >> >>> > > > change or >>>> >>>>> >>> >> >>> > > react >>>> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >>>> >>>>> >>> >> >>> > > > physically. >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >>>> >>>>> >>> >> >>> > > > either >>>> >>>>> >>> >> >>> > > > alsamixer >>>> >>>>> >>> >> >>> or >>>> >>>>> >>> >> >>> > > > hdaanalyzer. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > It's a mixer mute switch. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > > The microphone works perfectly fine under >>>> >>>>> >>> >> >>> > > > Windows, so I >>>> >>>>> >>> >> >>> > > > don't >>>> >>>>> >>> >> >>> > > > think >>>> >>>>> >>> >> >>> it is >>>> >>>>> >>> >> >>> > > > the mic pin. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >>>> >>>>> >>> >> >>> > > wrong >>>> >>>>> >>> >> >>> > > pin, >>>> >>>>> >>> >> >>> > > too. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > Takashi >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > -Håvard >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi >>>> >>>>> >>> >> >>> > > > Iwai >>>> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >>>> >>>>> >>> >> >>> >: >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >>>> >>>>> >>> >> >>> > > > > Håvard wrote: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Additional relevant info: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three >>>> >>>>> >>> >> >>> > > > > > years ago >>>> >>>>> >>> >> >>> > > > > > on >>>> >>>>> >>> >> >>> > > > > > Sun >>>> >>>>> >>> >> >>> > > > > > Jun >>>> >>>>> >>> >> >>> > > 17:15:54 >>>> >>>>> >>> >> >>> > > > > CEST >>>> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >>>> >>>>> >>> >> >>> > > > > > but did >>>> >>>>> >>> >> >>> > > > > > not >>>> >>>>> >>> >> >>> > > > > > touch >>>> >>>>> >>> >> >>> on the >>>> >>>>> >>> >> >>> > > > > > external microphone problem: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > alsa-info.sh: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >>>> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help >>>> >>>>> >>> >> >>> > > > > > the >>>> >>>>> >>> >> >>> > > > > > issue. >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > There are several of similar bug reports >>>> >>>>> >>> >> >>> > > > > > around the >>>> >>>>> >>> >> >>> > > > > > web >>>> >>>>> >>> >> >>> experiencing >>>> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >>>> >>>>> >>> >> >>> > > > > > seems to >>>> >>>>> >>> >> >>> > > > > > happen >>>> >>>>> >>> >> >>> > > > > > with >>>> >>>>> >>> >> >>> all >>>> >>>>> >>> >> >>> > > > > laptops >>>> >>>>> >>> >> >>> > > > > > under the G751 name. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >>>> >>>>> >>> >> >>> > > > > and >>>> >>>>> >>> >> >>> > > > > unmute/adjust >>>> >>>>> >>> >> >>> "Mic >>>> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >>>> >>>>> >>> >> >>> > > > > you >>>> >>>>> >>> >> >>> > > > > hear >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> > > > > input >>>> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from >>>> >>>>> >>> >> >>> > > > > NID 0x18 >>>> >>>>> >>> >> >>> > > > > to >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> mixer >>>> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >>>> >>>>> >>> >> >>> > > > > outputs. >>>> >>>>> >>> >> >>> > > > > So >>>> >>>>> >>> >> >>> > > > > this >>>> >>>>> >>> >> >>> > > > > can >>>> >>>>> >>> >> >>> be >>>> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> > > > > input >>>> >>>>> >>> >> >>> > > > > from >>>> >>>>> >>> >> >>> the ext >>>> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >>>> >>>>> >>> >> >>> > > > > something >>>> >>>>> >>> >> >>> > > > > outside >>>> >>>>> >>> >> >>> > > > > HD-audio codec. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > Takashi >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> > _______________________________________________ >>>> >>>>> >>> >> > Alsa-devel mailing list >>>> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 26

[alsa-devel] [PATCH 1/2] ALSA: hda - Add quirk for ASUS G751 laptop
by Takashi Iwai 07 Oct '18

07 Oct '18

ASUS G751 requires the extra COEF initialization to make it microphone working properly. Reported-and-tested-by: Håvard <hovardslill(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69283633ae91..389e8e996589 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7738,6 +7738,7 @@ enum { ALC662_FIXUP_ASUS_Nx50, ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE, ALC668_FIXUP_ASUS_Nx51, + ALC668_FIXUP_ASUS_G751, ALC891_FIXUP_HEADSET_MODE, ALC891_FIXUP_DELL_MIC_NO_PRESENCE, ALC662_FIXUP_ACER_VERITON, @@ -8007,6 +8008,14 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE, }, + [ALC668_FIXUP_ASUS_G751] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4000 }, + {} + }, + }, [ALC891_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -8080,6 +8089,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), + SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), -- 2.18.0

1 1

Re: [alsa-devel] WARNING in port_delete
by syzbot 06 Oct '18

06 Oct '18

syzbot has found a reproducer for the following crash on: HEAD commit: 091a1eaa0e30 Merge branch 'akpm' git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=16623e59400000 kernel config: https://syzkaller.appspot.com/x/.config?x=c0af03fe452b65fb dashboard link: https://syzkaller.appspot.com/bug?extid=e4c8abb920efa77bace9 compiler: gcc (GCC) 8.0.1 20180413 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=102e2c91400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+e4c8abb920efa77bace9(a)syzkaller.appspotmail.com WARNING: CPU: 1 PID: 23693 at sound/core/seq/seq_ports.c:275 port_delete+0x177/0x1b0 sound/core/seq/seq_ports.c:275 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 23693 Comm: syz-executor0 Not tainted 4.19.0-rc6+ #47 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c4/0x2b4 lib/dump_stack.c:113 panic+0x238/0x4e7 kernel/panic.c:184 __warn.cold.8+0x163/0x1ba kernel/panic.c:536 report_bug+0x254/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1fc/0x4d0 arch/x86/kernel/traps.c:296 kobject: 'loop1' (00000000baf940ef): kobject_uevent_env kobject: 'loop1' (00000000baf940ef): fill_kobj_path: path = '/devices/virtual/block/loop1' do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:993 RIP: 0010:port_delete+0x177/0x1b0 sound/core/seq/seq_ports.c:275 Code: e7 99 fb 45 85 e4 75 14 e8 66 e6 99 fb 48 89 df e8 ee 29 dd fb 31 c0 5b 41 5c 5d c3 e8 52 e6 99 fb 0f 0b eb e3 e8 49 e6 99 fb <0f> 0b eb 9b e8 a0 55 dd fb eb bd e8 99 55 dd fb e9 77 ff ff ff e8 RSP: 0018:ffff8801a7477890 EFLAGS: 00010293 RAX: ffff8801a84e05c0 RBX: ffff8801cf0be4c0 RCX: ffffffff85e4eee1 RDX: 0000000000000000 RSI: ffffffff85e4ef47 RDI: 0000000000000005 RBP: ffff8801a74778a0 R08: ffff8801a84e05c0 R09: 00000000deede79a R10: 00000000da8ba726 R11: 0000000000000000 R12: 0000000000000001 R13: ffff8801cf0be510 R14: 1ffff10034e8ef20 R15: ffff8801cb447b00 snd_seq_delete_port+0x3cb/0x4a0 sound/core/seq/seq_ports.c:303 snd_seq_ioctl_delete_port+0xba/0x190 sound/core/seq/seq_clientmgr.c:1324 snd_seq_kernel_client_ctl+0x15a/0x190 sound/core/seq/seq_clientmgr.c:2353 snd_seq_event_port_detach+0xfc/0x170 sound/core/seq/seq_ports.c:705 delete_port+0x7d/0xc0 sound/core/seq/oss/seq_oss_init.c:354 snd_seq_oss_release+0xf2/0x130 sound/core/seq/oss/seq_oss_init.c:433 odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:153 __fput+0x385/0xa30 fs/file_table.c:278 ____fput+0x15/0x20 fs/file_table.c:309 task_work_run+0x1e8/0x2a0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:193 [inline] exit_to_usermode_loop+0x318/0x380 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x6be/0x820 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457579 Code: 1d b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 eb b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f85fa204c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000001 RCX: 0000000000457579 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000006 RBP: 000000000072bfa0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f85fa2056d4 R13: 00000000004ef912 R14: 00000000004cc460 R15: 00000000ffffffff Kernel Offset: disabled Rebooting in 86400 seconds..

1 0

[alsa-devel] [PATCH] ALSA: hda/sigmatel - Disable automute for Elo VuPoint
by Michael Pobega 05 Oct '18

05 Oct '18

The Elo VuPoint 15MX has two headphone jacks of which neither work by default. Disabling automute allows ALSA to work normally with the speakers & left headphone jack. Future pin configuration changes may be required in the future to get the right headphone jack working in tandem. Signed-off-by: Michael Pobega <mpobega(a)neverware.com> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 046705b4691a..d8168aa2cef3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -77,6 +77,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_ELO_VUPOINT_15MX, STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, STAC_92HD73XX_ASUS_MOBO, @@ -1879,6 +1880,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec, codec->no_jack_detect = 1; } + +static void stac92hd73xx_disable_automute(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.suppress_auto_mute = 1; +} + static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_REF] = { .type = HDA_FIXUP_FUNC, @@ -1904,6 +1917,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_alienware_m17x, }, + [STAC_ELO_VUPOINT_15MX] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd73xx_disable_automute, + }, [STAC_92HD73XX_INTEL] = { .type = HDA_FIXUP_PINS, .v.pins = intel_dg45id_pin_configs, @@ -1942,6 +1959,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = { { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" }, { .id = STAC_DELL_EQ, .name = "dell-eq" }, { .id = STAC_ALIENWARE_M17X, .name = "alienware" }, + { .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" }, { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" }, {} }; @@ -1991,6 +2009,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(0x1059, 0x1011, + "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927, "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, -- 2.17.1

2 2

[alsa-devel] Applied "ASoC: adau1761: Use the standard fall-through annotation" to the asoc tree
by Mark Brown 05 Oct '18

05 Oct '18

The patch ASoC: adau1761: Use the standard fall-through annotation 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 7e29317928bd79a03a9c35816afa709988b5d31b Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 4 Oct 2018 20:30:02 +0200 Subject: [PATCH] ASoC: adau1761: Use the standard fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, replace with the standard "fall through" annotation at the right place. It has to be put right before the next label. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau1761.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e981653..bef3e9e74c26 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 0/5] ASoC: Fix fall-through annotations
by Takashi Iwai 05 Oct '18

05 Oct '18

Hi, this is a patch set to add / fix the "fall through" annotations to be compliant with gcc's -Wimplicit-fallthrough option. Takashi === Takashi Iwai (5): ASoC: adau1761: Use the standard fall-through annotation ASoC: pcm186x: Use the standard fall-through annotation ASoC: rt274: Add fall-through annotations ASoC: intel: skylake: Add fall-through annotation ASoC: topology: Use the standard fall-through annotations sound/soc/codecs/adau1761.c | 3 ++- sound/soc/codecs/pcm186x.c | 3 ++- sound/soc/codecs/rt274.c | 2 ++ sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/soc-topology.c | 4 ++-- 5 files changed, 9 insertions(+), 4 deletions(-) -- 2.18.0

3 11

[alsa-devel] Applied "ASoC: intel: skylake: Add fall-through annotation" to the asoc tree
by Mark Brown 05 Oct '18

05 Oct '18

The patch ASoC: intel: skylake: Add fall-through annotation 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 e4bfd61571f5db5c69a7a49de401543cc7d6c87c Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 4 Oct 2018 20:30:05 +0200 Subject: [PATCH] ASoC: intel: skylake: Add fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, add the "fall through" annotation in Intel SST skylake driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 00b7a91b18c9..557f80c0bfe5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -495,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: dt-bindings: add dt bindings for adau1977 audio codec
by Alexandru Ardelean 05 Oct '18

05 Oct '18

Add device-tree bindings documentation file for ADAU1977 audio codec. This describes device-tree fields that are already supported by the driver. The driver supports both I2C AND and SPI, and this doc covers both aspects of the DT configuration. Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- .../bindings/sound/adi,adau1977.txt | 54 +++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt new file mode 100644 index 000000000000..e79aeef73f28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -0,0 +1,54 @@ +Analog Devices ADAU1977/ADAU1978/ADAU1979 + +Datasheets: +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf + +This driver supports both the I2C and SPI bus. + +Required properties: + - compatible: Should contain one of the following: + "adi,adau1977" + "adi,adau1978" + "adi,adau1979" + + - AVDD-supply: analog power supply for the device, please consult + Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + - reset-gpio: the reset pin for the chip, for more details consult + Documentation/devicetree/bindings/gpio/gpio.txt + + - DVDD-supply: supply voltage for the digital core, please consult + Documentation/devicetree/bindings/regulator/regulator.txt + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Examples: + + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + spi-max-frequency = <600000>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + }; -- 2.17.1

2 1

[alsa-devel] [PATCH v5] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Adds several vendor specific mixer quirks for RME's Class Compliant USB devices. These provide extra status information from the device otherwise not available. These include AES/SPDIF rate and status information, current system sampling rate and measured frequency. This information is especially useful in cases where device's clock is slaved to external clock source. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/mixer_quirks.c | 381 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 381 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..85ae0ff2382a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +/* maximum number of items for AES and S/PDIF rates for above table */ +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; -- 2.17.1

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 05 Oct '18

05 Oct '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Changes since v1 (https://patchwork.kernel.org/cover/10585805/) - added support to snd_compr_set_runtime_buffer() - make use of generic copy function - few trivial cleanups suggested by Vinod Koul Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ include/sound/compress_driver.h | 19 + sound/core/compress_offload.c | 18 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 377 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 6 files changed, 443 insertions(+), 4 deletions(-) -- 2.19.0

2 9

[alsa-devel] [PATCH v4] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Adds several vendor specific mixer quirks for RME's Class Compliant USB devices. These provide extra status information from the device otherwise not available. These include AES/SPDIF rate and status information, current system sampling rate and measured frequency. This information is especially useful in cases where device's clock is slaved to external clock source. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/mixer_quirks.c | 380 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 380 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..6988dc21c31e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,379 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2278,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; -- 2.17.1

1 0

[alsa-devel] [PATCH v3 0/1] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Here are the requested updates. I have kept the locking separate from the helper, because in frequency calculation result depends on both values. I can move locking to the helper too if necessary, but I thought it would be logically cleaner this way...

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Nicolas Huaman 05 Oct '18

05 Oct '18

A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..849953e5775c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.19.0

3 3

[alsa-devel] [PATCH v2 0/1] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

This patch has been updated with following changes and should be now good for merging: - Replace 64-bit division with suitable function to make it available on all architectures - Use specific VID:PID pairs instead of just VID to avoid confusions Last item was changed after discussion with RME.

2 4

[alsa-devel] [PATCH 0/5] ALSA: Fix fall-through annotations
by Takashi Iwai 04 Oct '18

04 Oct '18

Hi, here is some trivial patches to add / fix the fall-through annotations. The patchset for ASoC will follow later. Takashi === Takashi Iwai (5): ALSA: memalloc: Add fall-through annotation ALSA: caiaq: Add fall-through annotation ALSA: seq: oss: Use the standard fall-through annotation ALSA: opti92xx-ad1848: Use the standard fall-through annotation ALSA: au88xx: Add fall-through annotations sound/core/memalloc.c | 1 + sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 6 ++++-- sound/pci/au88x0/au88x0_core.c | 6 ++++++ sound/usb/caiaq/device.c | 1 + 5 files changed, 13 insertions(+), 3 deletions(-) -- 2.18.0

1 5

[alsa-devel] [PATCH] ASoC: ts3a227e: fix a typo in documentation
by Tzung-Bi Shih 04 Oct '18

04 Oct '18

There is no "MACBIAS". It should be "MICBIAS". Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Documentation/devicetree/bindings/sound/ts3a227e.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt index 3ed8359144d3..21ab45bc7e8f 100644 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -14,7 +14,7 @@ Required properties: Optional properies: - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). - Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage + Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V Default value is "1" (2.2V). -- 2.19.0.605.g01d371f741-goog

2 2

[alsa-devel] [PATCH] ASoC: intel: skylake: Add missing break in skl_tplg_get_token()
by Takashi Iwai 04 Oct '18

04 Oct '18

skl_tplg_get_token() misses a break in the big switch() block for SKL_TKN_U8_CORE_ID entry. Spotted nicely by -Wimplicit-fallthrough compiler option. Fixes: 6277e83292a2 ("ASoC: Intel: Skylake: Parse vendor tokens to build module data") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/skylake/skl-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..f99c600f86e4 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2461,6 +2461,7 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U8_CORE_ID: mconfig->core_id = tkn_elem->value; + break; case SKL_TKN_U8_MOD_TYPE: mconfig->m_type = tkn_elem->value; -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6asm-dai: checking NULL vs IS_ERR()
by Dan Carpenter 04 Oct '18

04 Oct '18

The q6asm_audio_client_alloc() doesn't return NULL, it returns error pointers. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c3806d7037fc..a16c71c03058 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -318,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { pr_info("%s: Could not allocate memory\n", __func__); + ret = PTR_ERR(prtd->audio_client); kfree(prtd); - return -ENOMEM; + return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

2 1

[alsa-devel] [PATCH] ASoC: pxa-ssp: enable and disable extclk if given
by Daniel Mack 04 Oct '18

04 Oct '18

If a "extclk" clock is given, enable and disable it when appropriate. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/pxa-ssp.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a84e6..adcf8ba9d287 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } + if (priv->extclk) + clk_prepare_enable(priv->extclk); + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; @@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } + if (priv->extclk) + clk_disable_unprepare(priv->extclk); + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: add fault detect recovery property to DT bindings
by Daniel Mack 04 Oct '18

04 Oct '18

The driver already has support for setting the FDRB bit in the CONFA register through platform data, but there was no property to set it in the device-tree bindings. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- Documentation/devicetree/bindings/sound/st,sta32x.txt | 3 +++ sound/soc/codecs/sta32x.c | 2 ++ 2 files changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index 255de3ae5b2f..ff4a685a4303 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -39,6 +39,9 @@ Optional properties: - st,thermal-warning-recover: If present, thermal warning recovery is enabled. + - st,fault-detect-recovery: + If present, fault detect recovery is enabled. + - st,thermal-warning-adjustment: If present, thermal warning adjustment is enabled. diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index d5035f2f2b2b..22de1593443c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1038,6 +1038,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x) of_property_read_u8(np, "st,ch3-output-mapping", &pdata->ch3_output_mapping); + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; if (of_get_property(np, "st,thermal-warning-recovery", NULL)) pdata->thermal_warning_recovery = 1; if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) -- 2.17.1

2 1

[alsa-devel] [BUG] sound: pci: trident: a possible data race
by Jia-Ju Bai 04 Oct '18

04 Oct '18

CPU0: snd_trident_hw_free snd_trident_free_voice line 3870: spin_lock_irqsave() line 3881: voice->substream = NULL; [WRITE] CPU1: snd_trident_interrupt line 3798: snd_pcm_period_elapsed(voice->substream); [READ] As for voice->substream, the WRITE operation in CPU0 is performed with holding a spinlock, but the READ operation in CPU1 is performed without holding this spinlock, so there may exist a data race. Best wishes, Jia-Ju Bai

2 4

[alsa-devel] Problem and Patch for B&W PX Bluetooth Headphones
by Nicolas Herochao 04 Oct '18

04 Oct '18

Hello, I'm writing this mail because of a Crash when connecting the B&W PX headphones via usb. Half a year ago this used to work with Linux after setting the default sample rate of pulseaudio to 48000k. A Launchpad Bug seems to describe this exact issue (detailed description below): https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1773697. It pointed me to the right direction, and today I have successfully compiled and tested a small patch that fixes the errors while also setting the sample rate to 48000k, making these headphones plug&play on any newer firmware. They sound awesome via USB! I would like to submit this patch, but it would be my first contribution and it could potentially break the B&W PX on older firmware. I can't test it since I see no obvious ways to downgrade the firmware. The automatic detection of these headphones always worked except for the sample-rate and perhaps the worst result is that old firmware will show a non-functional microphone, like on Windows. So I decided to first present the patch on this mailing list and ask for feedback and help. If anyone wants to pick the patch up, I'd appreciate that as well. My questions: * Can a quirk somehow handle multiple firmware versions of a usb device? * If not: Will this quirk allow old firmware to work out of the box but show a non-functional microphone? (And is that okay?) Thanks for your time and any feedback you can provide! Sincerely, Nicolas PS: more detail on the cause of the issue: ------ A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active with the following dmesg errors: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) --------------- Finally, here is the patch (based on the current mainline kernel and generated according to all Guidelines I could find) which fixes the quirk by removing the microphone part: ---------------- From 748587283db85e219902abf4ac012864005943c8 Mon Sep 17 00:00:00 2001 From: Nicolas Huaman <nicolas(a)herochao.de> Date: Fri, 28 Sep 2018 18:05:15 +0200 Subject: [PATCH 1/1] SOUND: update quirk for B&W PX to remove microphone Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..849953e5775c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.19.0

2 1

[alsa-devel] [PATCH 0/5] ALSA: use managed-resource to maintain allocated buffers
by Takashi Sakamoto 04 Oct '18

04 Oct '18

Hi, Linux Device Driver subsystem has a feature of 'devres', which is resource allocator/deallocator synchronously to reference counter of kobject corresponding to device structure. This feature avoid memory leak from programming mistakes. This patchset replaces calls of allocator/deallocator with devres helper functions. There're two types of target device structure; firewire unit device and sound card device. All drivers in ALSA firewire stack use reference counting to serialize timing to release the type of two device, therefore no worries of dereferencing to released memory objects. This patchset is for v4.20 and based on my series of fix patches in 'sound-4.19-rc5' branch. But current 'for-next' branch doesn't include them. So this patchset is rebased to upstream 'master' branch (HEAD=067284726963). I'm OK to take them pending till the '-rc5' is merged. Takashi Sakamoto (5): ALSA: firewire: use managed-resource of fw unit device for private data ALSA: bebob: use managed-resource to maintain data specific to M-Audio FW-1814/ProjectMix I/O ALSA: fireworks: use managed-resource to maintain response buffer ALSA: oxfw: use managed-resource to maintain model-specific data ALSA: oxfw: use managed-resource to maintain cache of stream formats sound/firewire/bebob/bebob.c | 17 ++++++----------- sound/firewire/bebob/bebob_maudio.c | 5 +++-- sound/firewire/dice/dice.c | 8 +++----- sound/firewire/digi00x/digi00x.c | 9 ++++----- sound/firewire/fireface/ff.c | 10 +++------- sound/firewire/fireworks/fireworks.c | 15 +++++---------- sound/firewire/motu/motu.c | 10 ++++------ sound/firewire/oxfw/oxfw-scs1x.c | 5 +++-- sound/firewire/oxfw/oxfw-spkr.c | 5 +++-- sound/firewire/oxfw/oxfw-stream.c | 13 ++++++++----- sound/firewire/oxfw/oxfw.c | 28 ++++------------------------ sound/firewire/tascam/tascam.c | 10 +++------- 12 files changed, 49 insertions(+), 86 deletions(-) -- 2.17.1

2 6

[alsa-devel] [PATCH] ALSA: hda/realtek - Cannot adjust speaker's volume on Dell XPS 27 7760
by Kai-Heng Feng 04 Oct '18

04 Oct '18

The issue is the same as commit dd9aa335c880 ("ALSA: hda/realtek - Can't adjust speaker's volume on a Dell AIO"), the output requires to connect to a node with Amp-out capability. Applying the same fixup ALC298_FIXUP_SPK_VOLUME can fix the issue. BugLink: https://bugs.launchpad.net/bugs/1775068 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f3c8e888c2a..69283633ae91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6409,6 +6409,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), -- 2.17.1

2 1

[alsa-devel] [PATCH 0/7] Add devres support to card resources
by Takashi Iwai 04 Oct '18

04 Oct '18

Hi, here is the patchset containing the changes to support a device-managed resource management for ALSA card and memory pages. The actual code changes are identical with the previous RFC series, but a few documentation updates are included in this official set. Takashi === Takashi Iwai (7): ALSA: core: Add device-managed page allocator helper ALSA: core: Add managed card creation ALSA: intel8x0: Allocate resources with device-managed APIs ALSA: atiixp: Allocate resources with device-managed APIs ALSA: hda: Allocate resources with device-managed APIs ALSA: doc: Brush up the old writing-an-alsa-driver ALSA: doc: Add device-managed resource section .../kernel-api/writing-an-alsa-driver.rst | 340 ++++++++++-------- include/sound/core.h | 5 + include/sound/memalloc.h | 4 + sound/core/init.c | 95 ++++- sound/core/memalloc.c | 88 ++++- sound/pci/atiixp.c | 104 ++---- sound/pci/atiixp_modem.c | 104 ++---- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 43 +-- sound/pci/intel8x0.c | 143 +++----- sound/pci/intel8x0m.c | 143 +++----- 11 files changed, 514 insertions(+), 556 deletions(-) -- 2.18.0

2 19

[alsa-devel] [PATCH linux-next v2 0/9] ASoC: rsnd: support to use different BUSIF for GEN3
by jiada_wang＠mentor.com 04 Oct '18

04 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Rcar audio DMA/SSIU already support to use different BUSIF to transfer data, but BUSIF is fixed at 0. This patchset adds busif property to dai stream, GEN3 series can automatically select BUSIF to transfer data, based on channel and SSI mode (currently only BUSIF0 is selected). Device tree files are updated to use extended audio dmac register, and add dma request number for busif0 ~ busif7 for GEN3. --- v2: - Instead of configure BUSIF by kctl interface "SSI Out/In BUSIF", use rsnd_ssi_select_busif() to automatically select BUSIF. - Keep 'rxu/txu' dma request, to keep compatibility with GEN2 series - Re-order patch-set to satisfy git bisect v1: initial version Jiada Wang (9): arm64: r8a7795: add dma request number for busif0 ~ busif7 arm64: r8a7796: add dma request number for busif0 ~ busif7 arm64: r8a774a1: add dma request number for busif0 ~ busif7 arm64: r8a7795: change to use extended audio dmac register arm64: r8a7796: change to use extended audio dmac register arm64: r8a774a1: change to use extended audio dmac register arm64: r8a77965: change to use extended audio dmac register ASoC: rsnd: ssi: Request dedicated dma channels for busif0 to 7 ASoC: rsnd: add busif property to dai stream arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 166 +++++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 166 +++++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 166 +++++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- sound/soc/sh/rcar/core.c | 3 + sound/soc/sh/rcar/dma.c | 31 ++++ sound/soc/sh/rcar/rsnd.h | 3 + sound/soc/sh/rcar/ssi.c | 58 +++++++- 8 files changed, 525 insertions(+), 70 deletions(-) -- 2.17.0

3 6

[alsa-devel] [PATCH linux-next v2 8/9] ASoC: rsnd: ssi: Request dedicated dma channels for busif0 to 7
by jiada_wang＠mentor.com 04 Oct '18

04 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Currently ssi driver only request dma channel for SSI_0, which is used to transfer data to/from busif0. But in GEN3 busif1 to busif7 also maybe used, dedicated dma channels are requested for data transfer through these busif. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/ssi.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..992aeac09e76 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -76,6 +76,8 @@ #define SSI_NAME "ssi" +#define SSI_DMA_NAME_SIZE 10 + struct rsnd_ssi { struct rsnd_mod mod; @@ -938,12 +940,28 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); int is_play = rsnd_io_is_play(io); - char *name; + char name[SSI_DMA_NAME_SIZE]; - if (rsnd_ssi_use_busif(io)) - name = is_play ? "rxu" : "txu"; - else - name = is_play ? "rx" : "tx"; + if (rsnd_ssi_use_busif(io)) { + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rxu"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "txu"); + + if (rsnd_is_gen3(priv)) { + /* + * Gen3 supports to use all available BUSIF + */ + int busif = rsnd_ssi_get_busif(io); + + snprintf(name, SSI_DMA_NAME_SIZE, "%s%d", name, busif); + } + } else { + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rx"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "tx"); + } return rsnd_dma_request_channel(rsnd_ssi_of_node(priv), mod, name); -- 2.17.0

3 2

[alsa-devel] [PATCH] ALSA: intel8x0: Fix fall-through annotations
by Gustavo A. R. Silva 03 Oct '18

03 Oct '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/intel8x0.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9517f9b..ffddcdf 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -825,7 +825,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -833,7 +833,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -867,7 +867,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -884,7 +884,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ -- 2.7.4

2 5

[alsa-devel] [PATCH linux-next v2 7/9] arm64: r8a77965: change to use extended audio dmac register
by jiada_wang＠mentor.com 03 Oct '18

03 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index fd00010a5ed7..6c0d5053a81c 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1405,7 +1405,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ /* placeholder */ rcar_sound,dvc { -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next v2 6/9] arm64: r8a774a1: change to use extended audio dmac register
by jiada_wang＠mentor.com 03 Oct '18

03 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi index ad2c9c0148ae..c04ca03f2987 100644 --- a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi @@ -1120,7 +1120,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next v2 5/9] arm64: r8a7796: change to use extended audio dmac register
by jiada_wang＠mentor.com 03 Oct '18

03 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index 53b361c5b16d..4526f62ee1f0 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -1670,7 +1670,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] snd_pcm_poll_descriptors() error return
by Alan Horstmann 02 Oct '18

02 Oct '18

Hi, A user of the Portaudio library is reporting that from Alsa version 1.1.4 the function snd_pcm_poll_descriptors() can return -32 (Broken Pipe), and this fails an assert within Portaudio. Earlier 1.1.3 does not seem to show this issue. For more details, see https://app.assembla.com/spaces/portaudio/support/tickets/268. The Alsa docs don't mention any possible negative error code from this function - has the behaviour changed, or is it likely something else is triggering the difference? Regards Alan

3 3

[alsa-devel] Thread spinning in kernel snd_pcm_link()/snd_pcm_unlink()
by Rob Duncan 02 Oct '18

02 Oct '18

I'm trying to address a bug where we end up with a thread spinning and consuming an entire cpu. The issue seems to be this code in sound/core/pcm_native.c: /* Writer in rwsem may block readers even during its waiting in queue, * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to * spin until it gets the lock. */ static inline void down_write_nonblock(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) cond_resched(); } The original commit for this is 67ec1072b053c15564e6090ab30127895dc77a89 What we're suspecting is that a normal thread (SCHED_OTHER) has a reader lock and a real-time thread using SCHED_RR or SCHED_FIFO is trying to take the writer lock. If both threads are pinned to the same CPU for some reason then the reader thread will never get scheduled (because the real-time writer thread is still runnable), and we will never make progress. Does this sound right? What can we do to fix this? Thanks, Rob.

2 2

[alsa-devel] [PATCH alsa-lib] ioplug: Fix comment in struct snd_pcm_ioplug
by Andre Guedes 02 Oct '18

02 Oct '18

This patch fixes the comment about 'pcm' field in 'struct snd_pcm_ioplug' which wrongly refers to snd_pcm_extplug_create(). Signed-off-by: Andre Guedes <andre.guedes(a)intel.com> --- include/pcm_ioplug.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm_ioplug.h b/include/pcm_ioplug.h index b16fc8b8..81ac8613 100644 --- a/include/pcm_ioplug.h +++ b/include/pcm_ioplug.h @@ -105,7 +105,7 @@ struct snd_pcm_ioplug { */ void *private_data; /** - * PCM handle filled by #snd_pcm_extplug_create() + * PCM handle filled by #snd_pcm_ioplug_create() */ snd_pcm_t *pcm; -- 2.14.4

3 2

[alsa-devel] [PATCH - Multiple plugins 0/4] Fix leaks on multiple plugins
by Andre Guedes 02 Oct '18

02 Oct '18

Hi all, While I was working on AAF plugin RFC v2 series [1], I spotted several leaks in some plugins when handling *_hw_constraint() error cases. This series fixes these issues. Regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… Andre Guedes (4): a52: Fix leaks when a52_set_hw_constraint() fails jack: Fix leaks when jack_set_hw_constraint() fails oss: Fix leaks when oss_hw_constraint() fails usb_stream: Fix leaks when us_set_hw_constraint() fails a52/pcm_a52.c | 2 +- jack/pcm_jack.c | 1 + oss/pcm_oss.c | 2 +- usb_stream/pcm_usb_stream.c | 1 + 4 files changed, 4 insertions(+), 2 deletions(-) -- 2.14.4

3 6

[alsa-devel] [PATCH] ALSA: sb8: Fix fall-through annotations
by Gustavo A. R. Silva 02 Oct '18

02 Oct '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/isa/sb/sb8_main.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 4817977..8288fae 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -130,13 +130,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -287,7 +287,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -387,7 +387,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fallthru */ + /* fall through */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -397,7 +397,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fallthru */ + /* fall through */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL
by Takashi Sakamoto 02 Oct '18

02 Oct '18

In blocking mode of IEC 61883-1/6, when one isochronous packet includes data for events, the data is for the same number of events as SYT_INTERVAL decided according to sampling transmission frequency (SFC). IEC 61883-1/6 engine of ALSA firewire stack applies constraints of period and buffer size of PCM intermediate buffer of PCM substream. At present, this constraint is designed to round the size up/down to 32 frames. This value comes from the least common multiple (LCM) of SYT_INTERVAL. Although this looks to work well, in lower sampling rate, applications are not allowed to set size of period quite near period time constraint (at present 5 msec per period). This commit adds PCM rules for period/buffer size and rate to obsoletes the constraints based on LCM. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 78 +++++++++++++++++++++++++++++++---- 1 file changed, 71 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index cb9acfe60f6a..fcd965f1d69e 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = { }; EXPORT_SYMBOL(amdtp_rate_table); +static int apply_constraint_to_size(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *s = hw_param_interval(params, rule->var); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval t = { + .min = s->min, .max = s->max, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int rate = amdtp_rate_table[i]; + unsigned int step = amdtp_syt_intervals[i]; + + if (!snd_interval_test(r, rate)) + continue; + + t.min = roundup(t.min, step); + t.max = rounddown(t.max, step); + } + + if (snd_interval_checkempty(&t)) + return -EINVAL; + + return snd_interval_refine(s, &t); +} + +static int apply_constraint_to_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int step = amdtp_syt_intervals[i]; + unsigned int rate = amdtp_rate_table[i]; + + if (s->min % step || s->max % step) + continue; + + t.min = min(t.min, rate); + t.max = max(t.max, rate); + } + + return snd_interval_refine(r, &t); +} + /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's * preferrable to align period/buffer sizes to current SYT_INTERVAL. - * - * TODO: These constraints can be improved with proper rules. - * Currently apply LCM of SYT_INTERVALs. */ - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) goto end; - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); end: return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH][alsa-lib] timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 6041cab2..4d35c84d 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -750,7 +750,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ed0a120d4f08..404d4b9ffe76 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -752,7 +752,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix mismatch for register mask and value in ext controller.
by Keyon Jie 02 Oct '18

02 Oct '18

E.g. for snd_hdac_ext_bus_link_power_up(), we should set mask to be AZX_MLCTL_SPA(it was 0), and AZX_MLCTL_SPA as value to power up it, here correct it and several similar mismatches. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 5bc4a1d587d4..60cb00fd0c69 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); @@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, AZX_PPCTL_PIE); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); @@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) */ int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, - AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; -- 2.17.1

2 1

[alsa-devel] Re-reading alsa-lib configuration when opening sound device
by Pavel Hofman 02 Oct '18

02 Oct '18

Hi, I am playing with virtual balanced inputs https://www.diyaudio.com/forums/pc-based/326747-virtual-balanced-regular-so… . For proper functionality it needs a regular recalibration of the ttable multiplication coefficients of route plugin in .asoundrc. The audio stack is alsa -> wine -> Arta measurement software. When I stop measuring in arta (without exiting the software), the soundcard gets closed - confirmed in /proc/asound/.../hw_params. Now I change the ttable coeffs. After re-opening the device (continuing with measurement in the running arta instance) the changes in .asoundrc are ignored. I have to restart the arta application for the changes to take effect. When I run e.g. aplay inbetween, it accepts the changes right away. Upon continuing in arta, the original ttable coeffs are still loaded for this instance. I understand it is the way wine handles the device somewhere in https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.c . Please when is the alsa-lib config loaded by the library/application? Is there any way to force a config reload from outside? Would e.g. calling snd_config_update before snd_pcm_open on line https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.… help? If so, what overhead does it impose? Perhaps I could ask the wine guys to consider such change if it worked and required minimum overhead. I very much appreciate any help. Best regards, Pavel.

2 4

[alsa-devel] alsactl monitor spins at 100% cpu after unplugging USB headphones
by Thomas Gläßle 02 Oct '18

02 Oct '18

Hi, on both my desktop and laptop (archlinux), `alsactl monitor` beginns spinning at 100% cpu when I unplug my USB headset. This happens every time and can be reproduced as follows: - plug in USB headphones - start `alsactl monitor` - unplug headphones It happens only if the headphones are already plugged in when starting alsactl. The increased activity does not lead to spamming stdout or similar, alsactl continues to perform seemingly correct, i.e. reports when I change volume levels etc, but nothing apart from the ordinary. I'm sorry if this is not the right place to report this issue or if the problem is already, I couldn't find anything about it. Below is the output from alsa-info.sh: with headphones: http://www.alsa-project.org/db/?f=0dfc632e3adafb8bc78a182e0a225bcb13bea9fa after unplugging: http://www.alsa-project.org/db/?f=41dbf0735fdd7cd2416a6b4a602505715d25b44f Please let me know if you need further info. Best regards, Thomas

2 2

[alsa-devel] [PATCH linux-next 10/10] ASoC: rsnd: Add kctrl to configure dai's busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> ssi0, ssi1, ssi2, ssi3, ssi4 and ssi9 have busif0 to busif7, currently only busif0 is used by all ssi. But tdm split and tdm ex-split mode, which require busif1 to busif7. This patch adds kctrl in ssi for dai-link, to make it possible to configure the busif used by each dai-link. Also adds rsnd_dma_addr_update() interface to make it possible to update dma data address when there is change in condition of dai connection. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/dma.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 31 ++++++++++++++++++++++++++++++- 3 files changed, 64 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 6d1947515dc8..8c284fd73fc1 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -792,6 +792,38 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, + struct rsnd_mod *dma_mod) +{ + struct rsnd_mod *mod_from = NULL; + struct rsnd_mod *mod_to = NULL; + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int is_play = rsnd_io_is_play(io); + struct rsnd_dma *dma = rsnd_mod_to_dma(dma_mod); + + if (!dmac) + return; + + rsnd_dma_of_path(mod, io, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + + if (mod_from && mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + + dmapp->chcr = rsnd_dmapp_get_chcr(io, mod_from, mod_to) | + PDMACHCR_DE; + } + + dev_dbg(dev, "%s[%d] %pad -> %pad\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + &dma->src_addr, &dma->dst_addr); +} + int rsnd_dma_probe(struct rsnd_priv *priv) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 59b6d89d8edc..6c16d8403f7a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -233,6 +233,8 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); */ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, struct rsnd_mod **dma_mod); +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, struct rsnd_mod *dma_mod); int rsnd_dma_probe(struct rsnd_priv *priv); struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4ac4b5b75ae2..b469db238697 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -160,6 +160,15 @@ int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) return io->busif->val; } +static void rsnd_ssi_set_busif(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + if (!rsnd_ssi_use_busif(io)) + return; + + rsnd_dma_addr_update(io, mod, io->dma); +} + static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -718,6 +727,10 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd) { + int is_play = rsnd_io_is_play(io); + int id = rsnd_mod_id(mod); + int ret = 0; + /* * rsnd_rdai_is_clk_master() will be enabled after set_fmt, * and, pcm_new will be called after it. @@ -725,7 +738,23 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, */ rsnd_ssi_parent_attach(mod, io); - return 0; + if (!rsnd_ssi_use_busif(io)) + return 0; + + if (rsnd_ssi_is_multi_slave(mod, io)) + return 0; + + /* SSI5 to SSI8 only have one BUSIF */ + if (id <= 4 || id == 9) + ret = rsnd_kctrl_new_s(mod, io, rtd, + is_play ? + "SSI Out BUSIF" : + "SSI In BUSIF", + rsnd_kctrl_accept_anytime, + rsnd_ssi_set_busif, + io->busif, 7); + + return ret; } static int rsnd_ssi_common_probe(struct rsnd_mod *mod, -- 2.17.0

3 2

[alsa-devel] [PATCH linux-next 09/10] ASoC: rsnd: add busif property to dai stream
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> SSI may use different busif for data transfer, this patch adds busif property to each dai stream, to indicate the busif used by playback/capture stream. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 16 +++++++++++++++- sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 5 ++++- 3 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..aee3132c72b9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -560,6 +560,14 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +struct rsnd_kctrl_cfg_s *rsnd_busif_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= 2 * rsnd_rdai_nr(priv))) + return NULL; + + return priv->busif + id; +} + static struct snd_soc_dai_driver *rsnd_daidrv_get(struct rsnd_priv *priv, int id) { @@ -1045,6 +1053,9 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, io_playback = &rdai->playback; io_capture = &rdai->capture; + io_playback->busif = rsnd_busif_get(priv, dai_i); + io_capture->busif = rsnd_busif_get(priv, rsnd_rdai_nr(priv) + dai_i); + snprintf(rdai->name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", dai_i); rdai->priv = priv; @@ -1108,6 +1119,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct snd_soc_dai_driver *rdrv; struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; int nr; int is_graph; int dai_i; @@ -1123,12 +1135,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) rdrv = devm_kcalloc(dev, nr, sizeof(*rdrv), GFP_KERNEL); rdai = devm_kcalloc(dev, nr, sizeof(*rdai), GFP_KERNEL); - if (!rdrv || !rdai) + busif = devm_kcalloc(dev, 2 * nr, sizeof(*busif), GFP_KERNEL); + if (!rdrv || !rdai || !busif) return -ENOMEM; priv->rdai_nr = nr; priv->daidrv = rdrv; priv->rdai = rdai; + priv->busif = busif; /* * parse all dai diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 4464d1d0a042..59b6d89d8edc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -456,6 +456,7 @@ struct rsnd_dai_stream { struct rsnd_mod *dma; struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ + struct rsnd_kctrl_cfg_s *busif; u32 parent_ssi_status; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) @@ -629,6 +630,8 @@ struct rsnd_priv { */ struct snd_soc_dai_driver *daidrv; struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; + int rdai_nr; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 75a3d15084c2..4ac4b5b75ae2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -154,7 +154,10 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) { - return 0; /* BUSIF0 only for now */ + if (!rsnd_ssi_use_busif(io)) + return 0; + + return io->busif->val; } static void rsnd_ssi_status_clear(struct rsnd_mod *mod) -- 2.17.0

3 2

[alsa-devel] [PATCH linux-next 00/10] ASoC: rsnd: support to set different busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Rcar audio DMA/SSIU already support to use different BUSIF to transfer data, but busif is fixed at 0. This patch set adds kctrl interface "SSI Out/In BUSIF", which enables to configure busif for each dai-stream used in transfer. device tree files are updated to use extended audio dmac register, and add dma request number for busif1 ~ busif7 Jiada Wang (10): ASoC: rsnd: ssi: Request dedicated dma channels for busif1 to 7 arm64: r8a7795: add dma request number for busif1 ~ busif7 arm64: r8a7796: add dma request number for busif1 ~ busif7 arm64: r8a774a1: add dma request number for busif1 ~ busif7 arm64: r8a7795: change to use extended audio dmac register arm64: r8a7796: change to use extended audio dmac register arm64: r8a774a1: change to use extended audio dmac register arm64: r8a77965: change to use extended audio dmac register ASoC: rsnd: add busif property to dai stream ASoC: rsnd: Add kctrl to configure dai's busif arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 139 ++++++++++++++++++---- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 132 +++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 138 +++++++++++++++++---- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- sound/soc/sh/rcar/core.c | 16 ++- sound/soc/sh/rcar/dma.c | 32 +++++ sound/soc/sh/rcar/rsnd.h | 5 + sound/soc/sh/rcar/ssi.c | 56 +++++++-- 8 files changed, 451 insertions(+), 69 deletions(-) -- 2.17.0

3 6

[alsa-devel] [PATCH] ASoC: Convert a few more users to using %pOFn instead of device_node.name
by Rob Herring 02 Oct '18

02 Oct '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/stm/stm32_sai.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d597eba61992..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; -- 2.17.1

1 0

[alsa-devel] [PATCH] slimbus: ngd: QCOM_QMI_HELPERS has to be selected
by Niklas Cassel 01 Oct '18

01 Oct '18

QCOM_QMI_HELPERS is a hidden kconfig, so the proper usage is to select it, not depend upon it. Signed-off-by: Niklas Cassel <niklas.cassel(a)linaro.org> --- drivers/slimbus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/Kconfig b/drivers/slimbus/Kconfig index 9d73ad806698..9cca2c3559f2 100644 --- a/drivers/slimbus/Kconfig +++ b/drivers/slimbus/Kconfig @@ -22,8 +22,8 @@ config SLIM_QCOM_CTRL config SLIM_QCOM_NGD_CTRL tristate "Qualcomm SLIMbus Satellite Non-Generic Device Component" - depends on QCOM_QMI_HELPERS depends on HAS_IOMEM && DMA_ENGINE + select QCOM_QMI_HELPERS help Select driver if Qualcomm's SLIMbus Satellite Non-Generic Device Component is programmed using Linux kernel. -- 2.17.1

3 2

[alsa-devel] KASAN: use-after-free Read in snd_seq_ioctl_get_subscription
by syzbot 01 Oct '18

01 Oct '18

Hello, syzbot found the following crash on: HEAD commit: e704966c45e4 Merge branch 'for-linus' of git://git.kernel... git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=1277814e400000 kernel config: https://syzkaller.appspot.com/x/.config?x=a8212f992609a887 dashboard link: https://syzkaller.appspot.com/bug?extid=9437020c82413d00222d compiler: gcc (GCC) 8.0.1 20180413 (experimental) userspace arch: i386 Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+9437020c82413d00222d(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_seq_ioctl_get_subscription+0x241/0x250 sound/core/seq/seq_clientmgr.c:1913 Read of size 80 at addr ffff880198190300 by task syz-executor1/22402 CPU: 0 PID: 22402 Comm: syz-executor1 Not tainted 4.19.0-rc5+ #162 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c4/0x2b4 lib/dump_stack.c:113 print_address_description.cold.8+0x9/0x1ff mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report.cold.9+0x242/0x309 mm/kasan/report.c:412 __asan_report_load_n_noabort+0xf/0x20 mm/kasan/report.c:443 snd_seq_ioctl_get_subscription+0x241/0x250 sound/core/seq/seq_clientmgr.c:1913 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 snd_seq_ioctl_compat+0x16c/0x2d0 sound/core/seq/seq_compat.c:123 __do_compat_sys_ioctl fs/compat_ioctl.c:1419 [inline] __se_compat_sys_ioctl fs/compat_ioctl.c:1365 [inline] __ia32_compat_sys_ioctl+0x20e/0x630 fs/compat_ioctl.c:1365 do_syscall_32_irqs_on arch/x86/entry/common.c:326 [inline] do_fast_syscall_32+0x34d/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f2bca9 Code: 85 d2 74 02 89 0a 5b 5d c3 8b 04 24 c3 8b 0c 24 c3 8b 1c 24 c3 90 90 90 90 90 90 90 90 90 90 90 90 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f5f270cc EFLAGS: 00000296 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 00000000c0505350 RDX: 0000000020000140 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Allocated by task 22402: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] kasan_kmalloc+0xc7/0xe0 mm/kasan/kasan.c:553 kmem_cache_alloc_trace+0x152/0x750 mm/slab.c:3620 kmalloc include/linux/slab.h:513 [inline] kzalloc include/linux/slab.h:707 [inline] snd_seq_port_connect+0xe4/0x730 sound/core/seq/seq_ports.c:571 snd_seq_ioctl_subscribe_port+0x1d8/0x310 sound/core/seq/seq_clientmgr.c:1458 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 snd_seq_ioctl_compat+0x16c/0x2d0 sound/core/seq/seq_compat.c:123 __do_compat_sys_ioctl fs/compat_ioctl.c:1419 [inline] __se_compat_sys_ioctl fs/compat_ioctl.c:1365 [inline] __ia32_compat_sys_ioctl+0x20e/0x630 fs/compat_ioctl.c:1365 do_syscall_32_irqs_on arch/x86/entry/common.c:326 [inline] do_fast_syscall_32+0x34d/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 Freed by task 22399: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/kasan.c:521 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:528 __cache_free mm/slab.c:3498 [inline] kfree+0xcf/0x230 mm/slab.c:3813 clear_subscriber_list+0x1ba/0x390 sound/core/seq/seq_ports.c:254 port_delete+0x7c/0x1b0 sound/core/seq/seq_ports.c:270 snd_seq_delete_all_ports+0x5fb/0x940 sound/core/seq/seq_ports.c:333 seq_free_client1.part.9+0xcb/0x270 sound/core/seq/seq_clientmgr.c:262 seq_free_client1 sound/core/seq/seq_clientmgr.c:256 [inline] seq_free_client+0x82/0x100 sound/core/seq/seq_clientmgr.c:285 snd_seq_release+0x51/0xe0 sound/core/seq/seq_clientmgr.c:366 __fput+0x385/0xa30 fs/file_table.c:278 ____fput+0x15/0x20 fs/file_table.c:309 task_work_run+0x1e8/0x2a0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:193 [inline] exit_to_usermode_loop+0x318/0x380 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:341 [inline] do_fast_syscall_32+0xcd5/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 The buggy address belongs to the object at ffff880198190300 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 0 bytes inside of 128-byte region [ffff880198190300, ffff880198190380) The buggy address belongs to the page: page:ffffea0006606400 count:1 mapcount:0 mapping:ffff8801da800640 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffffea000713cc48 ffffea0006e7eb08 ffff8801da800640 raw: 0000000000000000 ffff880198190000 0000000100000015 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff880198190200: fc fc fc fc fc fc fc fc 00 00 00 00 00 00 00 00 ffff880198190280: 00 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc > ffff880198190300: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff880198190380: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb ffff880198190400: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

1 0

[alsa-devel] i2s\pcm\spdif\slimbus
by Muni Sekhar 29 Sep '18

29 Sep '18

Hi All, Can someone share the links to i2s\pcm\spdif\slimbus bus specifications. -- Thanks, Sekhar

1 0

[alsa-devel] Fwd: Help with sound on Cherry Trail laptop with ES8316 sound chip
by Sundar Nagarajan 29 Sep '18

29 Sep '18

Hello, I have (separate) issues with two different Cherry Trail laptops that have the ES8316 chipset for sound. I already posted the following issues on Pierre Bossart's Intel UCM repository(https://github.com/plbossart/UCM): Issue 42: RDP Thinbook 1430 laptop (ES8316): Can't select between built-in speakers and headphone output (https://github.com/plbossart/UCM/issues/42) Pierre's input was as follows: The UCM file you are using gives the clue already: #Codec IC has only 1 analog out, switching SPK/HP is done externally, #probably by means of some analog switch controlled by gpio There is nothing we can do at the UCM level, this has to be handled at a lower level (i.e. the kernel) In addition, on a different laptop, I have a different problem with the ES3816 chipset. Issue 41: No sound on speakers on Chuwi Hi12 laptop - headphone sound works (es8316) (https://github.com/plbossart/UCM/issues/41) I reached out to Mark Brown, Daniel Drake and David Yang who were listed on the kernel.org commit for the ytcht_es8316.c driver (https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/…) and Mark suggested reaching out to you folks. What is the right place for me to post my issues to get your ideas and help? I have access to these machines, and I can compile kernels and perform testing, collect logs etc. Regards, Sundar ---------- Forwarded message --------- From: Mark Brown <broonie(a)kernel.org> Date: Thu, Sep 27, 2018 at 3:14 PM Subject: Re: Help with sound on Cherry Trail laptop with ES8316 sound chip To: Sundar Nagarajan <sun.nagarajan(a)gmail.com> Cc: Daniel Drake <drake(a)endlessm.com>, David Yang <yangxiaohua(a)everest-semi.com> On Wed, Sep 26, 2018 at 12:53:44PM -0700, Sundar Nagarajan wrote: > )I found your email from the kernel.org commit log for the > bytcht_es8316.c driver > (https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/…. > I have (separate) issues with two different Cherry Trail laptops that > have the ES8316 chipset. I posted the following issues on Pierre > Bossart's Intel UCM repository(https://github.com/plbossart/UCM): He's on sabbatical at the minute FWIW. > What is the right place for me to post my issues to get your ideas and help? You'd be best contacting the Intel maintainers about x86 specific issues: INTEL ASoC DRIVERS M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> M: Jie Yang <yang.jie(a)linux.intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/intel/ That includes Pierre but also a few other people, hopefully at least one of them will be able to help or point you to someone who can. You should also copy in the alsa-devel list (alsa-devel(a)alsa-project.org) in case someone else can help and so people can google the discussion later, though that'll mean your e-mail address ends up in the archives so you might not be happy with that.

1 0

[alsa-devel] [PATCH v2] ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops
by Matthias Kaehlcke 28 Sep '18

28 Sep '18

The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_dma_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - use 'snd_dummy_dma_ops' instead of 'snd_dummy_ops' --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..e3b9dd634c6d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_dma_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_dma_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- 2.19.0.605.g01d371f741-goog

2 1

[alsa-devel] [PATCH] ASoC: pxa: avoid AC97_BUS build warning
by Arnd Bergmann 28 Sep '18

28 Sep '18

Selecting AC97_BUS_NEW from SND_PXA2XX_SOC_AC97 leads to a Kconfig warning if any other driver selects AC97_BUS: WARNING: unmet direct dependencies detected for AC97_BUS_COMPAT Depends on [n]: SOUND [=y] && !UML && SND [=y] && AC97_BUS_NEW [=y] && !AC97_BUS [=y] Selected by [y]: - SND_SOC_WM9713 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && AC97_BUS_NEW [=y] I don't know if that combination is supposed to work. Assuming it is not, this adds a dependency on all users for PXA to avoids the combination. Fixes: 1c8bc7b3de5e ("ASoC: pxa: switch to new ac97 bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 29f577e6dfc0..943b44de1464 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -79,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -88,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,6 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,6 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 + depends on !AC97_BUS select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -116,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -126,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,6 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -194,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: atmel: add SND_SOC_I2C_AND_SPI dependency
by Arnd Bergmann 28 Sep '18

28 Sep '18

Selecting SND_SOC_WM8731 is only allowed when all its dependencies are already there: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] Selected by [m]: - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && (ARCH_AT91 [=y] || COMPILE_TEST [=y]) && ATMEL_SSC [=y] && SND_SOC_I2C_AND_SPI [=m] - SND_SOC_ALL_CODECS [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && COMPILE_TEST [=y] && SND_SOC_I2C_AND_SPI [=m] Fixes: a45f8853a5f9 ("ASoC: Add driver for PROTO Audio CODEC (with a WM8731)") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 81a0712d4f14..64f86f0b87e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -101,6 +101,7 @@ config SND_ATMEL_SOC_I2S config SND_SOC_MIKROE_PROTO tristate "Support for Mikroe-PROTO board" depends on OF + depends on SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 help Say Y or M if you want to add support for MikroElektronika PROTO Audio -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: wm9712: fix unused variable warning
by Arnd Bergmann 28 Sep '18

28 Sep '18

The 'ret' variable is now only used in an #ifdef, and causes a warning if it is declared outside of that block: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable] Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ade34c26ad2f..e873baa9e778 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component) { struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); struct regmap *regmap; - int ret; if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; } else { #ifdef CONFIG_SND_SOC_AC97_BUS + int ret; + wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Add r8a7744 support
by Biju Das 28 Sep '18

28 Sep '18

Document RZ/G1N (R8A7744) SoC bindings. Signed-off-by: Biju Das <biju.das(a)bp.renesas.com> Reviewed-by: Chris Paterson <Chris.Paterson2(a)renesas.com> --- This patch is tested against linux-next next-20180927. --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index f4688c5..d92b705 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -343,6 +343,7 @@ Required properties: "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) Examples with soctypes are: - "renesas,rcar_sound-r8a7743" (RZ/G1M) + - "renesas,rcar_sound-r8a7744" (RZ/G1N) - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - "renesas,rcar_sound-r8a7778" (R-Car M1A) -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include" to the asoc tree
by Mark Brown 28 Sep '18

28 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include 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 06da26e5ce157bf2af09ef4a9f5dc30af4599fa0 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Wed, 26 Sep 2018 15:33:03 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include Remove duplicated includes linux/of_platform.h and linux/wait.h Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 1 - sound/soc/qcom/qdsp6/q6core.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 2b2c7233bb5f..e1cfa846a1dc 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -11,7 +11,6 @@ #include <linux/spinlock.h> #include <linux/kref.h> #include <linux/of.h> -#include <linux/of_platform.h> #include <uapi/sound/asound.h> #include <linux/delay.h> #include <linux/slab.h> diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index ca1be7305524..cdfc8ab6cfc0 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -10,7 +10,6 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/jiffies.h> -#include <linux/wait.h> #include <linux/soc/qcom/apr.h> #include "q6core.h" #include "q6dsp-errno.h" -- 2.19.0

1 0

Re: [alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by AC30 CTLin0 28 Sep '18

28 Sep '18

OK, I will resend a new patch. However, from my last patch file, I can see DT file where is located at "Documentation/devicetree/bindings/sound/nau8822.txt". And don't you see any DT file? On 9/26/2018 1:52 AM, Mark Brown wrote: > On Tue, Sep 25, 2018 at 09:56:13AM -0700, Mark Brown wrote: >> On Tue, Sep 25, 2018 at 10:50:19AM +0800, AC30 CTLin0 wrote: >>> May I learn the status of patch review due to our customer ask us about >>> upstream status? >> I don't know, I don't seem to have a copy of this patch in my review >> queue. Please resend: > Found a copy in the list archives, but still no idea where it went in my > review queue. My best guess is that it got eaten by spam filtering > somewhere but I'm not 100% sure. Sorry about that. :( The driver > looks pretty clean, a few small points: > > - The Mute Switch controls should just be called Switch so that > applications know how to handle them. > - PGA Boost +20dB should just be "PGA Boost Volume" with the +20dB > specified by having it be a SOC_DOUBLE_R_TLV(). > - The switch statements in hw_params() should have default: > cases that return errors so that if an incorrect parameter makes it > through we detect it. > - There's a DT compatible string but no DT binding file is supplied, > should just be a very simple one as there's no parameters. > > These are all small, hopefully should be quick to update. =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

2 1

[alsa-devel] [PATCH linux-next 08/10] arm64: r8a77965: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index fd00010a5ed7..6c0d5053a81c 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1405,7 +1405,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ /* placeholder */ rcar_sound,dvc { -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 07/10] arm64: r8a774a1: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi index c9aad3e3b1b0..c3e0f2151176 100644 --- a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi @@ -1120,7 +1120,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 06/10] arm64: r8a7796: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index e9b05d997c4e..827a704154cd 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -1670,7 +1670,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 05/10] arm64: r8a7795: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7795.dtsi b/arch/arm64/boot/dts/renesas/r8a7795.dtsi index 0a66bda7124f..32d191102a81 100644 --- a/arch/arm64/boot/dts/renesas/r8a7795.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7795.dtsi @@ -1796,7 +1796,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH 0/5] MAX98088/9 of features
by Marco Felsch 26 Sep '18

26 Sep '18

Hi, I picked up the max98088 patches from Andreas series [1] and ported them to the new kernel. Regards, Marco [1] http://lists.infradead.org/pipermail/linux-arm-kernel/ \ 2015-February/324440.html Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable .../bindings/sound/maxim,max98088.txt | 20 ++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98088.c | 39 +++++++++++++++++++ 3 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt -- 2.19.0

2 11

Re: [alsa-devel] linux-next: Tree for Sep 25 (sound/soc/codecs/wm8731)
by Randy Dunlap 25 Sep '18

25 Sep '18

On 9/24/18 10:39 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20180924: > on i386: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] and then: ld: sound/soc/codecs/wm8731.o: in function `wm8731_i2c_probe': wm8731.c:(.text+0x609): undefined reference to `__devm_regmap_init_i2c' ld: sound/soc/codecs/wm8731.o: in function `wm8731_modinit': wm8731.c:(.init.text+0x8): undefined reference to `i2c_register_driver' ld: sound/soc/codecs/wm8731.o: in function `wm8731_exit': wm8731.c:(.exit.text+0x6): undefined reference to `i2c_del_driver' but CONFIG_I2C=m. Full randconfig file is attached. Note: I only see the kconfig warnings when I build for i386 on an x86_64 machine. If I build on an i386 system, I don't see those warnings, but I do see the linker errors. -- ~Randy

1 0

[alsa-devel] [PATCH v2] ASoC: cs4265: Add a MIC pre. route
by Matt Flax 25 Sep '18

25 Sep '18

The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- v2: Removed duplicate MIC lines. --- sound/soc/codecs/cs4265.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..4e55ac46 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"LINEOUTR", NULL, "DAC"}, {"SPDIFOUT", NULL, "SPDIF"}, + {"Pre-amp MIC", NULL, "MICL"}, + {"Pre-amp MIC", NULL, "MICR"}, + {"ADC Mux", "MIC", "Pre-amp MIC"}, {"ADC Mux", "LINEIN", "LINEINL"}, {"ADC Mux", "LINEIN", "LINEINR"}, - {"ADC Mux", "MIC", "MICL"}, - {"ADC Mux", "MIC", "MICR"}, {"ADC", NULL, "ADC Mux"}, {"DOUT", NULL, "ADC"}, {"DAI1 Capture", NULL, "DOUT"}, -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_ops
by Matthias Kaehlcke 25 Sep '18

25 Sep '18

The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..ab44133bc0bf 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- 2.19.0.605.g01d371f741-goog

3 2

[alsa-devel] [PATCH - Fix typo in configure.ac 1/1] PATCH - Fix typo in configure.ac
by plevine457＠gmail.com 25 Sep '18

25 Sep '18

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/configure.ac b/configure.ac index 3e2d233..cc3740b 100644 --- a/configure.ac +++ b/configure.ac @@ -90,7 +90,7 @@ if test "$use_maemo_rm" = "yes"; then fi AC_ARG_ENABLE([libav], - AS_HELP_STRING([--disable-avlib], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) + AS_HELP_STRING([--disable-libav], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) if test "x$enable_libav" != "xno"; then PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libavresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no]) -- 2.19.0

2 1

Re: [alsa-devel] [PATCH 1/3] ARM: configs: omap2plus: Add tlv320aic23 as module
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 08:41]: > The AM3517 EVM has an expander board which has two tlv320aic23 > codecs. This enables the driver for these codecs as a module. Applying into omap-for-v4.20/defconfig thanks. Tony

1 0

Re: [alsa-devel] [PATCH 3/3] ARM: OMAP2PLUS: Remove references to AM3517-EVM since it's in DT
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 08:41]: > Since the audio codecs used on the AM3517-EVM are now available > in device tree, and the audio driver wasn't really being used > anyway, this patch removes the remaining AM3517 legacy board > options and sound soc drivers. Mark do you want to pick this one? If so: Acked-by: Tony Lindgren <tony(a)atomide.com> Maybe update the subject line according too. Regards, Tony

1 0

Re: [alsa-devel] [PATCH 2/3] ARM: DTS: AM3517-EVM: Add support for UI board and Audio
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 15:41]: > The full AM3517-EVM shipped with an add-on board which contained > two Audio codecs, a GPIO expander with a variety of buttons, and > some other features. This patch enables these portions of the UI > board because they don't directly conflict with existing features. Applying this one into omap-for-v4.20/dt thanks. Tony

1 0

[alsa-devel] [PATCH] ASoC: cs4265: Add a MIC pre. route
by Matt Flax 24 Sep '18

24 Sep '18

The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..332d714a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -230,6 +230,10 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"DAI1 Capture", NULL, "DOUT"}, {"DAI2 Capture", NULL, "DOUT"}, + { "Pre-amp MIC", NULL, "MICL" }, + { "Pre-amp MIC", NULL, "MICR" }, + { "ADC Mux", "MIC", "Pre-amp MIC" }, + /* Loopback */ {"Loopback", "Switch", "ADC"}, {"DAC", NULL, "Loopback"}, -- 2.17.1

2 1

Re: [alsa-devel] [PATCH 3/3] ARM: OMAP2PLUS: Remove references to AM3517-EVM since it's in DT
by Jarkko Nikula 23 Sep '18

23 Sep '18

On 09/23/18 18:37, Adam Ford wrote: > Since the audio codecs used on the AM3517-EVM are now available > in device tree, and the audio driver wasn't really being used > anyway, this patch removes the remaining AM3517 legacy board > options and sound soc drivers. > > Signed-off-by: Adam Ford <aford173(a)gmail.com> > For sound/soc/omap/ changes: Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com>

1 0

[alsa-devel] Pinpoint audio audio corruption (Layla3G)
by Mark Hills 23 Sep '18

23 Sep '18

I have a bug in ALSA somewhere; playback is corrupted. Nothing is pro-actively telling me of a problem (eg. underrun) I could use some assistance getting to the bottom of this. Listen to [1] for what it sounds like -- analogue recording from the output. Look at the waveform [2] in Baudlien and it seems both L+R channels geting zero'd for some buffer period of 2.5 or 5 milliseconds. This is Layla3G, snd-echo3g. This problem has been rare, but persisted for many years. I've finally got something reproducable to investigate. It began ~10 years ago where very occasionally recordings or playback would do this. This was a simple self-contained application in production use; written by me, using plughw:Layla3G,0,0 (no desktops, browsers, dmix etc.) Problems were frequent enough to be annoying, but infrequent enough that I just had to put it down to 'cosmic rays', with no useful bug report. These days I am on a desktop using dmix; and more recently with Google Chrome and found I can reliably 'trigger' it. To reset, I must close all applications using /dev/snd, and re-open. It seems to be: * whether or not Google Chrome is first to open the device * about 10-60 minutes after Chrome has opened the device * more prone to happening if other applications are using the dmix * not related to buffer sizes; I use larger sizes in the asoundrc it doesn't affect the likelihood, or the 'sound' when it does happen * I've seen two Layla's affected * I've had many kernels and PCs over the years There's a bug somewhere, and with nothing printed to dmesg, console etc., I'm at a loss here, imagining my problem could be anywhere between dmix, alsa-lib, snd-echo3g or even the firmware itself. Whatever the case, I've finally got some combination I can do in userland as a trigger. Below is: - .asoundrc describing today's setup (previously I used stereo with no route, still with problems) - output of the 'aplay' that produced the recording above; and - some other stats. [1] http://www.pogo.org.uk/~mark/tmp/layla-glitch.flac [2] http://www.pogo.org.uk/~mark/tmp/layla-glitch.png Many thanks -- Mark pcm.!default { type plug slave.pcm layla_multi } pcm.layla_multi { type route slave.pcm { type dmix ipc_key 1742 slave { pcm "hw:Layla3G,0,0" period_size 992 buffer_size 10912 channels 5 rate 48000 format S16_LE } } ttable { # Headphones 0.0 1 1.1 1 # Speakers 0.2 1 1.3 1 # Subwoofer; mono mix 0.4 0.5 1.4 0.5 } } $ uname -a Linux stax 4.18.5 #113 SMP PREEMPT Mon Sep 3 23:14:59 BST 2018 x86_64 Intel(R) Core(TM) i5-6500 CPU @ 3.20GHz GenuineIntel GNU/Linux $ head -99l /proc/asound/Layla3G/pcm0p/sub0/* ==> /proc/asound/Layla3G/pcm0p/sub0/hw_params <== access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 6 rate: 48000 (48000/1) period_size: 992 buffer_size: 10912 ==> /proc/asound/Layla3G/pcm0p/sub0/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: Analog PCM name: Layla3G subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 8 subdevices_avail: 7 ==> /proc/asound/Layla3G/pcm0p/sub0/prealloc <== 128 ==> /proc/asound/Layla3G/pcm0p/sub0/prealloc_max <== 256 ==> /proc/asound/Layla3G/pcm0p/sub0/status <== state: RUNNING owner_pid : 9250 trigger_time: 663739.627559736 tstamp : 680208.555428028 delay : -790500256 avail : 790511168 avail_max : 790511168 ----- hw_ptr : 790500256 appl_ptr : 0 ==> /proc/asound/Layla3G/pcm0p/sub0/sw_params <== tstamp_mode: ENABLE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 6142909891733356544 silence_threshold: 0 silence_size: 6142909891733356544 boundary: 6142909891733356544 $ aplay -vv all48000.wav Playing WAVE 'all48000.wav' : Float 32 bit Little Endian, Rate 48000 Hz, Stereo Plug PCM: Linear Integer <-> Linear Float conversion PCM (S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 1 2 <- none 3 <- none 4 <- none 5 <- none Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Route conversion PCM Transformation table: 0 <- 0 1 <- 1 2 <- 0 3 <- 1 4 <- 0*0.5 + 1*0.5 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Hardware PCM card 0 'Layla3G' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 appl_ptr : 0 hw_ptr : 807111136 ############################ + | 56%^C Aborted by signal Interrupt... aplay: pcm_write:2004: write error: Interrupted system call

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: ensure that cluster header size is enough
by Dan Carpenter 22 Sep '18

22 Sep '18

I think we need a check to make sure that "hc_header.wLength" is large enough for a struct struct uac3_cluster_header_descriptor. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 67cf849aa16b..f7179ce39a19 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -968,6 +968,12 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * and request Cluster Descriptor */ wLength = le16_to_cpu(hc_header.wLength); + if (wLength < sizeof(*cluster)) { + dev_err(&dev->dev, + "%u:%d : cluster header size too small %d\n", + iface_no, altno, wLength); + return ERR_PTR(-EIO); + } cluster = kzalloc(wLength, GFP_KERNEL); if (!cluster) return ERR_PTR(-ENOMEM);

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 22 Sep '18

22 Sep '18

Hi Takashi, Please hold applying the patch, I'm having final checks with RME regarding the VID or VID:PID based approach when to use this functionality. If it seems better to use strict VID:PID table instead, I will send an updated patch. - Jussi

1 0

[alsa-devel] [RFC] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 22 Sep '18

22 Sep '18

This patch accessing vendor specific registers of RME Class Compliant devices works otherwise fine, but there is still one small problem. It is supposed to be a polled from the hardware, since the values indicate device status and are subject to change at all times. Behavior is correct if I keep running "amixer" all the time to retrieve values - sampling rate changes are visible and values get updated. However, when I poll the values from an application without reopening the mixer, they are being cached somewhere and the changes are not reflected. This problem doesn't exist for example with PCIe based RME HDSPe AIO and application gets updated values, but in this USB based device case they don't get updated. I have flagged all the mixer controls as read-only and volatile, but the volatile flag is not honored in this (assuming I've understood it correctly). Any guidance on how to fix the caching problem is welcome!

3 10

[alsa-devel] [PATCH 0/2] ASoC: for_each_component_xx() macro
by Kuninori Morimoto 21 Sep '18

21 Sep '18

Hi Mark These are v2 of for_each_component_xx() macro. v1 -> v2 is renamed comment_all to comment. Kuninori Morimoto (2): ASoC: add for_each_component() macro ASoC: add for_each_component_dais() macro include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 35 ++++++++++++++++++++--------------- 2 files changed, 25 insertions(+), 15 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI clock during suspend/resume modes
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Prepare <-> Cleanup functions pair has balanced calls. But in case of suspend mode no call to rsnd_soc_dai_shutdown() function, so cleanup isn't called. OTOH during resume mode function rsnd_soc_dai_prepare() is called, but calling rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero, bacause was not cleanup before). We need to call rsnd_ssi_prepare(), because it enables SSI clocks by calling rsnd_ssi_master_clk_start(). This patch allows to call prepare/cleanup functions always. Signed-off-by: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> [kuninori: adjusted to upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 7 +++---- sound/soc/sh/rcar/rsnd.h | 14 +++++++------- 2 files changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0bbc4b0..6d19475 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -134,10 +134,9 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); - if (dmaen->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } + /* maybe suspended */ + if (dmaen->chan) + return 0; /* * DMAEngine request uses mutex lock. diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e857311..4464d1d 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -318,9 +318,8 @@ struct rsnd_mod { /* * status * - * 0xH0000CBA + * 0xH0000CB0 * - * A 0: prepare 1: cleanup * B 0: init 1: quit * C 0: start 1: stop * @@ -331,9 +330,8 @@ struct rsnd_mod { * H 0: hw_params * H 0: pointer * H 0: prepare + * H 0: cleanup */ -#define __rsnd_mod_shift_prepare 0 -#define __rsnd_mod_shift_cleanup 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -345,11 +343,13 @@ struct rsnd_mod { #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ +#define __rsnd_mod_shift_prepare 28 /* always called */ +#define __rsnd_mod_shift_cleanup 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_prepare 1 -#define __rsnd_mod_add_cleanup -1 +#define __rsnd_mod_add_prepare 0 +#define __rsnd_mod_add_cleanup 0 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -363,7 +363,7 @@ struct rsnd_mod { #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 #define __rsnd_mod_call_prepare 0 -#define __rsnd_mod_call_cleanup 1 +#define __rsnd_mod_call_cleanup 0 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: tidyup for_each_card_prelinks() dai_link
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 7fe072b4df5d0 ("ASoC: add for_each_card_prelinks() macro") added new for_each_card_prelinks() macro, but it had typo. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 8 ++++---- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 875f846..97f9f38 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -311,9 +311,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = dev; @@ -326,9 +326,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index c67f629..6bc1d3d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -107,9 +107,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -122,9 +122,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index ff2e0ca..cc41eb5 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -172,7 +172,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) for_each_card_prelinks(card, i, dai_link) { if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -183,9 +183,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include from q6adm.c" to the asoc tree
by Mark Brown 21 Sep '18

21 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include from q6adm.c 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 c78d42c7fbd6f2a27a665bcd1ab60c9df617f7c9 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Fri, 21 Sep 2018 18:24:58 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include from q6adm.c We include wait.h twice in q6adm.c. it is unnecessary. hence remove it. Further, order the include files as alphabet. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6adm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 932c3ebfd252..da242515e146 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -2,25 +2,24 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited -#include <linux/slab.h> -#include <linux/wait.h> -#include <linux/kernel.h> #include <linux/device.h> -#include <linux/module.h> -#include <linux/sched.h> #include <linux/jiffies.h> +#include <linux/kernel.h> +#include <linux/kref.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> -#include <linux/kref.h> -#include <linux/wait.h> -#include <linux/soc/qcom/apr.h> #include <linux/platform_device.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/soc/qcom/apr.h> +#include <linux/wait.h> #include <sound/asound.h> #include "q6adm.h" #include "q6afe.h" #include "q6core.h" -#include "q6dsp-errno.h" #include "q6dsp-common.h" +#include "q6dsp-errno.h" #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 -- 2.19.0

1 0

[alsa-devel] [PATCH] ASoC: fix a use after free case
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

Unloading ASoC modules as used by the SOF driver leads to an object being used after it's been freed. Fix this be clearing a reference to it and making sure to check for its presence. This fixes issue #144. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-core.c | 6 +++--- sound/soc/soc-topology.c | 5 +++++ 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9cfe10d8040c..dce9d53ff283 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -942,9 +942,9 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { + if (dai && dai->probed) { + if (dai->driver && dai->driver->remove_order == order && + dai->driver->remove) { err = dai->driver->remove(dai); if (err < 0) dev_err(dai->dev, diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..6d438ec8c0ec 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp, { struct snd_soc_dai_driver *dai_drv = container_of(dobj, struct snd_soc_dai_driver, dobj); + struct snd_soc_dai *dai; if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->dai_unload) dobj->ops->dai_unload(comp, dobj); + list_for_each_entry(dai, &comp->dai_list, list) + if (dai->driver == dai_drv) + dai->driver = NULL; + kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); -- 2.17.1

1 0

[alsa-devel] [PATCH] ASoC:topology:bug fix:check return value avoid oops.
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

From: Wu Zhigang <zhigang.wu(a)linux.intel.com> check the return value to free the kcontrols instance to avoid oops caused by the pointer dereference. Signed-off-by: Wu Zhigang <zhigang.wu(a)linux.intel.com> [guennadi.liakhovetski(a)linux.intel.com add handling of .autodisable=1 cases] Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-dapm.c | 99 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 90 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..8dc31a7a42fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2402,12 +2402,11 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } -void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +static void snd_soc_dapm_free_widget_data(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *p, *next_p; enum snd_soc_dapm_direction dir; - list_del(&w->list); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both @@ -2420,6 +2419,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w->kcontrols); kfree_const(w->name); +} + +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + list_del(&w->list); + snd_soc_dapm_free_widget_data(w); kfree(w); } @@ -3038,11 +3043,16 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w, *last; unsigned int val; + int ret = 0; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + /* + * widgets with the snd_soc_dapm_kcontrol ID and .num_controls = 0 can + * be appended to the list while scanning it, this is safe. + */ list_for_each_entry(w, &card->widgets, list) { if (w->new) @@ -3053,8 +3063,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { - mutex_unlock(&card->dapm_mutex); - return -ENOMEM; + ret = -ENOMEM; + goto out_free; } } @@ -3062,23 +3072,29 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - dapm_new_mixer(w); + ret = dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - dapm_new_mux(w); + ret = dapm_new_mux(w); break; case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - dapm_new_pga(w); + ret = dapm_new_pga(w); break; case snd_soc_dapm_dai_link: - dapm_new_dai_link(w); + ret = dapm_new_dai_link(w); break; default: break; } + if (ret < 0) { + kfree(w->kcontrols); + w->kcontrols = NULL; + goto out_free; + } + /* Read the initial power state from the device */ if (w->reg >= 0) { soc_dapm_read(w->dapm, w->reg, &val); @@ -3087,6 +3103,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) if (val == w->on_val) w->power = 1; } + } + + list_for_each_entry(w, &card->widgets, list) { + if (w->new) + continue; w->new = 1; @@ -3097,6 +3118,66 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&card->dapm_mutex); return 0; + +out_free: + last = w; + + /* + * If any new widgets have been created above for .autodisable = 1 + * controls, they are also on this list, but at its very end. We're + * processing an error case, so the loop above was interrupted before + * reaching dynamically added widgets, since the latter cannot fail - + * their .num_controls = 0, so allocation cannot fail, and their type is + * snd_soc_dapm_kcontrol, those cannot fail either. Therefore "last" + * points to a widget before the first dynamic one. + */ + list_for_each_entry(w, &card->widgets, list) { + unsigned int i; + + if (w->new) + continue; + + if (w == last) + break; + + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_mixer_control *mc = + (struct soc_mixer_control *) + kcontrol->private_value; + + if (mc->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_enum *e = (struct soc_enum *) + kcontrol->private_value; + + if (e->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + default: + break; + } + + snd_soc_dapm_free_widget_data(w); + } + + mutex_unlock(&card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); -- 2.17.1

1 0

[alsa-devel] [PATCH v2] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 21 Sep '18

21 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to increase readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- Changes v2: - Fix typo in commit description, ncrease to increase. sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

2 3

[alsa-devel] [PATCH 00/11] add add for_each_xx() macro
by Kuninori Morimoto 20 Sep '18

20 Sep '18

Hi Mark Current ALSA SoC is using many list_for_each_entry() and/or for() loop for each lists. But, these need to use same pattern, and it is not readable. Now, "codec list loop for rtd", and "codec list loop for dai_link" had been replaced to for_each_xxx() macro. These patches replace "xxx list loop for yyy" by using for_each_yyy_xxx() macro. # Maybe some of them are overkill replacement, and/or # using not good naming. I'm happy to get feedback 1) - 2) : fixup previous for_each_xxx() macro patch 3) - 11) : new for_each_xxx() macro Kuninori Morimoto (11): 1) ASoC: convert for_each_rtd_codec_dai() for missing part 2) ASoC: rename for_each_rtd_codec_dai_reverse to rollback 3) ASoC: add for_each_card_prelinks() macro 4) ASoC: add for_each_card_links() macro 5) ASoC: add for_each_card_rtds() macro 6) ASoC: add for_each_card_components() macro 7) ASoC: add for_each_component_all() macro 8) ASoC: add for_each_component_dais() macro 9) ASoC: add for_each_comp_order() macro 10) ASoC: add for_each_dpcm_fe() macro 11) ASoC: add for_each_dpcm_be() macro include/sound/soc-dpcm.h | 10 ++ include/sound/soc.h | 29 ++++- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 5 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 5 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 13 +-- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 7 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/qcom/apq8096.c | 7 +- sound/soc/qcom/sdm845.c | 7 +- sound/soc/samsung/tm2_wm5110.c | 13 +-- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 130 +++++++++++------------ sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 101 +++++++++--------- 29 files changed, 230 insertions(+), 193 deletions(-) -- 2.7.4

2 21

[alsa-devel] Applied "ASoC: rt5663: Remove the boost volume in the beginning of playback" to the asoc tree
by Mark Brown 20 Sep '18

20 Sep '18

The patch ASoC: rt5663: Remove the boost volume in the beginning of playback 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 fc795bf7224efda8c35e505966c2757856064247 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 19 Sep 2018 09:56:47 +0800 Subject: [PATCH] ASoC: rt5663: Remove the boost volume in the beginning of playback The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad42240..2444fad7c2df 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.19.0

1 0

[alsa-devel] [PATCH v2 00/12] Sound Open Firmware Core
by Liam Girdwood 20 Sep '18

20 Sep '18

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core, later series will add host platform support for SOF. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since V1 :- o Added PM source. o Updated header comments to C++ style o Used generic private data in snd_soc_pcm_runtime o remove sof_get_pages() o Added comments for compressed SG pages. o Check return values for runtime_pm calls. Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware trace event support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add PM support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add Build support for SOF core include/sound/soc.h | 3 + include/sound/sof.h | 82 ++ include/uapi/sound/sof-abi.h | 29 + include/uapi/sound/sof-eq.h | 102 ++ include/uapi/sound/sof-fw.h | 67 + include/uapi/sound/sof-ipc.h | 938 ++++++++++++ include/uapi/sound/sof-tone.h | 27 + include/uapi/sound/sof-topology.h | 94 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 36 + sound/soc/sof/Makefile | 6 + sound/soc/sof/control.c | 270 ++++ sound/soc/sof/core.c | 381 +++++ sound/soc/sof/debug.c | 157 ++ sound/soc/sof/ipc.c | 788 ++++++++++ sound/soc/sof/loader.c | 287 ++++ sound/soc/sof/ops.c | 210 +++ sound/soc/sof/ops.h | 298 ++++ sound/soc/sof/pcm.c | 733 ++++++++++ sound/soc/sof/pm.c | 345 +++++ sound/soc/sof/sof-priv.h | 519 +++++++ sound/soc/sof/topology.c | 2222 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 293 ++++ 24 files changed, 7889 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/uapi/sound/sof-abi.h create mode 100644 include/uapi/sound/sof-eq.h create mode 100644 include/uapi/sound/sof-fw.h create mode 100644 include/uapi/sound/sof-ipc.h create mode 100644 include/uapi/sound/sof-tone.h create mode 100644 include/uapi/sound/sof-topology.h create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c -- 2.17.1

2 22

[alsa-devel] SoftVol in user space
by Riccardo Casagrande 20 Sep '18

20 Sep '18

Hello devs, I have to add a volume control to my custom card (It has an i.MX6 chip and 3 FDA2100 from TI as audio chip) because the FDAs have no volume controller. This below is the .asoundrc file I made. It works great if I manually made a speaker-test of one of devicefdaX sound card. Then I can see on alsamixer the volume control, but my need is to access and control softvol from user space. pcm.softvolfda0 { type softvol slave { pcm "hw:0,0" } control { name "volume0" card 0 device 0 } } pcm.softvolfda1 { type softvol slave { pcm "hw:0,1" } control { name "volume1" card 0 device 1 } } pcm.softvolfda2 { type softvol slave { pcm "hw:0,2" } control { name "volume2" card 0 device 2 } } pcm.devicefda0 { type plug slave.pcm "softvolfda0" } pcm.devicefda1 { type plug slave.pcm "softvolfda1" } pcm.devicefda2 { type plug slave.pcm "softvolfda2" } I saw that on alsalib there is this command int snd_pcm_softvol_open (snd_pcm_t ** pcmp, const char * name, snd_pcm_format_t sformat, int ctl_card, snd_ctl_elem_id_t * ctl_id, int cchannels, double min_dB, double max_dB, int resolution, snd_pcm_t * slave, int close_slave ) but I can't find any example of usage.This is the log of the error: mp && slave' failed. Aborted Any suggestion is welcome Riccardo Casagrande

1 0

[alsa-devel] [PATCH 00/15] Add support for Sound BlasterX AE-5
by Connor McAdams 20 Sep '18

20 Sep '18

This patch series adds support for the Sound BlasterX AE-5. I have personally tested it, and had others test it as well. All outputs work, front headphone, rear headphone, rear line-out, surround, digital. The only issue currently is that the microphone suffers from inconsistency like the Sound Blaster Z does, and Creative may be willing to help fix that, but if they're not, I have a few ideas. I also have the ZxR working currently. That will require some more complicated changes of the driver, so I have left it out and just focused on the AE-5. Connor McAdams (15): ALSA: hda/ca0132 - Define new verbs and control params ALSA: hda/ca0132 - Add quirk for Sound BlasterX AE-5 ALSA: hda/ca0132 - Add pincfg for AE-5 ALSA: hda/ca0132 - Set AE-5 bools and select mixer ALSA: hda/ca0132 - Change ca0132_mmio_init for AE-5 ALSA: hda/ca0132 - Add AE-5 pre-init and ca0113 functions ALSA: hda/ca0132 - Add AE-5 regular init setup ALSA: hda/ca0132 - Change firmware name and usage ALSA: hda/ca0132 - Merge post-dsp functions + cleanup ALSA: hda/ca0132 - Add DSP setup functions for AE-5 ALSA: hda/ca0132 - Clean up ca0132_alt_out_select ALSA: hda/ca0132 - Add output set commands for AE-5 ALSA: hda/ca0132 - Add input selection commands for AE-5 ALSA: hda/ca0132 - Add AE-5 specific controls ALSA: hda/ca0132 - Add AE-5 exit function sound/pci/hda/patch_ca0132.c | 1283 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 1038 insertions(+), 245 deletions(-) -- 2.7.4

2 17

[alsa-devel] [PATCH - hwmixvolume v2 1/7] hwmixvolume: use four spaces instead of one tab for indent
by Emmanuel Gil Peyrot 20 Sep '18

20 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This is the recommended coding style for all Python programs, as specified in PEP-0008[1]. [1] https://www.python.org/dev/peps/pep-0008/ Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index ef80bc8..038bcb3 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -26,285 +26,285 @@ EVENT_INFO = alsahcontrol.event_mask['INFO'] EVENT_REMOVE = alsahcontrol.event_mask_remove class Stream: - def __init__(self, element, parent): - self.element = element - self.element.set_callback(self) - self.parent = parent - self.label = None - self.scales = [] - self.adjustments = [] - self.callback(self.element, EVENT_INFO) - - def destroy(self): - self.deactivate() - - def callback(self, e, mask): - if mask == EVENT_REMOVE: - self.deactivate() - elif (mask & EVENT_INFO) != 0: - info = alsahcontrol.Info(self.element) - if info.is_inactive: - self.deactivate() - else: - self.activate() - elif (mask & EVENT_VALUE) != 0: - self.update_scales_from_ctl() - - def activate(self): - if self.label: - return - info = alsahcontrol.Info(self.element) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) - self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) - for i in range(info.count): - adj = gtk.Adjustment(value=values[i], - lower=info.min, upper=info.max, - step_incr=1, - page_incr=(info.max-info.min+1)/8) - adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) - scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) - self.scales.append(scale) - self.adjustments.append(adj) - self.parent.scales_vbox.show_all() - self.parent.update_msg_label() - - def deactivate(self): - if not self.label: - return - self.label.destroy() - for s in self.scales: - s.destroy() - self.label = None - self.scales = [] - self.adjustments = [] - self.parent.update_msg_label() - - def update_scales_from_ctl(self): - if not self.label: - return - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, count) - for i in range(count): - self.adjustments[i].set_value(values[i]) - - def update_ctl_from_scale(self, adj, index): - scale_value = adj.get_value() - value_to_set = int(round(adj.get_value())) - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - if self.parent.lock_check.get_active(): - values = [value_to_set for i in range(count)] - else: - value.read() - values = value.get_array(TYPE_INTEGER, count) - values[index] = value_to_set - value.set_array(TYPE_INTEGER, values) - value.write() - if value_to_set != scale_value: - adj.set_value(value_to_set) - - def get_label(self, info): - pid = self.get_pid(info) - if pid: - cmdline = self.get_pid_cmdline(pid) - if cmdline: - return cmdline - else: - return "PID %d" % pid - else: - name = info.name - if name[-7:] == " Volume": - name = name[:-7] - if name[-9:] == " Playback": - name = name[:-9] - return name - - def get_pid(self, info): - card = self.parent.current_card - device = info.device - subdevice = info.subdevice - if subdevice == 0: - subdevice = info.index - filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) - try: - f = open(filename, "r") - except IOError: - return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() - return None - - def get_pid_cmdline(self, pid): - try: - f = open("/proc/%d/cmdline" % pid, "r") - except IOError: - return None - try: - cmdline = f.read() - finally: - f.close() - return cmdline.replace('\x00', ' ').strip() + def __init__(self, element, parent): + self.element = element + self.element.set_callback(self) + self.parent = parent + self.label = None + self.scales = [] + self.adjustments = [] + self.callback(self.element, EVENT_INFO) + + def destroy(self): + self.deactivate() + + def callback(self, e, mask): + if mask == EVENT_REMOVE: + self.deactivate() + elif (mask & EVENT_INFO) != 0: + info = alsahcontrol.Info(self.element) + if info.is_inactive: + self.deactivate() + else: + self.activate() + elif (mask & EVENT_VALUE) != 0: + self.update_scales_from_ctl() + + def activate(self): + if self.label: + return + info = alsahcontrol.Info(self.element) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, info.count) + self.label = gtk.Label(self.get_label(info)) + self.label.set_single_line_mode(True) + self.parent.scales_vbox.pack_start(self.label, expand=False) + for i in range(info.count): + adj = gtk.Adjustment(value=values[i], + lower=info.min, upper=info.max, + step_incr=1, + page_incr=(info.max-info.min+1)/8) + adj.connect('value-changed', self.update_ctl_from_scale, i) + scale = gtk.HScale(adj) + scale.set_draw_value(False) + self.parent.scales_vbox.pack_start(scale, expand=False) + self.scales.append(scale) + self.adjustments.append(adj) + self.parent.scales_vbox.show_all() + self.parent.update_msg_label() + + def deactivate(self): + if not self.label: + return + self.label.destroy() + for s in self.scales: + s.destroy() + self.label = None + self.scales = [] + self.adjustments = [] + self.parent.update_msg_label() + + def update_scales_from_ctl(self): + if not self.label: + return + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, count) + for i in range(count): + self.adjustments[i].set_value(values[i]) + + def update_ctl_from_scale(self, adj, index): + scale_value = adj.get_value() + value_to_set = int(round(adj.get_value())) + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + if self.parent.lock_check.get_active(): + values = [value_to_set for i in range(count)] + else: + value.read() + values = value.get_array(TYPE_INTEGER, count) + values[index] = value_to_set + value.set_array(TYPE_INTEGER, values) + value.write() + if value_to_set != scale_value: + adj.set_value(value_to_set) + + def get_label(self, info): + pid = self.get_pid(info) + if pid: + cmdline = self.get_pid_cmdline(pid) + if cmdline: + return cmdline + else: + return "PID %d" % pid + else: + name = info.name + if name[-7:] == " Volume": + name = name[:-7] + if name[-9:] == " Playback": + name = name[:-9] + return name + + def get_pid(self, info): + card = self.parent.current_card + device = info.device + subdevice = info.subdevice + if subdevice == 0: + subdevice = info.index + filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) + try: + f = open(filename, "r") + except IOError: + return None + try: + for line in f.readlines(): + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) + finally: + f.close() + return None + + def get_pid_cmdline(self, pid): + try: + f = open("/proc/%d/cmdline" % pid, "r") + except IOError: + return None + try: + cmdline = f.read() + finally: + f.close() + return cmdline.replace('\x00', ' ').strip() class MixerWindow(gtk.Window): - card_numbers = alsacard.card_list() - current_card = -1 - hcontrol = None - scales_vbox = None - msg_label = None - streams = [] - hctl_sources = [] - - def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) - self.set_title("Hardware Mixer Volumes") - - vbox = gtk.VBox() - self.add(vbox) - - hbox = gtk.HBox() - vbox.pack_start(hbox, expand=False) - - label = gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) - - combo = gtk.combo_box_new_text() - for i in self.card_numbers: - str = "%d: %s" % (i, alsacard.card_get_name(i)) - combo.append_text(str) - if len(self.card_numbers) > 0: - combo.set_active(0) - combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) - label.set_mnemonic_widget(combo) - - self.lock_check = gtk.CheckButton(label="_Lock Channels") - self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) - - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) - - self.scales_vbox = gtk.VBox() - scrollwin.add_with_viewport(self.scales_vbox) - - label = gtk.Label() - label.set_single_line_mode(True) - line_height = label.size_request()[1] - label.destroy() - scale = gtk.HScale() - scale.set_draw_value(False) - line_height += scale.size_request()[1] - scale.destroy() - # always have space for at least four sliders - scrollwin.set_size_request(width=-1, height=line_height*4+4) - - # TODO: select the default card or the first card with stream controls - if len(self.card_numbers) > 0: - self.change_card(self.card_numbers[0]) - self.update_msg_label() - - self.show_all() - - def change_card(self, cardnum): - for s in self.hctl_sources: - gobject.source_remove(s) - self.hctl_sources = [] - - self.hcontrol = self.open_hcontrol_for_card(cardnum) - - for s in self.streams: - s.destroy() - self.streams = [] - - self.current_card = cardnum - - if not self.hcontrol: - self.update_msg_label() - return - - for id in self.hcontrol.list(): - if not self.is_stream_elem(id): - continue - elem = alsahcontrol.Element(self.hcontrol, id[0]) - info = alsahcontrol.Info(elem) - if not self.is_stream_info(info): - continue - stream = Stream(elem, self) - self.streams.append(stream) - - for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) - - self.update_msg_label() - - self.scales_vbox.show_all() - - def update_msg_label(self): - needs_msg = len(self.scales_vbox.get_children()) < 2 - has_msg = self.msg_label - if has_msg and not needs_msg: - self.msg_label.destroy() - self.msg_label = None - elif needs_msg: - if len(self.streams) > 0: - msg = "There are no open streams." - else: - msg = "This card does not have stream controls." - if not has_msg: - self.msg_label = gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) - self.scales_vbox.show_all() - elif self.msg_label.get_text() != msg: - self.msg_label.set_text(msg) - - def open_hcontrol_for_card(self, cardnum): - devname = "hw:CARD=" + str(cardnum) - try: - hc = alsahcontrol.HControl(name=devname, - mode=alsahcontrol.open_mode['NONBLOCK']) - except: - # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, - "Cannot open sound card control device.") - dlg.run() - dlg.destroy() - return None - return hc - - def is_stream_elem(self, id): - return ((id[1] == INTF_PCM and - id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or - (id[1] == INTF_MIXER and - id[4] == "VIA DXS Playback Volume")) - - def is_stream_info(self, info): - return info.is_readable and info.is_writable and info.type == TYPE_INTEGER - - def hctl_io_callback(self, source, condition): - self.hcontrol.handle_events() - return True + card_numbers = alsacard.card_list() + current_card = -1 + hcontrol = None + scales_vbox = None + msg_label = None + streams = [] + hctl_sources = [] + + def __init__(self): + gtk.Window.__init__(self) + self.connect('destroy', lambda w: gtk.main_quit()) + self.set_title("Hardware Mixer Volumes") + + vbox = gtk.VBox() + self.add(vbox) + + hbox = gtk.HBox() + vbox.pack_start(hbox, expand=False) + + label = gtk.Label("_Sound Card: ") + label.set_use_underline(True) + hbox.pack_start(label, expand=False) + + combo = gtk.combo_box_new_text() + for i in self.card_numbers: + str = "%d: %s" % (i, alsacard.card_get_name(i)) + combo.append_text(str) + if len(self.card_numbers) > 0: + combo.set_active(0) + combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) + hbox.pack_start(combo) + label.set_mnemonic_widget(combo) + + self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check.set_active(True) + vbox.pack_start(self.lock_check, expand=False) + + scrollwin = gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(gtk.SHADOW_NONE) + vbox.pack_start(scrollwin) + + self.scales_vbox = gtk.VBox() + scrollwin.add_with_viewport(self.scales_vbox) + + label = gtk.Label() + label.set_single_line_mode(True) + line_height = label.size_request()[1] + label.destroy() + scale = gtk.HScale() + scale.set_draw_value(False) + line_height += scale.size_request()[1] + scale.destroy() + # always have space for at least four sliders + scrollwin.set_size_request(width=-1, height=line_height*4+4) + + # TODO: select the default card or the first card with stream controls + if len(self.card_numbers) > 0: + self.change_card(self.card_numbers[0]) + self.update_msg_label() + + self.show_all() + + def change_card(self, cardnum): + for s in self.hctl_sources: + gobject.source_remove(s) + self.hctl_sources = [] + + self.hcontrol = self.open_hcontrol_for_card(cardnum) + + for s in self.streams: + s.destroy() + self.streams = [] + + self.current_card = cardnum + + if not self.hcontrol: + self.update_msg_label() + return + + for id in self.hcontrol.list(): + if not self.is_stream_elem(id): + continue + elem = alsahcontrol.Element(self.hcontrol, id[0]) + info = alsahcontrol.Info(elem) + if not self.is_stream_info(info): + continue + stream = Stream(elem, self) + self.streams.append(stream) + + for fd,condition in self.hcontrol.poll_fds: + self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + + self.update_msg_label() + + self.scales_vbox.show_all() + + def update_msg_label(self): + needs_msg = len(self.scales_vbox.get_children()) < 2 + has_msg = self.msg_label + if has_msg and not needs_msg: + self.msg_label.destroy() + self.msg_label = None + elif needs_msg: + if len(self.streams) > 0: + msg = "There are no open streams." + else: + msg = "This card does not have stream controls." + if not has_msg: + self.msg_label = gtk.Label(msg) + self.scales_vbox.pack_start(self.msg_label) + self.scales_vbox.show_all() + elif self.msg_label.get_text() != msg: + self.msg_label.set_text(msg) + + def open_hcontrol_for_card(self, cardnum): + devname = "hw:CARD=" + str(cardnum) + try: + hc = alsahcontrol.HControl(name=devname, + mode=alsahcontrol.open_mode['NONBLOCK']) + except: + # TODO: alsa error msg + dlg = gtk.MessageDialog(self, + gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, + gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + "Cannot open sound card control device.") + dlg.run() + dlg.destroy() + return None + return hc + + def is_stream_elem(self, id): + return ((id[1] == INTF_PCM and + id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or + (id[1] == INTF_MIXER and + id[4] == "VIA DXS Playback Volume")) + + def is_stream_info(self, info): + return info.is_readable and info.is_writable and info.type == TYPE_INTEGER + + def hctl_io_callback(self, source, condition): + self.hcontrol.handle_events() + return True def main(): - MixerWindow() - gtk.main() + MixerWindow() + gtk.main() main() -- 2.19.0

3 5

[alsa-devel] [PATCH] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 20 Sep '18

20 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to ncrease readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: q6afe: dt-bindings: Update input range for qcom, sd-lines
by Rohit kumar 19 Sep '18

19 Sep '18

Input to qcom,sd-lines should be between 0 and 3 instead of 1 to 4 as 0 corresponds to BIT(0) which is MI2S_SD0 line. Bit 1 to 3 corresponds to SD1 to SD3 lines respectively. Updated documentation for the same. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Documentation/devicetree/bindings/sound/qcom,q6afe.txt | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index a817940..d74888b 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -49,7 +49,7 @@ configuration of each dai. Must contain the following properties. Usage: required for mi2s interface Value type: <prop-encoded-array> Definition: Must be list of serial data lines used by this dai. - should be one or more of the 1-4 sd lines. + should be one or more of the 0-3 sd lines. - qcom,tdm-sync-mode: Usage: required for tdm interface @@ -137,42 +137,42 @@ q6afe@4 { prim-mi2s-rx@16 { reg = <16>; - qcom,sd-lines = <1 3>; + qcom,sd-lines = <0 2>; }; prim-mi2s-tx@17 { reg = <17>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; sec-mi2s-rx@18 { reg = <18>; - qcom,sd-lines = <1 4>; + qcom,sd-lines = <0 3>; }; sec-mi2s-tx@19 { reg = <19>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; tert-mi2s-rx@20 { reg = <20>; - qcom,sd-lines = <2 4>; + qcom,sd-lines = <1 3>; }; tert-mi2s-tx@21 { reg = <21>; - qcom,sd-lines = <1>; + qcom,sd-lines = <0>; }; quat-mi2s-rx@22 { reg = <22>; - qcom,sd-lines = <1>; + qcom,sd-lines = <0>; }; quat-mi2s-tx@23 { reg = <23>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; }; }; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

Re: [alsa-devel] [PATCH - hwmixvolume v2 0/7] Migrate to GTK+ 3.0 and allow python3
by Takashi Sakamoto 19 Sep '18

19 Sep '18

Hi, On Sep 18 2018 22:42, Emmanuel Gil Peyrot wrote: > From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> > > hwmixvolume was one of the last few programs on my system to depend on > PyGTK. This series ports it to GTK+ 3.x using gobject-introspection, > which allows it to run on Python 3.x too. > > My audio card does not have stream controls so I’ve been unable to test > whether the functionality is working properly, but from my reading of > the documentation it should be fine. > > Changes since v1: > - Reordered tab→space patch before everything else, to make it possible > to merge it as is without any other change. > - Merged fixup commits into the main switch patch, as they had no reason > to stand by themselves. > - Replaced iteration on file.readlines() with file.__iter__() directly > (thanks Sakamoto-san). > - Fix a typo in the README (thanks Sakamoto-san). > - Explicit more the reason for each patch (thanks Iwai-san), except for > trivial ones. > > Emmanuel Gil Peyrot (7): > hwmixvolume: use four spaces instead of one tab for indent > hwmixvolume: replace PyGTK with gobject-introspection > hwmixvolume: switch to GTK+ 3.0 and GLib 2.0 > hwmixvolume: use a with context to open files > hwmixvolume: switch the shebang to python > hwmixvolume: add my copyright > hwmixvolume: mention the new dependencies in the README > > hwmixvolume/README | 3 +- > hwmixvolume/hwmixvolume | 560 ++++++++++++++++++++-------------------- > 2 files changed, 283 insertions(+), 280 deletions(-) I reviewed all of the above patches. Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Thanks Takashi Sakamoto

1 0

[alsa-devel] [PATCH - hwmixvolume v2 3/7] hwmixvolume: switch to GTK+ 3.0 and GLib 2.0
by Emmanuel Gil Peyrot 19 Sep '18

19 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This replaces VBox and HBox with Grid (using Gtk.Orientation), HScale with Scale, creates labels with mnemonics, set hexpand and vexpand properly, use the correct enum container classes, use the correct getter for size request, and finally update to the correct GLib watch function. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 64d232c..8e0b6b8 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -60,18 +60,18 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = Gtk.Label(self.get_label(info)) + self.label = Gtk.Label.new(self.get_label(info)) self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) + self.parent.scales_vbox.add(self.label) for i in range(info.count): adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = Gtk.HScale(adj) + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL, adjustment=adj) scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) + self.parent.scales_vbox.add(scale) self.scales.append(scale) self.adjustments.append(adj) self.parent.scales_vbox.show_all() @@ -174,45 +174,48 @@ class MixerWindow(Gtk.Window): self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = Gtk.VBox() + vbox = Gtk.Grid() + vbox.set_orientation(Gtk.Orientation.VERTICAL) self.add(vbox) - hbox = Gtk.HBox() - vbox.pack_start(hbox, expand=False) + hbox = Gtk.Grid() + vbox.add(hbox) - label = Gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) + label = Gtk.Label.new_with_mnemonic("_Sound Card: ") + hbox.add(label) - combo = Gtk.combo_box_new_text() + combo = Gtk.ComboBoxText() + combo.set_hexpand(True) for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) if len(self.card_numbers) > 0: combo.set_active(0) combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) + hbox.add(combo) label.set_mnemonic_widget(combo) - self.lock_check = Gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton.new_with_mnemonic(label="_Lock Channels") self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) + vbox.add(self.lock_check) scrollwin = Gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(Gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) + scrollwin.set_policy(hscrollbar_policy=Gtk.PolicyType.NEVER, vscrollbar_policy=Gtk.PolicyType.AUTOMATIC) + scrollwin.set_shadow_type(Gtk.ShadowType.NONE) + scrollwin.set_vexpand(True) + vbox.add(scrollwin) - self.scales_vbox = Gtk.VBox() - scrollwin.add_with_viewport(self.scales_vbox) + self.scales_vbox = Gtk.Grid() + self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) + scrollwin.add(self.scales_vbox) label = Gtk.Label() label.set_single_line_mode(True) - line_height = label.size_request()[1] + line_height = max(label.get_size_request().height, 0) label.destroy() - scale = Gtk.HScale() + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL) scale.set_draw_value(False) - line_height += scale.size_request()[1] + line_height += max(scale.get_size_request().height, 0) scale.destroy() # always have space for at least four sliders scrollwin.set_size_request(width=-1, height=line_height*4+4) @@ -252,7 +255,7 @@ class MixerWindow(Gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, 0, GLib.IOCondition(condition), self.hctl_io_callback)) self.update_msg_label() @@ -270,8 +273,9 @@ class MixerWindow(Gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = Gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) + self.msg_label = Gtk.Label.new(msg) + self.msg_label.set_vexpand(True) + self.scales_vbox.add(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: self.msg_label.set_text(msg) @@ -284,8 +288,8 @@ class MixerWindow(Gtk.Window): except: # TODO: alsa error msg dlg = Gtk.MessageDialog(self, - Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, - Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, + Gtk.DialogFlags.MODAL | Gtk.DialogFlags.DESTROY_WITH_PARENT, + Gtk.MessageType.ERROR, Gtk.ButtonsType.OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() -- 2.19.0

2 3

[alsa-devel] [PATCH 2/2] sound: soc: wm8903: Cut gpio_base data
by Linus Walleij 19 Sep '18

19 Sep '18

This variable is only used inside the file, and we should never hard-code the GPIO base, so delete this. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/codecs/wm8903.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 23e43ff40ded..545512dc4c45 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -144,7 +144,6 @@ struct wm8903_platform_data { int micdet_delay; /* Delay after microphone detection (ms) */ - int gpio_base; u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ }; @@ -1876,17 +1875,12 @@ static const struct gpio_chip wm8903_template_chip = { static void wm8903_init_gpio(struct wm8903_priv *wm8903) { - struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; wm8903->gpio_chip.parent = wm8903->dev; - - if (pdata->gpio_base) - wm8903->gpio_chip.base = pdata->gpio_base; - else - wm8903->gpio_chip.base = -1; + wm8903->gpio_chip.base = -1; ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903); if (ret != 0) -- 2.17.1

2 4

[alsa-devel] [PATCH Resend] ASoC: rt5663: Remove the boost volume in the beginning of playback
by Oder Chiou 19 Sep '18

19 Sep '18

The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad..2444fad 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.7.4

1 0

[alsa-devel] [PATCH] sound: soc: rt5677-spi: Drop unused GPIO include
by Linus Walleij 18 Sep '18

18 Sep '18

This SPI driver does not use the legacy GPIO header so just delete it. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/codecs/rt5677-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..84501c2020c7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -18,7 +18,6 @@ #include <linux/interrupt.h> #include <linux/irq.h> #include <linux/slab.h> -#include <linux/gpio.h> #include <linux/sched.h> #include <linux/uaccess.h> #include <linux/regulator/consumer.h> -- 2.17.1

2 2

[alsa-devel] [PATCH 1/2] sound: soc: wm8903: Pull in platform data
by Linus Walleij 18 Sep '18

18 Sep '18

No platform in the kernel is using platform data to populate the WM8903 so pull it into the source file. This is done to handle further simplifications. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- include/sound/wm8903.h | 24 ------------------------ sound/soc/codecs/wm8903.c | 24 ++++++++++++++++++++++++ 2 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h index b310c5a3a958..b2ef76d5d00d 100644 --- a/include/sound/wm8903.h +++ b/include/sound/wm8903.h @@ -11,12 +11,6 @@ #ifndef __LINUX_SND_WM8903_H #define __LINUX_SND_WM8903_H -/* - * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of - * zero in platform data means "don't touch this pin". - */ -#define WM8903_GPIO_CONFIG_ZERO 0x8000 - /* * R6 (0x06) - Mic Bias Control 0 */ @@ -245,22 +239,4 @@ #define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ #define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ -#define WM8903_NUM_GPIO 5 - -struct wm8903_platform_data { - bool irq_active_low; /* Set if IRQ active low, default high */ - - /* Default register value for R6 (Mic bias), used to configure - * microphone detection. In conjunction with gpio_cfg this - * can be used to route the microphone status signals out onto - * the GPIOs for use with snd_soc_jack_add_gpios(). - */ - u16 micdet_cfg; - - int micdet_delay; /* Delay after microphone detection (ms) */ - - int gpio_base; - u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ -}; - #endif diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 6cb3c153ba19..23e43ff40ded 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -124,6 +124,30 @@ static const char *wm8903_supply_names[WM8903_NUM_SUPPLIES] = { "DCVDD", }; +/* + * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of + * zero in platform data means "don't touch this pin". + */ +#define WM8903_GPIO_CONFIG_ZERO 0x8000 + +#define WM8903_NUM_GPIO 5 + +struct wm8903_platform_data { + bool irq_active_low; /* Set if IRQ active low, default high */ + + /* Default register value for R6 (Mic bias), used to configure + * microphone detection. In conjunction with gpio_cfg this + * can be used to route the microphone status signals out onto + * the GPIOs for use with snd_soc_jack_add_gpios(). + */ + u16 micdet_cfg; + + int micdet_delay; /* Delay after microphone detection (ms) */ + + int gpio_base; + u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ +}; + struct wm8903_priv { struct wm8903_platform_data *pdata; struct device *dev; -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: remove redundant include" to the asoc tree
by Mark Brown 18 Sep '18

18 Sep '18

The patch ASoC: remove redundant include 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 29ca7d32d7f10737e8d165fcf40fe31d44b06bee Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Tue, 18 Sep 2018 16:16:24 +0800 Subject: [PATCH] ASoC: remove redundant include module.h already contained moduleparam.h, so it is safe to remove the redundant include. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/soc-dapm.c | 1 - 4 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852fd9723..b613103d801b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -10,7 +10,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1965635ec07c..2a3e5fbd04e4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -13,7 +13,6 @@ #include <linux/clk.h> #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 43edaf8cd276..593a11960888 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -11,7 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 43983c69f6aa..ee6b9758ec15 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -18,7 +18,6 @@ // device reopen. #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/async.h> #include <linux/delay.h> -- 2.19.0

1 0

[alsa-devel] [PATCH 2/5] ASoC: rt5682: Improve HP performance
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> We change the settings while HP power-up for better performance. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 32 +++++++++++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 14 ++++++++++++++ 2 files changed, 43 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 31ec7c0..bf7d910 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1437,6 +1437,28 @@ static const struct snd_kcontrol_new hpor_switch = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, RT5682_R_MUTE_SFT, 1, 1); +static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV); + break; + default: + return 0; + } + + return 0; +} + static int rt5682_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1449,8 +1471,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_LOGIC_CTRL_2, 0x0012); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x6000); - snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, - RT5682_NG2_EN_MASK, RT5682_NG2_EN); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); break; @@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, RT5682_PWR_HA_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, - RT5682_PUMP_EN_SFT, 0, NULL, 0), + RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, RT5682_CAPLESS_EN_SFT, 0, NULL, 0), @@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref1"}, {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, @@ -2616,6 +2638,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 8068140..d82a830 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1214,6 +1214,20 @@ #define RT5682_JDH_NO_PLUG (0x1 << 4) #define RT5682_JDH_PLUG (0x0 << 4) +/* Bias current control 8 (0x0111) */ +#define RT5682_HPA_CP_BIAS_CTRL_MASK (0x3 << 2) +#define RT5682_HPA_CP_BIAS_2UA (0x0 << 2) +#define RT5682_HPA_CP_BIAS_3UA (0x1 << 2) +#define RT5682_HPA_CP_BIAS_4UA (0x2 << 2) +#define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682_CP_CLK_HP_600KHZ (0x3 << 4) + /* Chopper and Clock control for DAC (0x013a)*/ #define RT5682_CKXEN_DAC1_MASK (0x1 << 13) #define RT5682_CKXEN_DAC1_SFT 13 -- 2.7.4

2 1

[alsa-devel] [PATCH 3/5] ASoC: rt5682: Remove HP volume control
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> This patch removed Headphone Playback Volume control. Due to codec settings, we don't want the user to change HP analog gain. The user could use DAC1 Playback Volume control to change playback volume. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index bf7d910..2725eb7 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -749,7 +749,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); @@ -1108,10 +1107,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work) } static const struct snd_kcontrol_new rt5682_snd_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, - RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), - /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), -- 2.7.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: rt5682: Update calibration function
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> The ADC/DAC path should open while calibration process. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 731c6a8..31ec7c0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2451,17 +2451,22 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); + regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2478,8 +2483,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); mutex_unlock(&rt5682->calibrate_mutex); -- 2.7.4

2 1

[alsa-devel] [PATCH 4/5] ASoC: rt5682: Fix the boost volume at the begining of playback
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> This patch fixed the boost volume at the begining of playback while DAC volume set to lower level. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 2725eb7..1809966 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -68,6 +68,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {0x01c1, 0x1000}, + {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; static const struct reg_default rt5682_reg[] = { @@ -1468,6 +1469,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_CTRL_2, 0x6000); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); break; case SND_SOC_DAPM_POST_PMD: @@ -1475,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x0); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x0000); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; default: -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 18 Sep '18

18 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: fix overrun issue and missed case where bytescount = rtd->bytescount sound/soc/amd/acp-pcm-dma.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b0e245c..0ac4b5b 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1035,16 +1035,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { period_bytes = frames_to_bytes(runtime, runtime->period_size); - dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); - if (dscr == rtd->dma_dscr_idx_1) - pos = period_bytes; - else - pos = 0; bytescount = acp_get_byte_count(rtd); - if (bytescount > rtd->bytescount) + if (bytescount >= rtd->bytescount) bytescount -= rtd->bytescount; - delay = do_div(bytescount, period_bytes); - runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) { + pos = 0; + } else { + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + } + if (bytescount > 0) { + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); + } } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

2 1

[alsa-devel] [PATCH 5/5] ASoC: rt5682: Minor code modification
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> Minor code changes are: - improve the readability in patch list - add i2c remove function - regmap_register_patch changes to regmap_multi_reg_write Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 1809966..340f904 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -67,7 +67,7 @@ struct rt5682_priv { }; static const struct reg_sequence patch_list[] = { - {0x01c1, 0x1000}, + {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; @@ -2584,7 +2584,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_calibrate(rt5682); - ret = regmap_register_patch(rt5682->regmap, patch_list, + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, ARRAY_SIZE(patch_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); @@ -2659,11 +2659,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return devm_snd_soc_register_component(&i2c->dev, - &soc_component_dev_rt5682, + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2694,6 +2700,7 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; -- 2.7.4

2 1

[alsa-devel] [PATCH - hwmixvolume v2 7/7] hwmixvolume: mention the new dependencies in the README
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/README b/hwmixvolume/README index 9884ec2..56c7554 100644 --- a/hwmixvolume/README +++ b/hwmixvolume/README @@ -8,7 +8,8 @@ that use hardware mixing, i.e., those based on the following chips: * Yamaha DS-1 (YMF-724/740/744/754) (driver: snd-ymfpci) -This tool requires Python, pygtk, and alsa-pyton 1.0.22 or later. +This tool requires Python, PyGObject, GTK+ 3.0 or later, and alsa-python 1.0.22 +or later. It is recommended to use at least Linux kernel 2.6.32 or alsa-driver 1.0.22; otherwise, the name of the program that is using a stream cannot be shown. -- 2.19.0

1 0

[alsa-devel] [PATCH - hwmixvolume v2 6/7] hwmixvolume: add my copyright
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 4804a9b..871c2c5 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -2,6 +2,7 @@ # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch +# Copyright (c) 2018 Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> # # Permission to use, copy, modify, and/or distribute this software for any # purpose with or without fee is hereby granted, provided that the above -- 2.19.0

1 0

[alsa-devel] [PATCH - hwmixvolume v2 5/7] hwmixvolume: switch the shebang to python
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This signifies that this code is now compatible with both Python 2 and Python 3, as per PEP-0394[1]. [1] https://www.python.org/dev/peps/pep-0394/ Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 7f8ba8e..4804a9b 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -1,4 +1,4 @@ -#!/usr/bin/python2 +#!/usr/bin/env python # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch -- 2.19.0

1 0

[alsa-devel] [PATCH - hwmixvolume v2 4/7] hwmixvolume: use a with context to open files
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This feature has been added in Python 2.5 and automatically closes an open file once the context exits. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 8e0b6b8..7f8ba8e 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -138,26 +138,20 @@ class Stream: subdevice = info.index filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) try: - f = open(filename, "r") + with open(filename, "r") as f: + for line in f: + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) except IOError: return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() return None def get_pid_cmdline(self, pid): try: - f = open("/proc/%d/cmdline" % pid, "r") + with open("/proc/%d/cmdline" % pid, "r") as f: + cmdline = f.read() except IOError: return None - try: - cmdline = f.read() - finally: - f.close() return cmdline.replace('\x00', ' ').strip() class MixerWindow(Gtk.Window): -- 2.19.0

1 0

[alsa-devel] [PATCH - hwmixvolume v2 2/7] hwmixvolume: replace PyGTK with gobject-introspection
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This doesn’t work yet, we require Gtk 3.0 rather than 2.0 and the API changed quite a lot, so this is but a preparatory patch. This is done so that we can get rid of GTK+ 2 which has been EOL for many years already, and to add Python 3 support because Python 2 will very soon be EOL as well. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 038bcb3..64d232c 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -15,7 +15,10 @@ # OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR # PERFORMANCE OF THIS SOFTWARE. -import gobject, gtk +import gi +gi.require_version('GLib', '2.0') +gi.require_version('Gtk', '3.0') +from gi.repository import GLib, Gtk from pyalsa import alsacard, alsahcontrol INTF_PCM = alsahcontrol.interface_id['PCM'] @@ -57,16 +60,16 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) + self.label = Gtk.Label(self.get_label(info)) self.label.set_single_line_mode(True) self.parent.scales_vbox.pack_start(self.label, expand=False) for i in range(info.count): - adj = gtk.Adjustment(value=values[i], + adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) + scale = Gtk.HScale(adj) scale.set_draw_value(False) self.parent.scales_vbox.pack_start(scale, expand=False) self.scales.append(scale) @@ -157,7 +160,7 @@ class Stream: f.close() return cmdline.replace('\x00', ' ').strip() -class MixerWindow(gtk.Window): +class MixerWindow(Gtk.Window): card_numbers = alsacard.card_list() current_card = -1 hcontrol = None @@ -167,21 +170,21 @@ class MixerWindow(gtk.Window): hctl_sources = [] def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) + Gtk.Window.__init__(self) + self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = gtk.VBox() + vbox = Gtk.VBox() self.add(vbox) - hbox = gtk.HBox() + hbox = Gtk.HBox() vbox.pack_start(hbox, expand=False) - label = gtk.Label("_Sound Card: ") + label = Gtk.Label("_Sound Card: ") label.set_use_underline(True) hbox.pack_start(label, expand=False) - combo = gtk.combo_box_new_text() + combo = Gtk.combo_box_new_text() for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) @@ -191,23 +194,23 @@ class MixerWindow(gtk.Window): hbox.pack_start(combo) label.set_mnemonic_widget(combo) - self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton(label="_Lock Channels") self.lock_check.set_active(True) vbox.pack_start(self.lock_check, expand=False) - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) + scrollwin = Gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(Gtk.SHADOW_NONE) vbox.pack_start(scrollwin) - self.scales_vbox = gtk.VBox() + self.scales_vbox = Gtk.VBox() scrollwin.add_with_viewport(self.scales_vbox) - label = gtk.Label() + label = Gtk.Label() label.set_single_line_mode(True) line_height = label.size_request()[1] label.destroy() - scale = gtk.HScale() + scale = Gtk.HScale() scale.set_draw_value(False) line_height += scale.size_request()[1] scale.destroy() @@ -223,7 +226,7 @@ class MixerWindow(gtk.Window): def change_card(self, cardnum): for s in self.hctl_sources: - gobject.source_remove(s) + GLib.source_remove(s) self.hctl_sources = [] self.hcontrol = self.open_hcontrol_for_card(cardnum) @@ -249,7 +252,7 @@ class MixerWindow(gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) self.update_msg_label() @@ -267,7 +270,7 @@ class MixerWindow(gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = gtk.Label(msg) + self.msg_label = Gtk.Label(msg) self.scales_vbox.pack_start(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: @@ -280,9 +283,9 @@ class MixerWindow(gtk.Window): mode=alsahcontrol.open_mode['NONBLOCK']) except: # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + dlg = Gtk.MessageDialog(self, + Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, + Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() @@ -304,7 +307,7 @@ class MixerWindow(gtk.Window): def main(): MixerWindow() - gtk.main() + Gtk.main() main() -- 2.19.0

1 0

[alsa-devel] [PATCH] ASoC: es8328: Fix fall-through annotations
by Gustavo A. R. Silva 18 Sep '18

18 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/es8328.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 4b5827d..04a3aa7 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: hisilicon: fix fall-through annotations
by Gustavo A. R. Silva 18 Sep '18

18 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/hisilicon/hi6210-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 53344a3..a69e5b11 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U16_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S16_LE: bits = HII2S_BITS_16; break; case SNDRV_PCM_FORMAT_U24_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S24_LE: bits = HII2S_BITS_24; break; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c" to the asoc tree
by Mark Brown 18 Sep '18

18 Sep '18

The patch ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c 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 24b7a0aa1abec41b62c9a29a75e511d29f95033b Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Fri, 14 Sep 2018 01:36:04 +0000 Subject: [PATCH] ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c Remove duplicated include. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c75fab38905d..c3806d7037fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -8,7 +8,6 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> #include <asm/dma.h> -- 2.19.0

1 0

[alsa-devel] [PATCH 0/2] ASoC: soc-core: adjust platform for new style
by Kuninori Morimoto 18 Sep '18

18 Sep '18

Hi Mark These are v2 of "ASoC: soc-core: use more generic method to find platform" but, renamed patch series naming. v1 patch set had bug under DPCM which can't probe sound card. The cause was "platform name" management and "component" finding loop. These patch set solved this issue. And then, patch set became reduced Now, ALSA SoC is using snd_soc_dai_link_component for platform instead of legacy style (= platform_name/platform_of_node/platform). OTOH, CPU/Codec are finding its DAI by using common snd_soc_find_dai() function which uses snd_soc_dai_link_component. Then, of course, we want to use same style for platform. These patch do it. Kuninori Morimoto (2): ASoC: soc-core: manage platform name under snd_soc_init_platform() ASoC: soc-core: add snd_soc_is_matching_component() sound/soc/soc-core.c | 73 +++++++++++++++++++++++++++------------------------- 1 file changed, 38 insertions(+), 35 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly
by Oder Chiou 18 Sep '18

18 Sep '18

The value of period_bytes will get the zero before the hw_params() is not run completely. Move the function snd_pcm_lib_period_bytes() to copy work, and make sure that is not zero. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 6478d10..4d46f45 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + if (!period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + + if (rt5514_dsp->buf_size % period_bytes) + rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * + period_bytes; if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, @@ -149,13 +157,11 @@ static void rt5514_spi_copy_work(struct work_struct *work) static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { - size_t period_bytes; u8 buf[8]; if (!rt5514_dsp->substream) return; - period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); rt5514_dsp->get_size = 0; /** @@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - if (rt5514_dsp->buf_size % period_bytes) - rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * - period_bytes; - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: fireworks: fix memory leak of response buffer at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating memory object for response buffer, ALSA fireworks driver has leak of the memory object at error path. This commit releases the object at the error path. Fixes: 7d3c1d5901aa('ALSA: fireworks: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 71a0613d3da0..f2d073365cf6 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -301,6 +301,8 @@ do_registration(struct work_struct *work) snd_efw_transaction_remove_instance(efw); snd_efw_stream_destroy_duplex(efw); snd_card_free(efw->card); + kfree(efw->resp_buf); + efw->resp_buf = NULL; dev_info(&efw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak of discovered stream formats at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After finishing discover of stream formats, ALSA OXFW driver has memory leak of allocated memory object at error path. This commit releases the memory object at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 1e5b2c802635..63b3e1ce986e 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -207,6 +207,7 @@ static int detect_quirks(struct snd_oxfw *oxfw) static void do_registration(struct work_struct *work) { struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); + int i; int err; if (oxfw->registered) @@ -269,6 +270,12 @@ static void do_registration(struct work_struct *work) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); if (oxfw->has_output) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) { + kfree(oxfw->tx_stream_formats[i]); + oxfw->tx_stream_formats[i] = NULL; + kfree(oxfw->rx_stream_formats[i]); + oxfw->rx_stream_formats[i] = NULL; + } snd_card_free(oxfw->card); dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak for model-dependent data at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data, ALSA OXFW driver has memory leak of the data at error path. This commit releases the data at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 63b3e1ce986e..9c8c527ebff9 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -277,6 +277,8 @@ static void do_registration(struct work_struct *work) oxfw->rx_stream_formats[i] = NULL; } snd_card_free(oxfw->card); + kfree(oxfw->spec); + oxfw->spec = NULL; dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: bebob: fix memory leak for M-Audio FW1814 and ProjectMix I/O at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data for M-Audio FW1814 and ProjectMix I/O, ALSA bebob driver has memory leak at error path. This commit releases the allocated data at the error path. Fixes: 04a2c73c97eb('ALSA: bebob: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 2 ++ sound/firewire/bebob/bebob_maudio.c | 4 ---- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 730ea91d9be8..93676354f87f 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -263,6 +263,8 @@ do_registration(struct work_struct *work) error: mutex_unlock(&devices_mutex); snd_bebob_stream_destroy_duplex(bebob); + kfree(bebob->maudio_special_quirk); + bebob->maudio_special_quirk = NULL; snd_card_free(bebob->card); dev_info(&bebob->unit->device, "Sound card registration failed: %d\n", err); diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index bd55620c6a47..164f2126a676 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -290,10 +290,6 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814) bebob->midi_output_ports = 2; } end: - if (err < 0) { - kfree(params); - bebob->maudio_special_quirk = NULL; - } mutex_unlock(&bebob->mutex); return err; } -- 2.17.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.19
by Mark Brown 17 Sep '18

17 Sep '18

The following changes since commit f2cf0ef7c0ce141bb38f315c34c56e6ef5667a27: ASoC: adav80x: mark expected switch fall-through (2018-08-09 14:41:50 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.19-rc4 for you to fetch changes up to 75383f8d39d4c0fb96083dd460b7b139fbdac492: sound: don't call skl_init_chip() to reset intel skl soc (2018-09-12 11:22:09 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.19 This is the usual set of small fixes scatterd around various drivers, plus one fix for DAPM and a UAPI build fix. There's not a huge amount that stands out here relative to anything else. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: AMD: Ensure reset bit is cleared before configuring Andrew F. Davis (1): ASoC: tas6424: Save last fault register even when clear Charles Keepax (1): ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs Danny Smith (1): ASoC: sigmadsp: safeload should not have lower byte limit Dmitry V. Levin (1): ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirks for 2 more devices Jiada Wang (1): ASoC: rsnd: fixup not to call clk_get/set under non-atomic Katsuhiro Suzuki (1): ASoC: uniphier: change status to orphan Kuninori Morimoto (2): ASoC: rsnd: adg: care clock-frequency size ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER Marcel Ziswiler (1): ASoC: wm9712: fix replace codec to component Oder Chiou (1): ASoC: rt5514: Fix the issue of the delay volume applied again Pierre-Louis Bossart (1): ASoC: wm8804: Add ACPI support Ryan Lee (2): ASoC: max98373: Added speaker FS gain cotnrol register to volatile. ASoC: max98373: Added 10ms sleep after amp software reset Shuming Fan (1): ASoC: rt5682: Change DAC/ADC volume scale Srinivas Kandagatla (1): ASoC: q6routing: initialize data correctly Sébastien Szymanski (1): ASoC: cs4265: fix MMTLR Data switch control Yong Zhi (1): ASoC: Intel: Skylake: Acquire irq after RIRB allocation Yu Zhao (3): Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" sound: enable interrupt after dma buffer initialization sound: don't call skl_init_chip() to reset intel skl soc MAINTAINERS | 3 +- include/sound/hdaudio.h | 1 + include/sound/soc-dapm.h | 1 + include/uapi/sound/skl-tplg-interface.h | 106 ++++++++++++++++---------------- sound/hda/hdac_controller.c | 15 +++-- sound/soc/amd/acp-pcm-dma.c | 21 +++++++ sound/soc/codecs/cs4265.c | 4 +- sound/soc/codecs/max98373.c | 3 + sound/soc/codecs/rt5514.c | 8 +-- sound/soc/codecs/rt5682.c | 8 +-- sound/soc/codecs/sigmadsp.c | 3 +- sound/soc/codecs/tas6424.c | 12 +++- sound/soc/codecs/wm8804-i2c.c | 15 ++++- sound/soc/codecs/wm9712.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++++++ sound/soc/intel/skylake/skl.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 4 +- sound/soc/sh/rcar/adg.c | 5 ++ sound/soc/sh/rcar/core.c | 21 ++++++- sound/soc/sh/rcar/dma.c | 4 ++ sound/soc/sh/rcar/rsnd.h | 7 +++ sound/soc/sh/rcar/ssi.c | 16 +++-- sound/soc/soc-core.c | 4 +- sound/soc/soc-dapm.c | 4 ++ 24 files changed, 208 insertions(+), 87 deletions(-)

2 1

[alsa-devel] Bug#908938: alsa-utils: amixer does not unmute pulseaudio
by Elimar Riesebieter 17 Sep '18

17 Sep '18

Control: forwarded -1 alsa-devel(a)alsa-project.org Hi alsa devs, please notice Debian Bug https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=908938 and let us know whether a fix is possible. Thanks in advance -- Elimar * Łukasz Stelmach <steelman(a)post.pl> [2018-09-16 11:23 +0200]: > Package: alsa-utils > Version: 1.1.3-1 > Severity: normal > > Dear Maintainer, > > I've got PulseAudio running, so amixer (and alsamixer) controls the > default PA output. I use the following command to control muting of > audio. > > amixer -q -c 0 sset Master toggle > > It mutes the audio but can't unmute. Neither can > > amixer -q -c 0 sset Master unmute > > Only alsamixer and pavucontrol can unmute. -- Do you smell something burning or is it me?

3 2

[alsa-devel] [PATCH] Documentation: soundwire: fix stream.rst markup warnings
by Randy Dunlap 17 Sep '18

17 Sep '18

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc markup warnings in soundwire/stream.rst: rc4/Documentation/driver-api/soundwire/stream.rst:177: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:203: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:248: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:277: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:304: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:328: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:352: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:364: WARNING: Explicit markup ends without a blank line; unexpected unindent. Fixes: 89634f99a83e ("Documentation: soundwire: Add more documentation") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: Shreyas NC <shreyas.nc(a)intel.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-doc(a)vger.kernel.org --- Documentation/driver-api/soundwire/stream.rst | 8 ++++++++ 1 file changed, 8 insertions(+) --- lnx-419-rc4.orig/Documentation/driver-api/soundwire/stream.rst +++ lnx-419-rc4/Documentation/driver-api/soundwire/stream.rst @@ -174,6 +174,7 @@ per stream. From ASoC DPCM framework, th .startup() operation. .. code-block:: c + int sdw_alloc_stream(char * stream_name); @@ -200,6 +201,7 @@ only be invoked once by respective Maste framework, this stream state is linked to .hw_params() operation. .. code-block:: c + int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, struct sdw_ports_config * ports_config, @@ -245,6 +247,7 @@ stream. From ASoC DPCM framework, this s .prepare() operation. .. code-block:: c + int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -274,6 +277,7 @@ stream. From ASoC DPCM framework, this s .trigger() start operation. .. code-block:: c + int sdw_enable_stream(struct sdw_stream_runtime * stream); SDW_STREAM_DISABLED @@ -301,6 +305,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -325,6 +330,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -349,6 +355,7 @@ all the Master(s) and Slave(s) associate framework, this stream state is linked to .hw_free() operation. .. code-block:: c + int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); int sdw_stream_remove_slave(struct sdw_slave * slave, @@ -361,6 +368,7 @@ stream assigned as part of ALLOCATED sta In .shutdown() the data structure maintaining stream state are freed up. .. code-block:: c + void sdw_release_stream(struct sdw_stream_runtime * stream); Not Supported

3 2

[alsa-devel] [PATCH] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 17 Sep '18

17 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index c7e972b..f27e20c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1050,6 +1050,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) bytescount -= rtd->bytescount; delay = do_div(bytescount, period_bytes); runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) + pos = 0; } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

1 0

[alsa-devel] [PATCH v3 00/13] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

Thankyou for reviewing v2 patchset, here is v3 addressing comments from v2. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v2 (https://lwn.net/Articles/761107/) - Updated dt bindings according to Robs Comments. - Added MBHC support with new bindings - Moved mfd irq in to single file as suggested by Lee Jones. - Addressed various minor comments on mfd driver by Lee Jones. - Rebased on top of 4.19-rc1 - renamed clsh as clsh-v2 as its v2 block Srinivas Kandagatla (13): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support .../bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 369 ++ include/linux/mfd/wcd9335/registers.h | 627 ++ include/linux/mfd/wcd9335/wcd9335.h | 65 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5234 +++++++++++++++++ sound/soc/qcom/apq8096.c | 122 +- 12 files changed, 7092 insertions(+), 6 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.18.0

4 27

[alsa-devel] [PATCH 0/7] slimbus: fix module loading
by srinivas.kandagatla＠linaro.org 17 Sep '18

17 Sep '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Hi Greg, Here are some of the fixes for issues found while testing module loading on DB820c debian setup. First 3 core fixes are to do with generating uevents, calling status callback in probe sequence if the device is up and one fix is able to match device id from device tree compatible strings. Second 4 ngd patches fixes few issues firstly with registration, secondly checking the validity of logical address assigned by remote. Lastly there are few typo fixes. Thanks, Srini Srinivas Kandagatla (7): slimbus: core: add support to uevent slimbus: core: update device status in probe slimbus: core: match device tree based devices correctly slimbus: ngd: validate logical address assigned by remote silmbus: ngd: register controller after power up. slimbus: ngd: return proper error code instead of zero slimbus: ngd: register ngd driver only once. drivers/slimbus/core.c | 37 +++++++++++++++++++++++++++++++++++-- drivers/slimbus/qcom-ngd-ctrl.c | 28 +++++++++++++++++----------- 2 files changed, 52 insertions(+), 13 deletions(-) -- 2.9.3

1 7

[alsa-devel] [PATCH 0/3] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Srinivas Kandagatla (3): ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 418 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 4 files changed, 450 insertions(+), 1 deletion(-) -- 2.18.0

3 7

[alsa-devel] [RFC v2 - AAF PCM plugin 0/5] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 15 Sep '18

15 Sep '18

Hi all, This is the version 2 of AAF plugin RFC series. For general context information about the AAF plugin, please refer to the cover letter from version 1 [1] and the documentation in doc/aaf.txt. Code-wise, the main changes in this new version are: 1) The media clock was generating timestamps in UTC coordinate. However, AVTP timestamps are in TAI coordinate system so the media clock was fixed to use TAI instead. 2) All plugin resources (e.g. buffers, fds) are now allocated/freed in hw_params/hw_free ioplug callbacks. 3) The plugin constraints are applied at entry point instead of hw_params callback. Also, in this new version, the commit message from patch "aaf: Implement Playback mode support" was improved so it discusses the AVTP protocol restrictions in terms of formats and rates as well as why the plugin uses the system clock to implement a media clock. For further information about what has been discussed in the previous version, please refer to the RFC v1 patch series which starts in [1]. All versions of this series can be also found in my alsa-plugins tree in github [2]. Best regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.Even [2] https://github.com/aguedes/alsa-plugins Andre Guedes (5): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1214 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 142 +++++++ 5 files changed, 1377 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

1 5

[alsa-devel] Wrong UCM profile for Dell TB16 / WD15
by KillerRex 14 Sep '18

14 Sep '18

Hello, I have a Dell TB16 (same sound hardware as the WD15) and since the update to kernel 4.18 pulseaudio does not recognise any more the audio ports in the dock. After some work tracking the problem I have found that pulseaudio reports a failure when using the UCM profile from Dell-WD15-Dock: ... ( 0.027| 0.000) I: [pulseaudio] module-alsa-card.c: Found UCM profiles ( 0.027| 0.000) D: [pulseaudio] alsa-ucm.c: UCM mapping: Mapping HiFi: hw:WD15Dock,0: source dev Mic ( 0.027| 0.000) D: [pulseaudio] alsa-ucm.c: UCM mapping: Mapping HiFi: hw:WD15Dock,1: sink dev LineOut ( 0.027| 0.000) D: [pulseaudio] alsa-ucm.c: UCM mapping: Mapping HiFi: hw:WD15Dock,0: sink dev Headphone ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Profile HiFi (Default), input=(null), output=(null) priority=8000, supported=yes n_input_mappings=1, n_output_mappings=2 ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Input HiFi: hw:WD15Dock,0: source ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Output HiFi: hw:WD15Dock,1: sink ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Output HiFi: hw:WD15Dock,0: sink ( 0.027| 0.000) I: [pulseaudio] alsa-ucm.c: Set ucm verb to HiFi ( 0.027| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:WD15Dock,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.028| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: Invalid value for card ( 0.028| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:WD15Dock,1: No such device ... The problem is that the device is called "Dock" and not "WD15Dock" so changing the file "alsa/ucm/Dell-WD15-Dock/HiFi.conf" with the right device name corrects the problem: --- alsa/ucm/Dell-WD15-Dock/HiFi.conf_ori 2018-09-14 17:54:35.022035850 +0200 +++ alsa/ucm/Dell-WD15-Dock/HiFi.conf 2018-09-14 17:54:55.722248500 +0200 @@ -3,7 +3,7 @@ Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,0" + PlaybackPCM "hw:Dock,0" } } @@ -12,7 +12,7 @@ Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,1" + PlaybackPCM "hw:Dock,1" } } @@ -21,6 +21,6 @@ Value { CaptureChannels "2" - CapturePCM "hw:WD15Dock,0" + CapturePCM "hw:Dock,0" } } Is this the right change or the real problem is that the device name is "Dock" instead "WD15Dock"? Removing the "WD15" from the name is better, in that way it is also valid for the TB16. Thanks Andres Ayala

1 0

[alsa-devel] [PATCH] ALSA: atiixp: fix fall-through annotations
by Gustavo A. R. Silva 14 Sep '18

14 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/atiixp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index a1e4944..1a41f8c 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -903,15 +903,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fallthru */ + /* fall through */ case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fallthru */ + /* fall through */ case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fallthru */ + /* fall through */ default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); -- 2.7.4

2 1

Re: [alsa-devel] [PATCH -next] ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c
by Srinivas Kandagatla 14 Sep '18

14 Sep '18

On 14/09/18 02:36, YueHaibing wrote: > Remove duplicated include. > > Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > --- > sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c > index c75fab3..c3806d7 100644 > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c > @@ -8,7 +8,6 @@ > #include <linux/platform_device.h> > #include <linux/slab.h> > #include <sound/soc.h> > -#include <sound/soc.h> > #include <sound/soc-dapm.h> > #include <sound/pcm.h> > #include <asm/dma.h> > > >

1 0

Re: [alsa-devel] of_dma_request_slave_channel() failed ?
by Geert Uytterhoeven 14 Sep '18

14 Sep '18

Hi Morimoto-san, CC asoc, dma, iommu On Tue, Sep 4, 2018 at 8:06 AM Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: > Hi Renesas ML > Cc Rob > > I noticed that Sound driver is using PIO mode > on latest kernel. > > ... > rcar_sound ec500000.sound: src[0] : probe error -11 > rcar_sound ec500000.sound: ssi[0] fallback to PIO mode > rcar_sound ec500000.sound: src[1] : probe error -11 > rcar_sound ec500000.sound: ssi[1] fallback to PIO mode > rcar_sound ec500000.sound: ssi[2] : probe error -11 > rcar_sound ec500000.sound: ssi[2] fallback to PIO mode > rcar_sound ec500000.sound: ssi[3] : probe error -11 > rcar_sound ec500000.sound: ssi[3] fallback to PIO mode > rcar_sound ec500000.sound: probed > ... > > Sound can use DMA mode and PIO mode. > It automatically fallbacks to PIO mode if it can't use DMA. > > I bisected this issue, and found the 1st bad commit > > ----------------------------------------------------- > commit ac6bbf0cdf4206c517ac9789814c23e372ebce4d > Author: Rob Herring <robh(a)kernel.org> > Date: Mon Jul 9 09:41:52 2018 -0600 > > iommu: Remove IOMMU_OF_DECLARE > > Now that we use the driver core to stop deferred probe for missing > drivers, IOMMU_OF_DECLARE can be removed. > > This is slightly less optimal than having a list of built-in drivers in > that we'll now defer probe twice before giving up. This shouldn't have a > significant impact on boot times as past discussions about deferred > probe have given no evidence of deferred probe having a substantial > impact. > ... > ----------------------------------------------------- > > In more detail, it seems it can't find DMA controller, > and of_dma_request_slave_channel() returns > -EPROBE_DEFER from this commit. > > struct dma_chan *of_dma_request_slave_channel(struct device_node *np, > const char *name) > { > ... > ofdma = of_dma_find_controller(&dma_spec); > > if (ofdma) { > chan = ofdma->of_dma_xlate(&dma_spec, ofdma); > } else { > => ret_no_channel = -EPROBE_DEFER; > chan = NULL; > } > ... > => return ERR_PTR(ret_no_channel); > } > > Is this known issue ? > Or sound shouldn't fallbacks to PIO mode ? I assume you wrote commit 6c92d5a2744e2761 ("ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER") to work around this? While this got rid of the error messages, and postpones sound initialization until the audio DMAC is available, it does mean that the driver will _never_ fall back to PIO. I.e. if CONFIG_RCAR_DMAC=n, audio will fail to probe instead of falling back to PIO. With CONFIG_RCAR_DMAC=y: rcar-dmac e6700000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac e7300000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac e7310000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac ec700000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac ec720000.dma-controller: ignoring dependency for device, assuming no driver So it seems the audio DMAC is deferred a second time, before the iommu driver probed. subsys_initcall(ipmmu_init); calls platform_driver_register module_platform_driver(rcar_dmac_driver); 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

4 9

Re: [alsa-devel] i915, HDMI/DP audio with multiple monitors
by Ville Syrjälä 13 Sep '18

13 Sep '18

On Tue, Sep 11, 2018 at 03:50:13PM +0200, Bruno Prémont wrote: > Hi, > > I have a system with multiple monitors and would like to send > notification sounds to the monitor on which corresponding > window is visible. > > For a workstation and a tiny computer things look different: > - workstation (Intel(R) Core(TM) i7-4770 CPU @ 3.40GHz): > 00:02.0 VGA compatible controller [0300]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller [8086:0412] (rev 06) > 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) > 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 04) > > Here alsa show me two cards: > - HDA Intel PCH (Realtek ALC671) > - HDA Intel HDMI (Intel Generic) > > **** List of PLAYBACK Hardware Devices **** > card 0: HDMI [HDA Intel HDMI], device 3: Generic Digital [Generic Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 There should normally be multiple HDMI devices (one for each HDMI/DP connector more or less). Eg. my hsw shows: card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 Looking at the hda_codec.c we see: static int audio_idx[HDA_PCM_NTYPES][5] = { ... [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 }, So you always get those device numbers, but I don't see an immediate relationship between those and the pin numbers (which do have some kind of relationship with the HDMI/DP port). I guess if they always get registered in order of the pin numbers then those would translate to 3 == port B, 7 == port C, etc. And after that the problem is figuring out which port is related to which connector, for which we have nothing at the moment. I suppose the ideal solution might be to have a sysfs symlink (or something) to connect the two together. But then there's MST where I think the pcm device correlates with the crtc rather than the connector. Maybe. I can't remember anymore, and I'm not sure it even works (I've recently heard people saying it doesn't). > card 1: PCH [HDA Intel PCH], device 0: ALC671 Analog [ALC671 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > - tiny computer (Intel(R) Core(TM) i5-6500T CPU @ 2.50GHz): > 00:02.0 VGA compatible controller [0300]: Intel Corporation HD Graphics 530 [8086:1912] (rev 06) > 00:1f.3 Audio device [0403]: Intel Corporation Sunrise Point-H HD Audio [8086:a170] (rev 31) > > Here alsa shows a single card: > - HDA Intel PCH (Realtek ALC671) > > **** List of PLAYBACK Hardware Devices **** > card 0: PCH [HDA Intel PCH], device 0: ALC671 Analog [ALC671 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: Generic Digital [Generic Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > How can I determine/set to which monitor the sound should go, and preferably send > different sounds to both monitors at same time? > > > Cheers, > Bruno > _______________________________________________ > dri-devel mailing list > dri-devel(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/dri-devel -- Ville Syrjälä Intel

5 4

[alsa-devel] [PATCH v2] ALSA: hda - Enable runtime PM only for discrete GPU
by Takashi Iwai 13 Sep '18

13 Sep '18

The recent change of vga_switcheroo allowed the runtime PM for HD-audio on AMD GPUs, but this also resulted in a regression. When the HD-audio controller driver gets runtime-suspended, HD-audio link is turned off, and the hotplug notification is ignored. This leads to the inconsistent audio state (the connection isn't notified and ELD is ignored). The best fix would be to implement the proper ELD notification via the audio component, but it's still not ready. As a quick workaround, this patch adds the check of runtime_idle and allows the runtime suspend only when the vga_switcheroo is bound with discrete GPU. That is, a system with a single GPU and APU would be again without runtime PM to keep the HD-audio link for the hotplug notification and ELD read out. Also, the codec->auto_runtime_pm flag is set only for the discrete GPU at the time GPU gets bound via vga_switcheroo (i.e. only dGPU is forcibly runtime-PM enabled), so that APU can still get the ELD notification. For identifying which GPU is bound, a new vga_switcheroo client callback, gpu_bound, is implemented. The vga_switcheroo simply calls this when GPU is bound, and tells whether it's dGPU or APU. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200945 Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 2 + include/linux/vga_switcheroo.h | 3 ++ sound/pci/hda/hda_intel.c | 86 +++++++++++++++++++++++--------- sound/pci/hda/hda_intel.h | 1 + 4 files changed, 69 insertions(+), 23 deletions(-) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index a96bf46bc483..cf2a18571d48 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -215,6 +215,8 @@ static void vga_switcheroo_enable(void) return; client->id = ret | ID_BIT_AUDIO; + if (client->ops->gpu_bound) + client->ops->gpu_bound(client->pdev, ret); } vga_switcheroo_debugfs_init(&vgasr_priv); diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index a34539b7f750..7e6ac0114d55 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -133,15 +133,18 @@ struct vga_switcheroo_handler { * @can_switch: check if the device is in a position to switch now. * Mandatory. The client should return false if a user space process * has one of its device files open + * @gpu_bound: notify the client id to audio client when the GPU is bound. * * Client callbacks. A client can be either a GPU or an audio device on a GPU. * The @set_gpu_state and @can_switch methods are mandatory, @reprobe may be * set to NULL. For audio clients, the @reprobe member is bogus. + * OTOH, @gpu_bound is only for audio clients, and not used for GPU clients. */ struct vga_switcheroo_client_ops { void (*set_gpu_state)(struct pci_dev *dev, enum vga_switcheroo_state); void (*reprobe)(struct pci_dev *dev); bool (*can_switch)(struct pci_dev *dev); + void (*gpu_bound)(struct pci_dev *dev, enum vga_switcheroo_client_id); }; #if defined(CONFIG_VGA_SWITCHEROO) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..aa4c672dbaf7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -365,8 +365,10 @@ enum { */ #ifdef SUPPORT_VGA_SWITCHEROO #define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) +#define needs_eld_notify_link(chip) ((chip)->need_eld_notify_link) #else #define use_vga_switcheroo(chip) 0 +#define needs_eld_notify_link(chip) false #endif #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ @@ -453,6 +455,7 @@ static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, #endif static int azx_acquire_irq(struct azx *chip, int do_disconnect); +static void set_default_power_save(struct azx *chip); /* * initialize the PCI registers @@ -1201,6 +1204,10 @@ static int azx_runtime_idle(struct device *dev) azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; + /* ELD notification gets broken when HD-audio bus is off */ + if (needs_eld_notify_link(hda)) + return -EBUSY; + return 0; } @@ -1298,6 +1305,36 @@ static bool azx_vs_can_switch(struct pci_dev *pci) return true; } +/* + * The discrete GPU cannot power down unless the HDA controller runtime + * suspends, so activate runtime PM on codecs even if power_save == 0. + */ +static void setup_vga_switcheroo_runtime_pm(struct azx *chip) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hda_codec *codec; + + if (hda->use_vga_switcheroo && !hda->need_eld_notify_link) { + list_for_each_codec(codec, &chip->bus) + codec->auto_runtime_pm = 1; + /* reset the power save setup */ + if (chip->running) + set_default_power_save(chip); + } +} + +static void azx_vs_gpu_bound(struct pci_dev *pci, + enum vga_switcheroo_client_id client_id) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip = card->private_data; + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + if (client_id == VGA_SWITCHEROO_DIS) + hda->need_eld_notify_link = 0; + setup_vga_switcheroo_runtime_pm(chip); +} + static void init_vga_switcheroo(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -1306,6 +1343,7 @@ static void init_vga_switcheroo(struct azx *chip) dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; + hda->need_eld_notify_link = 1; /* cleared in gpu_bound op */ chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } @@ -1314,6 +1352,7 @@ static void init_vga_switcheroo(struct azx *chip) static const struct vga_switcheroo_client_ops azx_vs_ops = { .set_gpu_state = azx_vs_set_state, .can_switch = azx_vs_can_switch, + .gpu_bound = azx_vs_gpu_bound, }; static int register_vga_switcheroo(struct azx *chip) @@ -1339,6 +1378,7 @@ static int register_vga_switcheroo(struct azx *chip) #define init_vga_switcheroo(chip) /* NOP */ #define register_vga_switcheroo(chip) 0 #define check_hdmi_disabled(pci) false +#define setup_vga_switcheroo_runtime_pm(chip) /* NOP */ #endif /* SUPPORT_VGA_SWITCHER */ /* @@ -1352,6 +1392,7 @@ static int azx_free(struct azx *chip) if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); + chip->running = 0; azx_del_card_list(chip); @@ -2230,6 +2271,25 @@ static struct snd_pci_quirk power_save_blacklist[] = { }; #endif /* CONFIG_PM */ +static void set_default_power_save(struct azx *chip) +{ + int val = power_save; + +#ifdef CONFIG_PM + if (pm_blacklist) { + const struct snd_pci_quirk *q; + + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } +#endif /* CONFIG_PM */ + snd_hda_set_power_save(&chip->bus, val * 1000); +} + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2241,9 +2301,7 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; - struct hda_codec *codec; int dev = chip->dev_index; - int val; int err; hda->probe_continued = 1; @@ -2322,31 +2380,13 @@ static int azx_probe_continue(struct azx *chip) if (err < 0) goto out_free; + setup_vga_switcheroo_runtime_pm(chip); + chip->running = 1; azx_add_card_list(chip); - val = power_save; -#ifdef CONFIG_PM - if (pm_blacklist) { - const struct snd_pci_quirk *q; - - q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); - if (q && val) { - dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", - q->subvendor, q->subdevice); - val = 0; - } - } -#endif /* CONFIG_PM */ - /* - * The discrete GPU cannot power down unless the HDA controller runtime - * suspends, so activate runtime PM on codecs even if power_save == 0. - */ - if (use_vga_switcheroo(hda)) - list_for_each_codec(codec, &chip->bus) - codec->auto_runtime_pm = 1; + set_default_power_save(chip); - snd_hda_set_power_save(&chip->bus, val * 1000); if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index e3a3d318d2e5..f59719e06b91 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -37,6 +37,7 @@ struct hda_intel { /* vga_switcheroo setup */ unsigned int use_vga_switcheroo:1; + unsigned int need_eld_notify_link:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ -- 2.18.0

3 4

[alsa-devel] [PATCH v2 10/17] compat_ioctl: remove translation for sound ioctls
by Arnd Bergmann 13 Sep '18

13 Sep '18

The SNDCTL_* and SOUND_* commands are the old OSS user interface. I checked all the sound ioctl commands listed in fs/compat_ioctl.c to see if we still need the translation handlers. Here is what I found: - sound/oss/ is (almost) gone from the kernel, this is what actually needed all the translations - The ALSA emulation for OSS correctly handles all compat_ioctl commands already. - sound/oss/dmasound/ is the last holdout of the original OSS code, this is only used on arch/m68k, which has no 64-bit mode and hence needs no compat handlers - arch/um/drivers/hostaudio_kern.c may run in 64-bit mode with 32-bit x86 user space underneath it. This rare corner case is the only one that still needs the compat handlers. By adding a simple redirect of .compat_ioctl to .unlocked_ioctl in the UML driver, we can remove all the COMPATIBLE_IOCTL() annotations without a change in functionality. For completeness, I'm adding the same thing to the dmasound file, knowing that it makes no difference. The compat_ioctl list contains one comment about SNDCTL_DSP_MAPINBUF and SNDCTL_DSP_MAPOUTBUF, which actually would need a translation handler if implemented. However, the native implementation just returns -EINVAL, so we don't care. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/um/drivers/hostaudio_kern.c | 1 + fs/compat_ioctl.c | 157 ----------------------------- sound/core/oss/pcm_oss.c | 4 + sound/oss/dmasound/dmasound_core.c | 2 + 4 files changed, 7 insertions(+), 157 deletions(-) diff --git a/arch/um/drivers/hostaudio_kern.c b/arch/um/drivers/hostaudio_kern.c index 7f9dbdbc4eb7..9dfd209ed316 100644 --- a/arch/um/drivers/hostaudio_kern.c +++ b/arch/um/drivers/hostaudio_kern.c @@ -298,6 +298,7 @@ static const struct file_operations hostaudio_fops = { .write = hostaudio_write, .poll = hostaudio_poll, .unlocked_ioctl = hostaudio_ioctl, + .compat_ioctl = generic_compat_ioctl_ptrarg, .mmap = NULL, .open = hostaudio_open, .release = hostaudio_release, diff --git a/fs/compat_ioctl.c b/fs/compat_ioctl.c index 001c9fd89d30..c28fd9f6b1e8 100644 --- a/fs/compat_ioctl.c +++ b/fs/compat_ioctl.c @@ -77,7 +77,6 @@ #include <linux/if_bonding.h> #include <linux/watchdog.h> -#include <linux/soundcard.h> #include <linux/lp.h> #include <linux/ppdev.h> @@ -775,162 +774,6 @@ COMPATIBLE_IOCTL(PPPIOCGL2TPSTATS) /* PPPOX */ COMPATIBLE_IOCTL(PPPOEIOCSFWD) COMPATIBLE_IOCTL(PPPOEIOCDFWD) -/* Big A */ -/* sparc only */ -/* Big Q for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_SEQ_RESET) -COMPATIBLE_IOCTL(SNDCTL_SEQ_SYNC) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_INFO) -COMPATIBLE_IOCTL(SNDCTL_SEQ_CTRLRATE) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETOUTCOUNT) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETINCOUNT) -COMPATIBLE_IOCTL(SNDCTL_SEQ_PERCMODE) -COMPATIBLE_IOCTL(SNDCTL_FM_LOAD_INSTR) -COMPATIBLE_IOCTL(SNDCTL_SEQ_TESTMIDI) -COMPATIBLE_IOCTL(SNDCTL_SEQ_RESETSAMPLES) -COMPATIBLE_IOCTL(SNDCTL_SEQ_NRSYNTHS) -COMPATIBLE_IOCTL(SNDCTL_SEQ_NRMIDIS) -COMPATIBLE_IOCTL(SNDCTL_MIDI_INFO) -COMPATIBLE_IOCTL(SNDCTL_SEQ_THRESHOLD) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_MEMAVL) -COMPATIBLE_IOCTL(SNDCTL_FM_4OP_ENABLE) -COMPATIBLE_IOCTL(SNDCTL_SEQ_PANIC) -COMPATIBLE_IOCTL(SNDCTL_SEQ_OUTOFBAND) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETTIME) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_ID) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_CONTROL) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_REMOVESAMPLE) -/* Big T for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_TMR_TIMEBASE) -COMPATIBLE_IOCTL(SNDCTL_TMR_START) -COMPATIBLE_IOCTL(SNDCTL_TMR_STOP) -COMPATIBLE_IOCTL(SNDCTL_TMR_CONTINUE) -COMPATIBLE_IOCTL(SNDCTL_TMR_TEMPO) -COMPATIBLE_IOCTL(SNDCTL_TMR_SOURCE) -COMPATIBLE_IOCTL(SNDCTL_TMR_METRONOME) -COMPATIBLE_IOCTL(SNDCTL_TMR_SELECT) -/* Little m for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_MIDI_PRETIME) -COMPATIBLE_IOCTL(SNDCTL_MIDI_MPUMODE) -COMPATIBLE_IOCTL(SNDCTL_MIDI_MPUCMD) -/* Big P for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_DSP_RESET) -COMPATIBLE_IOCTL(SNDCTL_DSP_SYNC) -COMPATIBLE_IOCTL(SNDCTL_DSP_SPEED) -COMPATIBLE_IOCTL(SNDCTL_DSP_STEREO) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETBLKSIZE) -COMPATIBLE_IOCTL(SNDCTL_DSP_CHANNELS) -COMPATIBLE_IOCTL(SOUND_PCM_WRITE_FILTER) -COMPATIBLE_IOCTL(SNDCTL_DSP_POST) -COMPATIBLE_IOCTL(SNDCTL_DSP_SUBDIVIDE) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETFRAGMENT) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETFMTS) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETFMT) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETOSPACE) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETISPACE) -COMPATIBLE_IOCTL(SNDCTL_DSP_NONBLOCK) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETCAPS) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETTRIGGER) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETTRIGGER) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETIPTR) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETOPTR) -/* SNDCTL_DSP_MAPINBUF, XXX needs translation */ -/* SNDCTL_DSP_MAPOUTBUF, XXX needs translation */ -COMPATIBLE_IOCTL(SNDCTL_DSP_SETSYNCRO) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETDUPLEX) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETODELAY) -COMPATIBLE_IOCTL(SNDCTL_DSP_PROFILE) -COMPATIBLE_IOCTL(SOUND_PCM_READ_RATE) -COMPATIBLE_IOCTL(SOUND_PCM_READ_CHANNELS) -COMPATIBLE_IOCTL(SOUND_PCM_READ_BITS) -COMPATIBLE_IOCTL(SOUND_PCM_READ_FILTER) -/* Big C for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_COPR_RESET) -COMPATIBLE_IOCTL(SNDCTL_COPR_LOAD) -COMPATIBLE_IOCTL(SNDCTL_COPR_RDATA) -COMPATIBLE_IOCTL(SNDCTL_COPR_RCODE) -COMPATIBLE_IOCTL(SNDCTL_COPR_WDATA) -COMPATIBLE_IOCTL(SNDCTL_COPR_WCODE) -COMPATIBLE_IOCTL(SNDCTL_COPR_RUN) -COMPATIBLE_IOCTL(SNDCTL_COPR_HALT) -COMPATIBLE_IOCTL(SNDCTL_COPR_SENDMSG) -COMPATIBLE_IOCTL(SNDCTL_COPR_RCVMSG) -/* Big M for sound/OSS */ -COMPATIBLE_IOCTL(SOUND_MIXER_READ_VOLUME) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_BASS) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_TREBLE) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_SYNTH) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_PCM) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_SPEAKER) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_MIC) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_CD) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_IMIX) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_ALTPCM) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECLEV) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_IGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_OGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE1) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE2) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE3) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL1)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL2)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL3)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_PHONEIN)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_PHONEOUT)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_VIDEO)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_RADIO)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_MONITOR)) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_MUTE) -/* SOUND_MIXER_READ_ENHANCE, same value as READ_MUTE */ -/* SOUND_MIXER_READ_LOUD, same value as READ_MUTE */ -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECSRC) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_DEVMASK) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECMASK) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_STEREODEVS) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_CAPS) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_VOLUME) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_BASS) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_TREBLE) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_SYNTH) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_PCM) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_SPEAKER) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_MIC) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_CD) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_IMIX) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_ALTPCM) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_RECLEV) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_IGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_OGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE1) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE2) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE3) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL1)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL2)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL3)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_PHONEIN)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_PHONEOUT)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_VIDEO)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_RADIO)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_MONITOR)) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_MUTE) -/* SOUND_MIXER_WRITE_ENHANCE, same value as WRITE_MUTE */ -/* SOUND_MIXER_WRITE_LOUD, same value as WRITE_MUTE */ -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_RECSRC) -COMPATIBLE_IOCTL(SOUND_MIXER_INFO) -COMPATIBLE_IOCTL(SOUND_OLD_MIXER_INFO) -COMPATIBLE_IOCTL(SOUND_MIXER_ACCESS) -COMPATIBLE_IOCTL(SOUND_MIXER_AGC) -COMPATIBLE_IOCTL(SOUND_MIXER_3DSE) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE1) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE2) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE3) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE4) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE5) -COMPATIBLE_IOCTL(SOUND_MIXER_GETLEVELS) -COMPATIBLE_IOCTL(SOUND_MIXER_SETLEVELS) -COMPATIBLE_IOCTL(OSS_GETVERSION) /* Raw devices */ COMPATIBLE_IOCTL(RAW_SETBIND) COMPATIBLE_IOCTL(RAW_GETBIND) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f8d4a419f3af..3f25c9d7b7df 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2732,6 +2732,10 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long static long snd_pcm_oss_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) { + /* + * Everything is compatbile except SNDCTL_DSP_MAPINBUF/SNDCTL_DSP_MAPOUTBUF, + * which are not implemented for the native case either + */ return snd_pcm_oss_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); } #else diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index fc9bcd47d6a4..cda6bbfedd22 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -384,6 +384,7 @@ static const struct file_operations mixer_fops = .owner = THIS_MODULE, .llseek = no_llseek, .unlocked_ioctl = mixer_unlocked_ioctl, + .compat_ioctl = generic_compat_ioctl_ptrarg, .open = mixer_open, .release = mixer_release, }; @@ -1167,6 +1168,7 @@ static const struct file_operations sq_fops = .write = sq_write, .poll = sq_poll, .unlocked_ioctl = sq_unlocked_ioctl, + .compat_ioctl = generic_compat_ioctl_ptrarg, .open = sq_open, .release = sq_release, }; -- 2.18.0

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak of private data
by Takashi Sakamoto 13 Sep '18

13 Sep '18

Although private data of sound card instance is usually allocated in the tail of the instance, drivers in ALSA firewire stack allocate the private data before allocating the instance. In this case, the private data should be released explicitly at .private_free callback of the instance. This commit fixes memory leak following to the above design. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 1e5b2c802635..fd34ef2ac679 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -130,6 +130,7 @@ static void oxfw_free(struct snd_oxfw *oxfw) kfree(oxfw->spec); mutex_destroy(&oxfw->mutex); + kfree(oxfw); } /* -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: firewire-tascam: fix memory leak of private data
by Takashi Sakamoto 13 Sep '18

13 Sep '18

Although private data of sound card instance is usually allocated in the tail of the instance, drivers in ALSA firewire stack allocate the private data before allocating the instance. In this case, the private data should be released explicitly at .private_free callback of the instance. This commit fixes memory leak following to the above design. Fixes: b610386c8afb ('ALSA: firewire-tascam: deleyed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/tascam/tascam.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 44ad41fb7374..d3fdc463a884 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -93,6 +93,7 @@ static void tscm_free(struct snd_tscm *tscm) fw_unit_put(tscm->unit); mutex_destroy(&tscm->mutex); + kfree(tscm); } static void tscm_card_free(struct snd_card *card) -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: firewire-digi00x: fix memory leak of private data
by Takashi Sakamoto 13 Sep '18

13 Sep '18

Although private data of sound card instance is usually allocated in the tail of the instance, drivers in ALSA firewire stack allocate the private data before allocating the instance. In this case, the private data should be released explicitly at .private_free callback of the instance. This commit fixes memory leak following to the above design. Fixes: 86c8dd7f4da3 ('ALSA: firewire-digi00x: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/digi00x/digi00x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f5e1d23f31a..ef689997d6a5 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -49,6 +49,7 @@ static void dg00x_free(struct snd_dg00x *dg00x) fw_unit_put(dg00x->unit); mutex_destroy(&dg00x->mutex); + kfree(dg00x); } static void dg00x_card_free(struct snd_card *card) -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: remove unneeded static set .owner field in platform_driver" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch ASoC: remove unneeded static set .owner field in platform_driver 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 e1e38ea14ea3294546e6350d05a1376197a73589 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Wed, 12 Sep 2018 11:41:49 +0800 Subject: [PATCH] ASoC: remove unneeded static set .owner field in platform_driver platform_driver_register will set the .owner field. So it is safe to remove the redundant assignment. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 1 - sound/soc/mediatek/mt6797/mt6797-mt6351.c | 1 - sound/soc/rockchip/rk3288_hdmi_analog.c | 1 - 3 files changed, 3 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 89f34efd9747..e5d49e6e2f99 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -150,7 +150,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = { static struct platform_driver mt2701_wm8960_machine = { .driver = { .name = "mt2701-wm8960", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt2701_wm8960_machine_dt_match, #endif diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index b1558c57b9ca..6e578e830e42 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -205,7 +205,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = { static struct platform_driver mt6797_mt6351_driver = { .driver = { .name = "mt6797-mt6351", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt6797_mt6351_dt_match, #endif diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index 929b3fe289b0..a472d5eb2950 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = { .probe = snd_rk_mc_probe, .driver = { .name = DRV_NAME, - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = rockchip_sound_of_match, }, -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2] ASoC: core: add support to card rebind
by Srinivas Kandagatla 13 Sep '18

13 Sep '18

Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - None, just rebased on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 85 ++++++++++++++++++++++++++++++-------------- 2 files changed, 60 insertions(+), 27 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 821bf99992b8..46521b416aa1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1098,6 +1098,7 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head component_dev_list; + struct list_head list; struct list_head widgets; struct list_head paths; @@ -1374,6 +1375,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); + INIT_LIST_HEAD(&card->list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e9367aacc0c..bc99bd5b2065 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,7 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -2681,6 +2682,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dai(rtd, j, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2690,7 +2718,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2722,28 +2749,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int j; - - for_each_rtd_codec_dai(rtd, j, codec_dai) { - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2753,12 +2775,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3102,7 +3120,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3142,6 +3160,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3169,6 +3199,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: soc-core: remove dai->driver NULL check
by Kuninori Morimoto 13 Sep '18

13 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is strange if it has "dai" but doesn't have "dai->driver". And more over "dai->driver->xxx" is used everywhere without "dai->driver" pointer NULL checking. It got Oops already if "dai->driver" was NULL. Let's remove un-needed "dai->driver" NULL check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 41fef33..1f859cc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2524,8 +2524,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->driver == NULL) - return -EINVAL; if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; return dai->driver->ops->set_fmt(dai, fmt); @@ -2672,9 +2670,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction) { - if (!dai->driver) - return -ENOTSUPP; - if (dai->driver->ops->mute_stream) return dai->driver->ops->mute_stream(dai, mute, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && -- 2.7.4

2 1

[alsa-devel] Applied "Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation"" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" 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 542cedec53c9e8b73f3f05bf8468823598c50489 Mon Sep 17 00:00:00 2001 From: Yu Zhao <yuzhao(a)google.com> Date: Tue, 11 Sep 2018 15:12:46 -0600 Subject: [PATCH] Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" This reverts commit 12eeeb4f4733bbc4481d01df35933fc15beb8b19. The patch doesn't fix accessing memory with null pointer in skl_interrupt(). There are two problems: 1) skl_init_chip() is called twice, before and after dma buffer is allocate. The first call sets bus->chip_init which prevents the second from initializing bus->corb.buf and rirb.buf from bus->rb.area. 2) snd_hdac_bus_init_chip() enables interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers. There is a small window which skl_interrupt() can be called if irq has been acquired. If so, it crashes when using null dma buffer pointers. Will fix the problems in the following patches. Also attaching the crash for future reference. [ 16.949148] general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI <snipped> [ 16.950903] Call Trace: [ 16.950906] <IRQ> [ 16.950918] skl_interrupt+0x19e/0x2d6 [snd_soc_skl] [ 16.950926] ? dma_supported+0xb5/0xb5 [snd_soc_skl] [ 16.950933] __handle_irq_event_percpu+0x27a/0x6c8 [ 16.950937] ? __irq_wake_thread+0x1d1/0x1d1 [ 16.950942] ? __do_softirq+0x57a/0x69e [ 16.950944] handle_irq_event_percpu+0x95/0x1ba [ 16.950948] ? _raw_spin_unlock+0x65/0xdc [ 16.950951] ? __handle_irq_event_percpu+0x6c8/0x6c8 [ 16.950953] ? _raw_spin_unlock+0x65/0xdc [ 16.950957] ? time_cpufreq_notifier+0x483/0x483 [ 16.950959] handle_irq_event+0x89/0x123 [ 16.950962] handle_fasteoi_irq+0x16f/0x425 [ 16.950965] handle_irq+0x1fe/0x28e [ 16.950969] do_IRQ+0x6e/0x12e [ 16.950972] common_interrupt+0x7a/0x7a [ 16.950974] </IRQ> <snipped> [ 16.951031] RIP: snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] RSP: ffff88015c807c08 [ 16.951036] ---[ end trace 58bf9ece1775bc92 ]--- Fixes: 2eeeb4f4733b ("ASoC: Intel: Skylake: Acquire irq after RIRB allocation") Signed-off-by: Yu Zhao <yuzhao(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cf09721ca13e..dce649485649 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -838,7 +838,11 @@ static int skl_first_init(struct hdac_bus *bus) snd_hdac_bus_parse_capabilities(bus); + if (skl_acquire_irq(bus, 0) < 0) + return -EBUSY; + pci_set_master(pci); + synchronize_irq(bus->irq); gcap = snd_hdac_chip_readw(bus, GCAP); dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); @@ -871,12 +875,6 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - err = skl_acquire_irq(bus, 0); - if (err < 0) - return err; - - synchronize_irq(bus->irq); - /* initialize chip */ skl_init_pci(skl); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "sound: enable interrupt after dma buffer initialization" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch sound: enable interrupt after dma buffer initialization 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 b61749a89f826eb61fc59794d9e4697bd246eb61 Mon Sep 17 00:00:00 2001 From: Yu Zhao <yuzhao(a)google.com> Date: Tue, 11 Sep 2018 15:14:04 -0600 Subject: [PATCH] sound: enable interrupt after dma buffer initialization In snd_hdac_bus_init_chip(), we enable interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers. If irq has been acquired and irq handler uses the dma buffer, kernel may crash when interrupt comes in. Fix the problem by postponing enabling irq after dma buffer initialization. And warn once on null dma buffer pointer during the initialization. Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Yu Zhao <yuzhao(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/hda/hdac_controller.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 560ec0986e1a..11057d9f84ec 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -40,6 +40,8 @@ static void azx_clear_corbrp(struct hdac_bus *bus) */ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) { + WARN_ON_ONCE(!bus->rb.area); + spin_lock_irq(&bus->reg_lock); /* CORB set up */ bus->corb.addr = bus->rb.addr; @@ -479,13 +481,15 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) /* reset controller */ azx_reset(bus, full_reset); - /* initialize interrupts */ + /* clear interrupts */ azx_int_clear(bus); - azx_int_enable(bus); /* initialize the codec command I/O */ snd_hdac_bus_init_cmd_io(bus); + /* enable interrupts after CORB/RIRB buffers are initialized above */ + azx_int_enable(bus); + /* program the position buffer */ if (bus->use_posbuf && bus->posbuf.addr) { snd_hdac_chip_writel(bus, DPLBASE, (u32)bus->posbuf.addr); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "sound: don't call skl_init_chip() to reset intel skl soc" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch sound: don't call skl_init_chip() to reset intel skl soc 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 75383f8d39d4c0fb96083dd460b7b139fbdac492 Mon Sep 17 00:00:00 2001 From: Yu Zhao <yuzhao(a)google.com> Date: Tue, 11 Sep 2018 15:15:16 -0600 Subject: [PATCH] sound: don't call skl_init_chip() to reset intel skl soc Internally, skl_init_chip() calls snd_hdac_bus_init_chip() which 1) sets bus->chip_init to prevent multiple entrances before device is stopped; 2) enables interrupt. We shouldn't use it for the purpose of resetting device only because 1) when we really want to initialize device, we won't be able to do so; 2) we are ready to handle interrupt yet, and kernel crashes when interrupt comes in. Rename azx_reset() to snd_hdac_bus_reset_link(), and use it to reset device properly. Fixes: 60767abcea3d ("ASoC: Intel: Skylake: Reset the controller in probe") Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Yu Zhao <yuzhao(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_controller.c | 7 ++++--- sound/soc/intel/skylake/skl.c | 2 +- 3 files changed, 6 insertions(+), 4 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ab5ee3ef2198..207e816ce6e1 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -384,6 +384,7 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus); void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus); void snd_hdac_bus_enter_link_reset(struct hdac_bus *bus); void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus); +int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset); void snd_hdac_bus_update_rirb(struct hdac_bus *bus); int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status, diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 11057d9f84ec..74244d8e2909 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -385,7 +385,7 @@ void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_bus_exit_link_reset); /* reset codec link */ -static int azx_reset(struct hdac_bus *bus, bool full_reset) +int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) { if (!full_reset) goto skip_reset; @@ -410,7 +410,7 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) skip_reset: /* check to see if controller is ready */ if (!snd_hdac_chip_readb(bus, GCTL)) { - dev_dbg(bus->dev, "azx_reset: controller not ready!\n"); + dev_dbg(bus->dev, "controller not ready!\n"); return -EBUSY; } @@ -425,6 +425,7 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) return 0; } +EXPORT_SYMBOL_GPL(snd_hdac_bus_reset_link); /* enable interrupts */ static void azx_int_enable(struct hdac_bus *bus) @@ -479,7 +480,7 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) return false; /* reset controller */ - azx_reset(bus, full_reset); + snd_hdac_bus_reset_link(bus, full_reset); /* clear interrupts */ azx_int_clear(bus); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index dce649485649..1d17be0f78a0 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -834,7 +834,7 @@ static int skl_first_init(struct hdac_bus *bus) return -ENXIO; } - skl_init_chip(bus, true); + snd_hdac_bus_reset_link(bus, true); snd_hdac_bus_parse_capabilities(bus); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ALSA: hda: Fix implicit definition of pci_iomap() on SH" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch ALSA: hda: Fix implicit definition of pci_iomap() on SH 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 d9b84a15892c02334ac8a5c28865ae54168d9b22 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 12 Sep 2018 12:31:32 +0100 Subject: [PATCH] ALSA: hda: Fix implicit definition of pci_iomap() on SH Include asm/io.h directly so we've got a definition of pci_iomap(), the current set of includes do this implicitly on most architectures but not on SH. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index a585d0ec6d77..6ea04f889809 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -31,6 +31,7 @@ #include <linux/types.h> #include <linux/io.h> #include <linux/pci.h> +#include <asm/io.h> #include <sound/core.h> #include <sound/hda_codec.h> #include "hda_local.h" -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 1/3] Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation"
by Yu Zhao 13 Sep '18

13 Sep '18

This reverts commit 12eeeb4f4733bbc4481d01df35933fc15beb8b19. The patch doesn't fix accessing memory with null pointer in skl_interrupt(). There are two problems: 1) skl_init_chip() is called twice, before and after dma buffer is allocate. The first call sets bus->chip_init which prevents the second from initializing bus->corb.buf and rirb.buf from bus->rb.area. 2) snd_hdac_bus_init_chip() enables interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers. There is a small window which skl_interrupt() can be called if irq has been acquired. If so, it crashes when using null dma buffer pointers. Will fix the problems in the following patches. Also attaching the crash for future reference. [ 16.949148] general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI [ 16.950829] gsmi: Log Shutdown Reason 0x03 [ 16.950830] Modules linked in: uvcvideo(+) videobuf2_vmalloc snd_soc_skl(+) videobuf2_memops videobuf2_v4l2 videobuf2_core snd_soc_skl_ipc snd_soc_sst_ipc snd_soc_sst_dsp snd_soc_acpi snd_hda_ext_core snd_hda_core snd_soc_max98357a acpi_als snd_soc_da7219 lzo lzo_compress zram snd_seq_dummy snd_seq snd_seq_device bridge stp llc ipt_MASQUERADE nf_nat_masquerade_ipv4 xt_mark fuse cfg80211 iio_trig_sysfs cros_ec_sensors cros_ec_sensors_ring cros_ec_sensors_core industrialio_triggered_buffer kfifo_buf industrialio smsc95xx usbnet mii joydev [ 16.950874] CPU: 0 PID: 1083 Comm: chrome Not tainted 4.14.64 #14 [ 16.950875] Hardware name: Google Yorp/Yorp, BIOS Google_Yorp.10985.0.2018_08_20_1648 08/17/2018 [ 16.950878] task: ffff88015b1c2b80 task.stack: ffff880155f30000 [ 16.950887] RIP: 0010:snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] [ 16.950889] RSP: 0000:ffff88015c807c08 EFLAGS: 00010003 [ 16.950891] RAX: 0000000000000101 RBX: 000000000000080c RCX: 1ffff10026822185 [ 16.950893] RDX: dffffc0000000000 RSI: ffff88015b1c2b80 RDI: ffffc90000514058 [ 16.950894] RBP: ffff88015c807c68 R08: 0000000000000000 R09: 0000000000000000 [ 16.950895] R10: 0000000000000000 R11: ffffffffc043074f R12: 0000000000000800 [ 16.950897] R13: 0000000000000001 R14: 0000000000000002 R15: 1ffff10026822119 [ 16.950899] FS: 00007d85924cc740(0000) GS:ffff88015c800000(0000) knlGS:0000000000000000 [ 16.950900] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 16.950902] CR2: 000058a54df16038 CR3: 00000001551c0000 CR4: 00000000003406f0 [ 16.950903] Call Trace: [ 16.950906] <IRQ> [ 16.950918] skl_interrupt+0x19e/0x2d6 [snd_soc_skl] [ 16.950926] ? dma_supported+0xb5/0xb5 [snd_soc_skl] [ 16.950933] __handle_irq_event_percpu+0x27a/0x6c8 [ 16.950937] ? __irq_wake_thread+0x1d1/0x1d1 [ 16.950942] ? __do_softirq+0x57a/0x69e [ 16.950944] handle_irq_event_percpu+0x95/0x1ba [ 16.950948] ? _raw_spin_unlock+0x65/0xdc [ 16.950951] ? __handle_irq_event_percpu+0x6c8/0x6c8 [ 16.950953] ? _raw_spin_unlock+0x65/0xdc [ 16.950957] ? time_cpufreq_notifier+0x483/0x483 [ 16.950959] handle_irq_event+0x89/0x123 [ 16.950962] handle_fasteoi_irq+0x16f/0x425 [ 16.950965] handle_irq+0x1fe/0x28e [ 16.950969] do_IRQ+0x6e/0x12e [ 16.950972] common_interrupt+0x7a/0x7a [ 16.950974] </IRQ> [ 16.950976] RIP: 0033:0x58097f61a5c0 [ 16.950978] RSP: 002b:00007ffe95c971a8 EFLAGS: 00000206 ORIG_RAX: ffffffffffffffbc [ 16.950980] RAX: 000058097f61a5c0 RBX: 00000e4ac5220560 RCX: 0000000000004e10 [ 16.950982] RDX: 000058098563df20 RSI: 00007ffe95c97250 RDI: 00000e4ac5220500 [ 16.950983] RBP: 00007ffe95c97410 R08: 0000000000000000 R09: 00007ffe95c97250 [ 16.950984] R10: 0000000000000000 R11: 0000000000000000 R12: 00000e4ac5220560 [ 16.950986] R13: 00000e4ac5220560 R14: 00000e4ac47c9650 R15: 0000580987646350 [ 16.950988] Code: 74 12 48 89 df e8 eb 2d 8e cd 48 ba 00 00 00 00 00 fc ff df 4c 8b 23 44 89 f0 83 c8 01 0f b7 c0 49 8d 1c 84 48 89 d8 48 c1 e8 03 <8a> 04 10 84 c0 0f 85 da 01 00 00 44 8b 3b 41 0f b7 c6 49 8d 1c [ 16.951031] RIP: snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] RSP: ffff88015c807c08 [ 16.951036] ---[ end trace 58bf9ece1775bc92 ]--- [ 16.956871] Kernel panic - not syncing: Fatal exception in interrupt [ 16.956888] Kernel Offset: 0xc800000 from 0xffffffff81000000 (relocation range: 0xffffffff80000000-0xffffffffbfffffff) Signed-off-by: Yu Zhao <yuzhao(a)google.com> --- sound/soc/intel/skylake/skl.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cf09721ca13e..dce649485649 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -838,7 +838,11 @@ static int skl_first_init(struct hdac_bus *bus) snd_hdac_bus_parse_capabilities(bus); + if (skl_acquire_irq(bus, 0) < 0) + return -EBUSY; + pci_set_master(pci); + synchronize_irq(bus->irq); gcap = snd_hdac_chip_readw(bus, GCAP); dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); @@ -871,12 +875,6 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - err = skl_acquire_irq(bus, 0); - if (err < 0) - return err; - - synchronize_irq(bus->irq); - /* initialize chip */ skl_init_pci(skl); -- 2.19.0.rc2.392.g5ba43deb5a-goog

4 19

[alsa-devel] [PATCH 0/2][RFC] ALSA: firewire-lib: change packet processing timing in process context
by Takashi Sakamoto 13 Sep '18

13 Sep '18

In IEC 61883-1/6 engine in ALSA firewire stack, a work to process packet is done in three places: 1. bottom-half of handler for interrupts scheduled on 1394OHCI controller 2. a callback of struct 'snd_pcm_ops.pointer()' 3. a callback of struct 'snd_pcm_ops.ack()' This patchset adds optimization for 3., and remove 2. for efficiency. In Linux kernel v2.6.39, 1. and 2. was committed in initial implementation of this engine at a commit 31ef9134eb52('ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver'). This engine schedules hardware interrupts 16 packets (=INTERRUPT_INTERVAL) thus 1. appears in software IRQ context every 2 msec (= 16packets * 1sec / 8,000cycle). In Linux kernel v3.5, an idea of 2. was introduced at a commit e9148dddc3c7('ALSA: firewire-lib: flush completed packets when reading PCM position') to reduce timing gap between queueing/dequeueing PCM frames and scheduling packet transmission. Additionally, it has an effect to reduce load of the software IRQ context. It's done in a call of ioctl(2) with SNDRV_PCM_IOCTL_HWSYNC. As a result, in mmap programming scenario, a call of snd_pcm_hwsync() can process packets for recent isochronous cycle. while (1): snd_pcm_hwsync() ->ioctl(HWSYNC) ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() In Linux kernel v4.13, ALSA PCM core was improved in an aspect of updating appl_ptr at a commit 9027c4639ef1('ALSA: pcm: Call ack() whenever appl_ptr is updated'). This commit ensures callback of 'struct snd_pcm_ops.ack()' when updating appl_ptr. This callback can be done in below ioctl(2) requests: - SNDRV_PCM_IOCTL_REWIND - SNDRV_PCM_IOCTL_FORWARD - SNDRV_PCM_IOCTL_SYNC_PTR - SNDRV_PCM_IOCTL_READ[I|N]_FRAMES - SNDRV_PCM_IOCTL_WRITE[I|N]_FRAMES As I notes in a commit 875becf8412c('ALSA: firewire: process packets in 'struct snd_pcm_ops.ack' callback'), 3. works less efficiently because in usual PCM operation the above ioctl(2) requests are not used in event loop. At the same time of the appl_ptr improvement, a new flag of PCM hardware definition was added at a commit 42f945970af9('ALSA: pcm: Add the explicit appl_ptr sync support'), and relevant changes were added to ALSA PCM core and alsa-lib v1.1.5. This flag expects userspace to call ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR request per iteration of PCM frame queueing/dequeueing in mmap programming scenario below: while (1): snd_pcm_hwsync() ->ioctl(HWSYNC) ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() ->ioctl(SYNC_PTR) This patchset aims to support the new flag, SNDRV_PCM_INFO_SYNC_APPLPTR, to tell userspace to call ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR after queueing/dequeueing PCM frames in each iteration of event dispatcher. As a result: while (1): snd_pcm_hwsync() ->ioctl(HWSYNC) ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() ->ioctl(SYNC_PTR) ->struct snd_pcm_ops.ack() = amdtp_stream_pcm_ack() ->fw_iso_context_flush_completions() This patchset also adds optimization for timing to process packets in process context. As the above call graph shows, support of the flag brings closest two calls to process packet against intervals of isochronous cycle and it is not enough efficient. This patchset also cancel packet processing in .pointer() callback. As a result: while (1): snd_pcm_hwsync() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() ->ioctl(SYNC_PTR) ->struct snd_pcm_ops.ack() = amdtp_stream_pcm_ack() ->fw_iso_context_flush_completions() Takashi Sakamoto (2): ALSA: firewire-lib: adopt SYNC_APPLPTR flag to flush packets after queueing/dequeueing PCM frames ALSA: firewire-lib: cancel processing packets in struct snd_pcm_ops.pointer() call sound/firewire/amdtp-stream.c | 34 ++-------------------------------- sound/firewire/amdtp-stream.h | 12 +++++++++++- 2 files changed, 13 insertions(+), 33 deletions(-) -- 2.17.1

2 4

[alsa-devel] What causes a STOP in the trigger?
by Rob 13 Sep '18

13 Sep '18

I'm seeing a weird issue where ALSA doesn't ever hit the pointer function, and we go straight from START to STOP. Debugging the DMA, we get IRQs, I call the period_elapsed. It looks like we hit STOP before we even get the first IRQ. I've checked my parentheses, and theres no overrun or underrun messages. I don't hit the close function, nor any other functions until i control-C. Once I do that, I get a large series of pointer() printks showing multiples of IRQ positions, as if it queued up the commands... I thought I was IRQing too fast maybe, but as mentioned above, the irqreturn function doesn't even get called before we STOP. What in aplay would instantly force a STOP? Audio plays a few frames (I can hear a chord), then stalls and repeats until i Ctrl-C. Thanks, Rob

2 1

[alsa-devel] [PATCH] ALSA: rawmidi: A lightweight function to discard pending bytes
by Takashi Iwai 13 Sep '18

13 Sep '18

For discarding the pending bytes on rawmidi, we process with a loop of snd_rawmidi_transmit() which is just a waste of CPU power. Implement a lightweight API function to discard the pending bytes and the proceed the ring buffer instantly, and use it instead of open codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/rawmidi.h | 1 + sound/core/rawmidi.c | 22 ++++++++++++++++++++++ sound/core/seq/seq_virmidi.c | 4 +--- sound/usb/midi.c | 3 +-- 4 files changed, 25 insertions(+), 5 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 6665cb29e1a2..3b5a061132b6 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -171,6 +171,7 @@ int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count); int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count); +int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream); /* main midi functions */ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 08d5662039e3..ee601d7f0926 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1236,6 +1236,28 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, } EXPORT_SYMBOL(snd_rawmidi_transmit); +/** + * snd_rawmidi_proceed - Discard the all pending bytes and proceed + * @substream: rawmidi substream + * + * Return: the number of discarded bytes + */ +int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + int count = 0; + + spin_lock_irqsave(&runtime->lock, flags); + if (runtime->avail < runtime->buffer_size) { + count = runtime->buffer_size - runtime->avail; + __snd_rawmidi_transmit_ack(substream, count); + } + spin_unlock_irqrestore(&runtime->lock, flags); + return count; +} +EXPORT_SYMBOL(snd_rawmidi_proceed); + static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, const unsigned char __user *userbuf, const unsigned char *kernelbuf, diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index cb988efd1ed0..e5a40795914a 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -149,9 +149,7 @@ static void snd_vmidi_output_work(struct work_struct *work) /* discard the outputs in dispatch mode unless subscribed */ if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - char buf[32]; - while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0) - ; /* ignored */ + snd_rawmidi_proceed(substream); return; } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index dcfc546d81b9..b737f0ec77d0 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1175,8 +1175,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, if (port->ep->umidi->disconnected) { /* gobble up remaining bytes to prevent wait in * snd_rawmidi_drain_output */ - while (!snd_rawmidi_transmit_empty(substream)) - snd_rawmidi_transmit_ack(substream, 1); + snd_rawmidi_proceed(substream); return; } tasklet_schedule(&port->ep->tasklet); -- 2.18.0

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable runtime PM on AMD/ATI GPU only for discrete GPU
by Takashi Iwai 13 Sep '18

13 Sep '18

The recent change of vga_switcheroo allowed the runtime PM for HD-audio on AMD GPU, but this also resulted in a regression. When the HD-audio controller driver gets runtime-suspended, HD-audio link is turned off, and the hotplug notification is ignored. This leads to the inconsistent audio state (the connection isn't notified and ELD is ignored). The best fix would be to implement the proper ELD notification via the audio component, but it's still not ready. As a quick workaround, this patch adds the check of runtime_idle and allows the runtime suspend only when the vga_switcheroo is bound with discrete GPU. That is, a system with a single GPU would be again without runtime PM (of the controller), as well as the APU that needs to keep the HD-audio link for the hotplug notification and ELD read out. For identifying the audio client connection type, a new API is added to vga_switcheroo. Other than that, the patch does a pretty simple job. Note that this patch won't influence on the HD-audio *codec* runtime PM. When a codec is unused and the system goes to S3/S4 and resumed, the codec won't be resumed immediately because it's unused. This likely keeps the hotplug notification broken, but this is basically the expected behavior, a cost needed for the power-saving. The hotplug state will be activated once when you access to the device (or triggering the power up via reading a codec#* proc file or such). Again, this issue will be solved by the proper audio component notification, but it's still not ready. Until then, user would need to disable the power-saving explicitly if the ELD hotplug notification is required. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200945 Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 20 ++++++++++++++++++++ include/linux/vga_switcheroo.h | 1 + sound/pci/hda/hda_intel.c | 8 ++++++++ 3 files changed, 29 insertions(+) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index a96bf46bc483..89598adbf6be 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -485,6 +485,26 @@ enum vga_switcheroo_state vga_switcheroo_get_client_state(struct pci_dev *pdev) } EXPORT_SYMBOL(vga_switcheroo_get_client_state); +/** + * vga_switcheroo_get_client_id() - obtain client id of a given client + * @pdev: client pci device + */ +enum vga_switcheroo_client_id vga_switcheroo_get_client_id(struct pci_dev *pdev) +{ + struct vga_switcheroo_client *client; + enum vga_switcheroo_client_id ret; + + mutex_lock(&vgasr_mutex); + client = find_client_from_pci(&vgasr_priv.clients, pdev); + if (!client) + ret = VGA_SWITCHEROO_UNKNOWN_ID; + else + ret = client_id(client); + mutex_unlock(&vgasr_mutex); + return ret; +} +EXPORT_SYMBOL(vga_switcheroo_get_client_id); + /** * vga_switcheroo_unregister_client() - unregister client * @pdev: client pci device diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index a34539b7f750..e52fcd7b68f3 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -167,6 +167,7 @@ int vga_switcheroo_process_delayed_switch(void); bool vga_switcheroo_client_probe_defer(struct pci_dev *pdev); enum vga_switcheroo_state vga_switcheroo_get_client_state(struct pci_dev *dev); +enum vga_switcheroo_client_id vga_switcheroo_get_client_id(struct pci_dev *dev); int vga_switcheroo_init_domain_pm_ops(struct device *dev, struct dev_pm_domain *domain); void vga_switcheroo_fini_domain_pm_ops(struct device *dev); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..48ab090648a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1201,6 +1201,14 @@ static int azx_runtime_idle(struct device *dev) azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; +#ifdef SUPPORT_VGA_SWITCHEROO + /* ELD notification gets broken on AMD GPUs when HD-audio bus is off */ + if ((chip->pci->vendor == PCI_VENDOR_ID_ATI || + chip->pci->vendor == PCI_VENDOR_ID_AMD) && + vga_switcheroo_get_client_id(chip->pci) != VGA_SWITCHEROO_DIS) + return -EBUSY; +#endif + return 0; } -- 2.18.0

1 1

[alsa-devel] [PATCH RFC 0/5] Add devres support to card resources
by Takashi Iwai 13 Sep '18

13 Sep '18

Hi, the devres or device-managed resource allocation is a popular method among many drivers but we didn't have such a mechanism for the core sound card objects. Here is an attempt to implement that. The first patch gives the devm-version of page allocator helper, and the second patch provides the devm-version of snd_card_new(). The rest patches are examples of the new API usages. As of now, this patch series is still an RFC. The usefulness of this is still unclear, but I believe this can be used in all legacy ALSA drivers pretty well. For ASoC, this would help very little, supposedly, though. The latest patches are found in topic/devres branch of my sound.git tree, too. thanks, Takashi === Takashi Iwai (5): ALSA: core: Add device-managed page allocator helper ALSA: core: Add managed card creation ALSA: intel8x0: Allocate resources with device-managed APIs ALSA: atiixp: Allocate resources with device-managed APIs ALSA: hda: Allocate resources with device-managed APIs include/sound/core.h | 5 ++ include/sound/memalloc.h | 4 + sound/core/init.c | 95 +++++++++++++++++++++- sound/core/memalloc.c | 88 +++++++++++++++----- sound/pci/atiixp.c | 104 +++++++----------------- sound/pci/atiixp_modem.c | 104 +++++++----------------- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 43 +++------- sound/pci/intel8x0.c | 143 +++++++++++---------------------- sound/pci/intel8x0m.c | 143 +++++++++++---------------------- 10 files changed, 332 insertions(+), 398 deletions(-) -- 2.18.0

1 5

[alsa-devel] [PATCH] ALSA: intel8x0m: Register irq handler after register initializations
by Takashi Iwai 13 Sep '18

13 Sep '18

The interrupt handler has to be acquired after the other resource initialization when allocated with IRQF_SHARED. Otherwise it's triggered before the resource gets ready, and may lead to unpleasant behavior. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/intel8x0m.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 943a726b1c1b..c84629190cba 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card, } port_inited: - if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { - dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_intel8x0m_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - pci_set_master(pci); - synchronize_irq(chip->irq); - /* initialize offsets */ chip->bdbars_count = 2; tbl = intel_regs; @@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card, chip->int_sta_reg = ICH_REG_GLOB_STA; chip->int_sta_mask = int_sta_masks; + pci_set_master(pci); + if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) { snd_intel8x0m_free(chip); return err; } + if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, + KBUILD_MODNAME, chip)) { + dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); + snd_intel8x0m_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_intel8x0m_free(chip); return err; -- 2.18.0

1 0

[alsa-devel] [RFC - AAF PCM plugin 0/5] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 13 Sep '18

13 Sep '18

Hi all, This patchset introduces the AAF plugin to the alsa-plugins project which enables TSN audio applications on Linux. For those not familiarized with TSN, it stands for Time-Sensitive Networking, and it's a set of IEEE technologies (enhancements to IEEE 802.1Q and Ethernet protocols) that provide precise time synchronization, bounded latency and application interoperability to bridged networks. Those technologies enable time-sensitive applications such as audio, video and control loops to run on top of bridged networks, co-existing with regular applications. TSN technologies are a super set of Audio Video Bridging (AVB) technologies also developed by IEEE. AVTP is the protocol defined to transport data in a TSN system, and AAF is one of the formats defined by AVTP to encapsulate audio data. AAF is specified in Chapter 7 from the AVTP spec [1]. This work is part of the effort to enable TSN technologies on upstream Linux ecosystem. All building-blocks required to enable TSN audio applications have been already developed and pushed to upstream projects. Time synchronization features are provided by linuxptp project [2], bounded latency features are provided by Linux Traffic Control subsystem since kernel version 4.15 [3], and AVTP packetization is provided by libavtp [4]. What is currently missing in the ecosystem to enable TSN audio applications is a piece of software that plumbs it all together and interfaces with Linux Audio system properly. That's the point of the AAF plugin introduced here. The AAF plugin is a PCM plugin that uses AVTP protocol to transmit/receive audio data through a TSN capable network. When operating in playback mode, the plugin reads PCM samples from the audio buffer, encapsulates into AVTP packets and transmits to the network, mimicking a typical AVTP Talker. When operating in capture mode, the plugin receives AVTP packets from the network, retrieves the PCM samples, and present them to alsa-lib layer at the presentation time, mimicking a typical AVTP Listener. Details about the plugin implementation are provided in the patches description, and information on how to configure and run the AAF plugin is provided in doc/aaf.txt file. Best regards, Andre [1] 1722-2016 - IEEE Standard for a Transport Protocol for Time-Sensitive Applications in Bridged Local Area Networks [2] http://linuxptp.sourceforge.net [3] https://patchwork.ozlabs.org/cover/826678/ [4] https://github.com/AVnu/libavtp Andre Guedes (5): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1172 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 110 ++++++ 5 files changed, 1303 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

4 26

Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Connor McAdams 12 Sep '18

12 Sep '18

You get that terminal output while running the program? Those vfio_region_writes shouldn't be visible from the terminal... they should be fed into QemuHDADump, and then it outputs it's own stuff. If that is the output you're seeing, you might not have set up logging to go to stderr. On Wed, Sep 12, 2018 at 1:05 PM, Håvard <hovardslill(a)gmail.com> wrote: > I think this is right! > > I first booted up, then tested speakers (That I could not hear) and then > tested microphone (Which worked perfectly) in windows settings. > > -Håvard > > Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams > <conmanx360(a)gmail.com>: >> >> Also, sorry to keep sending more replies, make sure the only trace you >> have on is vfio_region_write, and not the read one. The read spams up >> the console. >> >> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >> wrote: >> > Reason being, it may not have identified the CORB buffer properly or >> > something. That's very odd to have no data. Never seen that before. >> > >> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams <conmanx360(a)gmail.com> >> > wrote: >> >> Could you try again and make a copy of your terminal output? >> >> >> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> wrote: >> >>> Hi again. >> >>> Sorry for not noticing all files were empty, did it not run for long >> >>> enough? >> >>> >> >>> -Håvard >> >>> >> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard <hovardslill(a)gmail.com>: >> >>>> >> >>>> Sorry my bad again it was working perfectly, just missed a step. >> >>>> >> >>>> I'll add the dumps as an attatchment. I had both external headset and >> >>>> microphone plugged in and let the VM run in the background for ~30min >> >>>> while >> >>>> doing other stuff. I did not play music or test my microphone in the >> >>>> VM >> >>>> either, If i did anything wrong, please tell me! >> >>>> >> >>>> It stopped at 0x104f4 >> >>>> >> >>>> -Håvard >> >>>> >> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >> >>>> <conmanx360(a)gmail.com>: >> >>>>> >> >>>>> Hm.... that's odd. They should show up in the folder you ran the >> >>>>> command from. Does your console show any "DumpMem entered..." or >> >>>>> something like that? You may have a permissions error. I've had some >> >>>>> people report that as an issue before. >> >>>>> >> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >> >>>>> wrote: >> >>>>> > Hi! >> >>>>> > >> >>>>> > Took a while, but I think I got it working. Did not see any >> >>>>> > "frame[xx]" >> >>>>> > files though. What dumps do you want? >> >>>>> > >> >>>>> > -Håvard >> >>>>> > >> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard >> >>>>> > <hovardslill(a)gmail.com>: >> >>>>> >> >> >>>>> >> Sorry about that, I forgot to enable a kernel config... >> >>>>> >> >> >>>>> >> -Håvard >> >>>>> >> >> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >> >>>>> >> <conmanx360(a)gmail.com>: >> >>>>> >>> >> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >> >>>>> >>> controls. >> >>>>> >>> You can check what module is loaded by doing lspci -v . It >> >>>>> >>> should say >> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the vfio-bind >> >>>>> >>> script >> >>>>> >>> before trying it? >> >>>>> >>> >> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard <hovardslill(a)gmail.com> >> >>>>> >>> wrote: >> >>>>> >>> > Thank you for the offer! >> >>>>> >>> > >> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >> >>>>> >>> > seem to >> >>>>> >>> > bind my >> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >> >>>>> >>> > to >> >>>>> >>> > function as >> >>>>> >>> > normal when I (think I) have bound it to a stud. >> >>>>> >>> > >> >>>>> >>> > -Håvard >> >>>>> >>> > >> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >> >>>>> >>> > <conmanx360(a)gmail.com>: >> >>>>> >>> >> >> >>>>> >>> >> One thing you could try, is using the program I used to >> >>>>> >>> >> reverse >> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. If >> >>>>> >>> >> your >> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >> >>>>> >>> >> you >> >>>>> >>> >> could >> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >> >>>>> >>> >> capture >> >>>>> >>> >> the commands. I'd be willing to look through the dumps to see >> >>>>> >>> >> if >> >>>>> >>> >> there >> >>>>> >>> >> are any special verbs or anything. >> >>>>> >>> >> >> >>>>> >>> >> You can find the program here: >> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >> >>>>> >>> >> >> >>>>> >>> >> Let me know. >> >>>>> >>> >> >> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard >> >>>>> >>> >> <hovardslill(a)gmail.com> >> >>>>> >>> >> wrote: >> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response before >> >>>>> >>> >> > I >> >>>>> >>> >> > read >> >>>>> >>> >> > your >> >>>>> >>> >> > last one. >> >>>>> >>> >> > >> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >> >>>>> >>> >> > headset >> >>>>> >>> >> > multijacks. >> >>>>> >>> >> > >> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the tip! >> >>>>> >>> >> > >> >>>>> >>> >> > -Håvard >> >>>>> >>> >> > >> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >> >>>>> >>> >> > <hovardslill(a)gmail.com>: >> >>>>> >>> >> > >> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >> >>>>> >>> >> >> >> >>>>> >>> >> >> Do you know where I can ask around for more help on the >> >>>>> >>> >> >> issue? >> >>>>> >>> >> >> I >> >>>>> >>> >> >> don't >> >>>>> >>> >> >> want to give up yet. >> >>>>> >>> >> >> >> >>>>> >>> >> >> -Håvard >> >>>>> >>> >> >> >> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >> >>>>> >>> >> >> <tiwai(a)suse.de>: >> >>>>> >>> >> >> >> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >> >>>>> >>> >> >>> Håvard wrote: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >> >>>>> >>> >> >>> > automatically >> >>>>> >>> >> >>> > changes >> >>>>> >>> >> >>> > to >> >>>>> >>> >> >>> it >> >>>>> >>> >> >>> > in pavucontrol. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Then it likely requires some additional initialization >> >>>>> >>> >> >>> outside >> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >> >>>>> >>> >> >>> vendor-specific. >> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >> >>>>> >>> >> >>> Windows >> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Takashi >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >> >>>>> >>> >> >>> > > Håvard wrote: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Thank you for replying! >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic Playback >> >>>>> >>> >> >>> > > > Volume" >> >>>>> >>> >> >>> > > > in >> >>>>> >>> >> >>> > > hdaanalyzer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >> >>>>> >>> >> >>> > > > made white noise come from the headset. However it >> >>>>> >>> >> >>> > > > did >> >>>>> >>> >> >>> > > > not >> >>>>> >>> >> >>> > > > change or >> >>>>> >>> >> >>> > > react >> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >> >>>>> >>> >> >>> > > > physically. >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >> >>>>> >>> >> >>> > > > either >> >>>>> >>> >> >>> > > > alsamixer >> >>>>> >>> >> >>> or >> >>>>> >>> >> >>> > > > hdaanalyzer. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > It's a mixer mute switch. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > The microphone works perfectly fine under Windows, >> >>>>> >>> >> >>> > > > so I >> >>>>> >>> >> >>> > > > don't >> >>>>> >>> >> >>> > > > think >> >>>>> >>> >> >>> it is >> >>>>> >>> >> >>> > > > the mic pin. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >> >>>>> >>> >> >>> > > wrong >> >>>>> >>> >> >>> > > pin, >> >>>>> >>> >> >>> > > too. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Takashi >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > -Håvard >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi Iwai >> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >> >>>>> >>> >> >>> >: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >> >>>>> >>> >> >>> > > > > Håvard wrote: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Additional relevant info: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three years >> >>>>> >>> >> >>> > > > > > ago >> >>>>> >>> >> >>> > > > > > on >> >>>>> >>> >> >>> > > > > > Sun >> >>>>> >>> >> >>> > > > > > Jun >> >>>>> >>> >> >>> > > 17:15:54 >> >>>>> >>> >> >>> > > > > CEST >> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >> >>>>> >>> >> >>> > > > > > but did >> >>>>> >>> >> >>> > > > > > not >> >>>>> >>> >> >>> > > > > > touch >> >>>>> >>> >> >>> on the >> >>>>> >>> >> >>> > > > > > external microphone problem: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > alsa-info.sh: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help the >> >>>>> >>> >> >>> > > > > > issue. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > There are several of similar bug reports around >> >>>>> >>> >> >>> > > > > > the >> >>>>> >>> >> >>> > > > > > web >> >>>>> >>> >> >>> experiencing >> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >> >>>>> >>> >> >>> > > > > > seems to >> >>>>> >>> >> >>> > > > > > happen >> >>>>> >>> >> >>> > > > > > with >> >>>>> >>> >> >>> all >> >>>>> >>> >> >>> > > > > laptops >> >>>>> >>> >> >>> > > > > > under the G751 name. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >> >>>>> >>> >> >>> > > > > and >> >>>>> >>> >> >>> > > > > unmute/adjust >> >>>>> >>> >> >>> "Mic >> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >> >>>>> >>> >> >>> > > > > you >> >>>>> >>> >> >>> > > > > hear >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from NID >> >>>>> >>> >> >>> > > > > 0x18 >> >>>>> >>> >> >>> > > > > to >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> mixer >> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >> >>>>> >>> >> >>> > > > > outputs. >> >>>>> >>> >> >>> > > > > So >> >>>>> >>> >> >>> > > > > this >> >>>>> >>> >> >>> > > > > can >> >>>>> >>> >> >>> be >> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from >> >>>>> >>> >> >>> the ext >> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >> >>>>> >>> >> >>> > > > > something >> >>>>> >>> >> >>> > > > > outside >> >>>>> >>> >> >>> > > > > HD-audio codec. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > Takashi >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> >> >>>>> >>> >> >> >> >>>>> >>> >> > _______________________________________________ >> >>>>> >>> >> > Alsa-devel mailing list >> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >> >>>>> >>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 2

Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Håvard 12 Sep '18

12 Sep '18

I think this is right! I first booted up, then tested speakers (That I could not hear) and then tested microphone (Which worked perfectly) in windows settings. (My first email was dissaproved as it was too large) Den ons. 12. sep. 2018 kl. 19:05 skrev Håvard <hovardslill(a)gmail.com>: > I think this is right! > > I first booted up, then tested speakers (That I could not hear) and then > tested microphone (Which worked perfectly) in windows settings. > > -Håvard > > Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams < > conmanx360(a)gmail.com>: > >> Also, sorry to keep sending more replies, make sure the only trace you >> have on is vfio_region_write, and not the read one. The read spams up >> the console. >> >> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >> wrote: >> > Reason being, it may not have identified the CORB buffer properly or >> > something. That's very odd to have no data. Never seen that before. >> > >> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams <conmanx360(a)gmail.com> >> wrote: >> >> Could you try again and make a copy of your terminal output? >> >> >> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> >> wrote: >> >>> Hi again. >> >>> Sorry for not noticing all files were empty, did it not run for long >> enough? >> >>> >> >>> -Håvard >> >>> >> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard <hovardslill(a)gmail.com >> >: >> >>>> >> >>>> Sorry my bad again it was working perfectly, just missed a step. >> >>>> >> >>>> I'll add the dumps as an attatchment. I had both external headset and >> >>>> microphone plugged in and let the VM run in the background for >> ~30min while >> >>>> doing other stuff. I did not play music or test my microphone in the >> VM >> >>>> either, If i did anything wrong, please tell me! >> >>>> >> >>>> It stopped at 0x104f4 >> >>>> >> >>>> -Håvard >> >>>> >> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >> >>>> <conmanx360(a)gmail.com>: >> >>>>> >> >>>>> Hm.... that's odd. They should show up in the folder you ran the >> >>>>> command from. Does your console show any "DumpMem entered..." or >> >>>>> something like that? You may have a permissions error. I've had some >> >>>>> people report that as an issue before. >> >>>>> >> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >> wrote: >> >>>>> > Hi! >> >>>>> > >> >>>>> > Took a while, but I think I got it working. Did not see any >> "frame[xx]" >> >>>>> > files though. What dumps do you want? >> >>>>> > >> >>>>> > -Håvard >> >>>>> > >> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard < >> hovardslill(a)gmail.com>: >> >>>>> >> >> >>>>> >> Sorry about that, I forgot to enable a kernel config... >> >>>>> >> >> >>>>> >> -Håvard >> >>>>> >> >> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >> >>>>> >> <conmanx360(a)gmail.com>: >> >>>>> >>> >> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >> >>>>> >>> controls. >> >>>>> >>> You can check what module is loaded by doing lspci -v . It >> should say >> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the vfio-bind >> >>>>> >>> script >> >>>>> >>> before trying it? >> >>>>> >>> >> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard <hovardslill(a)gmail.com> >> >>>>> >>> wrote: >> >>>>> >>> > Thank you for the offer! >> >>>>> >>> > >> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >> seem to >> >>>>> >>> > bind my >> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >> to >> >>>>> >>> > function as >> >>>>> >>> > normal when I (think I) have bound it to a stud. >> >>>>> >>> > >> >>>>> >>> > -Håvard >> >>>>> >>> > >> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >> >>>>> >>> > <conmanx360(a)gmail.com>: >> >>>>> >>> >> >> >>>>> >>> >> One thing you could try, is using the program I used to >> reverse >> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. >> If your >> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >> you >> >>>>> >>> >> could >> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >> >>>>> >>> >> capture >> >>>>> >>> >> the commands. I'd be willing to look through the dumps to >> see if >> >>>>> >>> >> there >> >>>>> >>> >> are any special verbs or anything. >> >>>>> >>> >> >> >>>>> >>> >> You can find the program here: >> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >> >>>>> >>> >> >> >>>>> >>> >> Let me know. >> >>>>> >>> >> >> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard < >> hovardslill(a)gmail.com> >> >>>>> >>> >> wrote: >> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response >> before I >> >>>>> >>> >> > read >> >>>>> >>> >> > your >> >>>>> >>> >> > last one. >> >>>>> >>> >> > >> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >> headset >> >>>>> >>> >> > multijacks. >> >>>>> >>> >> > >> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the tip! >> >>>>> >>> >> > >> >>>>> >>> >> > -Håvard >> >>>>> >>> >> > >> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >> >>>>> >>> >> > <hovardslill(a)gmail.com>: >> >>>>> >>> >> > >> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >> >>>>> >>> >> >> >> >>>>> >>> >> >> Do you know where I can ask around for more help on the >> issue? >> >>>>> >>> >> >> I >> >>>>> >>> >> >> don't >> >>>>> >>> >> >> want to give up yet. >> >>>>> >>> >> >> >> >>>>> >>> >> >> -Håvard >> >>>>> >>> >> >> >> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >> >>>>> >>> >> >> <tiwai(a)suse.de>: >> >>>>> >>> >> >> >> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >> >>>>> >>> >> >>> Håvard wrote: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >> automatically >> >>>>> >>> >> >>> > changes >> >>>>> >>> >> >>> > to >> >>>>> >>> >> >>> it >> >>>>> >>> >> >>> > in pavucontrol. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Then it likely requires some additional initialization >> outside >> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >> >>>>> >>> >> >>> vendor-specific. >> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >> Windows >> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Takashi >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >> >>>>> >>> >> >>> > > Håvard wrote: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Thank you for replying! >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic Playback >> >>>>> >>> >> >>> > > > Volume" >> >>>>> >>> >> >>> > > > in >> >>>>> >>> >> >>> > > hdaanalyzer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >> >>>>> >>> >> >>> > > > made white noise come from the headset. However it >> did >> >>>>> >>> >> >>> > > > not >> >>>>> >>> >> >>> > > > change or >> >>>>> >>> >> >>> > > react >> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >> >>>>> >>> >> >>> > > > physically. >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >> either >> >>>>> >>> >> >>> > > > alsamixer >> >>>>> >>> >> >>> or >> >>>>> >>> >> >>> > > > hdaanalyzer. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > It's a mixer mute switch. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > The microphone works perfectly fine under Windows, >> so I >> >>>>> >>> >> >>> > > > don't >> >>>>> >>> >> >>> > > > think >> >>>>> >>> >> >>> it is >> >>>>> >>> >> >>> > > > the mic pin. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >> wrong >> >>>>> >>> >> >>> > > pin, >> >>>>> >>> >> >>> > > too. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Takashi >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > -Håvard >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi Iwai >> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >> >>>>> >>> >> >>> >: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >> >>>>> >>> >> >>> > > > > Håvard wrote: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Additional relevant info: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three years >> ago >> >>>>> >>> >> >>> > > > > > on >> >>>>> >>> >> >>> > > > > > Sun >> >>>>> >>> >> >>> > > > > > Jun >> >>>>> >>> >> >>> > > 17:15:54 >> >>>>> >>> >> >>> > > > > CEST >> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >> but did >> >>>>> >>> >> >>> > > > > > not >> >>>>> >>> >> >>> > > > > > touch >> >>>>> >>> >> >>> on the >> >>>>> >>> >> >>> > > > > > external microphone problem: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > alsa-info.sh: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help the >> >>>>> >>> >> >>> > > > > > issue. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > There are several of similar bug reports >> around the >> >>>>> >>> >> >>> > > > > > web >> >>>>> >>> >> >>> experiencing >> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >> seems to >> >>>>> >>> >> >>> > > > > > happen >> >>>>> >>> >> >>> > > > > > with >> >>>>> >>> >> >>> all >> >>>>> >>> >> >>> > > > > laptops >> >>>>> >>> >> >>> > > > > > under the G751 name. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >> and >> >>>>> >>> >> >>> > > > > unmute/adjust >> >>>>> >>> >> >>> "Mic >> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >> you >> >>>>> >>> >> >>> > > > > hear >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from >> NID 0x18 >> >>>>> >>> >> >>> > > > > to >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> mixer >> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >> outputs. >> >>>>> >>> >> >>> > > > > So >> >>>>> >>> >> >>> > > > > this >> >>>>> >>> >> >>> > > > > can >> >>>>> >>> >> >>> be >> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >> the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from >> >>>>> >>> >> >>> the ext >> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >> >>>>> >>> >> >>> > > > > something >> >>>>> >>> >> >>> > > > > outside >> >>>>> >>> >> >>> > > > > HD-audio codec. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > Takashi >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> >> >>>>> >>> >> >> >> >>>>> >>> >> > _______________________________________________ >> >>>>> >>> >> > Alsa-devel mailing list >> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >> >>>>> >>> >> > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >

1 0

Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Håvard 12 Sep '18

12 Sep '18

Additional relevant info: A similar issue was also discussed three years ago on Sun Jun 17:15:54 CEST 2015 and was about his surround sound setup, but did not touch on the external microphone problem: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html alsa-info.sh: http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 Graph of ALC668 chipset from hdaanalyzer: http://i.imgur.com/c08DNJW.png setting alsa-mode[1-8] does nothing to help the issue. There are several of similar bug reports around the web experiencing similar issues, and on different distros. Microphone works perfectly in windows I have a ASUS ROG G751JT, but this problem seems to happen with all laptops under the G751 name. These are my kernel configs: https://paste.pound-python.org/show/GRk4T6xNvmdp8W7K9slJ/ Thank you! -Håvard

3 19

[alsa-devel] [PATCH] ASoC: pxa: remove duplicated snd_soc_component_driver ops
by petrcvekcz＠gmail.com 11 Sep '18

11 Sep '18

From: Petr Cvek <petrcvekcz(a)gmail.com> The sound soc drivers which are using a platform driver pxa-pcm-audio and cpu dai drivers from the pxa tree have defined pxa2xx_soc_pcm_new() twice. The function will get called in soc_new_pcm() for two components (platform and cpu dai) which causes a double allocation of DMA resources and later during a close of the playback/capture a kernel warning: WARNING: CPU: 0 PID: 892 at drivers/dma/dmaengine.c:788 dma_release_channel+0x44/0xa0 chan reference count 0 != 1 Modules linked in: snd_soc_magician snd_soc_pxa2xx snd_pxa2xx_lib ... CPU: 0 PID: 892 Comm: aplay Tainted: G C 4.18.0-rc6-next-20180726-magician+ #16 Hardware name: HTC Magician [<c0107d90>] (unwind_backtrace) from [<c010594c>] (show_stack+0x10/0x14) [<c010594c>] (show_stack) from [<c0111680>] (__warn+0xd4/0xec) [<c0111680>] (__warn) from [<c01112d0>] (warn_slowpath_fmt+0x44/0x6c) [<c01112d0>] (warn_slowpath_fmt) from [<c034908c>] (dma_release_channel+0x44/0xa0) [<c034908c>] (dma_release_channel) from [<bf330528>] (snd_dmaengine_pcm_close_release_chan+0x44/0xb1c [snd_pcm_dmaengine]) [<bf330528>] (snd_dmaengine_pcm_close_release_chan [snd_pcm_dmaengine]) from [<bf26c29c>] (soc_pcm_components_close+0x8c/0xac [snd_soc_core]) [<bf26c29c>] (soc_pcm_components_close [snd_soc_core]) from [<bf26c3a0>] (soc_pcm_close+0xe4/0xd44 [snd_soc_core]) [<bf26c3a0>] (soc_pcm_close [snd_soc_core]) from [<bf167254>] (snd_pcm_release_substream+0x88/0x108 [snd_pcm]) [<bf167254>] (snd_pcm_release_substream [snd_pcm]) from [<bf167330>] (snd_pcm_release+0x5c/0xd2c [snd_pcm]) [<bf167330>] (snd_pcm_release [snd_pcm]) from [<c01c39e4>] (__fput+0xdc/0x1e8) [<c01c39e4>] (__fput) from [<c0129e7c>] (task_work_run+0xc4/0xd4) [<c0129e7c>] (task_work_run) from [<c01053ac>] (do_work_pending+0xc0/0xc8) [<c01053ac>] (do_work_pending) from [<c0101068>] (slow_work_pending+0xc/0x20) Exception stack(0xc2fdffb0 to 0xc2fdfff8) ffa0: 00000000 0154d1a0 01542a3c 00000000 ffc0: 0154d150 00000000 0009283c 00000006 000003f0 0154d4c8 00000000 0154d4c8 ffe0: b6ee7d70 bef13960 b6e547b8 b6cfe018 60000010 00000004 The problem was found for cpu dai drivers: pxa-i2s and pxa-ssp and by a calltrace sent from Robert Jarzmik on pxa-ac97 (although there was no observed kernel warning). The git grep shows these 3 cpu dai drivers are the only one which assign the .pcm_new to pxa2xx_soc_pcm_new() and almost every machine which is using one of these cpu dai drivers is using the platform driver pxa-pcm-audio too. The only exception is TTC_DKB board which is using a platform driver mmp-pcm-audio. A quick look in the code of mmp-pcm-audio shows it allocates the DMA in the similar way as PXA. The solution is to remove one of pxa2xx_soc_pcm_new() assignments and the patch does it for 3 cpu dai drivers (SSP/I2S/AC97). The result was tested on HTC Magician board (SSP for playback, I2S for capture) and the DMA buffers are correctly allocated. Signed-off-by: Petr Cvek <petrcvekcz(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 3 --- sound/soc/pxa/pxa2xx-ac97.c | 3 --- sound/soc/pxa/pxa2xx-i2s.c | 3 --- 3 files changed, 9 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a84e6..9689a1552d53 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -867,9 +867,6 @@ static struct snd_soc_dai_driver pxa_ssp_dai = { static const struct snd_soc_component_driver pxa_ssp_component = { .name = "pxa-ssp", - .ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_soc_pcm_new, - .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..c52b33802bf2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -214,9 +214,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { static const struct snd_soc_component_driver pxa_ac97_component = { .name = "pxa-ac97", - .ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_soc_pcm_new, - .pcm_free = pxa2xx_pcm_free_dma_buffers, }; #ifdef CONFIG_OF diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 42820121e5b9..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -364,9 +364,6 @@ static struct snd_soc_dai_driver pxa_i2s_dai = { static const struct snd_soc_component_driver pxa_i2s_component = { .name = "pxa-i2s", - .ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_soc_pcm_new, - .pcm_free = pxa2xx_pcm_free_dma_buffers, }; static int pxa2xx_i2s_drv_probe(struct platform_device *pdev) -- 2.18.0

2 2

[alsa-devel] [PATCH 03/11] compat_ioctl: remove translation for sound ioctls
by Arnd Bergmann 11 Sep '18

11 Sep '18

The SNDCTL_* and SOUND_* commands are the old OSS user interface. I checked all the sound ioctl commands listed in fs/compat_ioctl.c to see if we still need the translation handlers. Here is what I found: - sound/oss/ is (almost) gone from the kernel, this is what actually needed all the translations - The ALSA emulation for OSS correctly handles all compat_ioctl commands already. - sound/oss/dmasound/ is the last holdout of the original OSS code, this is only used on arch/m68k, which has no 64-bit mode and hence needs no compat handlers - arch/um/drivers/hostaudio_kern.c may run in 64-bit mode with 32-bit x86 user space underneath it. This rare corner case is the only one that still needs the compat handlers. By adding a simple redirect of .compat_ioctl to .unlocked_ioctl in the UML driver, we can remove all the COMPATIBLE_IOCTL() annotations without a change in functionality. For completeness, I'm adding the same thing to the dmasound file, knowing that it makes no difference. The compat_ioctl list contains one comment about SNDCTL_DSP_MAPINBUF and SNDCTL_DSP_MAPOUTBUF, which actually would need a translation handler if implemented. However, the native implementation just returns -EINVAL, so we don't care. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/um/drivers/hostaudio_kern.c | 1 + fs/compat_ioctl.c | 157 ----------------------------- sound/core/oss/pcm_oss.c | 4 + sound/oss/dmasound/dmasound_core.c | 2 + 4 files changed, 7 insertions(+), 157 deletions(-) diff --git a/arch/um/drivers/hostaudio_kern.c b/arch/um/drivers/hostaudio_kern.c index 7f9dbdbc4eb7..0278a642a622 100644 --- a/arch/um/drivers/hostaudio_kern.c +++ b/arch/um/drivers/hostaudio_kern.c @@ -298,6 +298,7 @@ static const struct file_operations hostaudio_fops = { .write = hostaudio_write, .poll = hostaudio_poll, .unlocked_ioctl = hostaudio_ioctl, + .compat_ioctl = hostaudio_ioctl, .mmap = NULL, .open = hostaudio_open, .release = hostaudio_release, diff --git a/fs/compat_ioctl.c b/fs/compat_ioctl.c index c2c73b802fb3..875516658c39 100644 --- a/fs/compat_ioctl.c +++ b/fs/compat_ioctl.c @@ -78,7 +78,6 @@ #include <linux/if_bonding.h> #include <linux/watchdog.h> -#include <linux/soundcard.h> #include <linux/lp.h> #include <linux/ppdev.h> @@ -605,162 +604,6 @@ COMPATIBLE_IOCTL(SG_GET_KEEP_ORPHAN) /* PPP stuff */ COMPATIBLE_IOCTL(PPPIOCGUNIT) COMPATIBLE_IOCTL(PPPIOCGCHAN) -/* Big A */ -/* sparc only */ -/* Big Q for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_SEQ_RESET) -COMPATIBLE_IOCTL(SNDCTL_SEQ_SYNC) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_INFO) -COMPATIBLE_IOCTL(SNDCTL_SEQ_CTRLRATE) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETOUTCOUNT) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETINCOUNT) -COMPATIBLE_IOCTL(SNDCTL_SEQ_PERCMODE) -COMPATIBLE_IOCTL(SNDCTL_FM_LOAD_INSTR) -COMPATIBLE_IOCTL(SNDCTL_SEQ_TESTMIDI) -COMPATIBLE_IOCTL(SNDCTL_SEQ_RESETSAMPLES) -COMPATIBLE_IOCTL(SNDCTL_SEQ_NRSYNTHS) -COMPATIBLE_IOCTL(SNDCTL_SEQ_NRMIDIS) -COMPATIBLE_IOCTL(SNDCTL_MIDI_INFO) -COMPATIBLE_IOCTL(SNDCTL_SEQ_THRESHOLD) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_MEMAVL) -COMPATIBLE_IOCTL(SNDCTL_FM_4OP_ENABLE) -COMPATIBLE_IOCTL(SNDCTL_SEQ_PANIC) -COMPATIBLE_IOCTL(SNDCTL_SEQ_OUTOFBAND) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETTIME) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_ID) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_CONTROL) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_REMOVESAMPLE) -/* Big T for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_TMR_TIMEBASE) -COMPATIBLE_IOCTL(SNDCTL_TMR_START) -COMPATIBLE_IOCTL(SNDCTL_TMR_STOP) -COMPATIBLE_IOCTL(SNDCTL_TMR_CONTINUE) -COMPATIBLE_IOCTL(SNDCTL_TMR_TEMPO) -COMPATIBLE_IOCTL(SNDCTL_TMR_SOURCE) -COMPATIBLE_IOCTL(SNDCTL_TMR_METRONOME) -COMPATIBLE_IOCTL(SNDCTL_TMR_SELECT) -/* Little m for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_MIDI_PRETIME) -COMPATIBLE_IOCTL(SNDCTL_MIDI_MPUMODE) -COMPATIBLE_IOCTL(SNDCTL_MIDI_MPUCMD) -/* Big P for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_DSP_RESET) -COMPATIBLE_IOCTL(SNDCTL_DSP_SYNC) -COMPATIBLE_IOCTL(SNDCTL_DSP_SPEED) -COMPATIBLE_IOCTL(SNDCTL_DSP_STEREO) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETBLKSIZE) -COMPATIBLE_IOCTL(SNDCTL_DSP_CHANNELS) -COMPATIBLE_IOCTL(SOUND_PCM_WRITE_FILTER) -COMPATIBLE_IOCTL(SNDCTL_DSP_POST) -COMPATIBLE_IOCTL(SNDCTL_DSP_SUBDIVIDE) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETFRAGMENT) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETFMTS) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETFMT) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETOSPACE) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETISPACE) -COMPATIBLE_IOCTL(SNDCTL_DSP_NONBLOCK) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETCAPS) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETTRIGGER) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETTRIGGER) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETIPTR) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETOPTR) -/* SNDCTL_DSP_MAPINBUF, XXX needs translation */ -/* SNDCTL_DSP_MAPOUTBUF, XXX needs translation */ -COMPATIBLE_IOCTL(SNDCTL_DSP_SETSYNCRO) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETDUPLEX) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETODELAY) -COMPATIBLE_IOCTL(SNDCTL_DSP_PROFILE) -COMPATIBLE_IOCTL(SOUND_PCM_READ_RATE) -COMPATIBLE_IOCTL(SOUND_PCM_READ_CHANNELS) -COMPATIBLE_IOCTL(SOUND_PCM_READ_BITS) -COMPATIBLE_IOCTL(SOUND_PCM_READ_FILTER) -/* Big C for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_COPR_RESET) -COMPATIBLE_IOCTL(SNDCTL_COPR_LOAD) -COMPATIBLE_IOCTL(SNDCTL_COPR_RDATA) -COMPATIBLE_IOCTL(SNDCTL_COPR_RCODE) -COMPATIBLE_IOCTL(SNDCTL_COPR_WDATA) -COMPATIBLE_IOCTL(SNDCTL_COPR_WCODE) -COMPATIBLE_IOCTL(SNDCTL_COPR_RUN) -COMPATIBLE_IOCTL(SNDCTL_COPR_HALT) -COMPATIBLE_IOCTL(SNDCTL_COPR_SENDMSG) -COMPATIBLE_IOCTL(SNDCTL_COPR_RCVMSG) -/* Big M for sound/OSS */ -COMPATIBLE_IOCTL(SOUND_MIXER_READ_VOLUME) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_BASS) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_TREBLE) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_SYNTH) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_PCM) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_SPEAKER) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_MIC) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_CD) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_IMIX) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_ALTPCM) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECLEV) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_IGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_OGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE1) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE2) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE3) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL1)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL2)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL3)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_PHONEIN)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_PHONEOUT)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_VIDEO)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_RADIO)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_MONITOR)) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_MUTE) -/* SOUND_MIXER_READ_ENHANCE, same value as READ_MUTE */ -/* SOUND_MIXER_READ_LOUD, same value as READ_MUTE */ -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECSRC) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_DEVMASK) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECMASK) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_STEREODEVS) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_CAPS) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_VOLUME) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_BASS) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_TREBLE) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_SYNTH) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_PCM) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_SPEAKER) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_MIC) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_CD) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_IMIX) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_ALTPCM) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_RECLEV) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_IGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_OGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE1) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE2) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE3) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL1)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL2)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL3)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_PHONEIN)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_PHONEOUT)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_VIDEO)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_RADIO)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_MONITOR)) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_MUTE) -/* SOUND_MIXER_WRITE_ENHANCE, same value as WRITE_MUTE */ -/* SOUND_MIXER_WRITE_LOUD, same value as WRITE_MUTE */ -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_RECSRC) -COMPATIBLE_IOCTL(SOUND_MIXER_INFO) -COMPATIBLE_IOCTL(SOUND_OLD_MIXER_INFO) -COMPATIBLE_IOCTL(SOUND_MIXER_ACCESS) -COMPATIBLE_IOCTL(SOUND_MIXER_AGC) -COMPATIBLE_IOCTL(SOUND_MIXER_3DSE) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE1) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE2) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE3) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE4) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE5) -COMPATIBLE_IOCTL(SOUND_MIXER_GETLEVELS) -COMPATIBLE_IOCTL(SOUND_MIXER_SETLEVELS) -COMPATIBLE_IOCTL(OSS_GETVERSION) /* Raw devices */ COMPATIBLE_IOCTL(RAW_SETBIND) COMPATIBLE_IOCTL(RAW_GETBIND) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f8d4a419f3af..3f25c9d7b7df 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2732,6 +2732,10 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long static long snd_pcm_oss_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) { + /* + * Everything is compatbile except SNDCTL_DSP_MAPINBUF/SNDCTL_DSP_MAPOUTBUF, + * which are not implemented for the native case either + */ return snd_pcm_oss_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); } #else diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index fc9bcd47d6a4..b6b5f5a5df9c 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -384,6 +384,7 @@ static const struct file_operations mixer_fops = .owner = THIS_MODULE, .llseek = no_llseek, .unlocked_ioctl = mixer_unlocked_ioctl, + .compat_ioctl = mixer_unlocked_ioctl, .open = mixer_open, .release = mixer_release, }; @@ -1167,6 +1168,7 @@ static const struct file_operations sq_fops = .write = sq_write, .poll = sq_poll, .unlocked_ioctl = sq_unlocked_ioctl, + .compat_ioctl = sq_unlocked_ioctl, .open = sq_open, .release = sq_release, }; -- 2.18.0

2 2

[alsa-devel] [PATCH] ASoC: rt5668: remove empty rt5668_i2c_remove()
by Kuninori Morimoto 11 Sep '18

11 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rt5668_i2c_remove() is empty, and no longer needed. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/rt5668.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 4412cd2..85ba04d 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2591,11 +2591,6 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, rt5668_dai, ARRAY_SIZE(rt5668_dai)); } -static int rt5668_i2c_remove(struct i2c_client *i2c) -{ - return 0; -} - static void rt5668_i2c_shutdown(struct i2c_client *client) { struct rt5668_priv *rt5668 = i2c_get_clientdata(client); @@ -2626,7 +2621,6 @@ static struct i2c_driver rt5668_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5668_acpi_match), }, .probe = rt5668_i2c_probe, - .remove = rt5668_i2c_remove, .shutdown = rt5668_i2c_shutdown, .id_table = rt5668_i2c_id, }; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: soc-core: avoid nested code on soc_remove_dai()
by Kuninori Morimoto 11 Sep '18

11 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Nested code is not readable. This patch avoid it on soc_remove_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8c73c6..41fef33 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -938,17 +938,18 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { - err = dai->driver->remove(dai); - if (err < 0) - dev_err(dai->dev, - "ASoC: failed to remove %s: %d\n", - dai->name, err); - } - dai->probed = 0; + if (!dai || !dai->probed || + dai->driver->remove_order != order) + return; + + if (dai->driver->remove) { + err = dai->driver->remove(dai); + if (err < 0) + dev_err(dai->dev, + "ASoC: failed to remove %s: %d\n", + dai->name, err); } + dai->probed = 0; } static void soc_remove_link_dais(struct snd_soc_card *card, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: soc-core: remove unused num_dai_links
by Kuninori Morimoto 11 Sep '18

11 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC is counting card->dai_link_list user, but no-one is using it. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 4 ---- 2 files changed, 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 821bf99..e17a7ae 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1060,7 +1060,6 @@ struct snd_soc_card { struct snd_soc_dai_link *dai_link; /* predefined links only */ int num_links; /* predefined links only */ struct list_head dai_link_list; /* all links */ - int num_dai_links; struct list_head rtd_list; int num_rtd; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1f859cc..f43c532 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1009,7 +1009,6 @@ static void soc_remove_dai_links(struct snd_soc_card *card) link->name); list_del(&link->list); - card->num_dai_links--; } } @@ -1185,7 +1184,6 @@ int snd_soc_add_dai_link(struct snd_soc_card *card, card->add_dai_link(card, dai_link); list_add_tail(&dai_link->list, &card->dai_link_list); - card->num_dai_links++; return 0; } @@ -1223,7 +1221,6 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, list_for_each_entry_safe(link, _link, &card->dai_link_list, list) { if (link == dai_link) { list_del(&link->list); - card->num_dai_links--; return; } } @@ -2710,7 +2707,6 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); INIT_LIST_HEAD(&card->dai_link_list); - card->num_dai_links = 0; INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda: fix unused variable warning
by Anders Roxell 11 Sep '18

11 Sep '18

When CONFIG_X86=n function azx_snoop doesn't use the variable chip it only returns true. sound/pci/hda/hda_intel.c: In function ‘dma_alloc_pages’: sound/pci/hda/hda_intel.c:2002:14: warning: unused variable ‘chip’ [-Wunused-variable] struct azx *chip = bus_to_azx(bus); ^~~~ Create a inline function of azx_snoop. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/pci/hda/hda_controller.h | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 0053b704ddc4..aa20048742d0 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -170,11 +170,10 @@ struct azx { #define azx_bus(chip) (&(chip)->bus.core) #define bus_to_azx(_bus) container_of(_bus, struct azx, bus.core) -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif +static inline bool azx_snoop(struct azx *chip) +{ + return IS_ENABLED(CONFIG_X86) || (chip->snoop); +} /* * macros for easy use -- 2.18.0

2 4

[alsa-devel] Applied "ASoC: dapm: Fix a couple uninitialized ret variables" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: dapm: Fix a couple uninitialized ret variables 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 2e558a8127de7b2ed3302f9adcf332ba3feeadb2 Mon Sep 17 00:00:00 2001 From: Dan Carpenter <dan.carpenter(a)oracle.com> Date: Fri, 7 Sep 2018 22:40:33 +0300 Subject: [PATCH] ASoC: dapm: Fix a couple uninitialized ret variables Smatch complains that these variables could be uninitialized. The first one in snd_soc_dai_link_event() is probably a false positive, because probably we know the lists are not empty. I would normally ignore the warning, but GCC complains here as well so I just silenced the warning. The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it leads to a bogus dereference in the caller. Fixes: 3bbf5d34fd4a ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e496bc642c99..0dcdcc23dcfd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3938,8 +3938,10 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) + if (IS_ERR(w)) { + ret = PTR_ERR(w); goto outfree_kcontrol_news; + } w->priv = rtd; -- 2.19.0.rc1

1 1

[alsa-devel] Applied "ASoC: qdsp6: q6asm-dai: clean up a return" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: qdsp6: q6asm-dai: clean up a return 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 0712e0288b7602efbde95d95bca4c651ccad01b8 Mon Sep 17 00:00:00 2001 From: Dan Carpenter <dan.carpenter(a)oracle.com> Date: Mon, 10 Sep 2018 11:40:48 +0300 Subject: [PATCH] ASoC: qdsp6: q6asm-dai: clean up a return Smatch complains that if both "psubstream" and "csubstream" are NULL then "ret" is uninitialized. That probably can't happen, but it's cleaner to just return zero anyway so let's do that. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9db9a2944ef2..c75fab38905d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) } } - return ret; + return 0; } static void q6asm_dai_pcm_free(struct snd_pcm *pcm) -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: atmel_ssc_dai: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: atmel_ssc_dai: use devm_snd_soc_register_component() 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 e14614dc5153ad41f7d1e5b125e4cd155ca79aa2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:00:15 +0000 Subject: [PATCH] ASoC: atmel_ssc_dai: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/atmel_ssc_dai.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index d3b69682d9c2..6291ec7f9dd6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -1005,11 +1005,11 @@ static int asoc_ssc_init(struct device *dev) struct ssc_device *ssc = dev_get_drvdata(dev); int ret; - ret = snd_soc_register_component(dev, &atmel_ssc_component, + ret = devm_snd_soc_register_component(dev, &atmel_ssc_component, &atmel_ssc_dai, 1); if (ret) { dev_err(dev, "Could not register DAI: %d\n", ret); - goto err; + return ret; } if (ssc->pdata->use_dma) @@ -1019,15 +1019,10 @@ static int asoc_ssc_init(struct device *dev) if (ret) { dev_err(dev, "Could not register PCM: %d\n", ret); - goto err_unregister_dai; + return ret; } return 0; - -err_unregister_dai: - snd_soc_unregister_component(dev); -err: - return ret; } static void asoc_ssc_exit(struct device *dev) @@ -1038,8 +1033,6 @@ static void asoc_ssc_exit(struct device *dev) atmel_pcm_dma_platform_unregister(dev); else atmel_pcm_pdc_platform_unregister(dev); - - snd_soc_unregister_component(dev); } /** -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: bcm: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: bcm: use devm_snd_soc_register_component() 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 570f75b93551a6d70ecbd0a7b6d962b4ca4722f0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:00:49 +0000 Subject: [PATCH] ASoC: bcm: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/cygnus-ssp.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index b733f1446353..b7c358b48d8d 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1334,7 +1334,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev) cygaud->active_ports = 0; dev_dbg(dev, "Registering %d DAIs\n", active_port_count); - err = snd_soc_register_component(dev, &cygnus_ssp_component, + err = devm_snd_soc_register_component(dev, &cygnus_ssp_component, cygnus_ssp_dai, active_port_count); if (err) { dev_err(dev, "snd_soc_register_dai failed\n"); @@ -1345,32 +1345,27 @@ static int cygnus_ssp_probe(struct platform_device *pdev) if (cygaud->irq_num <= 0) { dev_err(dev, "platform_get_irq failed\n"); err = cygaud->irq_num; - goto err_irq; + return err; } err = audio_clk_init(pdev, cygaud); if (err) { dev_err(dev, "audio clock initialization failed\n"); - goto err_irq; + return err; } err = cygnus_soc_platform_register(dev, cygaud); if (err) { dev_err(dev, "platform reg error %d\n", err); - goto err_irq; + return err; } return 0; - -err_irq: - snd_soc_unregister_component(dev); - return err; } static int cygnus_ssp_remove(struct platform_device *pdev) { cygnus_soc_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: hdac_hda: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: hdac_hda: use devm_snd_soc_register_component() 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 10ccaa39d7628470a3de4aae9d2346a55cbee46e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:01:19 +0000 Subject: [PATCH] ASoC: hdac_hda: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdac_hda.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 8c25a1332fa7..2aaa83028e55 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -448,7 +448,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) return -ENOMEM; /* ASoC specific initialization */ - ret = snd_soc_register_component(&hdev->dev, + ret = devm_snd_soc_register_component(&hdev->dev, &hdac_hda_codec, hdac_hda_dais, ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { @@ -464,7 +464,6 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) static int hdac_hda_dev_remove(struct hdac_device *hdev) { - snd_soc_unregister_component(&hdev->dev); return 0; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: rt5668: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: rt5668: use devm_snd_soc_register_component() 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 4fe1984ebc086ee39dd57983a7fee84c96c954a7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:01:34 +0000 Subject: [PATCH] ASoC: rt5668: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5668.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 3c19d03f2446..4412cd2910cd 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2587,14 +2587,12 @@ static int rt5668_i2c_probe(struct i2c_client *i2c, } - return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, + return devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, rt5668_dai, ARRAY_SIZE(rt5668_dai)); } static int rt5668_i2c_remove(struct i2c_client *i2c) { - snd_soc_unregister_component(&i2c->dev); - return 0; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: tscs454: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: tscs454: use devm_snd_soc_register_component() 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 007ac42db9ff4c36e91d192353421c6209058e06 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:01:50 +0000 Subject: [PATCH] ASoC: tscs454: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tscs454.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index ff85a0bf6170..93d84e5ae2d5 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3459,7 +3459,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c, /* Sync pg sel reg with cache */ regmap_write(tscs454->regmap, R_PAGESEL, 0x00); - ret = snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_tscs454, tscs454_dais, ARRAY_SIZE(tscs454_dais)); if (ret) { dev_err(&i2c->dev, "Failed to register component (%d)\n", ret); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: nuc900: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: nuc900: use devm_snd_soc_register_component() 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 bfacaa8c8956dd6076641043b7be848267a708eb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:02:38 +0000 Subject: [PATCH] ASoC: nuc900: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/nuc900/nuc900-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 81b09d740ed9..6384bb6dacfd 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -356,7 +356,7 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) if (ret) goto out; - ret = snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, + ret = devm_snd_soc_register_component(&pdev->dev, &nuc900_ac97_component, &nuc900_ac97_dai, 1); if (ret) goto out; @@ -373,8 +373,6 @@ static int nuc900_ac97_drvprobe(struct platform_device *pdev) static int nuc900_ac97_drvremove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); - nuc900_ac97_data = NULL; snd_soc_set_ac97_ops(NULL); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: omap: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: omap: use devm_snd_soc_register_component() 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 642a722d3116fbd22e59ac027f81b5ecd285f17c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:02:54 +0000 Subject: [PATCH] ASoC: omap: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/omap-hdmi-audio.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index 8a99a8837dc9..673a9eb153b2 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -348,7 +348,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) default: return -EINVAL; } - ret = snd_soc_register_component(ad->dssdev, &omap_hdmi_component, + ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component, dai_drv, 1); if (ret) return ret; @@ -383,7 +383,6 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "snd_soc_register_card failed (%d)\n", ret); - snd_soc_unregister_component(ad->dssdev); return ret; } @@ -400,7 +399,6 @@ static int omap_hdmi_audio_remove(struct platform_device *pdev) struct hdmi_audio_data *ad = platform_get_drvdata(pdev); snd_soc_unregister_card(ad->card); - snd_soc_unregister_component(ad->dssdev); return 0; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: sh: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: sh: use devm_snd_soc_register_component() 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 afa88ee37b1383490df003ea005d1d9bc8afa8a8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:03:25 +0000 Subject: [PATCH] ASoC: sh: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/hac.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index c2b496398e6b..17622ceb98c0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -319,13 +319,12 @@ static int hac_soc_platform_probe(struct platform_device *pdev) if (ret != 0) return ret; - return snd_soc_register_component(&pdev->dev, &sh4_hac_component, + return devm_snd_soc_register_component(&pdev->dev, &sh4_hac_component, sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } static int hac_soc_platform_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); return 0; } -- 2.19.0.rc2

1 0