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

[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

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[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

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[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

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[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

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[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

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[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

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[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

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[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

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[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

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[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

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