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

[alsa-devel] [PATCH] [v2] ASoC: wm97xx: fix uninitialized regmap pointer problem
by Arnd Bergmann 05 Nov '18

05 Nov '18

gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: fix a missing check that Charles pointed out --- include/sound/soc.h | 2 +- sound/soc/codecs/wm9705.c | 10 ++++------ sound/soc/codecs/wm9712.c | 10 ++++------ sound/soc/codecs/wm9713.c | 10 ++++------ 4 files changed, 13 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 70c10a8f3e90..3e0ac310a3df 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -553,12 +553,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -#ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_component(struct snd_ac97 *ac97); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e873baa9e778..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -642,8 +642,7 @@ static int wm9712_soc_probe(struct snd_soc_component *component) if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { int ret; wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, @@ -660,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -673,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..5a2fdf4f69bf 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9713->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { -- 2.18.0

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[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix uninitialized variable access
by Arnd Bergmann 05 Nov '18

05 Nov '18

Randconfig testing revealed a very old bug, with gcc-8: sound/soc/intel/atom/sst/sst_loader.c: In function 'sst_load_fw': sound/soc/intel/atom/sst/sst_loader.c:357:5: error: 'fw' may be used uninitialized in this function [-Werror=maybe-uninitialized] if (fw == NULL) { ^ sound/soc/intel/atom/sst/sst_loader.c:354:25: note: 'fw' was declared here const struct firmware *fw; We must check the return code of request_firmware() before we look at the pointer result that may be uninitialized when the function fails. Fixes: 9012c9544eea ("ASoC: Intel: mrfld - Add DSP load and management") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/atom/sst/sst_loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 27413ebae956..b8c456753f01 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); -- 2.18.0

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[alsa-devel] Applied "ASoC: nau8825: remove unnecessary unlikely()" to the asoc tree
by Mark Brown 05 Nov '18

05 Nov '18

The patch ASoC: nau8825: remove unnecessary unlikely() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0b6277e6343e192aaa7d452ab933281eb0d420dc Mon Sep 17 00:00:00 2001 From: Yangtao Li <tiny.windzz(a)gmail.com> Date: Sun, 4 Nov 2018 07:55:29 -0500 Subject: [PATCH] ASoC: nau8825: remove unnecessary unlikely() WARN_ON() already contains an unlikely(), so it's not necessary to use unlikely. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8825.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); -- 2.19.0.rc2

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[alsa-devel] [PATCH] ASoC: AMD: constify regulator_desc structure
by Julia Lawall 05 Nov '18

05 Nov '18

The regulator_desc structure can be const as it is only passed as the second argument of devm_regulator_register and the corresponding parameter is declared as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3f813ea5210a..a5daad973ce5 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -403,7 +403,7 @@ static struct regulator_config acp_da7219_cfg = { static struct regulator_ops acp_da7219_ops = { }; -static struct regulator_desc acp_da7219_desc = { +static const struct regulator_desc acp_da7219_desc = { .name = "reg-fixed-1.8V", .type = REGULATOR_VOLTAGE, .owner = THIS_MODULE,

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: On OMAP1 and OMAP2420 only 16bit format is supported
by Peter Ujfalusi 05 Nov '18

05 Nov '18

For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to receive or transmit the most significant part of the 32bit sample. We can not trick the system by using the DRR2/DXR2 register offset and use 32bit element size since these (and other McBSP) registers are 16bit ones. We would need support for port_window in cyclic mode from the sDMA driver, but it is not straight forward as we would need to have the sDMA frame to cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA period to be able to receive periodic interrupts. Since the 32bit samples are not working (DMA timeout), just remove it in case we have McBSP with reg_size of 2. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have noticed that on Nokia n810 32bit samples just don't work at all for audio Regards, Peter sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d0ebb6b9bfac..caa24c463b3e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -862,6 +862,11 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; + if (mcbsp->pdata->reg_size == 2) { + omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: pcm3168a: cleanup hw_constraint for LEFT_J/I2S
by Kuninori Morimoto 05 Nov '18

05 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> hw constraint for LEFT_J/I2S are same, but had duplicated code. This code cleanup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e1c3a18..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -497,6 +497,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, unsigned int fmt; unsigned int sample_min; unsigned int channel_max; + unsigned int channel_maxs[] = { + 6, /* rx */ + 8 /* tx */ + }; if (tx) fmt = pcm3168a->dac_fmt; @@ -521,18 +525,9 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, channel_max = 2; break; case PCM3168A_FMT_LEFT_J: - sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; - break; case PCM3168A_FMT_I2S: sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; + channel_max = channel_maxs[tx]; break; default: sample_min = 24; -- 2.7.4

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[alsa-devel] [PATCH 0/3] ASoC: rsnd: prepare patches for BUSIF support
by Kuninori Morimoto 05 Nov '18

05 Nov '18

Hi Mark These are prepare/cleanup for for Renesas rsnd driver BUSIF support. I will post BUSIF support patch soon, but it needs Jiada Wang <jiada_wang(a)mentor.com> 's check. If it was OK, I will focus to TDM split mode. Kuninori Morimoto (3): ASoC: rsnd: move .get_status under rsnd_mod_ops ASoC: rsnd: add .get_id/.get_id_sub ASoC: rsnd: rsnd_mod_name() handles both name and ID sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/cmd.c | 11 ++-- sound/soc/sh/rcar/core.c | 72 +++++++++++++++++------ sound/soc/sh/rcar/ctu.c | 23 +++++++- sound/soc/sh/rcar/dma.c | 47 +++++++-------- sound/soc/sh/rcar/dvc.c | 3 +- sound/soc/sh/rcar/gen.c | 12 ++-- sound/soc/sh/rcar/mix.c | 3 +- sound/soc/sh/rcar/rsnd.h | 21 +++---- sound/soc/sh/rcar/src.c | 29 +++++----- sound/soc/sh/rcar/ssi.c | 147 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 37 ++++++------ 12 files changed, 232 insertions(+), 175 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0
by Hans de Goede 05 Nov '18

05 Nov '18

Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the mclk instead of pmc_plt_clk_3. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Swanky that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++++++++++++++++-- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..9d9f6e41d81c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.19.0

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[alsa-devel] [PATCH] ASoC: qdsp6: q6afe: Fix wrong MI2S SD line mask
by Rohit kumar 05 Nov '18

05 Nov '18

SD line mask for MI2S starts from BIT 0 instead of BIT 1. Fix all bit mask for MI2S SD lines. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/qdsp6/q6afe.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 000775b..829b5e987 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -49,14 +49,14 @@ #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 -#define AFE_PORT_I2S_SD0_MASK BIT(0x1) -#define AFE_PORT_I2S_SD1_MASK BIT(0x2) -#define AFE_PORT_I2S_SD2_MASK BIT(0x3) -#define AFE_PORT_I2S_SD3_MASK BIT(0x4) -#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1) -#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3) -#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1) -#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1) +#define AFE_PORT_I2S_SD0_MASK BIT(0x0) +#define AFE_PORT_I2S_SD1_MASK BIT(0x1) +#define AFE_PORT_I2S_SD2_MASK BIT(0x2) +#define AFE_PORT_I2S_SD3_MASK BIT(0x3) +#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0) +#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2) +#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0) +#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0) #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH] ALSA: usb-audio: Define registers for CM6206
by Linus Walleij 05 Nov '18

05 Nov '18

The register map for CM6206 is known and we can define what the values written to the different registers actually mean. I tested to print the hex values before/after this change, there is no functional change. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/usb/quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 124 insertions(+), 5 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8a945ece9869..3d0f09108c98 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -19,6 +19,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/midi.h> +#include <linux/bits.h> #include <sound/control.h> #include <sound/core.h> @@ -668,15 +669,133 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) } /* - * C-Media CM6206 is based on CM106 with two additional - * registers that are not documented in the data sheet. - * Values here are chosen based on sniffing USB traffic - * under Windows. + * CM6206 registers from the CM6206 datasheet rev 2.1 */ +#define CM6206_REG0_DMA_MASTER BIT(15) +#define CM6206_REG0_SPDIFO_RATE_48K (2 << 12) +#define CM6206_REG0_SPDIFO_RATE_96K (7 << 12) +/* Bit 4 thru 11 is the S/PDIF category code */ +#define CM6206_REG0_SPDIFO_CAT_CODE_GENERAL (0 << 4) +#define CM6206_REG0_SPDIFO_EMPHASIS_CD BIT(3) +#define CM6206_REG0_SPDIFO_COPYRIGHT_NA BIT(2) +#define CM6206_REG0_SPDIFO_NON_AUDIO BIT(1) +#define CM6206_REG0_SPDIFO_PRO_FORMAT BIT(0) + +#define CM6206_REG1_TEST_SEL_CLK BIT(14) +#define CM6206_REG1_PLLBIN_EN BIT(13) +#define CM6206_REG1_SOFT_MUTE_EN BIT(12) +#define CM6206_REG1_GPIO4_OUT BIT(11) +#define CM6206_REG1_GPIO4_OE BIT(10) +#define CM6206_REG1_GPIO3_OUT BIT(9) +#define CM6206_REG1_GPIO3_OE BIT(8) +#define CM6206_REG1_GPIO2_OUT BIT(7) +#define CM6206_REG1_GPIO2_OE BIT(6) +#define CM6206_REG1_GPIO1_OUT BIT(5) +#define CM6206_REG1_GPIO1_OE BIT(4) +#define CM6206_REG1_SPDIFO_INVALID BIT(3) +#define CM6206_REG1_SPDIF_LOOP_EN BIT(2) +#define CM6206_REG1_SPDIFO_DIS BIT(1) +#define CM6206_REG1_SPDIFI_MIX BIT(0) + +#define CM6206_REG2_DRIVER_ON BIT(15) +#define CM6206_REG2_HEADP_SEL_SIDE_CHANNELS (0 << 13) +#define CM6206_REG2_HEADP_SEL_SURROUND_CHANNELS (1 << 13) +#define CM6206_REG2_HEADP_SEL_CENTER_SUBW (2 << 13) +#define CM6206_REG2_HEADP_SEL_FRONT_CHANNELS (3 << 13) +#define CM6206_REG2_MUTE_HEADPHONE_RIGHT BIT(12) +#define CM6206_REG2_MUTE_HEADPHONE_LEFT BIT(11) +#define CM6206_REG2_MUTE_REAR_SURROUND_RIGHT BIT(10) +#define CM6206_REG2_MUTE_REAR_SURROUND_LEFT BIT(9) +#define CM6206_REG2_MUTE_SIDE_SURROUND_RIGHT BIT(8) +#define CM6206_REG2_MUTE_SIDE_SURROUND_LEFT BIT(7) +#define CM6206_REG2_MUTE_SUBWOOFER BIT(6) +#define CM6206_REG2_MUTE_CENTER BIT(5) +#define CM6206_REG2_MUTE_RIGHT_FRONT BIT(3) +#define CM6206_REG2_MUTE_LEFT_FRONT BIT(3) +#define CM6206_REG2_EN_BTL BIT(2) +#define CM6206_REG2_MCUCLKSEL_1_5_MHZ (0) +#define CM6206_REG2_MCUCLKSEL_3_MHZ (1) +#define CM6206_REG2_MCUCLKSEL_6_MHZ (2) +#define CM6206_REG2_MCUCLKSEL_12_MHZ (3) + +/* Bit 11..13 sets the sensitivity to FLY tuner volume control VP/VD signal */ +#define CM6206_REG3_FLYSPEED_DEFAULT (2 << 11) +#define CM6206_REG3_VRAP25EN BIT(10) +#define CM6206_REG3_MSEL1 BIT(9) +#define CM6206_REG3_SPDIFI_RATE_44_1K BIT(0 << 7) +#define CM6206_REG3_SPDIFI_RATE_48K BIT(2 << 7) +#define CM6206_REG3_SPDIFI_RATE_32K BIT(3 << 7) +#define CM6206_REG3_PINSEL BIT(6) +#define CM6206_REG3_FOE BIT(5) +#define CM6206_REG3_ROE BIT(4) +#define CM6206_REG3_CBOE BIT(3) +#define CM6206_REG3_LOSE BIT(2) +#define CM6206_REG3_HPOE BIT(1) +#define CM6206_REG3_SPDIFI_CANREC BIT(0) + +#define CM6206_REG5_DA_RSTN BIT(13) +#define CM6206_REG5_AD_RSTN BIT(12) +#define CM6206_REG5_SPDIFO_AD2SPDO BIT(12) +#define CM6206_REG5_SPDIFO_SEL_FRONT (0 << 9) +#define CM6206_REG5_SPDIFO_SEL_SIDE_SUR (1 << 9) +#define CM6206_REG5_SPDIFO_SEL_CEN_LFE (2 << 9) +#define CM6206_REG5_SPDIFO_SEL_REAR_SUR (3 << 9) +#define CM6206_REG5_CODECM BIT(8) +#define CM6206_REG5_EN_HPF BIT(7) +#define CM6206_REG5_T_SEL_DSDA4 BIT(6) +#define CM6206_REG5_T_SEL_DSDA3 BIT(5) +#define CM6206_REG5_T_SEL_DSDA2 BIT(4) +#define CM6206_REG5_T_SEL_DSDA1 BIT(3) +#define CM6206_REG5_T_SEL_DSDAD_NORMAL 0 +#define CM6206_REG5_T_SEL_DSDAD_FRONT 4 +#define CM6206_REG5_T_SEL_DSDAD_S_SURROUND 5 +#define CM6206_REG5_T_SEL_DSDAD_CEN_LFE 6 +#define CM6206_REG5_T_SEL_DSDAD_R_SURROUND 7 + static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { int err = 0, reg; - int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + int val[] = { + /* + * Values here are chosen based on sniffing USB traffic + * under Windows. + * + * REG0: DAC is master, sample rate 48kHz, no copyright + */ + CM6206_REG0_SPDIFO_RATE_48K | + CM6206_REG0_SPDIFO_COPYRIGHT_NA, + /* + * REG1: PLL binary search enable, soft mute enable. + */ + CM6206_REG1_PLLBIN_EN | + CM6206_REG1_SOFT_MUTE_EN | + /* + * REG2: enable output drivers, + * select front channels to the headphone output, + * then mute the headphone channels, run the MCU + * at 1.5 MHz. + */ + CM6206_REG2_DRIVER_ON | + CM6206_REG2_HEADP_SEL_FRONT_CHANNELS | + CM6206_REG2_MUTE_HEADPHONE_RIGHT | + CM6206_REG2_MUTE_HEADPHONE_LEFT, + /* + * REG3: default flyspeed, set 2.5V mic bias + * enable all line out ports and enable SPDIF + */ + CM6206_REG3_FLYSPEED_DEFAULT | + CM6206_REG3_VRAP25EN | + CM6206_REG3_FOE | + CM6206_REG3_ROE | + CM6206_REG3_CBOE | + CM6206_REG3_LOSE | + CM6206_REG3_HPOE | + CM6206_REG3_SPDIFI_CANREC, + /* REG4 is just a bunch of GPIO lines */ + 0x0000, + /* REG5: de-assert AD/DA reset signals */ + CM6206_REG5_DA_RSTN | + CM6206_REG5_AD_RSTN }; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); -- 2.19.1

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