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

3 6

[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

3 4

[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

3 9

[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

2 1

[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

2 5

[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

1 0

[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

2 1

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

3 6

[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

2 1

[alsa-devel] [PATCH] sound: fsl_dma: remove dma_object path member
by Rob Herring 19 Jul '17

19 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 --- sound/soc/fsl/fsl_dma.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..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]; }; /* @@ -897,20 +896,19 @@ 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; } - 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

2 1

[alsa-devel] [PATCH] sound: Convert to using %pOF instead of full_name
by Rob Herring 18 Jul '17

18 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 --- sound/aoa/codecs/tas.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 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/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

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: add AVC configuration controls
by Richard Leitner 18 Jul '17

18 Jul '17

For the configuration of the automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening, following controls 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 These controls use the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers for configuration. This patch depends on "ASoC: sgtl5000: add avc enable control" Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:144: CHECK: Alignment should match open parenthesis patch:147: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- sound/soc/codecs/sgtl5000.c | 85 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0477f2e..2bb3235 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 */ { @@ -435,7 +512,15 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + /* Automatic Volume Control (DAP AVC) */ SOC_SINGLE("AVC Enable 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

3 2

[alsa-devel] [PATCH] ALSA: hda: constify pci_device_id.
by Arvind Yadav 18 Jul '17

18 Jul '17

pci_device_id are not supposed to change at runtime. All functions working with pci_device_id provided by <linux/pci.h> work with const pci_device_id. So mark the non-const structs as const. File size before: text data bss dec hex filename 2610 304 8 2922 b6a sound/hda/hdac_i915.o File size After adding 'const': text data bss dec hex filename 2674 240 8 2922 b6a sound/hda/hdac_i915.o Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/hda/hdac_i915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 0659bf3..038a180 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -336,7 +336,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); /* check whether intel graphics is present */ static bool i915_gfx_present(void) { - static struct pci_device_id ids[] = { + static const struct pci_device_id ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_BASE_CLASS_DISPLAY << 16, .class_mask = 0xff << 16 }, -- 2.7.4

2 1

[alsa-devel] [PATCH v2 1/7] ALSA: als100: fix format string overflow warning
by Arnd Bergmann 18 Jul '17

18 Jul '17

The compiler sees that the format string might overflow for the longname: sound/isa/als100.c: In function 'snd_als100_pnp_detect': sound/isa/als100.c:225:27: error: ', dma ' directive writing 6 bytes into a region of size between 0 and 64 [-Werror=format-overflow=] sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/isa/als100.c:225:3: note: 'sprintf' output between 24 and 113 bytes into a destination of size 80 sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", Open-coding "shortname" here gets us below the limit, and using snprintf() is a good idea too. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/isa/als100.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/isa/als100.c b/sound/isa/als100.c index bc9ea306ee02..6a2c5b48f3ae 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -222,15 +222,16 @@ static int snd_card_als100_probe(int dev, if (pid->driver_data == SB_HW_DT019X) { strcpy(card->driver, "DT-019X"); strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); + snprintf(card->longname, sizeof(card->longname), + "Diamond Tech. DT-019X, %s at 0x%lx, irq %d, dma %d", + chip->name, chip->port, irq[dev], dma8[dev]); } else { strcpy(card->driver, "ALS100"); strcpy(card->shortname, "Avance Logic ALS100"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev], dma16[dev]); + snprintf(card->longname, sizeof(card->longname), + "Avance Logic ALS100, %s at 0x%lx, irq %d, dma %d&%d", + chip->name, chip->port, irq[dev], dma8[dev], + dma16[dev]); } if ((error = snd_sb16dsp_pcm(chip, 0)) < 0) { -- 2.9.0

2 7

[alsa-devel] [PATCH] ASoC: rt5663: Modify the default value for ASRC function
by Oder Chiou 18 Jul '17

18 Jul '17

The patch modifies the default value for ASRC function. It could prevent the pop noise and recording no sound with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ac6dd25..c46d47f 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -60,6 +60,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8080 }, + { 0x0086, 0x0028 }, }; static const struct reg_default rt5663_v2_reg[] = { @@ -509,7 +510,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0082, 0x0000 }, { 0x0083, 0x0000 }, { 0x0084, 0x0000 }, - { 0x0086, 0x0008 }, + { 0x0086, 0x0028 }, { 0x0087, 0x0000 }, { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: rt274: add rt274 codec driver
by Bard Liao 18 Jul '17

18 Jul '17

RT274 is a HD-A/SOC dual mode codec. This is the initial codec driver of SOC mode. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt274.txt | 33 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt274.c | 1228 +++++++++++++++++++++ sound/soc/codecs/rt274.h | 217 ++++ 5 files changed, 1487 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt274.txt create mode 100644 sound/soc/codecs/rt274.c create mode 100644 sound/soc/codecs/rt274.h diff --git a/Documentation/devicetree/bindings/sound/rt274.txt b/Documentation/devicetree/bindings/sound/rt274.txt new file mode 100644 index 0000000..e9a6178 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt274.txt @@ -0,0 +1,33 @@ +RT274 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt274". + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + + +Pins on the device (for linking into audio routes) for RT274: + + * DMIC1 Pin + * DMIC2 Pin + * MIC + * LINE1 + * LINE2 + * HPO Pin + * SPDIF + * LINE3 + +Example: + +codec: rt274@1c { + compatible = "realtek,rt274"; + reg = <0x1c>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6c78b0b..024ddc9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -114,6 +114,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C select SND_SOC_RT5514 if I2C @@ -716,11 +717,17 @@ config SND_SOC_RL6231 config SND_SOC_RL6347A tristate + default y if SND_SOC_RT274=y default y if SND_SOC_RT286=y default y if SND_SOC_RT298=y + default m if SND_SOC_RT274=m default m if SND_SOC_RT286=m default m if SND_SOC_RT298=m +config SND_SOC_RT274 + tristate + depends on I2C + config SND_SOC_RT286 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1755a54..a8a4b07 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -113,6 +113,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o +snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o @@ -349,6 +350,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o +obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c new file mode 100644 index 0000000..81e29ba --- /dev/null +++ b/sound/soc/codecs/rt274.c @@ -0,0 +1,1228 @@ +/* + * rt274.c -- RT274 ALSA SoC audio codec driver + * + * Copyright 2017 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/dmi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> + +#include "rl6347a.h" +#include "rt274.h" + +#define RT274_VENDOR_ID 0x10ec0274 + +struct rt274_priv { + struct reg_default *index_cache; + int index_cache_size; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + int clk_id; + int fs; + bool master; +}; + +static const struct reg_default rt274_index_def[] = { + { 0x00, 0x1004 }, + { 0x01, 0xaaaa }, + { 0x02, 0x88aa }, + { 0x03, 0x0002 }, + { 0x04, 0xaa09 }, + { 0x05, 0x0700 }, + { 0x06, 0x6110 }, + { 0x07, 0x0200 }, + { 0x08, 0xa807 }, + { 0x09, 0x0021 }, + { 0x0a, 0x7770 }, + { 0x0b, 0x7770 }, + { 0x0c, 0x002b }, + { 0x0d, 0x2420 }, + { 0x0e, 0x65c0 }, + { 0x0f, 0x7770 }, + { 0x10, 0x0420 }, + { 0x11, 0x7418 }, + { 0x12, 0x6bd0 }, + { 0x13, 0x645f }, + { 0x14, 0x0400 }, + { 0x15, 0x8ccc }, + { 0x16, 0x4c50 }, + { 0x17, 0xff00 }, + { 0x18, 0x0003 }, + { 0x19, 0x2c11 }, + { 0x1a, 0x830b }, + { 0x1b, 0x4e4b }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + { 0x1e, 0x0000 }, + { 0x1f, 0x0000 }, + { 0x20, 0x51ff }, + { 0x21, 0x8000 }, + { 0x22, 0x8f00 }, + { 0x23, 0x88f4 }, + { 0x24, 0x0000 }, + { 0x25, 0x0000 }, + { 0x26, 0x0000 }, + { 0x27, 0x0000 }, + { 0x28, 0x0000 }, + { 0x29, 0x3000 }, + { 0x2a, 0x0000 }, + { 0x2b, 0x0000 }, + { 0x2c, 0x0f00 }, + { 0x2d, 0x100f }, + { 0x2e, 0x2902 }, + { 0x2f, 0xe280 }, + { 0x30, 0x1000 }, + { 0x31, 0x8400 }, + { 0x32, 0x5aaa }, + { 0x33, 0x8420 }, + { 0x34, 0xa20c }, + { 0x35, 0x096a }, + { 0x36, 0x5757 }, + { 0x37, 0xfe05 }, + { 0x38, 0x4901 }, + { 0x39, 0x110a }, + { 0x3a, 0x0010 }, + { 0x3b, 0x60d9 }, + { 0x3c, 0xf214 }, + { 0x3d, 0xc2ba }, + { 0x3e, 0xa928 }, + { 0x3f, 0x0000 }, + { 0x40, 0x9800 }, + { 0x41, 0x0000 }, + { 0x42, 0x2000 }, + { 0x43, 0x3d90 }, + { 0x44, 0x4900 }, + { 0x45, 0x5289 }, + { 0x46, 0x0004 }, + { 0x47, 0xa47a }, + { 0x48, 0xd049 }, + { 0x49, 0x0049 }, + { 0x4a, 0xa83b }, + { 0x4b, 0x0777 }, + { 0x4c, 0x065c }, + { 0x4d, 0x7fff }, + { 0x4e, 0x7fff }, + { 0x4f, 0x0000 }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0xbf5f }, + { 0x53, 0x3320 }, + { 0x54, 0xcc00 }, + { 0x55, 0x0000 }, + { 0x56, 0x3f00 }, + { 0x57, 0x0000 }, + { 0x58, 0x0000 }, + { 0x59, 0x0000 }, + { 0x5a, 0x1300 }, + { 0x5b, 0x005f }, + { 0x5c, 0x0000 }, + { 0x5d, 0x1001 }, + { 0x5e, 0x1000 }, + { 0x5f, 0x0000 }, + { 0x60, 0x5554 }, + { 0x61, 0xffc0 }, + { 0x62, 0xa000 }, + { 0x63, 0xd010 }, + { 0x64, 0x0000 }, + { 0x65, 0x3fb1 }, + { 0x66, 0x1881 }, + { 0x67, 0xc810 }, + { 0x68, 0x2000 }, + { 0x69, 0xfff0 }, + { 0x6a, 0x0300 }, + { 0x6b, 0x5060 }, + { 0x6c, 0x0000 }, + { 0x6d, 0x0000 }, + { 0x6e, 0x0c25 }, + { 0x6f, 0x0c0b }, + { 0x70, 0x8000 }, + { 0x71, 0x4008 }, + { 0x72, 0x0000 }, + { 0x73, 0x0800 }, + { 0x74, 0xa28f }, + { 0x75, 0xa050 }, + { 0x76, 0x7fe8 }, + { 0x77, 0xdb8c }, + { 0x78, 0x0000 }, + { 0x79, 0x0000 }, + { 0x7a, 0x2a96 }, + { 0x7b, 0x800f }, + { 0x7c, 0x0200 }, + { 0x7d, 0x1600 }, + { 0x7e, 0x0000 }, + { 0x7f, 0x0000 }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt274_index_def) + +static const struct reg_default rt274_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x00000057 }, + { 0x0023a000, 0x00000057 }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x00000097 }, + { 0x00936000, 0x00000097 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000400 }, + { 0x00b37200, 0x00000400 }, + { 0x00b37300, 0x00000400 }, + { 0x00c37000, 0x00000400 }, + { 0x00c37100, 0x00000400 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01937000, 0x00000000 }, + { 0x01970500, 0x00000400 }, + { 0x02050000, 0x0000001b }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000001 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x01970700, 0x00000020 }, + { 0x00830000, 0x00000097 }, + { 0x00930000, 0x00000097 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt274_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT274_GET_HP_SENSE: + case RT274_GET_MIC_SENSE: + case RT274_PROC_COEF: + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0): + return true; + default: + return false; + } + + +} + +static bool rt274_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT274_GET_HP_SENSE: + case RT274_GET_MIC_SENSE: + case RT274_SET_AUDIO_POWER: + case RT274_SET_HPO_POWER: + case RT274_SET_DMIC1_POWER: + case RT274_LOUT_MUX: + case RT274_HPO_MUX: + case RT274_ADC0_MUX: + case RT274_ADC1_MUX: + case RT274_SET_MIC: + case RT274_SET_PIN_HPO: + case RT274_SET_PIN_LOUT3: + case RT274_SET_PIN_DMIC1: + case RT274_SET_AMP_GAIN_HPO: + case RT274_SET_DMIC2_DEFAULT: + case RT274_DAC0L_GAIN: + case RT274_DAC0R_GAIN: + case RT274_DAC1L_GAIN: + case RT274_DAC1R_GAIN: + case RT274_ADCL_GAIN: + case RT274_ADCR_GAIN: + case RT274_MIC_GAIN: + case RT274_HPOL_GAIN: + case RT274_HPOR_GAIN: + case RT274_LOUTL_GAIN: + case RT274_LOUTR_GAIN: + case RT274_DAC_FORMAT: + case RT274_ADC_FORMAT: + case RT274_COEF_INDEX: + case RT274_PROC_COEF: + case RT274_SET_AMP_GAIN_ADC_IN1: + case RT274_SET_AMP_GAIN_ADC_IN2: + case RT274_SET_POWER(RT274_DAC_OUT0): + case RT274_SET_POWER(RT274_DAC_OUT1): + case RT274_SET_POWER(RT274_ADC_IN1): + case RT274_SET_POWER(RT274_ADC_IN2): + case RT274_SET_POWER(RT274_DMIC2): + case RT274_SET_POWER(RT274_MIC): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_STREAM_FORMAT, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN1, 0): + case VERB_CMD(AC_VERB_GET_CONNECT_SEL, RT274_MIXER_IN2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_DMIC2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE1, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_LINE2, 0): + case VERB_CMD(AC_VERB_GET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_HP_OUT, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_MIC, 0): + case VERB_CMD(AC_VERB_GET_UNSOLICITED_RESPONSE, RT274_INLINE_CMD, 0): + return true; + default: + return false; + } +} + +#ifdef CONFIG_PM +static void rt274_index_sync(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt274->index_cache[i].reg, + rt274->index_cache[i].def); + } +} +#endif + +static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) +{ + unsigned int buf; + + *hp = false; + *mic = false; + + if (!rt274->codec) + return -EINVAL; + + regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + *mic = buf & 0x80000000; + + pr_debug("*hp = %d *mic = %d\n", *hp, *mic); + + return 0; +} + +static void rt274_jack_detect_work(struct work_struct *work) +{ + struct rt274_priv *rt274 = + container_of(work, struct rt274_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + if (rt274_jack_detect(rt274, &hp, &mic) < 0) + return; + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt274->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +static irqreturn_t rt274_irq(int irq, void *data); + +static int rt274_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, void *data) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + if (jack == NULL) { + /* Disable jack detection */ + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_EN, RT274_IRQ_DIS); + + return 0; + } + rt274->jack = jack; + + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_EN, RT274_IRQ_EN); + + /* Send an initial report */ + rt274_irq(0, rt274); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt274_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT274_DAC0L_GAIN, + RT274_DAC0R_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", RT274_DAC1L_GAIN, + RT274_DAC1R_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT274_ADCL_GAIN, + RT274_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT274_ADCL_GAIN, + RT274_ADCR_GAIN, RT274_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("AMIC Volume", RT274_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), +}; + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOL_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_HPOR_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new loutl_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTL_GAIN, + RT274_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new loutr_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT274_LOUTR_GAIN, + RT274_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt274_adc_src[] = { + "Mic", "Line1", "Line2", "Dmic" +}; + +static SOC_ENUM_SINGLE_DECL( + rt274_adc0_enum, RT274_ADC0_MUX, RT274_ADC_SEL_SFT, + rt274_adc_src); + +static const struct snd_kcontrol_new rt274_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt274_adc0_enum); + +static SOC_ENUM_SINGLE_DECL( + rt274_adc1_enum, RT274_ADC1_MUX, RT274_ADC_SEL_SFT, + rt274_adc_src); + +static const struct snd_kcontrol_new rt274_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt274_adc1_enum); + +static const char * const rt274_dac_src[] = { + "DAC OUT0", "DAC OUT1" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt274_hpo_enum, RT274_HPO_MUX, + 0, rt274_dac_src); + +static const struct snd_kcontrol_new rt274_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt274_hpo_enum); + +/* Line out source */ +static SOC_ENUM_SINGLE_DECL(rt274_lout_enum, RT274_LOUT_MUX, + 0, rt274_dac_src); + +static const struct snd_kcontrol_new rt274_lout_mux = +SOC_DAPM_ENUM("LOUT source", rt274_lout_enum); + +static const struct snd_soc_dapm_widget rt274_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + + /* DMIC */ + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, RT274_SET_STREAMID_ADC1, 4, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, RT274_SET_STREAMID_ADC2, 4, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("ADC 0 Mux", SND_SOC_NOPM, 0, 0, + &rt274_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", SND_SOC_NOPM, 0, 0, + &rt274_adc1_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RXL", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF1RXR", "AIF1 Playback", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TXL", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TXR", "AIF1 Capture", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RXL", "AIF1 Playback", 2, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RXR", "AIF1 Playback", 3, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TXL", "AIF1 Capture", 2, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TXR", "AIF1 Capture", 3, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, RT274_SET_STREAMID_DAC0, 4, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, RT274_SET_STREAMID_DAC1, 4, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt274_hpo_mux), + SND_SOC_DAPM_MUX("LOUT Mux", SND_SOC_NOPM, 0, 0, &rt274_lout_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT274_SET_PIN_HPO, + RT274_SET_PIN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LOUT Power", RT274_SET_PIN_LOUT3, + RT274_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_PGA("DAC OUT0", SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("DAC OUT1", SND_SOC_NOPM, 0, 0, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH("LOUT L", SND_SOC_NOPM, 0, 0, + &loutl_enable_control), + SND_SOC_DAPM_SWITCH("LOUT R", SND_SOC_NOPM, 0, 0, + &loutr_enable_control), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), + SND_SOC_DAPM_OUTPUT("LINE3"), +}; + +static const struct snd_soc_dapm_route rt274_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + + {"ADC 0 Mux", "Mic", "MIC"}, + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Line1", "LINE1"}, + {"ADC 0 Mux", "Line2", "LINE2"}, + {"ADC 1 Mux", "Mic", "MIC"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Line1", "LINE1"}, + {"ADC 1 Mux", "Line2", "LINE2"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TXL", NULL, "ADC 0"}, + {"AIF1TXR", NULL, "ADC 0"}, + {"AIF2TXL", NULL, "ADC 1"}, + {"AIF2TXR", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RXL"}, + {"DAC 0", NULL, "AIF1RXR"}, + {"DAC 1", NULL, "AIF2RXL"}, + {"DAC 1", NULL, "AIF2RXR"}, + + {"DAC OUT0", NULL, "DAC 0"}, + + {"DAC OUT1", NULL, "DAC 1"}, + + {"LOUT Mux", "DAC OUT0", "DAC OUT0"}, + {"LOUT Mux", "DAC OUT1", "DAC OUT1"}, + + {"LOUT L", "Switch", "LOUT Mux"}, + {"LOUT R", "Switch", "LOUT Mux"}, + {"LOUT L", NULL, "LOUT Power"}, + {"LOUT R", NULL, "LOUT Power"}, + + {"LINE3", NULL, "LOUT L"}, + {"LINE3", NULL, "LOUT R"}, + + {"HPO Mux", "DAC OUT0", "DAC OUT0"}, + {"HPO Mux", "DAC OUT1", "DAC OUT1"}, + + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt274_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code = 0, c_len_code = 0; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt274->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt274->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt274->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + c_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + c_len_code = 3; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + c_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + c_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + c_len_code = 0; + break; + default: + return -EINVAL; + } + + if (rt274->master) + c_len_code = 0x3; + + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0xc018, d_len_code << 3 | c_len_code << 14); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt274_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_M); + rt274->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_I2S_MODE_MASK, RT274_I2S_MODE_S); + rt274->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_LJ); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMA); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, RT274_I2S_CTRL1, + RT274_I2S_FMT_MASK, RT274_I2S_FMT_PCMB); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT274_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT274_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + switch (source) { + case RT274_PLL2_S_MCLK: + snd_soc_update_bits(codec, RT274_PLL2_CTRL, + RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_MCLK); + break; + default: + dev_warn(codec->dev, "invalid pll source, use BCLK\n"); + case RT274_PLL2_S_BCLK: + snd_soc_update_bits(codec, RT274_PLL2_CTRL, + RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); + break; + } + + if (source == RT274_PLL2_S_BCLK) { + snd_soc_update_bits(codec, RT274_MCLK_CTRL, + (0x3 << 12), (0x3 << 12)); + switch (rt274->fs) { + case 50: + snd_soc_write(codec, 0x7a, 0xaab6); + snd_soc_write(codec, 0x7b, 0x0301); + snd_soc_write(codec, 0x7c, 0x04fe); + break; + case 64: + snd_soc_write(codec, 0x7a, 0xaa96); + snd_soc_write(codec, 0x7b, 0x8003); + snd_soc_write(codec, 0x7c, 0x081e); + break; + case 128: + snd_soc_write(codec, 0x7a, 0xaa96); + snd_soc_write(codec, 0x7b, 0x8003); + snd_soc_write(codec, 0x7c, 0x080e); + break; + default: + dev_warn(codec->dev, "invalid freq_in, assume 4.8M\n"); + case 100: + snd_soc_write(codec, 0x7a, 0xaab6); + snd_soc_write(codec, 0x7b, 0x0301); + snd_soc_write(codec, 0x7c, 0x047e); + break; + } + } + + return 0; +} + +static int rt274_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + unsigned int clk_src, mclk_en; + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + switch (clk_id) { + case RT274_SCLK_S_MCLK: + mclk_en = RT274_MCLK_MODE_EN; + clk_src = RT274_CLK_SRC_MCLK; + break; + case RT274_SCLK_S_PLL1: + mclk_en = RT274_MCLK_MODE_DIS; + clk_src = RT274_CLK_SRC_MCLK; + break; + case RT274_SCLK_S_PLL2: + mclk_en = RT274_MCLK_MODE_EN; + clk_src = RT274_CLK_SRC_PLL2; + break; + default: + mclk_en = RT274_MCLK_MODE_DIS; + clk_src = RT274_CLK_SRC_MCLK; + dev_warn(codec->dev, "invalid sysclk source, use PLL1\n"); + break; + } + snd_soc_update_bits(codec, RT274_MCLK_CTRL, + RT274_MCLK_MODE_MASK, mclk_en); + snd_soc_update_bits(codec, RT274_CLK_CTRL, + RT274_CLK_SRC_MASK, clk_src); + + switch (freq) { + case 19200000: + if (clk_id == RT274_SCLK_S_MCLK) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT274_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (clk_id == RT274_SCLK_S_MCLK) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT274_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT274_MCLK_CTRL, 0x1fcf, 0x0008); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT274_MCLK_CTRL, 0x1fcf, 0x1543); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt274->sys_clk = freq; + rt274->clk_id = clk_id; + + return 0; +} + +static int rt274_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + rt274->fs = ratio; + if ((ratio / 50) == 0) + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt274_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) + +{ + struct snd_soc_codec *codec = dai->codec; + + if (rx_mask || tx_mask) { + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_EN); + } else { + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_EN, RT274_TDM_DIS); + return 0; + } + + switch (slots) { + case 4: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_4CH); + break; + case 2: + snd_soc_update_bits(codec, + RT274_I2S_CTRL1, RT274_TDM_CH_NUM, RT274_TDM_2CH); + break; + default: + dev_err(codec->dev, + "Support 2 or 4 slots TDM only\n"); + return -EINVAL; + } + + return 0; +} + +static int rt274_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == + snd_soc_codec_get_bias_level(codec)) { + snd_soc_write(codec, + RT274_SET_AUDIO_POWER, AC_PWRST_D0); + } + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT274_SET_AUDIO_POWER, AC_PWRST_D3); + break; + + default: + break; + } + + return 0; +} + +static irqreturn_t rt274_irq(int irq, void *data) +{ + struct rt274_priv *rt274 = data; + bool hp = false; + bool mic = false; + int ret, status = 0; + + /* Clear IRQ */ + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_IRQ_CLR, RT274_IRQ_CLR); + + ret = rt274_jack_detect(rt274, &hp, &mic); + + if (ret == 0) { + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt274->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt274->i2c->dev, 300); + } + + return IRQ_HANDLED; +} + +static int rt274_probe(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + rt274->codec = codec; + + if (rt274->i2c->irq) { + INIT_DELAYED_WORK(&rt274->jack_detect_work, + rt274_jack_detect_work); + schedule_delayed_work(&rt274->jack_detect_work, + msecs_to_jiffies(1250)); + } + + return 0; +} + +static int rt274_remove(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt274->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt274_suspend(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt274->regmap, true); + regcache_mark_dirty(rt274->regmap); + + return 0; +} + +static int rt274_resume(struct snd_soc_codec *codec) +{ + struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt274->regmap, false); + rt274_index_sync(codec); + regcache_sync(rt274->regmap); + + return 0; +} +#else +#define rt274_suspend NULL +#define rt274_resume NULL +#endif + +#define RT274_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT274_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt274_aif_dai_ops = { + .hw_params = rt274_hw_params, + .set_fmt = rt274_set_dai_fmt, + .set_sysclk = rt274_set_dai_sysclk, + .set_pll = rt274_set_dai_pll, + .set_bclk_ratio = rt274_set_bclk_ratio, + .set_tdm_slot = rt274_set_tdm_slot, +}; + +static struct snd_soc_dai_driver rt274_dai[] = { + { + .name = "rt274-aif1", + .id = RT274_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT274_STEREO_RATES, + .formats = RT274_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT274_STEREO_RATES, + .formats = RT274_FORMATS, + }, + .ops = &rt274_aif_dai_ops, + .symmetric_rates = 1, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt274 = { + .probe = rt274_probe, + .remove = rt274_remove, + .suspend = rt274_suspend, + .resume = rt274_resume, + .set_bias_level = rt274_set_bias_level, + .idle_bias_off = true, + .component_driver = { + .controls = rt274_snd_controls, + .num_controls = ARRAY_SIZE(rt274_snd_controls), + .dapm_widgets = rt274_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt274_dapm_widgets), + .dapm_routes = rt274_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt274_dapm_routes), + }, + .set_jack = rt274_mic_detect, +}; + +static const struct regmap_config rt274_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x05bfffff, + .volatile_reg = rt274_volatile_register, + .readable_reg = rt274_readable_register, + .reg_write = rl6347a_hw_write, + .reg_read = rl6347a_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt274_reg, + .num_reg_defaults = ARRAY_SIZE(rt274_reg), +}; + +#ifdef CONFIG_OF +static const struct of_device_id rt274_of_match[] = { + {.compatible = "realtek,rt274"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt274_of_match); +#endif + +static const struct i2c_device_id rt274_i2c_id[] = { + {"rt274", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); + +static const struct acpi_device_id rt274_acpi_match[] = { + { "10EC0274", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt274_acpi_match); + +static int rt274_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt274_priv *rt274; + + int ret; + unsigned int val; + + rt274 = devm_kzalloc(&i2c->dev, sizeof(*rt274), + GFP_KERNEL); + if (rt274 == NULL) + return -ENOMEM; + + rt274->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt274_regmap); + if (IS_ERR(rt274->regmap)) { + ret = PTR_ERR(rt274->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt274->regmap, + RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (val != RT274_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt274\n", val); + return -ENODEV; + } + + rt274->index_cache = devm_kmemdup(&i2c->dev, rt274_index_def, + sizeof(rt274_index_def), GFP_KERNEL); + if (!rt274->index_cache) + return -ENOMEM; + + rt274->index_cache_size = INDEX_CACHE_SIZE; + rt274->i2c = i2c; + i2c_set_clientdata(i2c, rt274); + + /* reset codec */ + regmap_write(rt274->regmap, RT274_RESET, 0); + regmap_update_bits(rt274->regmap, 0x1a, 0x4000, 0x4000); + + /* Set Pad PDB is floating */ + regmap_update_bits(rt274->regmap, RT274_PAD_CTRL12, 0x3, 0x0); + regmap_write(rt274->regmap, RT274_COEF5b_INDEX, 0x01); + regmap_write(rt274->regmap, RT274_COEF5b_COEF, 0x8540); + regmap_update_bits(rt274->regmap, 0x6f, 0x0100, 0x0100); + /* Combo jack auto detect */ + regmap_write(rt274->regmap, 0x4a, 0x201b); + /* Aux mode off */ + 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; + regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); + regmap_write(rt274->regmap, RT274_COEF58_COEF, 0xb888); + msleep(500); + regmap_update_bits(rt274->regmap, 0x6f, 0xf, 0xb); + regmap_write(rt274->regmap, RT274_COEF58_INDEX, 0x00); + regmap_write(rt274->regmap, RT274_COEF58_COEF, 0x3888); + /* Set pin widget */ + regmap_write(rt274->regmap, RT274_SET_PIN_HPO, 0x40); + regmap_write(rt274->regmap, RT274_SET_PIN_LOUT3, 0x40); + regmap_write(rt274->regmap, RT274_SET_MIC, 0x20); + regmap_write(rt274->regmap, RT274_SET_PIN_DMIC1, 0x20); + + regmap_update_bits(rt274->regmap, RT274_I2S_CTRL2, 0xc004, 0x4004); + regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, + RT274_GPI2_SEL_MASK, RT274_GPI2_SEL_DMIC_CLK); + + /* jack detection */ + regmap_write(rt274->regmap, RT274_UNSOLICITED_HP_OUT, 0x81); + regmap_write(rt274->regmap, RT274_UNSOLICITED_MIC, 0x82); + + if (rt274->i2c->irq) { + ret = request_threaded_irq(rt274->i2c->irq, NULL, rt274_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt274", rt274); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt274, + rt274_dai, ARRAY_SIZE(rt274_dai)); + + return ret; +} + +static int rt274_i2c_remove(struct i2c_client *i2c) +{ + struct rt274_priv *rt274 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt274); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt274_i2c_driver = { + .driver = { + .name = "rt274", + .acpi_match_table = ACPI_PTR(rt274_acpi_match), +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(rt274_of_match), +#endif + }, + .probe = rt274_i2c_probe, + .remove = rt274_i2c_remove, + .id_table = rt274_i2c_id, +}; + +module_i2c_driver(rt274_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT274 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt274.h b/sound/soc/codecs/rt274.h new file mode 100644 index 0000000..4fd1bcb --- /dev/null +++ b/sound/soc/codecs/rt274.h @@ -0,0 +1,217 @@ +/* + * rt274.h -- RT274 ALSA SoC audio driver + * + * Copyright 2016 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT274_H__ +#define __RT274_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT274_AUDIO_FUNCTION_GROUP 0x01 +#define RT274_DAC_OUT0 0x02 +#define RT274_DAC_OUT1 0x03 +#define RT274_ADC_IN2 0x08 +#define RT274_ADC_IN1 0x09 +#define RT274_DIG_CVT 0x0a +#define RT274_DMIC1 0x12 +#define RT274_DMIC2 0x13 +#define RT274_MIC 0x19 +#define RT274_LINE1 0x1a +#define RT274_LINE2 0x1b +#define RT274_LINE3 0x16 +#define RT274_SPDIF 0x1e +#define RT274_VENDOR_REGISTERS 0x20 +#define RT274_HP_OUT 0x21 +#define RT274_MIXER_IN1 0x22 +#define RT274_MIXER_IN2 0x23 +#define RT274_INLINE_CMD 0x55 + +#define RT274_SET_PIN_SFT 6 +#define RT274_SET_PIN_ENABLE 0x40 +#define RT274_SET_PIN_DISABLE 0 +#define RT274_SET_EAPD_HIGH 0x2 +#define RT274_SET_EAPD_LOW 0 + +#define RT274_MUTE_SFT 7 + +/* Verb commands */ +#define RT274_RESET\ + VERB_CMD(AC_VERB_SET_CODEC_RESET, RT274_AUDIO_FUNCTION_GROUP, 0) +#define RT274_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT274_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT274_SET_AUDIO_POWER RT274_SET_POWER(RT274_AUDIO_FUNCTION_GROUP) +#define RT274_SET_HPO_POWER RT274_SET_POWER(RT274_HP_OUT) +#define RT274_SET_DMIC1_POWER RT274_SET_POWER(RT274_DMIC1) +#define RT274_LOUT_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_LINE3, 0) +#define RT274_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_HP_OUT, 0) +#define RT274_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN1, 0) +#define RT274_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT274_MIXER_IN2, 0) +#define RT274_SET_MIC\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_MIC, 0) +#define RT274_SET_PIN_LOUT3\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_LINE3, 0) +#define RT274_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_HP_OUT, 0) +#define RT274_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_DMIC1, 0) +#define RT274_SET_PIN_SPDIF\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT274_SPDIF, 0) +#define RT274_SET_PIN_DIG_CVT\ + VERB_CMD(AC_VERB_SET_DIGI_CONVERT_1, RT274_DIG_CVT, 0) +#define RT274_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0) +#define RT274_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0) +#define RT274_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN2, 0) +#define RT274_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_HP_OUT, 0) +#define RT274_GET_MIC_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT274_MIC, 0) +#define RT274_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_DMIC2, 0) +#define RT274_SET_SPDIF_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT274_SPDIF, 0) +#define RT274_DAC0L_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0xa000) +#define RT274_DAC0R_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT0, 0x9000) +#define RT274_DAC1L_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0xa000) +#define RT274_DAC1R_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_DAC_OUT1, 0x9000) +#define RT274_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x6000) +#define RT274_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_ADC_IN1, 0x5000) +#define RT274_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_MIC, 0x7000) +#define RT274_LOUTL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0xa000) +#define RT274_LOUTR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_LINE3, 0x9000) +#define RT274_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0xa000) +#define RT274_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT274_HP_OUT, 0x9000) +#define RT274_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_DAC_OUT0, 0) +#define RT274_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT274_ADC_IN1, 0) +#define RT274_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT274_VENDOR_REGISTERS, 0) +#define RT274_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT274_VENDOR_REGISTERS, 0) +#define RT274_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_INLINE_CMD, 0) +#define RT274_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_HP_OUT, 0) +#define RT274_UNSOLICITED_MIC\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT274_MIC, 0) +#define RT274_COEF58_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x58, 0) +#define RT274_COEF58_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, 0x58, 0) +#define RT274_COEF5b_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, 0x5b, 0) +#define RT274_COEF5b_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, 0x5b, 0) +#define RT274_SET_STREAMID_DAC0\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT0, 0) +#define RT274_SET_STREAMID_DAC1\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_DAC_OUT1, 0) +#define RT274_SET_STREAMID_ADC1\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN1, 0) +#define RT274_SET_STREAMID_ADC2\ + VERB_CMD(AC_VERB_SET_CHANNEL_STREAMID, RT274_ADC_IN2, 0) + +/* Index registers */ +#define RT274_EAPD_GPIO_IRQ_CTRL 0x10 +#define RT274_PAD_CTRL12 0x35 +#define RT274_I2S_CTRL1 0x63 +#define RT274_I2S_CTRL2 0x64 +#define RT274_MCLK_CTRL 0x71 +#define RT274_CLK_CTRL 0x72 +#define RT274_PLL2_CTRL 0x7b + + +/* EAPD GPIO IRQ control (Index 0x10) */ +#define RT274_IRQ_DIS (0x0 << 13) +#define RT274_IRQ_EN (0x1 << 13) +#define RT274_IRQ_CLR (0x1 << 12) +#define RT274_GPI2_SEL_MASK (0x3 << 7) +#define RT274_GPI2_SEL_GPIO2 (0x0 << 7) +#define RT274_GPI2_SEL_I2S (0x1 << 7) +#define RT274_GPI2_SEL_DMIC_CLK (0x2 << 7) +#define RT274_GPI2_SEL_CBJ (0x3 << 7) + +/* Front I2S_Interface control 1 (Index 0x63) */ +#define RT274_I2S_MODE_MASK (0x1 << 11) +#define RT274_I2S_MODE_S (0x0 << 11) +#define RT274_I2S_MODE_M (0x1 << 11) +#define RT274_TDM_DIS (0x0 << 10) +#define RT274_TDM_EN (0x1 << 10) +#define RT274_TDM_CH_NUM (0x1 << 7) +#define RT274_TDM_2CH (0x0 << 7) +#define RT274_TDM_4CH (0x1 << 7) +#define RT274_I2S_FMT_MASK (0x3 << 8) +#define RT274_I2S_FMT_I2S (0x0 << 8) +#define RT274_I2S_FMT_LJ (0x1 << 8) +#define RT274_I2S_FMT_PCMA (0x2 << 8) +#define RT274_I2S_FMT_PCMB (0x3 << 8) + +/* MCLK clock domain control (Index 0x71) */ +#define RT274_MCLK_MODE_MASK (0x1 << 14) +#define RT274_MCLK_MODE_DIS (0x0 << 14) +#define RT274_MCLK_MODE_EN (0x1 << 14) + +/* Clock control (Index 0x72) */ +#define RT274_CLK_SRC_MASK (0x7 << 3) +#define RT274_CLK_SRC_MCLK (0x0 << 3) +#define RT274_CLK_SRC_PLL2 (0x3 << 3) + +/* PLL2 control (Index 0x7b) */ +#define RT274_PLL2_SRC_MASK (0x1 << 13) +#define RT274_PLL2_SRC_MCLK (0x0 << 13) +#define RT274_PLL2_SRC_BCLK (0x1 << 13) + +/* HP-OUT (0x21) */ +#define RT274_M_HP_MUX_SFT 14 +#define RT274_HP_SEL_MASK 0x1 +#define RT274_HP_SEL_SFT 0 +#define RT274_HP_SEL_F 0 +#define RT274_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT274_ADC_SEL_MASK 0x7 +#define RT274_ADC_SEL_SFT 0 +#define RT274_ADC_SEL_MIC 0 +#define RT274_ADC_SEL_LINE1 1 +#define RT274_ADC_SEL_LINE2 2 +#define RT274_ADC_SEL_DMIC 3 + +#define RT274_SCLK_S_MCLK 0 +#define RT274_SCLK_S_PLL1 1 +#define RT274_SCLK_S_PLL2 2 + +#define RT274_PLL2_S_MCLK 0 +#define RT274_PLL2_S_BCLK 1 + +enum { + RT274_AIF1, + RT274_AIFS, +}; + +#endif /* __RT274_H__ */ + -- 2.7.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: fix pcm-creation regression
by Johan Hovold 18 Jul '17

18 Jul '17

This reverts commit 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free"), which started calling the pcm_new callback for every component in a *card* when creating a new pcm, something which does not seem to make any sense. This specifically led to memory leaks in systems with more than one platform component and where DMA memory is allocated in the platform-driver callback. For example, when both mcasp devices are being used on an am335x board, DMA memory would be allocated twice for every DAI link during probe. When CONFIG_SND_VERBOSE_PROCFS was set this fortunately also led to warnings such as: WARNING: CPU: 0 PID: 565 at ../fs/proc/generic.c:346 proc_register+0x110/0x154 proc_dir_entry 'sub0/prealloc' already registered Since there seems to be no users of the new component callbacks, and the current implementation introduced a regression, let's revert the offending commit for now. Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") Cc: stable <stable(a)vger.kernel.org> # 4.10 Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Brian Norris <briannorris(a)chromium.org> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- include/sound/soc.h | 6 ------ sound/soc/soc-core.c | 25 ------------------------- sound/soc/soc-pcm.c | 32 +++++++++----------------------- 3 files changed, 9 insertions(+), 54 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9c94b97c17f8..c4a8b1947566 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -795,10 +795,6 @@ struct snd_soc_component_driver { int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - /* pcm creation and destruction */ - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); - /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -874,8 +870,6 @@ struct snd_soc_component { void (*remove)(struct snd_soc_component *); int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a01944..c240e13ba057 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3171,8 +3171,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; component->suspend = component->driver->suspend; component->resume = component->driver->resume; - component->pcm_new = component->driver->pcm_new; - component->pcm_free = component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; @@ -3360,25 +3358,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_new) - return platform->driver->pcm_new(rtd); - else - return 0; -} - -static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_free) - platform->driver->pcm_free(pcm); -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -3402,10 +3381,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->pcm_new) - platform->component.pcm_new = snd_soc_platform_drv_pcm_new; - if (platform_drv->pcm_free) - platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcc5ece08668..553f7a76743c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2628,25 +2628,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } -static void soc_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_component *component; - - list_for_each_entry(component, &rtd->card->component_dev_list, - card_list) { - if (component->pcm_free) - component->pcm_free(pcm); - } -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; @@ -2756,18 +2743,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { - if (component->pcm_new) { - ret = component->pcm_new(rtd); - if (ret < 0) { - dev_err(component->dev, - "ASoC: pcm constructor failed: %d\n", - ret); - return ret; - } + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + dev_err(platform->dev, + "ASoC: pcm constructor failed: %d\n", + ret); + return ret; } } - pcm->private_free = soc_pcm_free; + + pcm->private_free = platform->driver->pcm_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, -- 2.13.2

3 12

[alsa-devel] [PATCH 20/22] sound: pci: avoid string overflow warnings
by Arnd Bergmann 18 Jul '17

18 Jul '17

With gcc-7, we get various warnings about a possible string overflow: sound/pci/rme9652/hdspm.c: In function 'snd_hdspm_create_alsa_devices': sound/pci/rme9652/hdspm.c:2123:17: error: ' MIDIoverMADI' directive writing 13 bytes into a region of size between 1 and 32 [-Werror=format-overflow=] sound/pci/pcxhr/pcxhr.c: In function 'pcxhr_probe': sound/pci/pcxhr/pcxhr.c:1647:28: error: ' [PCM #' directive writing 7 bytes into a region of size between 1 and 32 [-Werror=format-overflow=] sound/pci/mixart/mixart.c: In function 'snd_mixart_probe': sound/pci/mixart/mixart.c:1353:28: error: ' [PCM #' directive writing 7 bytes into a region of size between 1 and 32 [-Werror=format-overflow=] sprintf(card->shortname, "%s [PCM #%d]", mgr->shortname, i); ^~~~~~~~~~~~~~ sound/pci/mixart/mixart.c:1353:28: note: using the range [-2147483648, 2147483647] for directive argument sound/pci/mixart/mixart.c:1353:3: note: 'sprintf' output between 10 and 51 bytes into a destination of size 32 sprintf(card->shortname, "%s [PCM #%d]", mgr->shortname, i); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/mixart/mixart.c:1354:27: error: ' [PCM #' directive writing 7 bytes into a region of size between 1 and 80 [-Werror=format-overflow=] sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); ^~~~~~~~~~~~~~ sound/pci/mixart/mixart.c:1354:27: note: using the range [-2147483648, 2147483647] for directive argument sound/pci/mixart/mixart.c:1354:3: note: 'sprintf' output between 10 and 99 bytes into a destination of size 80 I have checked these all and found that the driver-private shortname strings for mixart and pcxhr are longer than necessary, and making them shorter will be safe while also making it clear that no overflow can happen when they get passed as a substring into the card shortname. For hdspm, we have a local buffer of the same size as its substring. In this case, making the buffer a little longer is safe as the functions that take it as an argument all use length checking and the strings we pass into it are actually short enough. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/mixart/mixart.h | 4 ++-- sound/pci/pcxhr/pcxhr.h | 4 ++-- sound/pci/rme9652/hdspm.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h index 426743871540..c8309e327663 100644 --- a/sound/pci/mixart/mixart.h +++ b/sound/pci/mixart/mixart.h @@ -75,8 +75,8 @@ struct mixart_mgr { struct mem_area mem[2]; /* share the name */ - char shortname[32]; /* short name of this soundcard */ - char longname[80]; /* name of this soundcard */ + char shortname[16]; /* short name of this soundcard */ + char longname[40]; /* name of this soundcard */ /* one and only blocking message or notification may be pending */ u32 pending_event; diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 9e39e509a3ef..4909a43ce3d9 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -75,8 +75,8 @@ struct pcxhr_mgr { unsigned long port[3]; /* share the name */ - char shortname[32]; /* short name of this soundcard */ - char longname[96]; /* name of this soundcard */ + char shortname[16]; /* short name of this soundcard */ + char longname[40]; /* name of this soundcard */ struct pcxhr_rmh *prmh; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 254c3d040118..a1cbf5938a0e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2061,7 +2061,7 @@ static int snd_hdspm_create_midi(struct snd_card *card, struct hdspm *hdspm, int id) { int err; - char buf[32]; + char buf[64]; hdspm->midi[id].id = id; hdspm->midi[id].hdspm = hdspm; -- 2.9.0

2 2

[alsa-devel] Reset to clean state after snd_ctl_add_integer_elem_set ?
by Fulup Ar Foll 18 Jul '17

18 Jul '17

I'm using snd_ctl_add_boolean_elem_set and snd_ctl_add_integer_elem_set to create some user kernel controls. While the operation work find I did not find any simple command to erase all those controls. Even "/usr/sbin/alsactl --init" does not remove those added controls. The only way I find is to prevent /usr/sbin/alsactlto happen during reboot to really reset the ALSA driver to an initial state, which is really not pratical when developing/debugging. Question: is there a simple method to return to a clean and fresh state after snd_ctl_add_xxxx without rebooting the machine ? Thank you Fulup

2 2

[alsa-devel] [PATCH v1] ALSA: fm801: Initialize chip after IRQ handler is registered
by Andy Shevchenko 18 Jul '17

18 Jul '17

The commit b56fa687e02b ("ALSA: fm801: detect FM-only card earlier") rearranged initialization calls, i.e. it makes snd_fm801_chip_init() to be called before we register interrupt handler and set PCI bus mastering. Somehow it prevents FM801-AU to work properly. Thus, partially revert initialization order changed by commit mentioned above. Fixes: b56fa687e02b ("ALSA: fm801: detect FM-only card earlier") Reported-by: Émeric MASCHINO <emeric.maschino(a)gmail.com> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/fm801.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 2e402ece4c86..8e6b04b39dcc 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1235,8 +1235,6 @@ static int snd_fm801_create(struct snd_card *card, } } - snd_fm801_chip_init(chip); - if ((chip->tea575x_tuner & TUNER_ONLY) == 0) { if (devm_request_irq(&pci->dev, pci->irq, snd_fm801_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { @@ -1248,6 +1246,8 @@ static int snd_fm801_create(struct snd_card *card, pci_set_master(pci); } + snd_fm801_chip_init(chip); + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); return err; -- 2.11.0

3 5

[alsa-devel] [PATCH v2] ASoC: rt5663: Modify the default value for ASRC function
by Oder Chiou 18 Jul '17

18 Jul '17

The patch modifies the default value for ASRC function. It could prevent the pop noise and recording no sound with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index b262b6f..d324817 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -60,6 +60,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8080 }, + { 0x0086, 0x0028 }, }; static const struct reg_default rt5663_v2_reg[] = { @@ -509,7 +510,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0082, 0x0000 }, { 0x0083, 0x0000 }, { 0x0084, 0x0000 }, - { 0x0086, 0x0008 }, + { 0x0086, 0x0028 }, { 0x0087, 0x0000 }, { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, -- 2.7.4

1 0

[alsa-devel] [PATCH v2 0/2] Add support for TI TAS6424 digital amplifier
by Michael Stecklein 17 Jul '17

17 Jul '17

Hello all, This two-patch series adds the dt bindings, driver, and other related changes for the TAS6424 digital amplifier. I am upstreaming this driver on behalf of Andreas. Regards, Michael Stecklein <m-stecklein(a)ti.com> Changes since v1: - rebased on 4.13-rc1 Original v1 patches: - https://patchwork.kernel.org/patch/9795235/ - https://patchwork.kernel.org/patch/9795237/ Andreas Dannenberg (1): ASoC: add support for TAS6424 digital amplifier Michael Stecklein (1): dt-bindings: sound: add bindings for TAS6424 .../devicetree/bindings/sound/ti,tas6424.txt | 31 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas6424.c | 755 +++++++++++++++++++++ sound/soc/codecs/tas6424.h | 127 ++++ 5 files changed, 923 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas6424.txt create mode 100644 sound/soc/codecs/tas6424.c create mode 100644 sound/soc/codecs/tas6424.h -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: remove unnecessary static in rsnd_ssiu_probe()
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

Remove unnecessary static on local variable ops. Such variable is initialized before being used, on every execution path throughout the function. The static has no benefit and, removing it reduces the code size. This issue was detected using Coccinelle and the following semantic patch: @bad exists@ position p; identifier x; type T; @@ static T x@p; ... x = <+...x...+> @@ identifier x; expression e; type T; position p != bad.p; @@ -static T x@p; ... when != x when strict ?x = e; In the following log you can see the difference in the code size. Also, there is a significant difference in the bss segment. This log is the output of the size command, before and after the code change: before: text data bss dec hex filename 3211 680 64 3955 f73 sound/soc/sh/rcar/ssiu.o after: text data bss dec hex filename 3207 592 0 3799 ed7 sound/soc/sh/rcar/ssiu.o Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/sh/rcar/ssiu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index bed2c9c..4d94875 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -250,7 +250,7 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssiu *ssiu; - static struct rsnd_mod_ops *ops; + struct rsnd_mod_ops *ops; int i, nr, ret; /* same number to SSI */ -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: samsung: Remove non-existing CONFIG_CPU_S3C2413
by Krzysztof Kozlowski 17 Jul '17

17 Jul '17

There is no CONFIG_CPU_S3C2413 so get rid of it. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/samsung/s3c-i2s-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 8f42deaa184b..ca522a95160b 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -27,7 +27,7 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ +#if defined(CONFIG_CPU_S3C2412) \ || defined(CONFIG_ARCH_S3C64XX) || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: fsi: constify dev_pm_ops structure" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: fsi: constify dev_pm_ops structure has been applied to the asoc tree at git://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 4b2983da22d30c53d778b2e5035dc347dc4b0037 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sun, 16 Jul 2017 21:12:21 -0500 Subject: [PATCH] ASoC: fsi: constify dev_pm_ops structure This dev_pm_ops structure is only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7c4bdd82bb95..005b21506556 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2109,7 +2109,7 @@ static int fsi_resume(struct device *dev) return 0; } -static struct dev_pm_ops fsi_pm_ops = { +static const struct dev_pm_ops fsi_pm_ops = { .suspend = fsi_suspend, .resume = fsi_resume, }; -- 2.13.2

1 0

[alsa-devel] Applied "ASoC: psc-ac97: constify dev_pm_ops structure" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: psc-ac97: constify dev_pm_ops structure has been applied to the asoc tree at git://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 da61a11823395c87db0471d8d54055fa5a961688 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sun, 16 Jul 2017 21:12:39 -0500 Subject: [PATCH] ASoC: psc-ac97: constify dev_pm_ops structure This dev_pm_ops structure is only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/au1x/psc-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index bb53c7059005..a2050ae5a3fe 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -474,7 +474,7 @@ static int au1xpsc_ac97_drvresume(struct device *dev) return 0; } -static struct dev_pm_ops au1xpscac97_pmops = { +static const struct dev_pm_ops au1xpscac97_pmops = { .suspend = au1xpsc_ac97_drvsuspend, .resume = au1xpsc_ac97_drvresume, }; -- 2.13.2

1 0

[alsa-devel] Applied "ASoC: au1x: psc-i2s: constify dev_pm_ops structure" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: au1x: psc-i2s: constify dev_pm_ops structure has been applied to the asoc tree at git://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 90982dcdda4f0163bc5e53f49301e155c2cd8547 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sun, 16 Jul 2017 21:12:49 -0500 Subject: [PATCH] ASoC: au1x: psc-i2s: constify dev_pm_ops structure This dev_pm_ops structure is only stored in the pm field of a device_driver structure. This field is declared const, so dev_pm_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/au1x/psc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 0bf9d62b91a0..e6eec081eaae 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -392,7 +392,7 @@ static int au1xpsc_i2s_drvresume(struct device *dev) return 0; } -static struct dev_pm_ops au1xpsci2s_pmops = { +static const struct dev_pm_ops au1xpsci2s_pmops = { .suspend = au1xpsc_i2s_drvsuspend, .resume = au1xpsc_i2s_drvresume, }; -- 2.13.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix type in debug message
by Colin King 17 Jul '17

17 Jul '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to typo in debug message, dst_instacne should be dst_instance Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-messages.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index fb2f1f603f3c..c5b4e0a9946c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -950,7 +950,7 @@ static void skl_dump_bind_info(struct skl_sst *ctx, struct skl_module_cfg { dev_dbg(ctx->dev, "%s: src module_id = %d src_instance=%d\n", __func__, src_module->id.module_id, src_module->id.pvt_id); - dev_dbg(ctx->dev, "%s: dst_module=%d dst_instacne=%d\n", __func__, + dev_dbg(ctx->dev, "%s: dst_module=%d dst_instance=%d\n", __func__, dst_module->id.module_id, dst_module->id.pvt_id); dev_dbg(ctx->dev, "src_module state = %d dst module state = %d\n", -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: constify snd_soc_dai_ops structures
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

These structures are only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 0ebea34a..b1f2dba 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -628,7 +628,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops skl_pcm_dai_ops = { +static const struct snd_soc_dai_ops skl_pcm_dai_ops = { .startup = skl_pcm_open, .shutdown = skl_pcm_close, .prepare = skl_pcm_prepare, @@ -637,15 +637,15 @@ static struct snd_soc_dai_ops skl_pcm_dai_ops = { .trigger = skl_pcm_trigger, }; -static struct snd_soc_dai_ops skl_dmic_dai_ops = { +static const struct snd_soc_dai_ops skl_dmic_dai_ops = { .hw_params = skl_be_hw_params, }; -static struct snd_soc_dai_ops skl_be_ssp_dai_ops = { +static const struct snd_soc_dai_ops skl_be_ssp_dai_ops = { .hw_params = skl_be_hw_params, }; -static struct snd_soc_dai_ops skl_link_dai_ops = { +static const struct snd_soc_dai_ops skl_link_dai_ops = { .prepare = skl_link_pcm_prepare, .hw_params = skl_link_hw_params, .hw_free = skl_link_hw_free, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Atom: constify snd_soc_dai_ops structures
by Gustavo A. R. Silva 17 Jul '17

17 Jul '17

These structures are only stored in the ops field of a snd_soc_dai_driver structure. That field is declared const, so snd_soc_dai_ops structures that have this property can be declared as const also. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index b082b31..49c7b88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -471,7 +471,7 @@ static void sst_disable_ssp(struct snd_pcm_substream *substream, } } -static struct snd_soc_dai_ops sst_media_dai_ops = { +static const struct snd_soc_dai_ops sst_media_dai_ops = { .startup = sst_media_open, .shutdown = sst_media_close, .prepare = sst_media_prepare, @@ -480,11 +480,11 @@ static struct snd_soc_dai_ops sst_media_dai_ops = { .mute_stream = sst_media_digital_mute, }; -static struct snd_soc_dai_ops sst_compr_dai_ops = { +static const struct snd_soc_dai_ops sst_compr_dai_ops = { .mute_stream = sst_media_digital_mute, }; -static struct snd_soc_dai_ops sst_be_dai_ops = { +static const struct snd_soc_dai_ops sst_be_dai_ops = { .startup = sst_enable_ssp, .hw_params = sst_be_hw_params, .set_fmt = sst_set_format, -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: rt5665: force using PLL if MCLK is not suitable
by Bard Liao 17 Jul '17

17 Jul '17

Idealy PLL will be set in machine driver if MCLK doesn't meet the requirement of codec. To let the codec driver be more general, we can use a common PLL setting once sysclk/pll doesn't set properly in machine driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 370ed54d..d3103ef 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4098,9 +4098,12 @@ static int rt5665_hw_params(struct snd_pcm_substream *substream, rt5665->lrck[dai->id] = params_rate(params); pre_div = rl6231_get_clk_info(rt5665->sysclk, rt5665->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", - rt5665->lrck[dai->id], dai->id); - return -EINVAL; + dev_warn(codec->dev, "Force using PLL"); + snd_soc_codec_set_pll(codec, 0, RT5665_PLL1_S_MCLK, + rt5665->sysclk, rt5665->lrck[dai->id] * 512); + snd_soc_codec_set_sysclk(codec, RT5665_SCLK_S_PLL1, 0, + rt5665->lrck[dai->id] * 512, 0); + pre_div = 1; } frame_size = snd_soc_params_to_frame_size(params); if (frame_size < 0) { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: do not close shared backend dailink
by Patrick Lai 17 Jul '17

17 Jul '17

From: Banajit Goswami <bgoswami(a)codeaurora.org> Multiple frontend dailinks may be connected to a backend dailink at the same time. When one of frontend dailinks is closed, the associated backend dailink should not be closed if it is connected to other active frontend dailinks. Change ensures that backend dailink is closed only after all connected frontend dailinks are closed. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index efc5831..3d2c69f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -181,6 +181,10 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + if ((event == SND_SOC_DAPM_STREAM_STOP) && + (be->dpcm[dir].users >= 1)) + continue; + snd_soc_dapm_stream_event(be, dir, event); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project

2 1

[alsa-devel] Applied "ASoC: rsnd: add missing of_node_put" to the asoc tree
by Mark Brown 17 Jul '17

17 Jul '17

The patch ASoC: rsnd: add missing of_node_put has been applied to the asoc tree at git://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 53ba2aa314164f7fcd3acf9594313811256a52c4 Mon Sep 17 00:00:00 2001 From: Julia Lawall <Julia.Lawall(a)lip6.fr> Date: Sat, 15 Jul 2017 09:19:07 +0200 Subject: [PATCH] ASoC: rsnd: add missing of_node_put for_each_child_of_node performs an of_node_get on each iteration, so a jump out of the loop requires an of_node_put. The semantic patch that fixes this problem is as follows (http://coccinelle.lip6.fr): // <smpl> @@ local idexpression n; expression e,e1; identifier l; @@ for_each_child_of_node(e1,n) { ... ( of_node_put(n); | e = n | + of_node_put(n); ? goto l; ) ... } ... l: ... when != n // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ctu.c | 5 ++++- sound/soc/sh/rcar/dvc.c | 5 ++++- sound/soc/sh/rcar/mix.c | 5 ++++- sound/soc/sh/rcar/src.c | 6 +++++- sound/soc/sh/rcar/ssi.c | 6 +++++- 5 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 4ba8f2fe7a4c..e7f53f44165d 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -394,13 +394,16 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_ctu_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(ctu), &rsnd_ctu_ops, clk, rsnd_mod_get_status, RSND_MOD_CTU, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_ctu_probe_done; + } i++; } diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 99d2d9459e75..1743ade3cc55 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -380,13 +380,16 @@ int rsnd_dvc_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_dvc_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(dvc), &rsnd_dvc_ops, clk, rsnd_mod_get_status, RSND_MOD_DVC, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_dvc_probe_done; + } i++; } diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 195fc7bb22af..6c4826c189a4 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -168,13 +168,16 @@ int rsnd_mix_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_mix_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(mix), &rsnd_mix_ops, clk, rsnd_mod_get_status, RSND_MOD_MIX, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_mix_probe_done; + } i++; } diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 7aa239e28491..8e9a1de0ec25 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -581,20 +581,24 @@ int rsnd_src_probe(struct rsnd_priv *priv) src->irq = irq_of_parse_and_map(np, 0); if (!src->irq) { ret = -EINVAL; + of_node_put(np); goto rsnd_src_probe_done; } clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_src_probe_done; } ret = rsnd_mod_init(priv, rsnd_mod_get(src), &rsnd_src_ops, clk, rsnd_mod_get_status, RSND_MOD_SRC, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_src_probe_done; + } skip: i++; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 46feddd78ee2..49cdc5e007b3 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1079,6 +1079,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) clk = devm_clk_get(dev, name); if (IS_ERR(clk)) { ret = PTR_ERR(clk); + of_node_put(np); goto rsnd_ssi_probe_done; } @@ -1091,6 +1092,7 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) ssi->irq = irq_of_parse_and_map(np, 0); if (!ssi->irq) { ret = -EINVAL; + of_node_put(np); goto rsnd_ssi_probe_done; } @@ -1101,8 +1103,10 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) ret = rsnd_mod_init(priv, rsnd_mod_get(ssi), ops, clk, rsnd_ssi_get_status, RSND_MOD_SSI, i); - if (ret) + if (ret) { + of_node_put(np); goto rsnd_ssi_probe_done; + } i++; } -- 2.13.2

1 0

[alsa-devel] [PATCH 1/8] ASoC: sun4i-codec: Remove unneeded gpiod NULL check
by Fabio Estevam 17 Jul '17

17 Jul '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> The gpiod API checks for NULL descriptors, so there is no need to duplicate the check in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/sunxi/sun4i-codec.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 1500699..4c37231 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1171,9 +1171,8 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w, { struct sun4i_codec *scodec = snd_soc_card_get_drvdata(w->dapm->card); - if (scodec->gpio_pa) - gpiod_set_value_cansleep(scodec->gpio_pa, - !!SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(scodec->gpio_pa, + !!SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- 2.7.4

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