July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] Motorola Droid 4 Audio Support
by Sebastian Reichel 25 Jul '17

25 Jul '17

Hi, I got working sound on Droid 4 with mainline \o/. The codec is currently missing support for detecting if something has been plugged into the 3.5mm connector, since that seems to require some closed source firmware and needs further investigation. I think this can be added later. Tested: - Playing via [HiFi] using Speaker, Earpiece, Headphone - Playing via [Voice] using Speaker, Earpiece, Headphone - Recording via [Voice] using Mic - Volume Controls - Checking that DAPM graph look correct -- Sebastian Sebastian Reichel (3): ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 6 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 76 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1422 ++++++++++++++++++++ 6 files changed, 1528 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

5 16

[alsa-devel] [PATCH] linux/kernel.h: move DIV_ROUND_DOWN_ULL() macro
by Masahiro Yamada 25 Jul '17

25 Jul '17

This macro is useful to avoid link error on 32-bit systems. We have the same definition in two drivers, so move it to include/linux/kernel.h While we are here, refactor DIV_ROUND_UP_ULL() by using DIV_ROUND_DOWN_ULL(). Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- drivers/mtd/nand/denali.c | 3 --- include/linux/kernel.h | 7 +++++-- sound/soc/codecs/pcm512x.c | 3 --- 3 files changed, 5 insertions(+), 8 deletions(-) diff --git a/drivers/mtd/nand/denali.c b/drivers/mtd/nand/denali.c index d723be352148..3087b0ba7b7f 100644 --- a/drivers/mtd/nand/denali.c +++ b/drivers/mtd/nand/denali.c @@ -980,9 +980,6 @@ static int denali_erase(struct mtd_info *mtd, int page) return irq_status & INTR__ERASE_COMP ? 0 : NAND_STATUS_FAIL; } -#define DIV_ROUND_DOWN_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) - static int denali_setup_data_interface(struct mtd_info *mtd, int chipnr, const struct nand_data_interface *conf) { diff --git a/include/linux/kernel.h b/include/linux/kernel.h index 96381f2b0e09..2a254ae6f7f7 100644 --- a/include/linux/kernel.h +++ b/include/linux/kernel.h @@ -78,8 +78,11 @@ #define FIELD_SIZEOF(t, f) (sizeof(((t*)0)->f)) #define DIV_ROUND_UP __KERNEL_DIV_ROUND_UP -#define DIV_ROUND_UP_ULL(ll,d) \ - ({ unsigned long long _tmp = (ll)+(d)-1; do_div(_tmp, d); _tmp; }) + +#define DIV_ROUND_DOWN_ULL(ll, d) \ + ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) + +#define DIV_ROUND_UP_ULL(ll, d) DIV_ROUND_DOWN_ULL((ll) + (d) - 1, (d)) #if BITS_PER_LONG == 32 # define DIV_ROUND_UP_SECTOR_T(ll,d) DIV_ROUND_UP_ULL(ll, d) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 72b19e62f626..ad3bfb894ee0 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -30,9 +30,6 @@ #include "pcm512x.h" -#define DIV_ROUND_DOWN_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) - #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", -- 2.7.4

1 0

[alsa-devel] I use manjaro but no device detected
by fulgor fulgor 22 Jul '17

22 Jul '17

hello, I have no sound The sound device is not detected I have logging informations http://www.alsa-project.org/db/?f=6fb68d0201f421b41fe65677fde2ec5b1c9d4429 thanks for all -- Lionel Van Winnendaele

1 0

[alsa-devel] [PATCH v2] sound: Convert to using %pOF instead of full_name
by Rob Herring 21 Jul '17

21 Jul '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- v2: - Add printf specifier change for fsl_dma.c sound/aoa/codecs/tas.c | 4 ++-- sound/soc/fsl/fsl_dma.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 16 ++++++++-------- 3 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 733b6365dad6..15c05755d270 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -905,8 +905,8 @@ static int tas_i2c_probe(struct i2c_client *client, goto fail; } printk(KERN_DEBUG - "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - (unsigned int)client->addr, node->full_name); + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); return 0; fail: mutex_destroy(&tas->mtx); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..ed8ea002902d 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -897,8 +897,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) ret = of_address_to_resource(ssi_np, 0, &res); if (ret) { - dev_err(&pdev->dev, "could not determine resources for %s\n", - ssi_np->full_name); + dev_err(&pdev->dev, "could not determine resources for %pOF\n", + ssi_np); of_node_put(ssi_np); return ret; } diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..392d5eef356d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -268,13 +268,13 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, ret = of_property_read_u32(child, "fsl,audmux-port", &port); if (ret) { - dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", - child->full_name); + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%pOF\"\n", + child); continue; } if (!of_property_read_bool(child, "fsl,port-config")) { - dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", - child->full_name); + dev_warn(&pdev->dev, "child node \"%pOF\" does not have property fsl,port-config\n", + child); continue; } @@ -292,15 +292,15 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } if (ret != -EOVERFLOW) { - dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", - i, child->full_name); + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %pOF\n", + i, child); continue; } if (audmux_type == IMX31_AUDMUX) { if (i % 2) { - dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", - child->full_name); + dev_err(&pdev->dev, "One pdcr value is missing in child node %pOF\n", + child); continue; } imx_audmux_v2_configure_port(port, ptcr, pdcr); -- 2.11.0

2 1

[alsa-devel] [PATCH v2] sound: fsl_dma: remove dma_object path member
by Rob Herring 21 Jul '17

21 Jul '17

dma_object.path is unused, so rather than fix it to work with DT full_name changes, just remove it. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> 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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org --- v2: - Move printf specifier change to correct patch sound/soc/fsl/fsl_dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ed8ea002902d..b38dd328c783 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -903,14 +902,13 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) return ret; } - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + dma = kzalloc(sizeof(*dma), GFP_KERNEL); if (!dma) { dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Use snd_soc_kcontrol_codec()
by Fabio Estevam 21 Jul '17

21 Jul '17

Since commit a72952672005 ("ASoC: sgtl5000: add avc support") the following kernel crash happens after running a 'reboot' command: ALSA: Storing mixer settings... [ 20.031604] Unable to handle kernel paging request at virtual address fffffffe [ 20.039268] pgd = de2a0000 [ 20.041999] [fffffffe] *pgd=8fffd861, *pte=00000000, *ppte=00000000 [ 20.048387] Internal error: Oops: 80000007 [#1] SMP ARM The function that takes a kcontrol parameter and returns the codec that registered the control is snd_soc_kcontrol_codec(), so use the correct function to fix the problem. Fixes: a72952672005 ("ASoC: sgtl5000: add avc support") Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8f6814c..80f6d1d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -409,7 +409,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, static int avc_get_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db, i; u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); @@ -442,7 +442,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol, static int avc_put_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db; u16 reg; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: samsung: odroid: Fix EPLL frequency values
by Sylwester Nawrocki 21 Jul '17

21 Jul '17

To prevent incorrect setting of the EPLL the clock frequency values are changed to exact values as possible to obtain on the EPLL output with given PLL coefficients. This patch is required after recent change of the EPLL rate table by patch "clk: samsung: exynos5420: The EPLL rate table corrections". Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- It would be good to apply this patch for 4.13-rc, I plan to get applied corresponding clk patch also to v4.13-rc. --- sound/soc/samsung/odroid.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 0c0b00e..0834319 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -42,17 +42,17 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 32000: case 64000: - pll_freq = 131072000U; + pll_freq = 131072006U; break; case 44100: case 88200: case 176400: - pll_freq = 180633600U; + pll_freq = 180633609U; break; case 48000: case 96000: case 192000: - pll_freq = 196608000U; + pll_freq = 196608001U; break; default: return -EINVAL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Oder Chiou 21 Jul '17

21 Jul '17

The patch uses the local variable to get the data from SPI device to prevent to get all zero data. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 950d1ff..e693204 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -289,7 +289,7 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) { u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; int status; - u8 write_buf[8]; + u8 write_buf[8], read_buf[RT5514_SPI_BUF_LEN]; unsigned int i, end, offset = 0; struct spi_message message; @@ -319,7 +319,7 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) spi_message_add_tail(&x[1], &message); x[2].len = end; - x[2].rx_buf = rxbuf + offset; + x[2].rx_buf = read_buf; spi_message_add_tail(&x[2], &message); status = spi_sync(rt5514_spi, &message); @@ -327,6 +327,8 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) if (status) return false; + memcpy(rxbuf + offset, read_buf, end); + offset += RT5514_SPI_BUF_LEN; } @@ -365,14 +367,9 @@ int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) { u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; - u8 *write_buf; + u8 write_buf[RT5514_SPI_BUF_LEN + 6]; unsigned int i, end, offset = 0; - write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); - - if (write_buf == NULL) - return -ENOMEM; - while (offset < len) { if (offset + RT5514_SPI_BUF_LEN <= len) end = RT5514_SPI_BUF_LEN; @@ -403,8 +400,6 @@ int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) offset += RT5514_SPI_BUF_LEN; } - kfree(write_buf); - return 0; } EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); -- 2.7.4

2 1

[alsa-devel] Boss BR 80 Problem
by Benjamin 21 Jul '17

21 Jul '17

Hello Alsa-Developpers! Thank you for your excellent work. It's great how many devices have worked flawlessly over the years of me using Linux;) There seems to be a problem with the Boss Micro Br 80 that me and others are experiencing: https://linuxmusicians.com/viewtopic.php?f=6&t=17248 It's recognised but will only produce errors. It is reporting working with older Kernels, so maybe someone could look into that! Thank you, Ben

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[alsa-devel] No Loopback for ALC225/ALC295
by Kailang 21 Jul '17

21 Jul '17

Hi Takashi, There are also no loopback mixer for ALC225 / ALC295 codecs. Attach is the update patch. Many Thanks. BR, Kailang

2 1

[alsa-devel] ALC225 headset mode update
by Kailang 21 Jul '17

21 Jul '17

Hi Takashi, Dell also create new layout for ALC225. Headset mode policy need to change to match new layout. Attach patch was the update. Many thanks. BR, Kailang

2 1

[alsa-devel] [Test Request] Component level pcm_new/pcm_free v2
by Kuninori Morimoto 21 Jul '17

21 Jul '17

Hi Mark, Johan, Brian These are v2 of Component level pcm_new/free patches. I think current ALSA SoC need to cleanup and move to new style. Then, current all CPU/Codec/Platform features needs to be merged into "component" (and remove CPU/Codec/Platform categorize). My previous pcm_new/free patch had wrong operation, thus Johan, Brian got damage from it. I'm so sorry about that. Current ALSA SoC has rtd->platform, but it should be removed in new style, I think. So, this patch adds new "component list" for rtd (= I named it as rtdcom, maybe we want to have more nice naming). This list includes all components (= CPU/Codec/Platform), not only platform component. I'm thinking that rtd->platform related operation will be replaced like below - if(rtd->platform->xxx) - rtd->platform->xxx(yyy); + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; /* rtd connected CPU/Codec/Platform */ + if (component->xxx) + component->xxx(yyy); + } Here, xxxx is pcm_new/free on this patch. If my understanding was correct, this v2 pcm_new/free works well for you, but I don't know. So, can you please test these patches ? These are based on mark/fix/multi-pcm Kuninori Morimoto (3): ASoC: soc-core: add snd_soc_rtdcom_xxx() ASoC: use snd_soc_rtdcom_add() and convert to consistent operation ASoC: add Component level pcm_new/pcm_free v2 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v3 00/12] AC97 device/driver model revamp
by Robert Jarzmik 20 Jul '17

20 Jul '17

Hi Lars, Mark, Charles, Lee, This is a resend of the v2, with patch 1 added (forgotten in v2). Patch 12 was slightly modified after module unloading tests, and Dmitry acks were added. This serie has been stalled for almost one year, as I wasn't feeling that well lately. I'm quite better now, so let's get back to it. The following sentences is a what was already said in v2. As it's been so long since the last review, I basically withdrew all my former objections and took into account all remarks regardless of my own judgement. This is especially the case with Lars's review on the bus code, where I tried to address all his comments. For this next round, let's sum up what happened : - I rebased against latest Linus's tree => as Lars had made ports to regmap of several impacted wm97xx drivers, all the better for me. - ALSA: ac97: add an ac97 bus => this addresses Lars's comments => history : https://patchwork.kernel.org/patch/9398229/ => there is one remark which was left out, the drop of the table ac97_controller->codecs. The reason is that there are traversals required in the bus scan and controller removal, and the code is way simpler with a table, and the data structure very compact. - Input: wm97xx: split out touchscreen registering => Charles's comment taken into account => history: http://https://patchwork.kernel.org/patch/9398223/ Have a happy review. -- Robert Robert Jarzmik (12): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic Input: wm97xx: split out touchscreen registering mfd: wm97xx-core: core support for wm97xx Codec Input: wm97xx: add new AC97 bus support ASoC: wm9713: add ac97 new bus support ASoC: wm9712: add ac97 new bus support ASoC: wm9705: add private structure ASoC: wm9705: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support drivers/input/touchscreen/Kconfig | 2 +- drivers/input/touchscreen/wm97xx-core.c | 252 ++++++++++----- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 1 + drivers/mfd/wm97xx-core.c | 282 +++++++++++++++++ include/linux/mfd/wm97xx.h | 31 ++ include/sound/ac97/codec.h | 118 +++++++ include/sound/ac97/compat.h | 21 ++ include/sound/ac97/controller.h | 85 ++++++ include/sound/ac97/regs.h | 262 ++++++++++++++++ include/sound/ac97_codec.h | 239 +-------------- include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 + sound/ac97/bus.c | 526 ++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 + sound/ac97/snd_ac97_compat.c | 105 +++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 37 ++- sound/arm/pxa2xx-ac97.c | 35 ++- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/wm9705.c | 59 ++-- sound/soc/codecs/wm9712.c | 37 ++- sound/soc/codecs/wm9713.c | 39 ++- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 46 ++- 30 files changed, 1866 insertions(+), 414 deletions(-) create mode 100644 drivers/mfd/wm97xx-core.c create mode 100644 include/linux/mfd/wm97xx.h create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.11.0

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[alsa-devel] [PATCH v3] ASoC: sgtl5000: add avc support
by Richard Leitner 20 Jul '17

20 Jul '17

The sgtl5000 features a automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening. This patch adds support for this AVC block to the driver. Apart from the "AVC Switch" control which enables the block following controls for the configuration of AVC are added: + AVC Threshold Volume: threshold where audio is compressed when the measured level is above or expanded when below + AVC Max Gain Volume: maximum gain which can be applied when the measured audio level is below threshold + AVC Hard Limiter Switch: when enabled the signal is limited to the programmed threshold. + AVC Integrator Response: response time of the integrator The AVC block is enabled and configured using the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers. Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:145: CHECK: Alignment should match open parenthesis patch:148: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- CHANGES v3: - rename "AVC Enable Switch" to "AVC Switch" - add AVC configuration controls CHANGES v2: - added trailing "Switch" as covered in ControlNames.txt --- sound/soc/codecs/sgtl5000.c | 87 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 87 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702e..5d4b69f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,63 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = clamp((int)ucontrol->value.integer.value[0], 0, 96); + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +467,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -434,6 +511,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + /* Automatic Volume Control (DAP AVC) */ + SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ -- 2.1.4

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[alsa-devel] [PATCH v2 0/3]ASoC: Intel: Skylake: Enable hdmi for GLK
by Pradeep Tewani 20 Jul '17

20 Jul '17

GLK vendor nid is different from rest of the skylake platforms. Add vendor nid as part of driver data and use it. Also enable support for GLK platform. Add the patch to remove the .owner initialization in bxt driver as well Changes in v2: Removed the .owner field in driver Replaced the strcmp with strncmp for bxt_rt298 probe Pradeep Tewani (3): ASoC: Intel: board: Remove .owner initialization in bxt_rt298 driver ASoC: hdac_hdmi: Add the vendor nid for Geminilake HDMI ASoC: Intel: board: Add Geminilake platform support sound/soc/codecs/hdac_hdmi.c | 39 ++++++++++++++--- sound/soc/intel/boards/bxt_rt298.c | 85 ++++++++++++++++++++++++++++++++++++-- 2 files changed, 115 insertions(+), 9 deletions(-) -- 2.7.4

3 6

[alsa-devel] [PATCH 0/2] ASoC: Intel: Skylake: Enable hdmi for GLK
by Pradeep Tewani 20 Jul '17

20 Jul '17

GLK vendor nid is different from rest of the skylake platforms. Add vendor nid as part of driver data and use it. Also enable support for GLK platform. Pradeep Tewani (2): ASoC: Intel: board: Add Geminilake platform support ASoC: hdac_hdmi: Add the vendor nid for Geminilake HDMI sound/soc/codecs/hdac_hdmi.c | 39 +++++++++++++++--- sound/soc/intel/boards/bxt_rt298.c | 82 ++++++++++++++++++++++++++++++++++++-- 2 files changed, 113 insertions(+), 8 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 0/3] ASoC: rt5677: Few clean ups to the driver
by Andy Shevchenko 20 Jul '17

20 Jul '17

This series removes legacy and unused platform data along with fixing code which compiler complains about. In v2: - fix compilation error (kbuild bot) - add couple more patches to fix the code according to compile complains Andy Shevchenko (3): ASoC: rt5677: Hide platform data in the module sources ASoC: rt5677: Refactor code to avoid comparison unsigned >= 0 ASoC: rt5677: Remove never used variables include/sound/rt5677.h | 45 -------------------------- sound/soc/codecs/rt5677.c | 81 ++++++++++++++++++++++------------------------- sound/soc/codecs/rt5677.h | 30 +++++++++++++++++- 3 files changed, 67 insertions(+), 89 deletions(-) delete mode 100644 include/sound/rt5677.h -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rt5514: Support the TDM docking mode
by Oder Chiou 20 Jul '17

20 Jul '17

The TDM ducking mode can share the bus to other devices simultaneously. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 33 ++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5514.h | 12 ++++++++++++ 2 files changed, 44 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 4ca73bf..a105c84 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -871,11 +871,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, { struct snd_soc_codec *codec = dai->codec; struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - unsigned int val = 0; + unsigned int val = 0, val2 = 0; if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; + switch (tx_mask) { + case 0x3: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0x30: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + case 0xf: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0xf0: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + default: + break; + } + + + switch (slots) { case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; @@ -921,6 +948,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | RT5514_TDM_MODE2, val); + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL2, + RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH_MASK | + RT5514_TDM_DOCKING_START_MASK, val2); + return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212..75024a4 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -164,6 +164,18 @@ #define RT5514_I2S_DL_24 (0x2 << 0) #define RT5514_I2S_DL_8 (0x3 << 0) +/* RT5514_I2S_CTRL2 (0x2014) */ +#define RT5514_TDM_DOCKING_MODE (0x1 << 31) +#define RT5514_TDM_DOCKING_MODE_SFT 31 +#define RT5514_TDM_DOCKING_VALID_CH_MASK (0x1 << 29) +#define RT5514_TDM_DOCKING_VALID_CH_SFT 29 +#define RT5514_TDM_DOCKING_VALID_CH2 (0x0 << 29) +#define RT5514_TDM_DOCKING_VALID_CH4 (0x1 << 29) +#define RT5514_TDM_DOCKING_START_MASK (0x1 << 28) +#define RT5514_TDM_DOCKING_START_SFT 28 +#define RT5514_TDM_DOCKING_START_SLOT0 (0x0 << 28) +#define RT5514_TDM_DOCKING_START_SLOT4 (0x1 << 28) + /* RT5514_DIG_SOURCE_CTRL (0x20a4) */ #define RT5514_AD1_DMIC_INPUT_SEL (0x1 << 1) #define RT5514_AD1_DMIC_INPUT_SEL_SFT 1 -- 2.7.4

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[alsa-devel] [PATCH 1/3] ASoC: rt5665: fix GPIO6 pin function define
by Bard Liao 20 Jul '17

20 Jul '17

The GPIO6 pin function select value was wrong. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 1db5c6a..d95249c 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1692,8 +1692,8 @@ #define RT5665_GP6_PIN_MASK (0x3 << 5) #define RT5665_GP6_PIN_SFT 5 #define RT5665_GP6_PIN_GPIO6 (0x0 << 5) -#define RT5665_GP6_PIN_BCLK3 (0x0 << 5) -#define RT5665_GP6_PIN_PDM_SCL (0x1 << 5) +#define RT5665_GP6_PIN_BCLK3 (0x1 << 5) +#define RT5665_GP6_PIN_PDM_SCL (0x2 << 5) #define RT5665_GP7_PIN_MASK (0x3 << 3) #define RT5665_GP7_PIN_SFT 3 #define RT5665_GP7_PIN_GPIO7 (0x0 << 3) -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: rt5514: Support the TDM docking mode
by Oder Chiou 20 Jul '17

20 Jul '17

The TDM docking mode can share the bus to other devices simultaneously. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 33 ++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5514.h | 12 ++++++++++++ 2 files changed, 44 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 4ca73bf..a105c84 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -871,11 +871,38 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, { struct snd_soc_codec *codec = dai->codec; struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - unsigned int val = 0; + unsigned int val = 0, val2 = 0; if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; + switch (tx_mask) { + case 0x3: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0x30: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH2 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + case 0xf: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT0; + break; + + case 0xf0: + val2 |= RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH4 | + RT5514_TDM_DOCKING_START_SLOT4; + break; + + default: + break; + } + + + switch (slots) { case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; @@ -921,6 +948,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | RT5514_TDM_MODE2, val); + regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL2, + RT5514_TDM_DOCKING_MODE | RT5514_TDM_DOCKING_VALID_CH_MASK | + RT5514_TDM_DOCKING_START_MASK, val2); + return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212..75024a4 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -164,6 +164,18 @@ #define RT5514_I2S_DL_24 (0x2 << 0) #define RT5514_I2S_DL_8 (0x3 << 0) +/* RT5514_I2S_CTRL2 (0x2014) */ +#define RT5514_TDM_DOCKING_MODE (0x1 << 31) +#define RT5514_TDM_DOCKING_MODE_SFT 31 +#define RT5514_TDM_DOCKING_VALID_CH_MASK (0x1 << 29) +#define RT5514_TDM_DOCKING_VALID_CH_SFT 29 +#define RT5514_TDM_DOCKING_VALID_CH2 (0x0 << 29) +#define RT5514_TDM_DOCKING_VALID_CH4 (0x1 << 29) +#define RT5514_TDM_DOCKING_START_MASK (0x1 << 28) +#define RT5514_TDM_DOCKING_START_SFT 28 +#define RT5514_TDM_DOCKING_START_SLOT0 (0x0 << 28) +#define RT5514_TDM_DOCKING_START_SLOT4 (0x1 << 28) + /* RT5514_DIG_SOURCE_CTRL (0x20a4) */ #define RT5514_AD1_DMIC_INPUT_SEL (0x1 << 1) #define RT5514_AD1_DMIC_INPUT_SEL_SFT 1 -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt274: correct comment style
by Bard Liao 19 Jul '17

19 Jul '17

There was a comment style issue in the driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt274.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 81e29ba..fb683ff 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1159,7 +1159,7 @@ static int rt274_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt274->regmap, 0x6f, 0x3000, 0x2000); /* HP DC Calibration */ regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0x0); - //Set NID=58h.Index 00h [15]= 1b; + /* Set NID=58h.Index 00h [15]= 1b; */ regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888); msleep(500); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Correct the mixer switch setting and remove redundant routing path
by Oder Chiou 19 Jul '17

19 Jul '17

The patch corrects the mixer siwtch setting that was the mistake in the previous commitment. And remove the redundant routing paths. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index d324817..38c4027 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -59,7 +59,7 @@ struct rt5663_priv { }; static const struct reg_sequence rt5663_patch_list[] = { - { 0x002a, 0x8080 }, + { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, }; @@ -482,7 +482,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0023, 0x0039 }, { 0x0026, 0xc0c0 }, { 0x0029, 0x8080 }, - { 0x002a, 0x8080 }, + { 0x002a, 0x8020 }, { 0x002c, 0x000c }, { 0x002d, 0x0000 }, { 0x0040, 0x0808 }, @@ -1967,8 +1967,8 @@ static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { }; static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = { - SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, - RT5663_M_DAC_L1_STO_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, + RT5663_M_DAC_R1_STO_R_SHIFT, 1, 1), }; /* Out Switch */ @@ -2358,11 +2358,9 @@ static const struct snd_soc_dapm_route rt5663_dapm_routes[] = { { "DAC R1", NULL, "ADDA MIXR" }, { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, - { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, - { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt274: add rt274 codec driver
by Mark Brown 19 Jul '17

19 Jul '17

On Wed, Jul 19, 2017 at 02:08:58AM +0000, Bard Liao wrote: > I have sent a patch for it. But I was wondering why checkpatch.pl can't > find the style issue? No idea, sorry.

1 0

[alsa-devel] Patch for Digidesign Mbox 2 (set sample rate)
by Viktor Radnai 19 Jul '17

19 Jul '17

Hi there, I have recently bought a Digidesign Mbox 2 Mini and found the lack of support for 44100Hz sample rate in the driver limiting (the sample rate was hard-set to 48000Hz by the original author even though the device supports it, because he couldn't get the frame rate switching to work reliably). I had a go at hacking the driver and came up with the following patch which appears to work on the MBox 2 Mini: https://pastebin.com/kzzTDj9T I have built and tested this with Linux 4.8.15 without any realtime patches. I am planning to pick up a "full size" Mbox 2 as well, in the meantime if anyone on the list owns one I would appreciate if you could test this and let me know the result. There is a possibility that it will not work without the "magic" init sequence that I have removed because it does not look like it's needed on the Mini. Although I would be a bit surprised because the two devices are apparently identical to the point that the Mini allows the use of it's physically nonexistent features. Once someone could confirm that this works, I would like to get it merged. This is my first attempt at submitting a patch for any Linux device driver so please let me know if I need to do anything else :) Cheers, Vik -- My other sig is hilarious

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[alsa-devel] Alsa library plugin license
by Jean-Francois Dockes 19 Jul '17

19 Jul '17

Hello, I was requested to study the possibility to implement an alsa library pcm plugin doing specific audio processing. I had a look around, but I could not find an explicit answer to this question: is a closed source alsa library plugin possible ? My guess would be that it has to be GPL'd, but I have to ask. Cheers, jf

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