August 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/8] ASoC: simple-scu-card: tidyup and synchronize patches
by Kuninori Morimoto 25 Aug '16

25 Aug '16

Hi Mark These are cleanup/tidyup and synchronize patches for simple-scu-card. We have now simple-card and simple-scu-card. These are almost same, difference is very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. These are based on mark/topic/rcar Kuninori Morimoto (8): ASoC: simple-scu-card: remove unused definitions ASoC: simple-scu-card: is GPL v2 ASoC: simple-scu-card: tidyup codec daifmt handling method ASoC: simple-scu-card: tidyup asoc_simple_card_parse_links() method ASoC: simple-scu-card: depends on CONFIG_OF ASoC: simple-scu-card: add 2 CPU 1 Codec example ASoC: simple-scu-card: code sync: move asoc_simple_card_of_match ASoC: simple-scu-card: code sync: tidyup TDM setting position .../devicetree/bindings/sound/simple-scu-card.txt | 32 +++++++++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-scu-card.c | 72 +++++++++------------- 3 files changed, 60 insertions(+), 45 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/9] ASoC: simple-card: tidyup and synchronize patches
by Kuninori Morimoto 25 Aug '16

25 Aug '16

Hi Mark These are cleanup/tidyup and synchronize patches for simple-card. We have now simple-card and simple-scu-card. These are almost same, difference is very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. Kuninori Morimoto (9): ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: code sync: use tag ASoC: simple-card: code sync: rename num_link to num ASoC: simple-card: use kzalloc() for dai_props / dai_link sound/soc/generic/simple-card.c | 87 ++++++++++++++++++++--------------------- 1 file changed, 43 insertions(+), 44 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8810: Fix memory leak in nau8810_eq_put error path
by Axel Lin 25 Aug '16

25 Aug '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/nau8810.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index f9bcdc3..e455186 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -221,6 +221,7 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, if (ret) { dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", reg + i, ret); + kfree(data); return ret; } } -- 2.7.4

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[alsa-devel] [PATCH v2 1/2] ASoC: simple-card-utils: add __printf attribute
by Nicolas Iooss 25 Aug '16

25 Aug '16

asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..3207b1a70d38 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *codec, char *prefix, unsigned int *retfmt); +__printf(3, 4) int asoc_simple_card_set_dailink_name(struct device *dev, struct snd_soc_dai_link *dai_link, const char *fmt, ...); -- 2.9.3

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[alsa-devel] [PATCH] ASoC: tegra_rt5640: Correct a copy and paste typo in the comments
by Nicolin Chen 25 Aug '16

25 Aug '16

This patch corrects a copy and paste typo. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/tegra/tegra_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 773daec..e5ef4e9 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -1,5 +1,5 @@ /* -* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec. +* tegra_rt5640.c - Tegra machine ASoC driver for boards using RT5640 codec. * * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. * -- 2.1.4

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[alsa-devel] Spam in the wiki
by Tanu Kaskinen 24 Aug '16

24 Aug '16

I got a notification, because a wiki page that I had created was edited by a spammer. Looking at the recent changes, it seems that there's quite a lot of spam activity: http://www.alsa-project.org/main/index.php/Special:RecentChanges Can the wiki user registrations be restricted? There's already a captcha, but that doesn't seem very effective. -- Tanu

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[alsa-devel] [PATCH] ASoC: rt5659: Enable IRQ output for GPIO1 pin
by Nicolin Chen 24 Aug '16

24 Aug '16

Since it's possible to have an IRQ without enabling JD3, the GPIO1 pin then would remain its default GPIO function which is set as an input direction (seeing from rt5659). Meanwhile, CPU would also listen this connection via its own GPIO input: [input] [input] CPU GPIO <--------> RT5659 GPIO1 The result for the IRQ on the CPU side will be unexpectable. So this patch enables the IRQ output for GPIO1 pin any way as long as there's an IRQ assigned from platform data or DT: [input] [IRQ output] CPU GPIO <--------< RT5659 GPIO1 Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/rt5659.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4c0d2b8..db54550 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4189,6 +4189,9 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, if (ret) dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + /* Enable IRQ output for GPIO1 pin any way */ + regmap_update_bits(rt5659->regmap, RT5659_GPIO_CTRL_1, + RT5659_GP1_PIN_MASK, RT5659_GP1_PIN_IRQ); } return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659, -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: simple-card-utils: add __printf attribute
by Nicolas Iooss 24 Aug '16

24 Aug '16

asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..3207b1a70d38 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *codec, char *prefix, unsigned int *retfmt); +__printf(3, 4) int asoc_simple_card_set_dailink_name(struct device *dev, struct snd_soc_dai_link *dai_link, const char *fmt, ...); -- 2.9.3

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[alsa-devel] [PATCH -next] sound: rockchip: fix build when SPI_MASTER is not enabled
by Randy Dunlap 24 Aug '16

24 Aug '16

From: Randy Dunlap <rdunlap(a)infradead.org> Fix multiple build errors in sound/soc/rockchip/ drivers by adding "depends on SPI_MASTER". Fixes these build errors: sound/built-in.o: In function `rt5514_spi_burst_read': (.text+0x6c26c): undefined reference to `spi_sync' sound/built-in.o: In function `rt5514_spi_burst_write': (.text+0x6c96e): undefined reference to `spi_sync' sound/built-in.o: In function `rockchip_sound_probe': rk3399_gru_sound.c:(.text+0x9f579): undefined reference to `spi_bus_type' sound/built-in.o: In function `rt5514_spi_driver_init': rt5514-spi.c:(.init.text+0x983): undefined reference to `__spi_register_driver' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-rockchip(a)lists.infradead.org --- sound/soc/rockchip/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20160824.orig/sound/soc/rockchip/Kconfig +++ linux-next-20160824/sound/soc/rockchip/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ROCKCHIP_RT5645 config SND_SOC_RK3399_GRU_SOUND tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + depends on SPI_MASTER select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A select SND_SOC_RT5514

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[alsa-devel] [PATCH -next] sound: rockchip: fix build when SPI is not enabled
by Randy Dunlap 24 Aug '16

24 Aug '16

From: Randy Dunlap <rdunlap(a)infradead.org> Fix build errors found in rt5514 drivers when SPI is not enabled. Fixes these build errors: ERROR: "__spi_register_driver" [sound/soc/codecs/snd-soc-rt5514-spi.ko] undefined! ERROR: "spi_sync" [sound/soc/codecs/snd-soc-rt5514-spi.ko] undefined! Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) This builds without errors, but if you always want to have the SPI interface available in this driver, change the first "depends" to include "&& SPI_MASTER". --- linux-next-20160822.orig/sound/soc/rockchip/Kconfig +++ linux-next-20160822/sound/soc/rockchip/Kconfig @@ -49,7 +49,7 @@ config SND_SOC_RK3399_GRU_SOUND select SND_SOC_MAX98357A select SND_SOC_RT5514 select SND_SOC_DA7219 - select SND_SOC_RT5514_SPI + select SND_SOC_RT5514_SPI if SPI_MASTER help Say Y or M here if you want to add support multiple codecs for SoC audio on Rockchip RK3399 GRU boards.

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[alsa-devel] Questions about DAPM Routing for STA309A DSP Driver
by Mauricio Cirelli 24 Aug '16

24 Aug '16

Dear ALSA Community, I am new to this community and have a basic knowledge on ASoC architecture. I have worked a few times with Freescale's boards and the SGTL5000 Codec driver. Now, time has come for me to develop a ASoC Codec Driver for an Audio DSP (ST Microelectronics STA309A). Obviously, this driver should work only on the Playback stream. This chip supports up to 8 playback channels (I2S format) and has up to 8 output channels. There are also 8 internal channels, whose inputs can be one of the 8 playback channels and outputs are connected to one of each 8 output channels like this: Playback Channel X --> Input Channel Y --> Output Channel Z This DSP has several controls to apply effects and modify volume on each channel internal channel individually. I have been able to implement snd_kcontrol_new for them successfully and I am able to control them from the userspace correctly. The part I am still in doubt on how to proceed is related to this channel routing. From what I have researched, this could be designed on top of DAPM Widgets and DAPM Routes in a way that the machine-level driver could configure this routing according to the specific board and the userspace would not be required to deal with this stuff. I want to develop a device driver following the ASoC architecture so I can use it in different boards and in a way it could be useful for other developers. Here is what I was sketching up for this snd_soc_codec_driver: static const char *input_names[8] = { "DIN1", "DIN2", "DIN3", "DIN4", "DIN5", "DIN6", "DIN7", "DIN8" }; static const char *internal_names[8] = { "INT1", "INT2", "INT3", "INT4", "INT5", "INT6", "INT7", "INT8" }; /* Playback Channel to Internal DSP Channel Mapping */ static SOC_ENUM_SINGLE_DECL(int1mux, STA309A_C12IM, 0, input_names); static const struct snd_kcontrol_new int1mux_ctrl = SOC_DAPM_ENUM("INT1MUX", int1mux); .... replicated to all 8 internal channels /* Internal DSP Channel to Output Channel Mapping */ static SOC_ENUM_SINGLE_DECL(out1mux, STA309A_C12OM, 0, internal_names); static const struct snd_kcontrol_new out1mux_ctrl = SOC_DAPM_ENUM("OUT1MUX", out1mux); .... replicated to all 8 output channels static const struct snd_soc_dapm_widget sta309a_dsp_dapm_widgets[] = { SND_SOC_DAPM_MUX("INT1MUX", SND_SOC_NOPM, 0, 0, &int1mux_ctrl), .... added 8 internal channel muxes SND_SOC_DAPM_MUX("OUT1MUX", SND_SOC_NOPM, 0, 0, &out1mux_ctrl), .... added 8 output channel muxes }; Now I am lost on how to declare the possible routes. It should be possible to route each playback channel to any of the 8 possible internal channels and the internal channels to any of the 8 possible output channels. Note that this DSP supports Cross-Overs, so one can route a playback channel to one or more internal channels. Analogously, one can route an internal channel to one or more output channels. I have not started to develop the machine-level driver, but any thoughts on how to design it would be very helpful. In the machine-level driver, I would like to define the DSP routing for an specific board. How can I do it? Thank you very much. Mauricio Cirelli

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[alsa-devel] [PATCH 0/2] ALSA: bebob: merge vendor-dependent codes for Yamaha/Terratec
by Takashi Sakamoto 24 Aug '16

24 Aug '16

Hi, This patchset merges vendor-dependent codes for Yamaha/Terratec, because each of these two pairs has the same configuration: - Yamaha GO44 / Terratec Phase 24 FW - Yamaha GO46 / Terratec Phase X24 FW The two companies might once have cooperated to develop some audio and music units on IEEE 1394 bus, based on DM1000E ASIC and BeBoB firmware. This conclusion comes from a reporter of a forum of Ubuntu Japanese Community. Thanks for his help to identify subunit formation and supported stream formats: Jack & Firewireオーディオインターフェイスの設定に関する質問です。 https://forums.ubuntulinux.jp/viewtopic.php?id=18657 Terratec has also produced the other models with DM ASIC series and BeBoB firmware. For the models, vendor-dependent codes are still left. Takashi Sakamoto (2): ALSA: bebob: rename file with vendor-dependent code so that it's for Yamaha/Terratec ALSA: bebob: unify configurations for some models produced by Yamaha/Terratec sound/firewire/bebob/Makefile | 5 ++-- sound/firewire/bebob/bebob.c | 8 +++--- sound/firewire/bebob/bebob.h | 3 +-- sound/firewire/bebob/bebob_terratec.c | 31 ---------------------- .../{bebob_yamaha.c => bebob_yamaha_terratec.c} | 9 ++++--- 5 files changed, 14 insertions(+), 42 deletions(-) rename sound/firewire/bebob/{bebob_yamaha.c => bebob_yamaha_terratec.c} (85%) -- 2.7.4

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[alsa-devel] [PATCH -next] ASoC: samsung: i2s: Add missing clk_disable_unprepare() on error in samsung_i2s_probe()
by Wei Yongjun 24 Aug '16

24 Aug '16

Add the missing clk_disable_unprepare() before return from samsung_i2s_probe() in the error handling case. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/samsung/i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fa3ff03..7e32cf4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1318,7 +1318,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err_disable_clk; } sec_dai->lock = &pri_dai->spinlock; @@ -1342,7 +1343,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); - return -EINVAL; + ret = -EINVAL; + goto err_disable_clk; } ret = devm_snd_soc_register_component(&pri_dai->pdev->dev, @@ -1366,6 +1368,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) err_free_dai: if (sec_dai) i2s_free_sec_dai(sec_dai); +err_disable_clk: + clk_disable_unprepare(pri_dai->clk); return ret; }

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[alsa-devel] [PATCH] ASoC: tas5086: fix typo: "Inavlid" -> "Invalid"
by Colin King 24 Aug '16

24 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> trivial typo fix in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index c297b9f..b7de857 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -387,7 +387,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), priv->mclk / priv->rate); if (val < 0) { - dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + dev_err(codec->dev, "Invalid MCLK / Fs ratio\n"); return -EINVAL; } -- 2.9.3

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[alsa-devel] [PATCH -next] ASoC: omap-abe-twl6040: Use devm_snd_soc_register_card()
by Wei Yongjun 24 Aug '16

24 Aug '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/omap/omap-abe-twl6040.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index f61b3b5..89fe95e 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -305,23 +305,14 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } -static int omap_abe_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id omap_abe_of_match[] = { {.compatible = "ti,abe-twl6040", }, { }, @@ -335,7 +326,6 @@ static struct platform_driver omap_abe_driver = { .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, - .remove = omap_abe_remove, }; static int __init omap_abe_init(void)

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[alsa-devel] [PATCH] ASoC: sunxi: depend on ARCH_SUNXI
by Peter Robinson 24 Aug '16

24 Aug '16

Depend on ARCH_SUNXI or COMPILE_TEST as it doesn't make sense to enable these if the appropriate SoC they're embedded in isn't enabled. Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> --- sound/soc/sunxi/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 2a954bd..a2b62dc5 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -2,6 +2,7 @@ menu "Allwinner SoC Audio support" config SND_SUN4I_CODEC tristate "Allwinner A10 Codec Support" + depends on ARCH_SUNXI || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help @@ -10,6 +11,7 @@ config SND_SUN4I_CODEC config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" + depends on ARCH_SUNXI || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help @@ -19,7 +21,7 @@ config SND_SUN4I_I2S config SND_SUN4I_SPDIF tristate "Allwinner A10 SPDIF Support" - depends on OF + depends on OF && (ARCH_SUNXI || COMPILE_TEST) select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 2.9.3

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[alsa-devel] [PATCH v2 0/4] ASoC: rsrc-card to simple-scu-card
by Kuninori Morimoto 24 Aug '16

24 Aug '16

Hi Mark Thesee are v2 of rsrc-card to simple-scu-card. rsrc-card was created for DPCM sound card for Renesas, but we can reuse this driver as more generic purpose. This driver is supporting Multi CPU and Single Codec DPCM system. Kuninori Morimoto (4): ASoC: rsrc-card: remove board specific compatible ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase3 .../bindings/sound/renesas,rsrc-card.txt | 75 --------- .../devicetree/bindings/sound/simple-scu-card.txt | 80 ++++++++++ sound/soc/generic/Kconfig | 7 + sound/soc/generic/Makefile | 2 + .../rcar/rsrc-card.c => generic/simple-scu-card.c} | 172 +++++++++------------ sound/soc/sh/Kconfig | 6 - sound/soc/sh/rcar/Makefile | 3 - 7 files changed, 162 insertions(+), 183 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt rename sound/soc/{sh/rcar/rsrc-card.c => generic/simple-scu-card.c} (57%) -- 1.9.1

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Re: [alsa-devel] good ideas
by lemeunier gilles 24 Aug '16

24 Aug '16

Hi! I have some brilliant ideas regarding the issues we've discussed last time, please read 'em here <http://ndicrugafru.whonue.net/e6wutsxl> Pardon my monkey thumbs, lemeunier gilles

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[alsa-devel] [PATCH] ALSA: hda/realtek - fix headset mic detection for MSI MS-B120
by Anisse Astier 24 Aug '16

24 Aug '16

MSI Cubi MS-B120 needs the same fixup as the Gigabyte BXBT-2807 for its mic to work. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- sound/pci/hda/patch_realtek.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 574b1b4..7100f05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4828,7 +4828,7 @@ enum { ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, ALC292_FIXUP_TPT440, - ALC283_FIXUP_BXBT2807_MIC, + ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, @@ -5321,7 +5321,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_TPT440_DOCK, }, - [ALC283_FIXUP_BXBT2807_MIC] = { + [ALC283_FIXUP_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x04a110f0 }, @@ -5651,7 +5651,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), + SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- 2.5.5

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[alsa-devel] [PATCH] ASoC: sunxi: depend on ARCH_SUNXI
by Peter Robinson 23 Aug '16

23 Aug '16

Depend on ARCH_SUNXI or COMPILE_TEST as it doesn't make sense to enable these if the appropriate SoC they're embedded in isn't enabled. Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> --- sound/soc/sunxi/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 2a954bd..dd23682 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -1,4 +1,5 @@ menu "Allwinner SoC Audio support" + depends on ARCH_SUNXI || COMPILE_TEST config SND_SUN4I_CODEC tristate "Allwinner A10 Codec Support" -- 2.9.3

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[alsa-devel] CVE-2014-9892 fix is not upstream
by Luis Henriques 23 Aug '16

23 Aug '16

Hi! Digging through some old CVEs I came across this one that doesn't seem be in mainline. Was there a good reason for not being sent upstream? Maybe it was rejected for some reason and I failed to find the discussion. References: https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-9892 https://source.android.com/security/bulletin/2016-08-01.html https://source.codeaurora.org/quic/la/kernel/msm-3.10/commit/?id=591b1f455c… Cheers, -- Luís

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[alsa-devel] [PATCH v5 0/2] amidi delay option
by Felipe F. Tonello 23 Aug '16

23 Aug '16

Small improvement to amidi, adding useful functionality. changes from v4: * simplified code changes from v3: * fixed a bug if memchr() returns NULL changes from v2: * only check of end of sysex (0xf7) * use memchr() instead of direct loop approach * fixed commit message changes from v1: * remove delay in between any MIDI message, only SysEx * added patch to use getopt_long() arguments Felipe F. Tonello (2): amidi: add delay option amidi: use GNU getopt_long() proper argument names amidi/amidi.1 | 14 ++++++++++++++ amidi/amidi.c | 61 +++++++++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 59 insertions(+), 16 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] ASoC: nau8810: fix compile warning in loopback switch control
by John Hsu 23 Aug '16

23 Aug '16

Thank Stephen Rothwell for the message. The patch is to fix the following compile error. sound/soc/codecs/nau8810.c:441:3: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, &nau8810_loopback), ^ sound/soc/codecs/nau8810.c:441:3: note: (near initialization for 'nau8810_dapm_widgets[11].kcontrol_news') Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 7617dda..f9bcdc3 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -391,10 +391,9 @@ static const struct snd_kcontrol_new nau8810_mic_boost_controls = NAU8810_PMICBSTGAIN_SFT, 0x7, 0); /* Loopback Switch */ -static const struct snd_kcontrol_new nau8810_loopback[] = { +static const struct snd_kcontrol_new nau8810_loopback = SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, - NAU8810_ADDAP_SFT, 1, 0), -}; + NAU8810_ADDAP_SFT, 1, 0); static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) -- 2.6.4

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[alsa-devel] [PATCH for 4.8] ASoC: omap-mcpdm: Fix irq resource handling
by Peter Ujfalusi 23 Aug '16

23 Aug '16

Fixes: ddd17531ad908 ("ASoC: omap-mcpdm: Clean up with devm_* function") Managed irq request will not doing any good in ASoC probe level as it is not going to free up the irq when the driver is unbound from the sound card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Russell King <linux(a)armlinux.org.uk> --- sound/soc/omap/omap-mcpdm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 74d6e6fdcfd0..64609c77a79d 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -394,8 +394,8 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", + (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -420,6 +420,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; -- 2.9.2

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[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Parse manifest data
by Vinod Koul 23 Aug '16

23 Aug '16

From: Shreyas NC <shreyas.nc(a)intel.com> Topology manifest has lib names and lib count info. So, define tokens to represent module private data and parse these tokens to fill up the manifest structure in the driver accordingly. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/intel/skylake/skl-topology.c | 194 ++++++++++++++++++++++++++++++++- 2 files changed, 200 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h index f56a932736ca..1ee2e943d66a 100644 --- a/include/uapi/sound/snd_sst_tokens.h +++ b/include/uapi/sound/snd_sst_tokens.h @@ -153,6 +153,10 @@ * * %SKL_TKN_U32_PROC_DOMAIN: Specify processing domain * + * %SKL_TKN_U32_LIB_COUNT: Specifies the number of libraries + * + * %SKL_TKN_STR_LIB_NAME: Specifies the library name + * * module_id and loadable flags dont have tokens as these values will be * read from the DSP FW manifest */ @@ -202,7 +206,9 @@ enum SKL_TKNS { SKL_TKN_U32_CAPS_PARAMS_ID, SKL_TKN_U32_CAPS_SIZE, SKL_TKN_U32_PROC_DOMAIN, - SKL_TKN_MAX = SKL_TKN_U32_PROC_DOMAIN, + SKL_TKN_U32_LIB_COUNT, + SKL_TKN_STR_LIB_NAME, + SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME, }; #endif diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 000482e6e2f7..108ebb9ab329 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2201,6 +2201,197 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, return 0; } +static int skl_tplg_fill_str_mfest_tkn(struct device *dev, + struct snd_soc_tplg_vendor_string_elem *str_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + static int ref_count; + + switch (str_elem->token) { + case SKL_TKN_STR_LIB_NAME: + if (ref_count > minfo->lib_count - 1) { + ref_count = 0; + return -EINVAL; + } + + strncpy(minfo->lib[ref_count].name, str_elem->string, + ARRAY_SIZE(minfo->lib[ref_count].name)); + ref_count++; + tkn_count++; + break; + + default: + dev_err(dev, "Not a string token %d", str_elem->token); + break; + } + + return tkn_count; +} + +static int skl_tplg_get_str_tkn(struct device *dev, + struct snd_soc_tplg_vendor_array *array, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0, ret; + struct snd_soc_tplg_vendor_string_elem *str_elem; + + str_elem = (struct snd_soc_tplg_vendor_string_elem *)array->value; + while (tkn_count < array->num_elems) { + ret = skl_tplg_fill_str_mfest_tkn(dev, str_elem, minfo); + str_elem++; + + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + } + + return tkn_count; +} + +static int skl_tplg_get_int_tkn(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + + switch (tkn_elem->token) { + case SKL_TKN_U32_LIB_COUNT: + minfo->lib_count = tkn_elem->value; + tkn_count++; + break; + + default: + dev_err(dev, "Not a manifest token %d", tkn_elem->token); + return -EINVAL; + } + + return tkn_count; +} + +/* + * Fill the manifest structure by parsing the tokens based on the + * type. + */ +static int skl_tplg_get_manifest_tkn(struct device *dev, + char *pvt_data, struct skl_dfw_manifest *minfo, + int block_size) +{ + int tkn_count = 0, ret; + int off = 0, tuple_size = 0; + struct snd_soc_tplg_vendor_array *array; + struct snd_soc_tplg_vendor_value_elem *tkn_elem; + + if (block_size <= 0) + return -EINVAL; + + while (tuple_size < block_size) { + array = (struct snd_soc_tplg_vendor_array *)(pvt_data + off); + off += array->size; + switch (array->type) { + case SND_SOC_TPLG_TUPLE_TYPE_STRING: + ret = skl_tplg_get_str_tkn(dev, array, minfo); + + if (ret < 0) + return ret; + tkn_count += ret; + + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_string_elem); + continue; + + case SND_SOC_TPLG_TUPLE_TYPE_UUID: + dev_warn(dev, "no uuid tokens for skl tplf manifest"); + continue; + + default: + tkn_elem = array->value; + tkn_count = 0; + break; + } + + while (tkn_count <= array->num_elems - 1) { + ret = skl_tplg_get_int_tkn(dev, + tkn_elem, minfo); + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + tkn_elem++; + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_value_elem); + break; + } + tkn_count = 0; + } + + return 0; +} + +/* + * Parse manifest private data for tokens. The private data block is + * preceded by descriptors for type and size of data block. + */ +static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, + struct device *dev, struct skl_dfw_manifest *minfo) +{ + struct snd_soc_tplg_vendor_array *array; + int num_blocks, block_size = 0, block_type, off = 0; + char *data; + int ret; + + /* Read the NUM_DATA_BLOCKS descriptor */ + array = (struct snd_soc_tplg_vendor_array *)manifest->priv.data; + ret = skl_tplg_get_desc_blocks(dev, array); + if (ret < 0) + return ret; + num_blocks = ret; + + off += array->size; + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + /* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */ + while (num_blocks > 0) { + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_type = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_size = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + data = (manifest->priv.data + off); + + if (block_type == SKL_TYPE_TUPLE) { + ret = skl_tplg_get_manifest_tkn(dev, data, minfo, + block_size); + + if (ret < 0) + return ret; + + --num_blocks; + } else { + return -EINVAL; + } + } + + return 0; +} + static int skl_manifest_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_manifest *manifest) { @@ -2211,7 +2402,8 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int ret = 0; minfo = &skl->skl_sst->manifest; - memcpy(minfo, manifest->priv.data, sizeof(struct skl_dfw_manifest)); + + skl_tplg_get_manifest_data(manifest, bus->dev, minfo); if (minfo->lib_count > HDA_MAX_LIB) { dev_err(bus->dev, "Exceeding max Library count. Got:%d\n", -- 1.9.1

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