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[alsa-devel] [PATCH 1148/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/core/misc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/misc.c b/sound/core/misc.c index f2e8226..4683c43 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -36,7 +36,7 @@ #endif static int debug = DEFAULT_DEBUG_LEVEL; -module_param(debug, int, 0644); +module_param(debug, int, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(debug, "Debug level (0 = disable)"); #endif /* CONFIG_SND_DEBUG */ -- 2.9.2

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[alsa-devel] [PATCH 1147/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/core/init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6bda843..993b21c 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -55,7 +55,7 @@ EXPORT_SYMBOL(snd_cards); static DEFINE_MUTEX(snd_card_mutex); static char *slots[SNDRV_CARDS]; -module_param_array(slots, charp, NULL, 0444); +module_param_array(slots, charp, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(slots, "Module names assigned to the slots."); /* return non-zero if the given index is reserved for the given -- 2.9.2

1 0

[alsa-devel] [PATCH 1146/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/aoa/soundbus/i2sbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 1cbf210..2b741a4 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -27,7 +27,7 @@ MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); static int force; -module_param(force, int, 0444); +module_param(force, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(force, "Force loading i2sbus even when" " no layout-id property is present"); -- 2.9.2

1 0

[alsa-devel] [PATCH 1145/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 02 Aug '16

02 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/aoa/core/alsa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/core/alsa.c b/sound/aoa/core/alsa.c index 4a7e4e6..6af6fb6 100644 --- a/sound/aoa/core/alsa.c +++ b/sound/aoa/core/alsa.c @@ -9,7 +9,7 @@ #include "alsa.h" static int index = -1; -module_param(index, int, 0444); +module_param(index, int, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(index, "index for AOA sound card."); static struct aoa_card *aoa_card; -- 2.9.2

1 0

[alsa-devel] [asoc:topic/intel 19/26] sound/soc/intel/skylake/skl-topology.c:480:24-30: ERROR: application of sizeof to pointer
by kbuild test robot 02 Aug '16

02 Aug '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: b1f89b197b3f7360239fc2120755252cead4e572 commit: 7c617ec3e8f6b614e63ea9c82ce2ea39e7daf579 [19/26] ASoC: Intel: Skylake: Move modules query to runtime coccinelle warnings: (new ones prefixed by >>) >> sound/soc/intel/skylake/skl-topology.c:480:24-30: ERROR: application of sizeof to pointer Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: cs53l30: Do not ignore errors if mclk is specified
by Nicolin Chen 01 Aug '16

01 Aug '16

When the clock is specified, there could be other errors besides the EPROBE_DEFER so don't ignore them. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 2c0d9c4..227c556 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -999,8 +999,8 @@ static int cs53l30_i2c_probe(struct i2c_client *client, /* Check if MCLK provided */ cs53l30->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs53l30->mclk)) { - if (PTR_ERR(cs53l30->mclk) == -EPROBE_DEFER) { - ret = -EPROBE_DEFER; + if (PTR_ERR(cs53l30->mclk) != -ENOENT) { + ret = PTR_ERR(cs53l30->mclk); goto error; } /* Otherwise mark the mclk pointer to NULL */ -- 2.1.4

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[alsa-devel] [PATCH] ASoC: da7219: Make more efficient use of MCLK within driver
by Adam Thomson 01 Aug '16

01 Aug '16

Currently, if the driver has control of MCLK then it remains enabled as long as the codec is in STANDBY or above. The MCLK is only really required in STANDBY when a 3-pole jack is inserted and the HP detect procedure is required to run. This patch updates the code to enable/disable the MCLK when moving between the STANDBY and PREPARE bias level, and when a 3-pole jack is inserted and HP detection is required, thus saving power at all other times. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219-aad.c | 18 +++++++++++++++++- sound/soc/codecs/da7219.c | 18 +++++++++++------- 2 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index f0057cd..4e369a1 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/property.h> #include <linux/pm_wakeirq.h> @@ -115,12 +116,23 @@ static void da7219_aad_hptest_work(struct work_struct *work) u16 tonegen_freq_hptest; u8 accdet_cfg8; - int report = 0; + int report = 0, ret = 0; /* Lock DAPM and any Kcontrols that are affected by this test */ snd_soc_dapm_mutex_lock(dapm); mutex_lock(&da7219->lock); + /* Ensure MCLK is available for HP test procedure */ + if (da7219->mclk) { + ret = clk_prepare_enable(da7219->mclk); + if (ret) { + dev_err(codec->dev, "Failed to enable mclk - %d\n", ret); + mutex_unlock(&da7219->lock); + snd_soc_dapm_mutex_unlock(dapm); + return; + } + } + /* Bypass cache so it saves current settings */ regcache_cache_bypass(da7219->regmap, true); @@ -250,6 +262,10 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, DA7219_HP_R_AMP_OE_MASK); + /* Remove MCLK, if previously enabled */ + if (da7219->mclk) + clk_disable_unprepare(da7219->mclk); + mutex_unlock(&da7219->lock); snd_soc_dapm_mutex_unlock(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 50ea943..737e914 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1508,11 +1508,10 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: break; - case SND_SOC_BIAS_STANDBY: - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { - /* MCLK */ + case SND_SOC_BIAS_PREPARE: + /* Enable MCLK for transition to ON state */ + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { if (da7219->mclk) { ret = clk_prepare_enable(da7219->mclk); if (ret) { @@ -1521,11 +1520,19 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, return ret; } } + } + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Master bias */ snd_soc_update_bits(codec, DA7219_REFERENCES, DA7219_BIAS_EN_MASK, DA7219_BIAS_EN_MASK); + } else { + /* Remove MCLK */ + if (da7219->mclk) + clk_disable_unprepare(da7219->mclk); } break; case SND_SOC_BIAS_OFF: @@ -1534,9 +1541,6 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, DA7219_REFERENCES, DA7219_BIAS_EN_MASK, 0); - /* MCLK */ - if (da7219->mclk) - clk_disable_unprepare(da7219->mclk); break; } -- 1.9.3

2 1

[alsa-devel] [PATCH v2] ASoC: rt5659: Add mclk controls
by Nicolin Chen 01 Aug '16

01 Aug '16

The codec driver should control the mclk. So this patch adds this support. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changlog: v1->v2 * Only set the mclk pointer to NULL when getting an ENOENT error code (i.e. no clock is specified) -- corrected by Lars-Peter Documentation/devicetree/bindings/sound/rt5659.txt | 3 +++ sound/soc/codecs/rt5659.c | 24 ++++++++++++++++++++++ sound/soc/codecs/rt5659.h | 1 + 3 files changed, 28 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 5f79e7f..1766e05 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -12,6 +12,9 @@ Required properties: Optional properties: +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + - realtek,in1-differential - realtek,in3-differential - realtek,in4-differential diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 1b30914..42c183f 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -3565,7 +3566,9 @@ static int rt5659_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) static int rt5659_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_PREPARE: @@ -3582,6 +3585,17 @@ static int rt5659_set_bias_level(struct snd_soc_codec *codec, RT5659_PWR_FV1 | RT5659_PWR_FV2); break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + ret = clk_prepare_enable(rt5659->mclk); + if (ret) { + dev_err(codec->dev, + "failed to enable MCLK: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: regmap_update_bits(rt5659->regmap, RT5659_PWR_DIG_1, RT5659_PWR_LDO, 0); @@ -3591,6 +3605,7 @@ static int rt5659_set_bias_level(struct snd_soc_codec *codec, RT5659_PWR_MB | RT5659_PWR_VREF2); regmap_update_bits(rt5659->regmap, RT5659_DIG_MISC, RT5659_DIG_GATE_CTRL, 0); + clk_disable_unprepare(rt5659->mclk); break; default: @@ -4020,6 +4035,15 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, regmap_write(rt5659->regmap, RT5659_RESET, 0); + /* Check if MCLK provided */ + rt5659->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(rt5659->mclk)) { + if (PTR_ERR(rt5659->mclk) != -ENOENT) + return PTR_ERR(rt5659->mclk); + /* Otherwise mark the mclk pointer to NULL */ + rt5659->mclk = NULL; + } + rt5659_calibrate(rt5659); /* line in diff mode*/ diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index d31c9e5..d69b0eb 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -1796,6 +1796,7 @@ struct rt5659_priv { struct gpio_desc *gpiod_reset; struct snd_soc_jack *hs_jack; struct delayed_work jack_detect_work; + struct clk *mclk; int sysclk; int sysclk_src; -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add clocking option
by Stefan Müller-Klieser 01 Aug '16

01 Aug '16

The AHCLKX clock output of the McASP can be used as audio MCLK, even if the McASP runs in slave mode. Add a DT binding to make this option available in a simple-card setup. Signed-off-by: Stefan Müller-Klieser <s.mueller-klieser(a)phytec.de> --- Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt | 4 ++++ include/linux/platform_data/davinci_asp.h | 5 +++++ sound/soc/davinci/davinci-mcasp.c | 8 ++++++++ 3 files changed, 17 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 46bc982..96d6244 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -39,6 +39,10 @@ Optional properties: please refer to pinctrl-bindings.txt - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +- ahclkx-en: In case the MCASP runs in slave mode, e.g. i2s SCK and WS slave, + we can still provide the MCLK to a codec PLL using the AHCLKX + transmit clock output. This setup can be activated by using this + property. Clock master mode remains unaffected. Example: diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 85ad68f..c878836 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -44,6 +44,11 @@ struct davinci_mcasp_pdata { int clk_input_pin; /* + * Activate AHCLKX clock output in slave mode. + */ + bool ahclkx_en; + + /* * This flag works when both clock and FS are outputs for the cpu * and makes clock more accurate (FS is not symmetrical and the * clock is very fast. diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 237dc67..aba888e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -94,6 +94,7 @@ struct davinci_mcasp { int sysclk_freq; bool bclk_master; + bool ahclkx_en; /* McASP FIFO related */ u8 txnumevt; @@ -605,6 +606,10 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); + } else if (mcasp->ahclkx_en) { + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); @@ -1610,6 +1615,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (ret >= 0) pdata->op_mode = val; + pdata->ahclkx_en = of_property_read_bool(np, "ahclkx-en"); + ret = of_property_read_u32(np, "tdm-slots", &val); if (ret >= 0) { if (val < 2 || val > 32) { @@ -1851,6 +1858,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; + mcasp->ahclkx_en = pdata->ahclkx_en; mcasp->dev = &pdev->dev; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - On-board speaker fixup on ACER Veriton
by Shrirang Bagul 01 Aug '16

01 Aug '16

On Acer Veriton machines, codec with subsystem-id 0x1b0a01b8 the port at 0x15 is configured by default as an Internal Speaker (0x90170120). However, no physical is speaker installed on-board. This patch adds a quirk which disables the physical connection on this pin. BugLink: https://bugs.launchpad.net/bugs/1607647 Signed-off-by: Shrirang Bagul <shrirang.bagul(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ddd29b9..498f199 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6549,6 +6549,7 @@ enum { ALC668_FIXUP_ASUS_Nx51, ALC891_FIXUP_HEADSET_MODE, ALC891_FIXUP_DELL_MIC_NO_PRESENCE, + ALC662_FIXUP_ACER_VERITON, }; static const struct hda_fixup alc662_fixups[] = { @@ -6818,6 +6819,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC891_FIXUP_HEADSET_MODE }, + [ALC662_FIXUP_ACER_VERITON] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x15, 0x50170120 }, /* no internal speaker */ + { } + } + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6857,6 +6865,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), + SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), #if 0 /* Below is a quirk table taken from the old code. -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda/realtek - Can't adjust speaker's volume on a Dell AIO
by Hui Wang 01 Aug '16

01 Aug '16

We have a Dell AIO on which we can't adjust its speaker's volume. The problem is it is connected to a Audio Output node without Amp-out capability. To fix it, we change it to be connnected to a node with Amp-out capability. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ddd29b9..7355054 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4689,6 +4689,22 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, } } +static void alc298_fixup_speaker_volume(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* The speaker is routed to the Node 0x06 by a mistake, as a result + we can't adjust the speaker's volume since this node does not has + Amp-out capability. we change the speaker's route to: + Node 0x02 (Audio Output) -> Node 0x0c (Audio Mixer) -> Node 0x17 ( + Pin Complex), since Node 0x02 has Amp-out caps, we can adjust + speaker's volume now. */ + + hda_nid_t conn1[1] = { 0x0c }; + snd_hda_override_conn_list(codec, 0x17, 1, conn1); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -4838,6 +4854,7 @@ enum { ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, + ALC298_FIXUP_SPK_VOLUME, }; static const struct hda_fixup alc269_fixups[] = { @@ -5493,6 +5510,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE, }, + [ALC298_FIXUP_SPK_VOLUME] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_speaker_volume, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5539,6 +5560,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.9.1

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[alsa-devel] [PATCH v2 0/3] ASoC: topology: Add support for BE DAIs
by mengdong.lin＠linux.intel.com 29 Jul '16

29 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is topology ABI udpate in this series. Topology core can check size of each ABI objects to detect version mismatch between user space and kernel. This series adds support for BE (Back End) DAIs: - Define the type and ABI struct for Backend DAIs. - The topology core can configure an existing BE DAI or create new BE DAIs. The user space series "topology: Add support for BE DAIs" is also submitted. There will be 2 pairs of kernel & user space series for PCM & BE links later. History: v2: Remove unused argument to fix the compiler warning. sound/soc/soc-topology.c: In function ‘soc_tplg_be_dai_config’: sound/soc/soc-topology.c:1789:22: warning: too many arguments for format [-Wformat-extra-args] dev_err(tplg->dev, "ASoC: Invalid BE DAI name\n", Mengdong Lin (3): ASoC: topology: ABI - Add the types for BE DAI ASoC: topology: Add support for configuring existing BE DAIs ASoC: topology: Able to create BE DAIs include/sound/soc-topology.h | 1 + include/uapi/sound/asoc.h | 34 +++++++++- sound/soc/soc-core.c | 3 +- sound/soc/soc-topology.c | 155 ++++++++++++++++++++++++++++++++++++++++++- 4 files changed, 188 insertions(+), 5 deletions(-) -- 2.5.0

2 4

[alsa-devel] Neverending HDA pin quirks, how does Windows cope?
by Daniel Drake 28 Jul '16

28 Jul '16

Hi, We're constantly having to support new laptop models from vendors and the majority of them always seem to need some kind of HDA quirk adding, especially for pin usage info where the BIOS data is clearly incorrect. From the vast collection of quirks present in Linux's HDA drivers I see we are not alone. We're trying to figure out if there is a way the BIOS can be fixed for the cases that come from our partners, but I am also curious as to how this is managed on Windows and if we can learn anything from that. I had a look at the windows driver for some affected hardware. I was unable to identify any hardware-specific quirks in the .inf files, but using "strings" I found that various binary .sys, .dll and .dat files refer to hundreds of different laptop models. That might suggest that for each new laptop model, they make a modification to the windows driver, recompile, and get the new version signed again by MS? Sounds insane, anyone know for sure how this is handled? Thanks Daniel

1 0

[alsa-devel] [PATCH v4] topology: Return -EINVAL at once on failure to find a reference
by mengdong.lin＠linux.intel.com 28 Jul '16

28 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> In building phase, return error -EINVAL at once if an element's references (e.g. TLV, text, control or private data) cannot be found. Move this error check right after looking up a reference, to make the code more clear. And checking the return value of tplg_copy_data() is enough, no need to check ref->elem for data references. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 54f8e44..d910230 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -135,24 +135,23 @@ static int tplg_build_mixer_control(snd_tplg_t *tplg, if (ref->type == SND_TPLG_TYPE_TLV) { ref->elem = tplg_elem_lookup(&tplg->tlv_list, ref->id, SND_TPLG_TYPE_TLV); - if (ref->elem) - err = copy_tlv(elem, ref->elem); + if (!ref->elem) + goto ref_missing; + copy_tlv(elem, ref->elem); } else if (ref->type == SND_TPLG_TYPE_DATA) { err = tplg_copy_data(tplg, elem, ref); if (err < 0) return err; } - - if (!ref->elem) { - SNDERR("error: cannot find '%s' referenced by" - " control '%s'\n", ref->id, elem->id); - return -EINVAL; - } else if (err < 0) - return err; } return 0; + +ref_missing: + SNDERR("error: cannot find '%s' referenced by control '%s'\n", + ref->id, elem->id); + return -EINVAL; } static void copy_enum_texts(struct tplg_elem *enum_elem, @@ -183,23 +182,23 @@ static int tplg_build_enum_control(snd_tplg_t *tplg, if (ref->type == SND_TPLG_TYPE_TEXT) { ref->elem = tplg_elem_lookup(&tplg->text_list, ref->id, SND_TPLG_TYPE_TEXT); - if (ref->elem) - copy_enum_texts(elem, ref->elem); + if (!ref->elem) + goto ref_missing; + copy_enum_texts(elem, ref->elem); } else if (ref->type == SND_TPLG_TYPE_DATA) { err = tplg_copy_data(tplg, elem, ref); if (err < 0) return err; } - if (!ref->elem) { - SNDERR("error: cannot find '%s' referenced by" - " control '%s'\n", ref->id, elem->id); - return -EINVAL; - } else if (err < 0) - return err; } return 0; + +ref_missing: + SNDERR("error: cannot find '%s' referenced by control '%s'\n", + ref->id, elem->id); + return -EINVAL; } /* check referenced private data for a byte control */ diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 4d343b2..988a5fc 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -160,7 +160,10 @@ static int tplg_build_widget(snd_tplg_t *tplg, base = &elem->ref_list; - /* for each ref in this control elem */ + /* Look up and merge each reference in this widget elem. + * For widgets added by C API, its control elements are already set + * and so no need to look up. + */ list_for_each(pos, base) { ref = list_entry(pos, struct tplg_ref, list); @@ -170,48 +173,50 @@ static int tplg_build_widget(snd_tplg_t *tplg, if (!ref->elem) ref->elem = tplg_elem_lookup(&tplg->mixer_list, ref->id, SND_TPLG_TYPE_MIXER); - if (ref->elem) - err = copy_dapm_control(elem, ref->elem); + if (!ref->elem) + goto ref_missing; + + err = copy_dapm_control(elem, ref->elem); break; case SND_TPLG_TYPE_ENUM: if (!ref->elem) ref->elem = tplg_elem_lookup(&tplg->enum_list, ref->id, SND_TPLG_TYPE_ENUM); - if (ref->elem) - err = copy_dapm_control(elem, ref->elem); + if (!ref->elem) + goto ref_missing; + + err = copy_dapm_control(elem, ref->elem); break; case SND_TPLG_TYPE_BYTES: if (!ref->elem) ref->elem = tplg_elem_lookup(&tplg->bytes_ext_list, ref->id, SND_TPLG_TYPE_BYTES); - if (ref->elem) - err = copy_dapm_control(elem, ref->elem); + if (!ref->elem) + goto ref_missing; + + err = copy_dapm_control(elem, ref->elem); break; case SND_TPLG_TYPE_DATA: err = tplg_copy_data(tplg, elem, ref); - if (err < 0) - return err; break; default: break; } - if (!ref->elem) { - SNDERR("error: cannot find control '%s'" - " referenced by widget '%s'\n", - ref->id, elem->id); - return -EINVAL; - } - if (err < 0) return err; } return 0; + +ref_missing: + SNDERR("error: cannot find '%s' referenced by widget '%s'\n", + ref->id, elem->id); + return -EINVAL; } int tplg_build_widgets(snd_tplg_t *tplg) -- 2.5.0

3 3

[alsa-devel] [PATCH] topology: Fix inaccurate comments when building a widget element
by mengdong.lin＠linux.intel.com 28 Jul '16

28 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> The original comments use 'control' by mistake when building a widget. Also describe the difference between the referenced controls defined by C API and text conf file. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/topology/dapm.c b/src/topology/dapm.c index e3c90d8..0141653 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -150,7 +150,7 @@ static int copy_dapm_control(struct tplg_elem *elem, struct tplg_elem *ref) return 0; } -/* check referenced controls for a widget */ +/* Merge referenced controls and private data for a widget */ static int tplg_build_widget(snd_tplg_t *tplg, struct tplg_elem *elem) { @@ -160,7 +160,12 @@ static int tplg_build_widget(snd_tplg_t *tplg, base = &elem->ref_list; - /* for each ref in this control elem */ + /* Look up and merge each control or data reference of this + * widget element. For a widget defined by C API, its control + * references are already bound and can be directly merged + * without lookup. Only control references defined by text + * conf file need lookup before being merged. + */ list_for_each(pos, base) { ref = list_entry(pos, struct tplg_ref, list); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: Fixup SRCm_IFSVR calculate method
by Kuninori Morimoto 27 Jul '16

27 Jul '16

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> This patch fixes the calculation accuracy degradation of SRCm_IFSVR register value. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e39f916..969a516 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -226,8 +226,12 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, ifscr = 0; fsrate = 0; if (fin != fout) { + u64 n; + ifscr = 1; - fsrate = 0x0400000 / fout * fin; + n = (u64)0x0400000 * fin; + do_div(n, fout); + fsrate = n; } /* -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Add a dummy snd_pcm_runtime to avoid NULL pointer access
by Nicolin Chen 27 Jul '16

27 Jul '16

The SND_SOC_DAPM_PRE_PMU case would call startup()/hw_params() that might access substream->runtime through other functions. For example: Unable to handle kernel NULL pointer dereference at virtual address [....] PC is at snd_pcm_hw_rule_add+0x24/0x1b0 LR is at snd_pcm_hw_constraint_list+0x20/0x28 [....] Process arecord (pid: 424, stack limit = 0xffffffc1ecaf0020) Call trace: [<ffffffc00086be68>] snd_pcm_hw_rule_add+0x24/0x1b0 [<ffffffc00086c014>] snd_pcm_hw_constraint_list+0x20/0x28 [<ffffffc0008b47a4>] cs53l30_pcm_startup+0x24/0x30 [<ffffffc0008a6260>] snd_soc_dai_link_event+0x290/0x354 [<ffffffc0008a7528>] dapm_seq_check_event.isra.31+0x134/0x2c8 [<ffffffc0008a7768>] dapm_seq_run_coalesced+0x94/0x1c8 [<ffffffc0008a7940>] dapm_seq_run+0xa4/0x404 [<ffffffc0008a8bac>] dapm_power_widgets+0x524/0x984 [<ffffffc0008ab1c4>] snd_soc_dapm_stream_event+0x8c/0xa8 [<ffffffc0008ac7f4>] soc_pcm_prepare+0x10c/0x1ec [<ffffffc000865b9c>] snd_pcm_do_prepare+0x1c/0x38 [<ffffffc000865600>] snd_pcm_action_single+0x40/0x88 [<ffffffc0008656b8>] snd_pcm_action_nonatomic+0x70/0x90 [<ffffffc000868d28>] snd_pcm_common_ioctl1+0xb6c/0xdd8 [<ffffffc000869508>] snd_pcm_capture_ioctl1+0x200/0x334 [<ffffffc00086a084>] snd_pcm_ioctl_compat+0x648/0x95c [<ffffffc0001ff4b4>] compat_SyS_ioctl+0xac/0xfc4 [<ffffffc000084cf0>] el0_svc_naked+0x24/0x28 ---[ end trace 0dc4f99c2759c35c ]--- So this patch adds a dummy runtime for the original dummy substream to merely avoid the NULL pointer access. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog: v1->v2 * Allocate the runtime from heap instead of stack due to limitation of the stack sizes on different platforms. sound/soc/soc-dapm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8698c26..d908ff8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3493,6 +3493,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; + struct snd_pcm_runtime *runtime = NULL; u64 fmt; int ret; @@ -3541,6 +3542,14 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, memset(&substream, 0, sizeof(substream)); + /* Allocate a dummy snd_pcm_runtime for startup() and other ops() */ + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) { + ret = -ENOMEM; + goto out; + } + substream.runtime = runtime; + switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; @@ -3606,6 +3615,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } out: + kfree(runtime); kfree(params); return ret; } -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: usb-us122l: compatible ioctl support and fix sparse warnings
by Takashi Sakamoto 27 Jul '16

27 Jul '16

Hi, This patchset is to fix some sparse warnings in ALSA USB US122L module. Additionally, compatible ioctl is supported. Takashi Sakamoto (3): usb-us122l: replace 'unsigned' with 'unsigned int' usb-us122l: add support for compatible ioctl usb-us122l: add qualifier into pointer to userspace sound/usb/usx2y/us122l.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) -- 2.7.4

2 5

[alsa-devel] da7213 questions
by Peter Meerwald-Stadler 27 Jul '16

27 Jul '16

Hello, I have a couple of questions/comments regarding the da7213 driver in mainline: (1) the da7213 driver enables mclk in BIAS_SUSPEND, so essentially the clock is always running (since idle_bias_off is not set); at least on my platform, clk_set_rate() fails when the clock is already enabled (2) da7213's srm_en variable is redundant and can be removed (3) for 32 kHz PLL mode (DAI master), section 13.28 of the datasheet indicates that PLL_SRM_EN should be asserted; the da7213 drivers goesn't (4) is only S16_LE actually supported? #define DA7213_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) but in da7213_set_dai_fmt() we have /* By default only 32 BCLK per WCLK is supported */ dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_32; thanks, regards, p. -- Peter Meerwald-Stadler +43-664-2444418 (mobile)

2 1

[alsa-devel] Is this bug ? dev_set/get_drvdata() of rtd
by Kuninori Morimoto 27 Jul '16

27 Jul '16

Hi ALSA SoC I'm now checking ALSA SoC framework, and I noticed strange operation. Is this bug ? I think device data of "rtd->dev" is set as "rtd" on soc_post_component_init() by using dev_set_drvdata(), and many soc-core functions are based on it (= they are using dev_get_drvdata()) But, soc.h has snd_soc_pcm_set_drvdata(). device data of rtd->dev will be overwrite by this function ? But is this OK ? or bug ? It seems below drivers are using this sound/soc/intel/haswell/sst-haswell-pcm.c sound/soc/qcom/lpass-platform.c linux/sound/soc/soc-core.c static int soc_post_component_init(xxx) { ... dev_set_drvdata(rtd->dev, rtd); ... } static ssize_t codec_reg_show(xxx) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); ... } static ssize_t pmdown_time_show(xxx) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); ... } static ssize_t pmdown_time_set(xxx) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); ... } static umode_t soc_dev_attr_is_visible(xxx) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); ... } linux/include/sound/soc.h static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd, void *data) { dev_set_drvdata(rtd->dev, data); } Best regards --- Kuninori Morimoto

2 1

[alsa-devel] [PATCH v2 0/2] topology: Fix for handling widgets' references
by mengdong.lin＠linux.intel.com 27 Jul '16

27 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series fixes an error when merging a widget's private data and an inaccurate output message. History: v2: Split v1 into 2 patches. And set the valid reference element value only on success. Mengdong Lin (2): topology: Fix inaccurate message on failure to find a widgets's reference topology: Fix the missing referenced elem ptr when merging private data src/topology/dapm.c | 2 +- src/topology/data.c | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) -- 2.5.0

4 6

[alsa-devel] [PATCH] ASoC: dapm: Add a dummy snd_pcm_runtime to avoid NULL pointer access
by Nicolin Chen 27 Jul '16

27 Jul '16

The SND_SOC_DAPM_PRE_PMU case would call startup()/hw_params() that might access substream->runtime through other functions. For example: Unable to handle kernel NULL pointer dereference at virtual address [....] PC is at snd_pcm_hw_rule_add+0x24/0x1b0 LR is at snd_pcm_hw_constraint_list+0x20/0x28 [....] Process arecord (pid: 424, stack limit = 0xffffffc1ecaf0020) Call trace: [<ffffffc00086be68>] snd_pcm_hw_rule_add+0x24/0x1b0 [<ffffffc00086c014>] snd_pcm_hw_constraint_list+0x20/0x28 [<ffffffc0008b47a4>] cs53l30_pcm_startup+0x24/0x30 [<ffffffc0008a6260>] snd_soc_dai_link_event+0x290/0x354 [<ffffffc0008a7528>] dapm_seq_check_event.isra.31+0x134/0x2c8 [<ffffffc0008a7768>] dapm_seq_run_coalesced+0x94/0x1c8 [<ffffffc0008a7940>] dapm_seq_run+0xa4/0x404 [<ffffffc0008a8bac>] dapm_power_widgets+0x524/0x984 [<ffffffc0008ab1c4>] snd_soc_dapm_stream_event+0x8c/0xa8 [<ffffffc0008ac7f4>] soc_pcm_prepare+0x10c/0x1ec [<ffffffc000865b9c>] snd_pcm_do_prepare+0x1c/0x38 [<ffffffc000865600>] snd_pcm_action_single+0x40/0x88 [<ffffffc0008656b8>] snd_pcm_action_nonatomic+0x70/0x90 [<ffffffc000868d28>] snd_pcm_common_ioctl1+0xb6c/0xdd8 [<ffffffc000869508>] snd_pcm_capture_ioctl1+0x200/0x334 [<ffffffc00086a084>] snd_pcm_ioctl_compat+0x648/0x95c [<ffffffc0001ff4b4>] compat_SyS_ioctl+0xac/0xfc4 [<ffffffc000084cf0>] el0_svc_naked+0x24/0x28 ---[ end trace 0dc4f99c2759c35c ]--- So this patch adds a dummy runtime for the original dummy substream to merely avoid the NULL pointer access. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8698c26..eed15ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3493,6 +3493,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; + struct snd_pcm_runtime runtime; u64 fmt; int ret; @@ -3540,6 +3541,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, = config->channels_max; memset(&substream, 0, sizeof(substream)); + memset(&runtime, 0, sizeof(runtime)); + substream.runtime = &runtime; switch (event) { case SND_SOC_DAPM_PRE_PMU: -- 2.1.4

1 1

[alsa-devel] [GIT PULL] ASoC updates for v4.8
by Mark Brown 26 Jul '16

26 Jul '16

The following changes since commit 523d939ef98fd712632d93a5a2b588e477a7565e: Linux 4.7 (2016-07-24 12:23:50 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.8 for you to fetch changes up to e7ca8fcd15049b1e48ae2ef1434a68a51ef0ead5: Merge remote-tracking branches 'asoc/topic/wm8753' and 'asoc/topic/wm8985' into asoc-next (2016-07-24 22:07:48 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.8 Not really any framework work this time around (though we have seen one of the Analog Devices drivers move more to the clock API which is good to see) but rather a lot of new drivers: - Lots of updates for the Intel drivers, mostly board support and bug fixing, and to the NAU8825 driver. - Work on generalizing bits of simple-card to allow more code sharing with the Renesas rsrc-card (which can't use simple-card due to DPCM). - Removal of the Odroid X2 driver due to replacement with simple-card. - Support for several new Mediatek platforms and associated boards. - New drivers for Allwinner A10, Analog Devices ADAU7002, Broadcom Cygnus, Cirrus Logic CS35L33 and CS53L30, Maxim MAX8960 and MAX98504, Realtek RT5514 and Wolfson WM8758 ---------------------------------------------------------------- Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions Alan Cox (2): ASoC: Intel: atom: fix missing breaks that would cause the wrong operation to execute ASoC: Intel: atom: fix missing breaks that would cause the wrong operation to execute Alexander Shiyan (1): ASoC: wm8753: Replace magic number Amitoj Kaur Chawla (3): ASoC: wm8753: Remove unneeded header file ASoC: Atmel: ClassD: Simplify use of devm_ioremap_resource ASoC: atmel-pdmic: Simplify use of devm_ioremap_resource Andrea Gelmini (2): ASoC: dt: Fix typo ASoC: fsl: Fix typo Arnd Bergmann (10): ASoC: wm8985: add i2c dependency ASoC: fix ABE_TWL6040 dependency ASoC: cs53l30: include gpio/consumer.h ASoC: pcm1681/pcm1791: fix typo in declaration ASoC: rcar: fix 'const static' variables ASoC: remove one extraneous 'const' ASoC: nau8825: mark pm functions __maybe_unused ASoC fix up SND_SOC_WM8985 dependency ASoC: dwc: make pcm support built-in when necessary ASoC: cs35l33: mark PM functions as __maybe_unused Axel Lin (5): ASoC: cs35l33: Remove unnecessary free_irq call ASoC: cs35l33: Remove setting dapm->bias_level in cs35l33_set_bias_level ASoC: cs35l33: Fix testing return value of devm_gpiod_get_optional ASoC: cs35l33: Fix display revision id ASoC: rt5514-spi: Convert to use devm_* API Bard Liao (5): ASoC: rt5670: fix HP Playback Volume control ASoC: rt5670: patch reg-0x8a ASoC: rt5645: patch reg-0x8a ASoC: rt5645: set RT5645_PRIV_INDEX as volatile ASoC: rt5645: add DAC1 soft volume func control Bastien Nocera (1): ASoC: tlv320aix31xx: Add ACPI match for Lenovo 100S Ben Zhang (1): ASoC: Intel: Fix conflicting pcm dev drvdata on haswell Charles Keepax (12): ASoC: arizona: Add a notifier chain for CODEC events ASoC: arizona: Add event notification on voice trigger events ASoC: arizona: Tie SYSCLK to DRC signal activity widgets ASoC: arizona: Add voice trigger output widget ASoC: wm_adsp: Specifically propagate voice trigger event to caller ASoC: arizona: Add data structure for voice trigger notifier ASoC: arizona: Add a couple of missing consts ASoC: wm5102: Revert manual speaker enable ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: wm_adsp: Treat missing compressed buffer as a fatal error ASoC: compress: Pass error out of soc_compr_pointer Chris Zhong (1): ASoC: rockchip: correct the spdif clk Clemens Gruber (1): ASoC: sgtl5000: Remove misleading comment Colin Ian King (3): SoC: dwc: trivial fix of spelling mistake "unsuppted" -> "unsupported" ASoC: samsung: fix spelling mistake: "unknwon" -> "unknown" ASoC: fsl_spdif: fix spelling mistake: "receivce" -> "receive" Dan Carpenter (1): ASoC: mediatek: mt2701: fix some error handling in probe Dharageswari R (5): ASoC: Intel: Skylake: Don't pause stopped pipeline while deleting ASoC: Intel: Skylake: Disable SRAM Retention before D3 ASoC: Intel: Skylake: Avoid freeing up of unallocated memory/mcps ASoC: Intel: Skylake: Clean up of driver resources in suspend ASoC: Intel: Skylake: Fix to use the actual size for TLV control Enric Balletbo i Serra (4): ASoC: rockchip-max98090: Fix NULL pointer dereference while accessing to jack. ASoC: rockchip-max98090: Fix the Headset Mic route. ASoC: rockchip-max98090: Fix jack detection and event reporting. ASoC: max9867: Fix unix permissions for source files. Eric Nelson (5): ASoC: sgtl5000: Fix regulator support ASoC: sgtl5000: Write all default registers ASoC: sgtl5000: Initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: Disable internal PLL early ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF Fabio Estevam (2): ASoC: pcm5102a: Remove owner assignment from platform_driver ASoC: sgtl5000: Place optional properties in the correct section Garlic Tseng (10): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: bt-sco: add config prompt ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add mt2701 platform driver implementation. ASoC: mediatek: Add mt2701-cs42448 driver and config option. ASoC: mediatek: add BT implementation Geert Uytterhoeven (2): ASoC: Add file patterns for sound device tree bindings ASoC: ak4613: Implement suspend callback Helen Koike (6): ASoC: max9877: Remove unused function declaration ASoC: tpa6130a2: Register component ASoC: tap6130a2: Use regmap ASoC: tpa6130a2: Use snd soc volsw functions ASoC: tpa6130a2: Add DAPM support ASoC: tpa6130a2: Remove goto err_gpio Javier Martinez Canillas (1): ASoC: nau8825: Export I2C module alias information Jayachandran B (5): ASoC: Intel: Skylake: Enable firmware reload in suspend ASoC: Intel: Skylake: Update DSP stall bits ASoC: Intel: Skylake: Add DSP muti-core infrastructure ASoC: Intel: Skylake: Support multi-core in Skylake ASoC: Intel: Skylake: Support multi-core in Broxton Jeeja KP (8): ASoC: Intel: Skylake: Reset DSP pipe when host/link DMA is reset ASoC: Intel: Skylake: Set the pipe state to paused when paused ASoC: Intel: Skylake: Create Pipe to widget list in soc probe ASoC: Intel: Skylake: Set the DSP pipe type ASoC: Intel: Skylake: Copy the pipe parameter by pipe type ASoC: Intel: Skylake: Report position in pointer query ASoC: Intel: Add support for PM ops in bxt-rt298 ASoC: dapm: Fix to return correct path list in is_connected_ep. John Hsu (11): ASoC: nau8825: support different clock source for FLL function ASoC: nau8825: improve FLL function for better performance ASoC: nau8825: reduce standby power consumption ASoC: nau8825: change output power for interrupt ASoC: nau8825: assign DAC Ch to match headset L/R ASoC: nau8825: add programmable biquad filter control ASoC: nau8825: non-clock jack detection for power saving at standby ASoC: nau8825: correct typo in biquad filter coefficients ASoC: nau8825: cross talk suppression measurement function ASoC: nau8825: jack connection decision with different insertion logic ASoC: nau8825: drop redundant idiom when converting integer to boolean Jose Abreu (3): ASoC: dwc: Add helper functions to disable/enable irqs ASoC: dwc: Add PIO PCM extension ASoC: dwc: Add irq parameter to DOCUMENTATION Kalle Kankare (1): sgtl5000: add Lineout volume control Krzysztof Kozlowski (1): MAINTAINERS: ASoC: samsung: Add Sylwester Nawrocki and Krzysztof Kozlowski Kuninori Morimoto (16): ASoC: rsrc-card: remove unused dai_num ASoC: simple-card: platform also uses asoc_simple_card_sub_parse_of() ASoC: rsnd: adg :: AUDIO-CLKOUTn asynchronizes support ASoC: simple-card: use common PREFIX for each DT property ASoC: hdmi-codec: callback function will be called with private data ASoC: simple-card: add new asoc_simple_jack and use it ASoC: add new simple-card-utils.c ASoC: simple-card: use asoc_simple_card_parse_daifmt() ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() ASoC: simple-card-utils: add asoc_simple_card_set_dailink_name() ASoC: simple-card: use asoc_simple_card_parse_dailink_name() ASoC: simple-card-utils: add asoc_simple_card_parse_card_name() ASoC: simple-card: use asoc_simple_card_parse_card_name() ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai ASoC: rsrc-card: use asoc_simple_card_parse_card_name() Lars-Peter Clausen (4): ASoC: Add ADAU7002 Stereo PDM-to-I2S/TDM Converter DT bindings ASoC: Add ADAU7002 Stereo PDM-to-I2S/TDM Converter driver ASoC: adau: Factor out shared PLL configuration code ASoC: adau17x1: Add support for specifying the MCLK using the CCF Mark Brown (20): Merge branch 'topic/compress' of git://git.kernel.org/.../broonie/sound into asoc-adsp Merge branch 'topic/simple' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm5102a' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/adau', 'asoc/topic/adau7002', 'asoc/topic/adsp', 'asoc/topic/ak4613' and 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/bt-sco', 'asoc/topic/compress' and 'asoc/topic/cs35l33' into asoc-next Merge remote-tracking branches 'asoc/topic/cs53l30', 'asoc/topic/cygnus', 'asoc/topic/da7219' and 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/dt', 'asoc/topic/dwc' and 'asoc/topic/fsl' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/hdmi', 'asoc/topic/maintainers' and 'asoc/topic/max8960' into asoc-next Merge remote-tracking branches 'asoc/topic/max98504', 'asoc/topic/max9867', 'asoc/topic/max9877', 'asoc/topic/mtk' and 'asoc/topic/nau8825' into asoc-next Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5514', 'asoc/topic/rt5614', 'asoc/topic/rt5670' and 'asoc/topic/s8' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/sti' and 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/tas571x', 'asoc/topic/tlv320aic31xx', 'asoc/topic/tpa6130a2', 'asoc/topic/twl6040' and 'asoc/topic/wm8731' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8753' and 'asoc/topic/wm8985' into asoc-next Markus Elfring (1): ASoC: Intel: Skylake: Delete an unnecessary check before the function call "release_firmware" Matt Flax (1): ASoc: wm8731: add 32bit mode. Maxime Ripard (2): ASoC: sunxi: Add A10 I2S controller binding documentation ASoC: sunxi: Add Allwinner A10 Digital Audio driver Nicolin Chen (8): ASoC: cs53l30: Add codec driver support for Cirrus CS53L30 ASoC: cs53l30: Rename the volume controls for preamplifier ASoC: cs53l30: Check return value of regcache_sync() ASoC: cs53l30: Correct clock inversion check ASoC: cs53l30: Set idle_bias_off true ASoC: cs53l30: Add MUTE pin control support via GPIO ASoC: cs53l30: Fix a bug for TDM slot location validation ASoC: cs53l30: Fix bit shift issue of TDM mode Oder Chiou (3): ASoC: rt5514: add rt5514 SPI driver ASoC: rt5514: Add the MCLK handling ASoC: rt5514: Fix the issue that the variable dereferenced before checking PC Liao (3): ASoC: mediatek: Change the order of MCLK clock configuration ASoC: mediatek: add MCLK source selection ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Paul Handrigan (2): ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver. ASoC: cs35l33: Add device tree bindings file for cs35l33 Peter Griffin (1): ASoC: sti: Update DT example to match the driver code Peter Meerwald (1): ASoC: atmel_ssc_dai: Fix DMA params for different SSC Peter Rosin (2): ASoC: max8960: add bindings for the max9860 codec ASoC: max9860: new driver Peter Ujfalusi (8): ASoC: omap: Kconfig: SND_OMAP_SOC_OMAP_ABE_TWL6040 to select CLK_TWL6040 ASoC: omap-mcpdm: Move the WD enable write inside omap_mcpdm_open_streams() ASoC: omap-mcpdm: Support for suspend resume ASoC: omap-mcpdm: Add support for pdmclk clock handling ASoC: tlv320aic31xx: Make the register values human readable ASoC: ak4642: Implement suspend callback ASoC: davinci-mcasp: Fix dra7 DMA offset when using CFG port ASoC: davinci-mcasp: Use a copy of pdata per instance during DT boot Petr Kulhavy (5): ASoC: tas571x: add biquad registers for TAS5717/19 ASoC: tas571x: add biquads for TAS5717/19 ASoC: wm8985: add register definitions for WM8758 ASoC: wm8985: add support for WM8758 ASoC: tas571x: add input channel mixer for TAS5717/19 Piotr Stankiewicz (2): ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets ASoC: dpcm: play nice with CODEC<->CODEC links Ramesh Babu (1): ASoC: Intel: Skylake: Add strip extended manifest utility Randy Dunlap (1): ASoC: intel: make function stub static Sathyanarayana Nujella (2): ASoC: Intel: Add Broxton-P Dialog Maxim machine driver ASoC: Intel: Add Broxton-P Dialog+Maxim machine driver entry Senthilnathan Veppur (5): ASoC: Intel: Add DMIC 4 channel support for bxt machine ASoC: Intel: Add FE rate & channel constraints for bxt-rt298 ASoC: Intel: Update ignore suspend for bxt-rt298 ASoC: Intel: Skylake: Update FW purge for Broxton ASoC: Intel: Skylake: Fix fw reload failure Shreyas NC (2): ASoC: Intel: Skylake: Add DSP firmware manifest parsing ASoC: hdac_hdmi: Add device id for Kabylake Simon Trimmer (1): ASoC: wm_adsp: Disable DMAs before clearing the transfer length Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver Srinivas Kandagatla (1): ASoC: Introduce SOC_SINGLE_S8_TLV() macro Subhransu S. Prusty (1): ASoC: dapm: Export snd_soc_dapm_new_control Sudip Mukherjee (2): ASoC: fsl: fix build failure ASoC: atmel: fix build failure Sugar Zhang (2): ASoC: rockchip: i2s: configure the sdio pins' iomux mode ASoC: rockchip: i2s: rename I2S_CKR_TRCM_TX/RXSHARE to I2S_CKR_TRCM_TX/RXONLY Sylwester Nawrocki (8): ASoC: samsung: Remove unused "samsung-i2sv4" platform_device_id entry ASoC: samsung: Remove definition of an unused data structure ASoC: samsung: Use of_device_get_match_data() helper ASoC: samsung: Remove unused Odroid x2/u3 machine driver ASoC: max98504: Add max98504 speaker amplifier driver ASoC: Add DT bindings documentation for max98504 amplifier ASoC: samsung: Fix error paths in the I2S driver's probe() ASoC: samsung: Specify DMA channels through struct snd_dmaengine_pcm_config Takashi Iwai (1): ASoC: intel: Fix sst-dsp dependency on dw stuff Vedang Patel (4): ASoC: hdac_hdmi: Increase loglevel of hex dump printed ASoC: Intel: common: increase the loglevel of "FW Poll Status". ASoC: Intel: Skylake: Increase loglevel of debug messages. ASoC: Intel: Skylake: Fix NULL Pointer exception in dynamic_debug. Vinod Koul (27): ASoC: Intel: Skylake: Don't use local pointer for firmware ASoC: Intel: Skylake: Strip manifest for Skylake platform ASoC: Intel: Skylake: Strip manifest for Broxton platform ASoC: Intel: Skylake: Find uuids for Skylake ASoC: Intel: Skylake: Find uuids for Broxton ASoC: Intel: Skylake: Check for module list being NULL ASoC: Intel: Kconfig: fix build when ACPI is not enabled ASoC: Intel: Kconfig: formatting update ASoC: Intel: Revert "ASoC: Intel: Add support for PM ops in bxt-rt298" ASoC: Intel: Skylake: Disable async suspend ASoC: Intel: Skylake: Add pm ops for broxton-rt298 machine ASoC: Intel: Skylake: Update comment style ASoC: Intel: Atom: Add quirk for Surface 3 ASoC: Intel: Add surface3 entry in CHT-RT5645 machine ASoC: rt5645: Add ACPI ID 10EC5640 ASoC: Intel: cht: fix uninit variable warning ASoC: Intel: atom: statify cht_quirk ASoC: hdac_hdmi: Fix potential NULL dereference ASoC: Intel: Skylake: Initialize module list for Broxton ASoC: Intel: add kablake device IDs ASoC: rt286: set combo jack for Kabylake ASoC: Intel: board: add kabylake machine id ASoC: Intel: Skylake: reduce machine name for skl_nau88l25_ssm4567 ASoC: Intel: Skylake: reduce machine name for skl_nau88l25_max98357a ASoC: Intel: Kbl: add kabylake additional machine entries ASoC: Intel: board: add kabylake nau88l25_max98357a machine id ASoC: Intel: board: add kabylake nau88l25_ssm4567 machine id Wei Yongjun (6): ASoC: sti: fix return value check in uni_player_parse_dt_audio_glue() ASoC: max9860: fix non static symbol warnings ASoC: cs53l30: Fix non static symbol warnings ASoC: sunxi: remove redundant dev_err call in sun4i_i2s_probe() ASoC: cs35l33: Remove unused including <linux/version.h> ASoC: mediatek: mt2701: fix non static symbol warning Yong Zhi (5): ASoC: Intel: Skylake: Add channel constraints for refcap ASoC: Intel: Skylake: Use refcap device for mono recording ASoC: Intel: Skylake: Add api to retrieve dmic array info from nhlt ASoC: Intel: boards: configure DMIC for machine sklnau8825adi ASoC: Intel: boards: configure DMIC for machine sklnau8825max kbuild test robot (2): ASoC: cs35l33: fix platform_no_drv_owner.cocci warnings ASoC: sunxi: fix semicolon.cocci warnings .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + .../devicetree/bindings/sound/adi,adau7002.txt | 19 + .../bindings/sound/brcm,cygnus-audio.txt | 67 + Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/cs35l33.txt | 126 ++ .../devicetree/bindings/sound/cs53l30.txt | 44 + .../devicetree/bindings/sound/designware-i2s.txt | 4 + .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 +- .../devicetree/bindings/sound/max98504.txt | 44 + .../devicetree/bindings/sound/max9860.txt | 28 + .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 +- .../devicetree/bindings/sound/omap-mcpdm.txt | 10 + .../devicetree/bindings/sound/renesas,rsnd.txt | 2 + .../devicetree/bindings/sound/rockchip-i2s.txt | 5 + Documentation/devicetree/bindings/sound/rt5514.txt | 5 + .../bindings/sound/samsung,odroidx2-max98090.txt | 35 - .../devicetree/bindings/sound/sgtl5000.txt | 18 +- .../devicetree/bindings/sound/st,sti-asoc-card.txt | 20 +- .../devicetree/bindings/sound/sun4i-i2s.txt | 34 + Documentation/sound/alsa/soc/machine.txt | 2 +- MAINTAINERS | 11 +- drivers/base/property.c | 28 + include/acpi/acpi_bus.h | 7 + include/linux/acpi.h | 6 + include/linux/mfd/arizona/core.h | 10 + include/linux/of.h | 14 +- include/linux/property.h | 3 + include/sound/compress_driver.h | 5 + include/sound/cs35l33.h | 48 + include/sound/hdmi-codec.h | 13 +- include/sound/simple_card.h | 11 +- include/sound/simple_card_utils.h | 36 + include/sound/soc-dapm.h | 8 +- include/sound/soc.h | 11 + sound/core/compress_offload.c | 67 +- sound/soc/atmel/Kconfig | 1 + sound/soc/atmel/atmel-classd.c | 5 - sound/soc/atmel/atmel-pdmic.c | 5 - sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 ++++++++++ sound/soc/bcm/cygnus-ssp.c | 1529 ++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ sound/soc/codecs/Kconfig | 38 +- sound/soc/codecs/Makefile | 14 + sound/soc/codecs/adau-utils.c | 61 + sound/soc/codecs/adau-utils.h | 7 + sound/soc/codecs/adau1373.c | 38 +- sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 251 +-- sound/soc/codecs/adau17x1.h | 6 + sound/soc/codecs/adau7002.c | 80 + sound/soc/codecs/ak4613.c | 12 +- sound/soc/codecs/ak4642.c | 13 +- sound/soc/codecs/arizona.c | 91 +- sound/soc/codecs/arizona.h | 22 +- sound/soc/codecs/bt-sco.c | 52 +- sound/soc/codecs/cs35l33.c | 1303 +++++++++++++++ sound/soc/codecs/cs35l33.h | 221 +++ sound/soc/codecs/cs47l24.c | 19 + sound/soc/codecs/cs53l30.c | 1143 ++++++++++++++ sound/soc/codecs/cs53l30.h | 459 ++++++ sound/soc/codecs/da7219-aad.c | 103 +- sound/soc/codecs/da7219.c | 34 +- sound/soc/codecs/hdac_hdmi.c | 7 +- sound/soc/codecs/hdmi-codec.c | 15 +- sound/soc/codecs/max98504.c | 383 +++++ sound/soc/codecs/max98504.h | 59 + sound/soc/codecs/max9860.c | 753 +++++++++ sound/soc/codecs/max9860.h | 162 ++ sound/soc/codecs/max9867.c | 0 sound/soc/codecs/max9867.h | 0 sound/soc/codecs/max9877.h | 2 - sound/soc/codecs/nau8825.c | 1256 +++++++++++++-- sound/soc/codecs/nau8825.h | 127 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm179x.c | 2 +- sound/soc/codecs/pcm5102a.c | 1 - sound/soc/codecs/rt286.c | 7 + sound/soc/codecs/rt5514-spi.c | 447 ++++++ sound/soc/codecs/rt5514-spi.h | 38 + sound/soc/codecs/rt5514.c | 168 +- sound/soc/codecs/rt5514.h | 7 + sound/soc/codecs/rt5645.c | 24 +- sound/soc/codecs/rt5645.h | 3 + sound/soc/codecs/rt5670.c | 3 +- sound/soc/codecs/sgtl5000.c | 431 ++--- sound/soc/codecs/sgtl5000.h | 2 + sound/soc/codecs/tas571x.c | 188 +++ sound/soc/codecs/tas571x.h | 40 + sound/soc/codecs/tlv320aic31xx.h | 134 +- sound/soc/codecs/tpa6130a2.c | 394 ++--- sound/soc/codecs/tpa6130a2.h | 14 +- sound/soc/codecs/wm5102.c | 1 + sound/soc/codecs/wm5110.c | 19 + sound/soc/codecs/wm8731.c | 5 +- sound/soc/codecs/wm8753.c | 3 +- sound/soc/codecs/wm8985.c | 143 +- sound/soc/codecs/wm8985.h | 38 + sound/soc/codecs/wm8998.c | 1 + sound/soc/codecs/wm_adsp.c | 34 +- sound/soc/codecs/wm_adsp.h | 4 + sound/soc/davinci/davinci-mcasp.c | 9 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 4 +- sound/soc/dwc/designware_i2s.c | 231 ++- sound/soc/dwc/designware_pcm.c | 225 +++ sound/soc/dwc/local.h | 128 ++ sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/generic/Kconfig | 4 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-card-utils.c | 97 ++ sound/soc/generic/simple-card.c | 276 ++-- sound/soc/intel/Kconfig | 73 +- sound/soc/intel/atom/sst/sst_acpi.c | 44 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 460 ++++++ sound/soc/intel/boards/bxt_rt298.c | 118 ++ sound/soc/intel/boards/cht_bsw_rt5645.c | 20 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 51 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 51 +- sound/soc/intel/boards/skl_rt286.c | 9 + sound/soc/intel/common/Makefile | 4 +- sound/soc/intel/common/sst-acpi.h | 4 +- sound/soc/intel/common/sst-dsp-priv.h | 4 - sound/soc/intel/common/sst-dsp.c | 69 +- sound/soc/intel/common/sst-dsp.h | 2 +- sound/soc/intel/common/sst-firmware.c | 68 + sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/bxt-sst.c | 202 ++- sound/soc/intel/skylake/skl-messages.c | 53 +- sound/soc/intel/skylake/skl-nhlt.c | 40 + sound/soc/intel/skylake/skl-nhlt.h | 22 + sound/soc/intel/skylake/skl-pcm.c | 93 +- sound/soc/intel/skylake/skl-sst-dsp.c | 260 ++- sound/soc/intel/skylake/skl-sst-dsp.h | 102 +- sound/soc/intel/skylake/skl-sst-ipc.c | 4 +- sound/soc/intel/skylake/skl-sst-ipc.h | 18 + sound/soc/intel/skylake/skl-sst-utils.c | 256 +++ sound/soc/intel/skylake/skl-sst.c | 124 +- sound/soc/intel/skylake/skl-topology.c | 230 ++- sound/soc/intel/skylake/skl-topology.h | 7 +- sound/soc/intel/skylake/skl.c | 39 +- sound/soc/intel/skylake/skl.h | 10 + sound/soc/mediatek/Kconfig | 38 +- sound/soc/mediatek/Makefile | 10 +- sound/soc/mediatek/common/Makefile | 16 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 172 ++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1656 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 412 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 925 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 73 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-mcpdm.c | 74 +- sound/soc/omap/rx51.c | 46 +- sound/soc/rockchip/rockchip_i2s.c | 71 +- sound/soc/rockchip/rockchip_i2s.h | 11 +- sound/soc/rockchip/rockchip_max98090.c | 65 +- sound/soc/rockchip/rockchip_spdif.c | 17 +- sound/soc/samsung/Kconfig | 8 - sound/soc/samsung/Makefile | 2 - sound/soc/samsung/ac97.c | 3 +- sound/soc/samsung/dma.h | 9 +- sound/soc/samsung/dmaengine.c | 31 +- sound/soc/samsung/i2s.c | 53 +- sound/soc/samsung/odroidx2_max98090.c | 185 --- sound/soc/samsung/pcm.c | 3 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 3 +- sound/soc/samsung/s3c24xx-i2s.c | 3 +- sound/soc/samsung/spdif.c | 3 +- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/adg.c | 18 + sound/soc/sh/rcar/gen.c | 12 +- sound/soc/sh/rcar/rsrc-card.c | 99 +- sound/soc/soc-compress.c | 5 +- sound/soc/soc-dapm.c | 61 +- sound/soc/soc-pcm.c | 43 +- sound/soc/sti/uniperif_player.c | 4 +- sound/soc/sunxi/Kconfig | 9 + sound/soc/sunxi/Makefile | 2 +- sound/soc/sunxi/sun4i-i2s.c | 701 +++++++++ 205 files changed, 19212 insertions(+), 3008 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.txt create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cs35l33.txt create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/sun4i-i2s.txt create mode 100644 include/sound/cs35l33.h create mode 100644 include/sound/simple_card_utils.h create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h create mode 100644 sound/soc/codecs/adau-utils.c create mode 100644 sound/soc/codecs/adau-utils.h create mode 100644 sound/soc/codecs/adau7002.c create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h mode change 100755 => 100644 sound/soc/codecs/max9867.c mode change 100755 => 100644 sound/soc/codecs/max9867.h create mode 100644 sound/soc/codecs/rt5514-spi.c create mode 100644 sound/soc/codecs/rt5514-spi.h create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/local.h create mode 100644 sound/soc/generic/simple-card-utils.c create mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c create mode 100644 sound/soc/intel/skylake/skl-sst-utils.c create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (80%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h delete mode 100644 sound/soc/samsung/odroidx2_max98090.c create mode 100644 sound/soc/sunxi/sun4i-i2s.c

2 1

[alsa-devel] v1.1.2 release?
by Takashi Iwai 26 Jul '16

26 Jul '16

Hi Jaroslav, shall we release v1.1.2 along with 4.7 kernel? We've got already a bunch of changes in alsa-lib and co. thanks, Takashi

3 4

[alsa-devel] [PATCH v4 4/4] ASoC: samsung: Add machine driver for Exynos5433 based TM2 board
by Sylwester Nawrocki 25 Jul '16

25 Jul '16

This patch adds the sound machine driver for TM2 and TM2E board. Speaker and headphone playback, Main Mic capture, Bluetooth, Voice call and external accessory are supported. Signed-off-by: Inha Song <ideal.song(a)samsung.com> [k.kozlowski: rebased on 4.1] Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> [s.nawrocki: rebased to 4.7, adjustment to the ASoC core changes, removed unused ops and direct calls to the max98504 function, added parsing of "audio-amplifier" and "audio-codec" properties, added TDM API calls, switched to gpiod API] Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v3: - removed SND_SOC_SAMSUNG_AUDSS from Kconfig. Changes since v2: - added missing Kconfig dependencies. Changes since initial version: - added PDM Tx channels setup through TDM API - adaptation to renamed 'samsung,model', 'samsung,i2s-controller', 'samsung,speaker-amplifier' properties, - removed some dev_dbg() calls, - cleaned up mic-bias GPIO handling and switched to gpiod API, - added parsing of 'audio-codec' property, - initialized codec_of_node of dai_link instead of codec_name, - switched to using clock, clock-names properties from the wm5110 codec node, - fixed error paths in probe() (of_node reference counting). --- sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 590 insertions(+) create mode 100644 sound/soc/samsung/tm2_wm5110.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 78baa26..a711605 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -237,3 +237,12 @@ config SND_SOC_ARNDALE_RT5631_ALC5631 depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631 + +config SND_SOC_SAMSUNG_TM2_WM5110 + tristate "SoC I2S Audio support for WM5110 on TM2 board" + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + select SND_SOC_MAX98504 + select SND_SOC_WM5110 + select SND_SAMSUNG_I2S + help + Say Y if you want to add support for SoC audio on the TM2 board. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71..c15a759 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -45,6 +45,7 @@ snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o snd-soc-arndale-rt5631-objs := arndale_rt5631.o +snd-soc-tm2-wm5110-objs := tm2_wm5110.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o +obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c new file mode 100644 index 0000000..9728b3c --- /dev/null +++ b/sound/soc/samsung/tm2_wm5110.c @@ -0,0 +1,579 @@ +/* + * Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd. + * + * Authors: Inha Song <ideal.song(a)samsung.com> + * Sylwester Nawrocki <s.nawrocki(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/module.h> +#include <linux/of.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "../codecs/wm5110.h" + +struct tm2_machine_priv { + struct snd_soc_codec *codec; + struct clk *codec_mclk1; + struct clk *codec_mclk2; + + unsigned int sysclk_rate; + + struct gpio_desc *gpio_mic_bias; +}; + +static int tm2_start_sysclk(struct snd_soc_card *card) +{ + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_codec *codec = priv->codec; + unsigned long mclk_rate = clk_get_rate(priv->codec_mclk1); + int ret; + + ret = clk_prepare_enable(priv->codec_mclk1); + if (ret < 0) { + dev_err(card->dev, "Failed to enable mclk: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5110_FLL1, + ARIZONA_FLL_SRC_MCLK1, + mclk_rate, + priv->sysclk_rate); + if (ret < 0) { + dev_err(codec->dev, "Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5110_FLL1_REFCLK, + ARIZONA_FLL_SRC_MCLK1, + mclk_rate, + priv->sysclk_rate); + if (ret < 0) { + dev_err(codec->dev, "Failed to set FLL1 Source: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, + ARIZONA_CLK_SRC_FLL1, + priv->sysclk_rate, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec->dev, "Failed to set SYSCLK Source: %d\n", ret); + return ret; + } + + return 0; +} + +static int tm2_stop_sysclk(struct snd_soc_card *card) +{ + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_codec *codec = priv->codec; + int ret; + + ret = snd_soc_codec_set_pll(codec, WM5110_FLL1, 0, 0, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to stop FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, + ARIZONA_CLK_SRC_FLL1, 0, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to stop SYSCLK: %d\n", ret); + return ret; + } + + clk_disable_unprepare(priv->codec_mclk1); + + return 0; +} + +static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card); + int ret; + + switch (params_rate(params)) { + case 4000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + case 96000: + case 192000: + /* Highest possible SYSCLK frequency: 147.456MHz */ + priv->sysclk_rate = 147456000U; + break; + case 11025: + case 22050: + case 44100: + case 88200: + case 176400: + /* Highest possible SYSCLK frequency: 135.4752 MHz */ + priv->sysclk_rate = 135475200U; + break; + default: + dev_err(codec->dev, "Not supported sample rate: %d\n", + params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + return tm2_start_sysclk(rtd->card); +} + +static struct snd_soc_ops tm2_aif1_ops = { + .hw_params = tm2_aif1_hw_params, +}; + +static int tm2_aif2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card); + unsigned long mclk_rate = clk_get_rate(priv->codec_mclk1); + unsigned int asyncclk_rate; + int ret; + + switch (params_rate(params)) { + case 8000: + case 12000: + case 16000: + /* Highest possible ASYNCCLK frequency: 49.152MHz */ + asyncclk_rate = 49152000U; + break; + case 11025: + /* Highest possible ASYNCCLK frequency: 45.1584 MHz */ + asyncclk_rate = 45158400U; + break; + default: + dev_err(codec->dev, "Not supported sample rate: %d\n", + params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_codec_set_pll(codec, WM5110_FLL2, + ARIZONA_FLL_SRC_MCLK1, + mclk_rate, + asyncclk_rate); + if (ret < 0) { + dev_err(codec->dev, "Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5110_FLL2_REFCLK, + ARIZONA_FLL_SRC_MCLK1, + mclk_rate, + asyncclk_rate); + if (ret < 0) { + dev_err(codec->dev, "Failed to set FLL1 Source: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); + + if (ret < 0) { + dev_err(codec_dai->dev, "Failed to set ASYNCCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK, + ARIZONA_CLK_SRC_FLL2, + asyncclk_rate, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec->dev, "Failed to set ASYNCCLK Source: %d\n", ret); + return ret; + } + + return 0; +} + +static struct snd_soc_ops tm2_aif2_ops = { + .hw_params = tm2_aif2_hw_params, +}; + +static int tm2_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + gpiod_set_value_cansleep(priv->gpio_mic_bias, 1); + break; + case SND_SOC_DAPM_POST_PMD: + gpiod_set_value_cansleep(priv->gpio_mic_bias, 0); + break; + } + + return 0; +} + +static int tm2_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + + if (dapm->dev != rtd->codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) + tm2_start_sysclk(card); + break; + case SND_SOC_BIAS_OFF: + tm2_stop_sysclk(card); + break; + case SND_SOC_BIAS_PREPARE: + break; + default: + break; + } + + card->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_aux_dev tm2_speaker_amp_dev; + +static int tm2_late_probe(struct snd_soc_card *card) +{ + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link_component dlc = { 0 }; + struct snd_soc_dai *amp_pdm_dai; + struct snd_soc_pcm_runtime *rtd; + unsigned int ch_map[] = { 0, 1 }; + int ret; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + priv->codec = rtd->codec; + + /* 32 kHz must be enabled for jack detection */ + if (!IS_ERR(priv->codec_mclk2)) + clk_prepare_enable(priv->codec_mclk2); + + dlc.of_node = tm2_speaker_amp_dev.codec_of_node; + amp_pdm_dai = snd_soc_find_dai(&dlc); + if (!amp_pdm_dai) + return -ENODEV; + + /* Set the MAX98504 V/I sense PDM Tx DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(amp_pdm_dai, ARRAY_SIZE(ch_map), + ch_map, 0, NULL); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(amp_pdm_dai, 0x3, 0x0, 2, 16); + if (ret < 0) + return ret; + + return 0; +} + +static int tm2_suspend_post(struct snd_soc_card *card) +{ + return tm2_stop_sysclk(card); +} + +static int tm2_resume_pre(struct snd_soc_card *card) +{ + return tm2_start_sysclk(card); +} + +static const struct snd_kcontrol_new tm2_controls[] = { + SOC_DAPM_PIN_SWITCH("HP"), + SOC_DAPM_PIN_SWITCH("SPK"), + SOC_DAPM_PIN_SWITCH("RCV"), + SOC_DAPM_PIN_SWITCH("VPS"), + SOC_DAPM_PIN_SWITCH("HDMI"), + + SOC_DAPM_PIN_SWITCH("Main Mic"), + SOC_DAPM_PIN_SWITCH("Sub Mic"), + SOC_DAPM_PIN_SWITCH("Third Mic"), + + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +const struct snd_soc_dapm_widget tm2_dapm_widgets[] = { + SND_SOC_DAPM_HP("HP", NULL), + SND_SOC_DAPM_SPK("SPK", NULL), + SND_SOC_DAPM_SPK("RCV", NULL), + SND_SOC_DAPM_LINE("VPS", NULL), + SND_SOC_DAPM_LINE("HDMI", NULL), + + SND_SOC_DAPM_MIC("Main Mic", tm2_mic_bias), + SND_SOC_DAPM_MIC("Sub Mic", NULL), + SND_SOC_DAPM_MIC("Third Mic", NULL), + + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_component_driver tm2_component = { + .name = "tm2-audio", +}; + +static struct snd_soc_dai_driver tm2_ext_dai[] = { + { + .name = "Voice call", + .playback = { + .channels_min = 1, + .channels_max = 4, + .rate_min = 8000, + .rate_max = 48000, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 4, + .rate_min = 8000, + .rate_max = 48000, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, + { + .name = "Bluetooth", + .playback = { + .channels_min = 1, + .channels_max = 4, + .rate_min = 8000, + .rate_max = 16000, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 16000, + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +static struct snd_soc_dai_link tm2_dai_links[] = { + { + .name = "WM5110 AIF1", + .stream_name = "HiFi Primary", + .codec_dai_name = "wm5110-aif1", + .ops = &tm2_aif1_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, { + .name = "WM5110 Voice", + .stream_name = "Voice call", + .codec_dai_name = "wm5110-aif2", + .ops = &tm2_aif2_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + }, { + .name = "WM5110 BT", + .stream_name = "Bluetooth", + .codec_dai_name = "wm5110-aif3", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + } +}; + +static struct snd_soc_card tm2_card = { + .owner = THIS_MODULE, + + .dai_link = tm2_dai_links, + .num_links = ARRAY_SIZE(tm2_dai_links), + .controls = tm2_controls, + .num_controls = ARRAY_SIZE(tm2_controls), + .dapm_widgets = tm2_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tm2_dapm_widgets), + + .aux_dev = &tm2_speaker_amp_dev, + .num_aux_devs = 1, + + .late_probe = tm2_late_probe, + + .set_bias_level = tm2_set_bias_level, + + .suspend_post = tm2_suspend_post, + .resume_pre = tm2_resume_pre, +}; + +static int tm2_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_soc_card *card = &tm2_card; + struct tm2_machine_priv *priv; + struct device_node *cpu_dai_node, *codec_dai_node; + int ret, i; + + if (!dev->of_node) { + dev_err(dev, "DT node is missing\n"); + return -ENODEV; + } + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + card->dev = dev; + + priv->gpio_mic_bias = devm_gpiod_get(dev, "mic-bias", + GPIOF_OUT_INIT_LOW); + if (IS_ERR(priv->gpio_mic_bias)) { + dev_err(dev, "Failed to get mic bias gpio\n"); + return PTR_ERR(priv->gpio_mic_bias); + } + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret < 0) { + dev_err(dev, "Card name is not specified\n"); + return ret; + } + + ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); + if (ret < 0) { + dev_err(dev, "Audio routing is not specified or invalid\n"); + return ret; + } + + card->aux_dev[0].codec_of_node = of_parse_phandle(dev->of_node, + "audio-amplifier", 0); + if (!card->aux_dev[0].codec_of_node) { + dev_err(dev, "audio-amplifier property invalid or missing\n"); + return -EINVAL; + } + + cpu_dai_node = of_parse_phandle(dev->of_node, "i2s-controller", 0); + if (!cpu_dai_node) { + dev_err(dev, "i2s-controllers property invalid or missing\n"); + ret = -EINVAL; + goto err_put_amp; + } + + codec_dai_node = of_parse_phandle(dev->of_node, "audio-codec", 0); + if (!codec_dai_node) { + dev_err(dev, "audio-codec property invalid or missing\n"); + ret = -EINVAL; + goto err_put_cpu_dai; + } + + for (i = 0; i < card->num_links; i++) { + card->dai_link[i].cpu_dai_name = NULL; + card->dai_link[i].cpu_name = NULL; + card->dai_link[i].platform_name = NULL; + card->dai_link[i].codec_of_node = codec_dai_node; + card->dai_link[i].cpu_of_node = cpu_dai_node; + card->dai_link[i].platform_of_node = cpu_dai_node; + } + + priv->codec_mclk1 = of_clk_get_by_name(codec_dai_node, "mclk1"); + if (IS_ERR(priv->codec_mclk1)) { + dev_err(dev, "Failed to get mclk1 clock\n"); + ret = PTR_ERR(priv->codec_mclk1); + goto err_put_codec_dai; + } + + /* mclk2 is optional */ + priv->codec_mclk2 = of_clk_get_by_name(codec_dai_node, "mclk2"); + if (IS_ERR(priv->codec_mclk2)) + dev_info(dev, "Not using mclk2 clock\n"); + + ret = devm_snd_soc_register_component(dev, &tm2_component, + tm2_ext_dai, ARRAY_SIZE(tm2_ext_dai)); + if (ret < 0) { + dev_err(dev, "Failed to register component: %d\n", ret); + goto err_put_mclk; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret < 0) { + dev_err(dev, "Failed to register card: %d\n", ret); + goto err_put_mclk; + } + + return 0; + +err_put_mclk: + clk_put(priv->codec_mclk1); + if (!IS_ERR(priv->codec_mclk2)) + clk_put(priv->codec_mclk2); +err_put_codec_dai: + of_node_put(codec_dai_node); +err_put_cpu_dai: + of_node_put(cpu_dai_node); +err_put_amp: + of_node_put(card->aux_dev[0].codec_of_node); + return ret; +} + +static int tm2_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = &tm2_card; + struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(card); + + clk_put(priv->codec_mclk1); + if (!IS_ERR(priv->codec_mclk2)) + clk_put(priv->codec_mclk2); + + of_node_put(card->dai_link[0].codec_of_node); + of_node_put(card->dai_link[0].cpu_of_node); + of_node_put(card->aux_dev[0].codec_of_node); + + return 0; +} + +static const struct of_device_id tm2_of_match[] = { + { .compatible = "samsung,tm2-audio" }, + { }, +}; +MODULE_DEVICE_TABLE(of, tm2_of_match); + +static struct platform_driver tm2_driver = { + .driver = { + .name = "tm2-audio", + .pm = &snd_soc_pm_ops, + .of_match_table = tm2_of_match, + }, + .probe = tm2_probe, + .remove = tm2_remove, +}; + +module_platform_driver(tm2_driver); + +MODULE_AUTHOR("Inha Song <ideal.song(a)samsung.com>"); +MODULE_DESCRIPTION("ALSA SoC Exynos TM2 Audio Support"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

3 6

[alsa-devel] [PATCH v3 0/2] topology: Fix for handling widgets' references
by mengdong.lin＠linux.intel.com 25 Jul '16

25 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series fixes an error when merging a widget's private data and an inaccurate output message. History: v2: Split v1 into 2 patches. And set the valid reference element value only on success. v3: Also set valid refernced element ptr when skipping data copy for an empty data reference, which is not an error. Mengdong Lin (2): topology: Fix inaccurate message on failure to find a widgets's reference topology: Fix the missing referenced elem ptr when merging private data src/topology/dapm.c | 2 +- src/topology/data.c | 6 +++++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.5.0

3 4

[alsa-devel] [PATCH] sound: oss: Use kernel_read_file_from_path() for mod_firmware_load()
by Takashi Iwai 25 Jul '16

25 Jul '16

Since recently we have kernel_read_file_from_path(), and it's doing the same thing as our own home-baked mod_firmware_load(). Let's use the official API function and clean up the old code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/Makefile | 1 - sound/oss/sound_firmware.h | 29 ++++++++++++++++- sound/sound_firmware.c | 77 ---------------------------------------------- 3 files changed, 28 insertions(+), 79 deletions(-) delete mode 100644 sound/sound_firmware.c diff --git a/sound/Makefile b/sound/Makefile index 77320709fd26..c41bdf5fdf24 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -2,7 +2,6 @@ # obj-$(CONFIG_SOUND) += soundcore.o -obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ diff --git a/sound/oss/sound_firmware.h b/sound/oss/sound_firmware.h index 0a0cbfdfb855..da4c67e005ed 100644 --- a/sound/oss/sound_firmware.h +++ b/sound/oss/sound_firmware.h @@ -1,2 +1,29 @@ -extern int mod_firmware_load(const char *fn, char **fp); +#include <linux/fs.h> +/** + * mod_firmware_load - load sound driver firmware + * @fn: filename + * @fp: return for the buffer. + * + * Load the firmware for a sound module (up to 128K) into a buffer. + * The buffer is returned in *fp. It is allocated with vmalloc so is + * virtually linear and not DMAable. The caller should free it with + * vfree when finished. + * + * The length of the buffer is returned on a successful load, the + * value zero on a failure. + * + * Caution: This API is not recommended. Firmware should be loaded via + * request_firmware. + */ +static inline int mod_firmware_load(const char *fn, char **fp) +{ + loff_t size; + int err; + + err = kernel_read_file_from_path((char *)fn, (void **)fp, &size, + 131072, READING_FIRMWARE); + if (err < 0) + return 0; + return size; +} diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c deleted file mode 100644 index 026347643c81..000000000000 --- a/sound/sound_firmware.c +++ /dev/null @@ -1,77 +0,0 @@ -#include <linux/vmalloc.h> -#include <linux/module.h> -#include <linux/fs.h> -#include <linux/file.h> -#include <linux/mm.h> -#include <linux/sched.h> -#include <asm/uaccess.h> -#include "oss/sound_firmware.h" - -static int do_mod_firmware_load(const char *fn, char **fp) -{ - struct file* filp; - long l; - char *dp; - - filp = filp_open(fn, 0, 0); - if (IS_ERR(filp)) - { - printk(KERN_INFO "Unable to load '%s'.\n", fn); - return 0; - } - l = i_size_read(file_inode(filp)); - if (l <= 0 || l > 131072) - { - printk(KERN_INFO "Invalid firmware '%s'\n", fn); - fput(filp); - return 0; - } - dp = vmalloc(l); - if (dp == NULL) - { - printk(KERN_INFO "Out of memory loading '%s'.\n", fn); - fput(filp); - return 0; - } - if (kernel_read(filp, 0, dp, l) != l) - { - printk(KERN_INFO "Failed to read '%s'.\n", fn); - vfree(dp); - fput(filp); - return 0; - } - fput(filp); - *fp = dp; - return (int) l; -} - -/** - * mod_firmware_load - load sound driver firmware - * @fn: filename - * @fp: return for the buffer. - * - * Load the firmware for a sound module (up to 128K) into a buffer. - * The buffer is returned in *fp. It is allocated with vmalloc so is - * virtually linear and not DMAable. The caller should free it with - * vfree when finished. - * - * The length of the buffer is returned on a successful load, the - * value zero on a failure. - * - * Caution: This API is not recommended. Firmware should be loaded via - * request_firmware. - */ - -int mod_firmware_load(const char *fn, char **fp) -{ - int r; - mm_segment_t fs = get_fs(); - - set_fs(get_ds()); - r = do_mod_firmware_load(fn, fp); - set_fs(fs); - return r; -} -EXPORT_SYMBOL(mod_firmware_load); - -MODULE_LICENSE("GPL"); -- 2.9.2

1 0

[alsa-devel] M-Audio Fast track problem
by Yann Collette 25 Jul '16

25 Jul '16

Hello, I've got a M-Audio fast track. If I open alsamixer and select the fast track sound card, alsamixer just displays the message "This device has no command" Is it a bug or a normal behavior ? When I dig into google for this problem I feel iti's more a problem than a normal behavior. The 'lsusb' part for Fasttrack: Bus 003 Device 003: ID 0763:2024 Midiman M-Audio Fast Track MKII

3 5

Re: [alsa-devel] [PATCH 3/3] ASoC: topology: Able to create BE DAIs
by Lin, Mengdong 25 Jul '16

25 Jul '16

Sorry for the mistake. I'll fix this warning. Thanks Mengdong > -----Original Message----- > From: Wu, Fengguang > Sent: Friday, July 22, 2016 8:48 PM > To: mengdong.lin(a)linux.intel.com > Cc: kbuild-all(a)01.org; alsa-devel(a)alsa-project.org; broonie(a)kernel.org; > Mengdong Lin; tiwai(a)suse.de; Shah, Hardik T; Singh, Guneshwor O; > liam.r.girdwood(a)linux.intel.com; Lin, Mengdong > Subject: Re: [PATCH 3/3] ASoC: topology: Able to create BE DAIs > > Hi Mengdong, > > [auto build test WARNING on asoc/for-next] [also build test WARNING on > v4.7-rc7] > > url: > https://github.com/0day-ci/linux/commits/mengdong-lin-linux-intel-com/AS > oC-topology-Add-support-for-BE-DAIs/20160722-160125 > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git > for-next > config: x86_64-randconfig-s0-07221711 (attached as .config) > compiler: gcc-4.4 (Debian 4.4.7-8) 4.4.7 > reproduce: > # save the attached .config to linux build tree > make ARCH=x86_64 > > All warnings (new ones prefixed by >>): > > sound/soc/soc-topology.c: In function 'soc_tplg_be_dai_config': > >> sound/soc/soc-topology.c:1790: warning: too many arguments for > format > At top level: > cc1: warning: unrecognized command line option > "-Wno-maybe-uninitialized" > > git remote add linux-review https://github.com/0day-ci/linux git remote > update linux-review git checkout > 234d57f817caf65062dd97dcd7e44be8f5e721e3 > vim +1790 sound/soc/soc-topology.c > > 234d57f8 Mengdong Lin 2016-07-22 1774 * soc_tplg_be_dai_config - > Create a new BE DAI or configure an existing one. > 234d57f8 Mengdong Lin 2016-07-22 1775 * @tplg: topology context > 234d57f8 Mengdong Lin 2016-07-22 1776 * @be: topology BE DAI > configs. > 234d57f8 Mengdong Lin 2016-07-22 1777 * > 234d57f8 Mengdong Lin 2016-07-22 1778 * The BE dai should already > be registered by the platform driver. The > 234d57f8 Mengdong Lin 2016-07-22 1779 * platform driver should > specify the BE DAI name and ID for matching. > 234d57f8 Mengdong Lin 2016-07-22 1780 */ > 234d57f8 Mengdong Lin 2016-07-22 1781 static int > soc_tplg_be_dai_config(struct soc_tplg *tplg, > 234d57f8 Mengdong Lin 2016-07-22 1782 struct > snd_soc_tplg_be_dai *be) > 234d57f8 Mengdong Lin 2016-07-22 1783 { > 234d57f8 Mengdong Lin 2016-07-22 1784 struct > snd_soc_dai_link_component dai_component = {0}; > 234d57f8 Mengdong Lin 2016-07-22 1785 struct snd_soc_dai *dai; > 234d57f8 Mengdong Lin 2016-07-22 1786 struct snd_soc_dai_driver > *dai_drv; > 234d57f8 Mengdong Lin 2016-07-22 1787 > 234d57f8 Mengdong Lin 2016-07-22 1788 if (!strlen(be->dai_name)) { > 234d57f8 Mengdong Lin 2016-07-22 1789 dev_err(tplg->dev, > "ASoC: Invalid BE DAI name\n", > 234d57f8 Mengdong Lin 2016-07-22 @1790 be->dai_name); > 234d57f8 Mengdong Lin 2016-07-22 1791 return -EINVAL; > 234d57f8 Mengdong Lin 2016-07-22 1792 } > 234d57f8 Mengdong Lin 2016-07-22 1793 > 234d57f8 Mengdong Lin 2016-07-22 1794 dai_component.dai_name = > be->dai_name; > 234d57f8 Mengdong Lin 2016-07-22 1795 dai = > snd_soc_find_dai(&dai_component); > 234d57f8 Mengdong Lin 2016-07-22 1796 if (!dai) /* BE DAI doesn't > exist, create it */ > 234d57f8 Mengdong Lin 2016-07-22 1797 return > soc_tplg_be_dai_create(tplg, be); > 234d57f8 Mengdong Lin 2016-07-22 1798 > > --- > 0-DAY kernel test infrastructure Open Source Technology > Center > https://lists.01.org/pipermail/kbuild-all Intel > Corporation

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix NULL Pointer exception in dynamic_debug.
by vedang.patel＠intel.com 25 Jul '16

25 Jul '16

From: Vedang Patel <vedang.patel(a)intel.com> The following bug was reported by sometime back: https://lkml.org/lkml/2016/6/29/795 This commit fixes this bug by setting value for the prefix string. Signed-off-by: Vedang Patel <vedang.patel(a)intel.com> --- sound/soc/intel/skylake/skl-sst-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index c141f24cae05..96f2f6889b18 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -692,7 +692,7 @@ int skl_ipc_init_instance(struct sst_generic_ipc *ipc, /* param_block_size must be in dwords */ u16 param_block_size = msg->param_data_size / sizeof(u32); - print_hex_dump_debug(NULL, DUMP_PREFIX_NONE, + print_hex_dump_debug("Param data:", DUMP_PREFIX_NONE, 16, 4, buffer, param_block_size, false); header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); -- 2.6.6

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Delete an unnecessary check before the function call "release_firmware"
by SF Markus Elfring 25 Jul '16

25 Jul '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 22 Jul 2016 18:58:14 +0200 The release_firmware() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/intel/skylake/skl.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 2337748..cd59536 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -781,8 +781,7 @@ static void skl_remove(struct pci_dev *pci) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); - if (skl->tplg) - release_firmware(skl->tplg); + release_firmware(skl->tplg); if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); -- 2.9.2

2 1

[alsa-devel] duplicate timestamps when capturing from multiple sources with a single gst-launch-0.1 command (2 pipelines in a single command)
by Przemek Gajos 22 Jul '16

22 Jul '16

Hello, I hope this is the right place to describe this problem. I am an ALSA driver developer I have encountered an issue of duplicate timestamps when simultaneously capturing from multiple alsa sources with gst-launch-0.1. Capture command: gst-launch-0.10 -v alsasrc device=hw:2 num-buffers=5 ! fakesink name=dig alsasrc device=hw:1 num-buffers=5 ! audiorate ! fakesink name=ANALOG | grep last-message | grep timestamp These are the initial messages from gst-launch-0.1 app (note that first and second timestamp for the ANALOG source is 0): /GstPipeline:pipeline0/GstFakeSink:ANALOG: last-message = "chain ******* (ANALOG:sink) (32768 bytes, timestamp: 0:00:00.000000000, duration: 0:00:00.185759637, offset: 0, offset_end: 8192, flags: 160 discont gap ) 0x1a89f20" /GstPipeline:pipeline0/GstFakeSink:ANALOG: last-message = "chain ******* (ANALOG:sink) (32768 bytes, timestamp: 0:00:00.000000000, duration: 0:00:00.185759637, offset: 0, offset_end: 8192, flags: 160 discont gap ) 0x1a89f20" /GstPipeline:pipeline0/GstFakeSink:ANALOG: last-message = "chain ******* (ANALOG:sink) (16384 bytes, timestamp: 0:00:00.185759637, duration: 0:00:00.092879818, offset: 8192, offset_end: 12288, flags: 0 ) 0x1a89e80" /GstPipeline:pipeline0/GstFakeSink:dig: last-message = "chain ******* (dig:sink) (16384 bytes, timestamp: 0:00:00.092879818, duration: 0:00:00.092879819, offset: 4096, offset_end: 8192, flags: 0 ) 0x7f6164002dc0" But it does not have to be zero. Sometimes it will have a value though still it will be duplicated. I never see the issue when capturing form a single source. Also, timestamps are CORRECT when splitting the above command into 2 separate commands and trying to run them simultaneously from script (commands run in background are run as close to one another as possible). Therefore, I am thinking: could there be any problems in the layers of ALSA between the driver and an app to set timestamps correctly when the single app does multiple captures from multiple devices (like the first command I wrote at the top)? I am "looking" into this direction because in my driver I see the following calls: 2016-07-22T13:28:49.788041+01:00 ubuntu-12 kernel: [13830.118539] rgb133: In rgb133_mychip_pcm_trigger(1) OEM_VISIONRGB_AV Analog: DMA START, chip->last_ptr(0) 2016-07-22T13:28:49.788043+01:00 ubuntu-12 kernel: [13830.118545] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(0) 2016-07-22T13:28:49.788045+01:00 ubuntu-12 kernel: [13830.118581] rgb133: In rgb133_mychip_pcm_trigger(1) OEM_VISIONRGB_AVHD Dig Node1: DMA START, chip->last_ptr(0) 2016-07-22T13:28:49.788047+01:00 ubuntu-12 kernel: [13830.118587] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(0) 2016-07-22T13:28:49.868018+01:00 ubuntu-12 kernel: [13830.195580] rgb133: -> SndPcmNewDataNotify OEM_VISIONRGB_AV Analog: cur_ptr(16384) 2016-07-22T13:28:49.868031+01:00 ubuntu-12 kernel: [13830.195585] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(16383) 2016-07-22T13:28:49.876011+01:00 ubuntu-12 kernel: [13830.203555] rgb133: -> SndPcmNewDataNotify OEM_VISIONRGB_AVHD Dig Node1: cur_ptr(16384) 2016-07-22T13:28:49.876023+01:00 ubuntu-12 kernel: [13830.203559] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(16383) 2016-07-22T13:28:49.960057+01:00 ubuntu-12 kernel: [13830.287300] rgb133: -> SndPcmNewDataNotify OEM_VISIONRGB_AV Analog: cur_ptr(0) 2016-07-22T13:28:49.960069+01:00 ubuntu-12 kernel: [13830.287304] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(32767) 2016-07-22T13:28:49.960071+01:00 ubuntu-12 kernel: [13830.287336] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(32767) 2016-07-22T13:28:49.960074+01:00 ubuntu-12 kernel: [13830.287342] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(32767) 2016-07-22T13:28:49.960076+01:00 ubuntu-12 kernel: [13830.287395] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(28671) 2016-07-22T13:28:49.960077+01:00 ubuntu-12 kernel: [13830.287409] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(28671) 2016-07-22T13:28:49.960079+01:00 ubuntu-12 kernel: [13830.287457] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(28671) 2016-07-22T13:28:49.960081+01:00 ubuntu-12 kernel: [13830.287464] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(28671) 2016-07-22T13:28:49.960083+01:00 ubuntu-12 kernel: [13830.287514] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(28671) 2016-07-22T13:28:49.968030+01:00 ubuntu-12 kernel: [13830.295282] rgb133: -> SndPcmNewDataNotify OEM_VISIONRGB_AVHD Dig Node1: cur_ptr(0) 2016-07-22T13:28:49.968041+01:00 ubuntu-12 kernel: [13830.295287] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AVHD Dig Node1: return(32767) 2016-07-22T13:28:50.052020+01:00 ubuntu-12 kernel: [13830.379040] rgb133: -> SndPcmNewDataNotify OEM_VISIONRGB_AV Analog: cur_ptr(16384) 2016-07-22T13:28:50.052032+01:00 ubuntu-12 kernel: [13830.379045] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(16383) 2016-07-22T13:28:50.052035+01:00 ubuntu-12 kernel: [13830.379101] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(16383) 2016-07-22T13:28:50.052037+01:00 ubuntu-12 kernel: [13830.379108] rgb133: <- rgb133_mychip_pcm_pointer OEM_VISIONRGB_AV Analog: return(16383) Let me explain: - SndPcmNewDataNotify is my (driver's) notification to the ALSA middle layer that there is new data ready for it (calling snd_pcm_period_elapsed); period size is 16384 bytes. - rgb133_mychip_pcm_pointer is the ALSA layer querying the pointer position in the buffer (I guess this is involved in timestamp creation, correct?). (note that I actually return bytes_to_frames() of what you see above as ALSA expects) Above, I have marked what I think is "shady": I notify ALSA of new data for the ANALOG source (blue), ALSA layer calls back to the driver asking for pointer position for ANALOG (blue), ALSA layer calls back to the driver asking for pointer position for dig source(red) but the driver has not notified ALSA that there is new data for the dig source! In the working case, the call stack in the driver is what I would expect, that is: - SndPcmNewDataNotify for ANALOG - .pointer callback(s) for ANALOG - SndPcmNewDataNotify for dig - .pointer callback(s) for dig - SndPcmNewDataNotify for ANALOG - .pointer callback(s) for ANALOG ... and so on... Thanks a lot for any suggestions, Przemek Gajos

1 0

[alsa-devel] [PATCH 0/4] topology: Add support for BE DAIs
by mengdong.lin＠linux.intel.com 22 Jul '16

22 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series is to work with the kernel series 'ASoC: topology: Add support for BE DAIs' This series adds support for BE (Back End) DAIs in user space: - Define the type and ABI struct for Backend DAIs in UAPI header file. - User can define BE DAIs either by the text conf file or by C API. There is topology ABI udpate in this series. Topology core can check size of each ABI objects to detect version mismatch between user space and kernel. There will be 2 pairs of kernel & user space series for PCM & BE links later. Guneshwor Singh (3): topology: Parse BE DAIs in text conf file topology: Support adding BE DAIs by C API topology: Export BE DAIs to the binary for kernel Mengdong Lin (1): topology: ABI - Add the types for BE DAI include/sound/asoc.h | 34 ++++++- include/topology.h | 44 +++++++++ src/topology/builder.c | 11 +++ src/topology/data.c | 4 + src/topology/elem.c | 4 + src/topology/parser.c | 16 ++++ src/topology/pcm.c | 237 ++++++++++++++++++++++++++++++++++++++++++++++ src/topology/tplg_local.h | 7 ++ 8 files changed, 355 insertions(+), 2 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] topology: Fix the missing referenced elem ptr when merging private data
by mengdong.lin＠linux.intel.com 22 Jul '16

22 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> tplg_copy_data() should set the valid referenced data element pointer. The caller will double check this pointer for all kinds of references, including TLV, controls, text and data. Also fix an inaccurate dmesg. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 4d343b2..e3c90d8 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -201,7 +201,7 @@ static int tplg_build_widget(snd_tplg_t *tplg, } if (!ref->elem) { - SNDERR("error: cannot find control '%s'" + SNDERR("error: cannot find '%s'" " referenced by widget '%s'\n", ref->id, elem->id); return -EINVAL; diff --git a/src/topology/data.c b/src/topology/data.c index 768fc27..397f6a5 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -1050,6 +1050,7 @@ int tplg_copy_data(snd_tplg_t *tplg, struct tplg_elem *elem, " element '%s'\n", ref->id, elem->id); return -EINVAL; } + ref->elem = ref_elem; tplg_dbg("Data '%s' used by '%s'\n", ref->id, elem->id); /* overlook empty private data */ -- 2.5.0

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Re: [alsa-devel] Could I put a patch to 4.7 final?
by Takashi Iwai 21 Jul '16

21 Jul '16

On Thu, 21 Jul 2016 05:12:10 +0200, Takashi Sakamoto wrote: > > Iwai-san, > > Yesterday, I found an user of additional OXFW device, and we're under > testing a patch to support it. > https://sourceforge.net/p/ffado/mailman/message/35229260/ > > I need one day more to post corresponding patch. The prototype is: > https://github.com/takaswie/snd-firewire-improve/commit/690a2419fc95766826c… > > This week is final week for 4.7, but do I have a rest to add the patch > to it in time of your work for mainline? If the patch is a regression fix or a stable fix, I can queue it for 4.7-final. Others will go to 4.8-rc1. I'm going to send a pull request tomorrow, so please submit in time. thanks, Takashi

2 1

[alsa-devel] [PATCH 0/2] ASoC: samsung: Deferred probing support fixes
by Sylwester Nawrocki 21 Jul '16

21 Jul '16

This patch series is an attempt to fix sound card initialization isssues observed in Odroid XU3 board after conversion of the AUDSS clock controller driver to proper loadable module. Related patch can be found here: https://patchwork.kernel.org/patch/9221051, it's also already in -next. It would be good to have these two patches merged in 4.8-rc cycle so as to avoid regressions on on boards using ADMA. The sound breakage happens only when ADMA is used, as ADMA gets apb_pclk clock from the AUDSS clock controller. Previously the clock controller was initialized early in core_initcall which ensured the ADMA PL330 driver initialization before the I2S probe. Probably as a followup the SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME could be dropped and it's usage replaced with struct snd_dmaengine_pcm_config ::dma_names, I didn't want to make too invasive changes for a bug fix patch and don't have the older s3c non-dt boards for testing. Sylwester Nawrocki (2): ASoC: samsung: Fix error paths in the I2S driver's probe() ASoC: samsung: Specify DMA channels through struct snd_dmaengine_pcm_config sound/soc/samsung/ac97.c | 3 ++- sound/soc/samsung/dma.h | 9 ++++++--- sound/soc/samsung/dmaengine.c | 31 ++++++++++++++++++++----------- sound/soc/samsung/i2s.c | 28 ++++++++++++++++++++++------ sound/soc/samsung/pcm.c | 3 ++- sound/soc/samsung/s3c2412-i2s.c | 3 ++- sound/soc/samsung/s3c24xx-i2s.c | 3 ++- sound/soc/samsung/spdif.c | 3 ++- 8 files changed, 58 insertions(+), 25 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: cs53l30: Fix a bug for TDM slot location validation
by Nicolin Chen 21 Jul '16

21 Jul '16

The maximum slot number of CS53L30 is 4 while it should support the situation that's less than 4 channels based on the rx_mask. So when the driver validates the last slot location, it should check the last active slot instead of always the 4th one. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 5988b5c..fd5502e 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -809,8 +809,8 @@ static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai, return -EINVAL; } - /* Validate the last CS53L30 slot */ - slot_next = loc[CS53L30_TDM_SLOT_MAX - 1] + slot_step - 1; + /* Validate the last active CS53L30 slot */ + slot_next = loc[i - 1] + slot_step - 1; if (slot_next > 47) { dev_err(dai->dev, "slot selection out of bounds: %u\n", slot_next); -- 2.1.4

2 3

[alsa-devel] [PATCH v2 0/2] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 21 Jul '16

21 Jul '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting include/sound/hda_register.h | 23 ++++ sound/pci/hda/hda_controller.c | 244 ++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 27 +++++ sound/pci/hda/hda_intel.c | 12 ++ 4 files changed, 305 insertions(+), 1 deletion(-) -- 1.9.1

2 7

[alsa-devel] [PATCH] ASoC: rockchip: correct the spdif clk
by Chris Zhong 20 Jul '16

20 Jul '16

The spdif mclk should be 128 times of sample rate, and there is a internal divider, the real rate of spdif mclk is mclk / (div + 1). Hence, the original driver always get the good frequency for 48000/96000/44100/192000. But for 32000, the mclk is incorrect, it should be 32000*128, but get 48000*128. Do not use the internal divider here, just set all mclk to 128 * sample rate directly. Signed-off-by: Chris Zhong <zyw(a)rock-chips.com> --- sound/soc/rockchip/rockchip_spdif.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 100781e..4ca2657 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -101,21 +101,7 @@ static int rk_spdif_hw_params(struct snd_pcm_substream *substream, int ret; srate = params_rate(params); - switch (srate) { - case 32000: - case 48000: - case 96000: - mclk = 96000 * 128; /* 12288000 hz */ - break; - case 44100: - mclk = 44100 * 256; /* 11289600 hz */ - break; - case 192000: - mclk = 192000 * 128; /* 24576000 hz */ - break; - default: - return -EINVAL; - } + mclk = srate * 128; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -139,7 +125,6 @@ static int rk_spdif_hw_params(struct snd_pcm_substream *substream, return ret; } - val |= SPDIF_CFGR_CLK_DIV(mclk/(srate * 256)); ret = regmap_update_bits(spdif->regmap, SPDIF_CFGR, SPDIF_CFGR_CLK_DIV_MASK | SPDIF_CFGR_HALFWORD_ENABLE | SDPIF_CFGR_VDW_MASK, -- 2.6.3

2 1

[alsa-devel] [PATCH 0/3] ASoC: Intel: Add support for surface3 tablet
by Vinod Koul 20 Jul '16

20 Jul '16

Surface3 tablet is CHT based device which used RT5645 codec. But the ACPI ID is 10EC5640 which belong to RT5640 codec :( So we add quirk support to load cht-machine driver for surface tablet combination, add the ACPI id and DMI fix for jack detect in codec and finally machine update. This fixes Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98001 Please verify and send your tested-by. Vinod Koul (3): ASoC: Intel: Atom: Add quirk for Surface 3 ASoC: rt5645: Add ACPI ID 10EC5640 ASoC: Intel: Add surface3 entry in CHT-RT5645 machine sound/soc/codecs/rt5645.c | 7 ++++++ sound/soc/intel/atom/sst/sst_acpi.c | 44 ++++++++++++++++++++++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 20 ++++++++++++++- sound/soc/intel/common/sst-acpi.h | 2 +- 4 files changed, 70 insertions(+), 3 deletions(-) -- 1.9.1

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[alsa-devel] [alsa-lib][PATCH] ctl: use condition statements instead of assert() for new APIs to add an element set
by Takashi Sakamoto 20 Jul '16

20 Jul '16

Usage of assert() is not better practice of programming as shared library APIs. They should return appropriate error code to promote applications to handle error state. This commit applies condition statements with return value of -EINVAL, instead of assert(). As a backward compatibility for existent applications, old APIs still call assert(). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 24 +++++++++++++++++++----- 1 file changed, 19 insertions(+), 5 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index aa05130..6c00b8e 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -382,7 +382,8 @@ int snd_ctl_add_integer_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int numid; int err; - assert(ctl && info && info->id.name[0]); + if (ctl == NULL || info == NULL || info->id.name[0] == '\0') + return -EINVAL; info->type = SND_CTL_ELEM_TYPE_INTEGER; info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -471,7 +472,8 @@ int snd_ctl_add_integer64_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int numid; int err; - assert(ctl && info && info->id.name[0]); + if (ctl == NULL || info == NULL || info->id.name[0] == '\0') + return -EINVAL; info->type = SND_CTL_ELEM_TYPE_INTEGER64; info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -549,7 +551,8 @@ int snd_ctl_add_boolean_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count) { - assert(ctl && info && info->id.name[0]); + if (ctl == NULL || info == NULL || info->id.name[0] == '\0') + return -EINVAL; info->type = SND_CTL_ELEM_TYPE_BOOLEAN; info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -619,7 +622,9 @@ int snd_ctl_add_enumerated_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, char *buf, *p; int err; - assert(ctl && info && info->id.name[0] && labels); + if (ctl == NULL || info == NULL || info->id.name[0] == '\0' || + labels == NULL) + return -EINVAL; info->type = SND_CTL_ELEM_TYPE_ENUMERATED; info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -698,7 +703,8 @@ int snd_ctl_add_bytes_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count) { - assert(ctl && info && info->id.name[0]); + if (ctl == NULL || info == NULL || info->id.name[0] == '\0') + return -EINVAL; info->type = SND_CTL_ELEM_TYPE_BYTES; info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -726,6 +732,8 @@ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, { snd_ctl_elem_info_t info = {0}; + assert(ctl && id && id->name[0]); + info.id = *id; return snd_ctl_add_integer_elem_set(ctl, &info, 1, member_count, @@ -745,6 +753,8 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, { snd_ctl_elem_info_t info = {0}; + assert(ctl && id && id->name[0]); + info.id = *id; return snd_ctl_add_integer64_elem_set(ctl, &info, 1, member_count, @@ -763,6 +773,8 @@ int snd_ctl_elem_add_boolean(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, { snd_ctl_elem_info_t info = {0}; + assert(ctl && id && id->name[0]); + info.id = *id; return snd_ctl_add_boolean_elem_set(ctl, &info, 1, member_count); @@ -783,6 +795,8 @@ int snd_ctl_elem_add_enumerated(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, { snd_ctl_elem_info_t info = {0}; + assert(ctl && id && id->name[0] && labels); + info.id = *id; return snd_ctl_add_enumerated_elem_set(ctl, &info, 1, member_count, -- 2.7.4

2 1

[alsa-devel] problem with copy function
by Shahram Shabpareh 20 Jul '16

20 Jul '16

Hi I have a board with ti am3358 processor. it has a sound port called McASP and I've used IDT821034 as codec. The Linux kernel source package that is included by board company is 3.2.0 and I wrote a basic driver for IDT821034 codec and changed the code of davinci-mcasp.c . Now the basic functionality of system is ok and i have signals on oscilloscope. Another thing is i have to setup a specific configuration for this system. the target system must have 32 tdm slots in each frame and 32 PCM channels and each slot is 8 bits. this is also done in kernel source code. currently the problem is i want that data of channel 1 be mapped to slot 1, channel 2 be mapped to slot 2, .... and channel 32 be mapped to slot 32. but when i run aplay to send to only channel 1, data will be send to all slots! i want that all slots be 0 except slot 1. How can i do that? I searched in source tree and found the function that i need to override this behaviour is copy that is defined in snd_pcm_ops structure. I defined a function called davinci_pcm_copy in davinci-pcm.c file and added the function to structure like below: static struct snd_pcm_ops davinci_pcm_ops = { .... .copy = davinci_pcm_copy, }; the problem is, this copy function is not called at all by kernel pcm subsystem. Again i searched the kernel code and found that copy function has a calling path as below: alsa-lib -> snd_pcm_playback_ioctl -> snd_pcm_playback_ioctl1 -> snd_pcm_lib_write -> snd_pcm_lib_write1 -> snd_pcm_lib_write_transfer -> copy but my calling path is: alsa-lib -> snd_pcm_playback_ioctl -> snd_pcm_playback_ioctl1 -> snd_pcm_common_ioctl1 -> snd_pcm_sync_ptr I created a wav file from a mp3 file with this command: ffmpeg -i file.mp3 -ac 1 -ar 8000 -ab 64k -f u8 file.wav and played with this command: aplay -vv -c 1 file.wav Strange thing is when i run this aplay command in my PC the first calling path is executed but when i run the same aplay with same file in my board the second calling path is executed. Is it related to software and hardware params of sound devices? if yes, what should i do to copy function being called in my board? thanks

2 2

[alsa-devel] [PATCH v2 0/3] Remove support for SNDRV_CTL_ELEM_ACCESS_USER & minor fix
by mengdong.lin＠linux.intel.com 20 Jul '16

20 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. Remove support for control flag SNDRV_CTL_ELEM_ACCESS_USER as Takashi Sakamoto suggested. Also fix 2 compiler warnings and add check before string cpy. History: v2: Revise commit messages. Mengdong Lin (3): topology: Remove support for control flag SNDRV_CTL_ELEM_ACCESS_USER topology: Add ATTRIBUTE_UNUSED for unused parameters to fix compiler warning topology: Check address and length before string copy src/topology/ctl.c | 1 - src/topology/pcm.c | 8 ++++---- src/topology/tplg_local.h | 3 +++ 3 files changed, 7 insertions(+), 5 deletions(-) -- 2.5.0

4 6

[alsa-devel] [PATCH] alsabat: make snr_is_valid static
by Vinod Koul 20 Jul '16

20 Jul '16

The compilation fails due to multiple defination of snr_is_valid common.o: In function `snr_is_valid': bat/common.h:99: multiple definition of `snr_is_valid' bat.o:bat/common.h:99: first defined here signal.o: In function `snr_is_valid': bat/common.h:99: multiple definition of `snr_is_valid' bat.o:bat/common.h:99: first defined here latencytest.o: In function `snr_is_valid': bat/common.h:99: multiple definition of `snr_is_valid' bat.o:bat/common.h:99: first defined here analyze.o: In function `snr_is_valid': bat/common.h:99: multiple definition of `snr_is_valid' bat.o:bat/common.h:99: first defined here alsa.o: In function `snr_is_valid': bat/common.h:99: multiple definition of `snr_is_valid' bat.o:bat/common.h:99: first defined here Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- bat/common.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/bat/common.h b/bat/common.h index db49868a419a..1b07fbeb7755 100644 --- a/bat/common.h +++ b/bat/common.h @@ -94,7 +94,7 @@ #define SNR_DB_MIN 0.0 #define SNR_DB_MAX 200.0 -inline bool snr_is_valid(float db) +static inline bool snr_is_valid(float db) { return (db > SNR_DB_MIN && db < SNR_DB_MAX); } -- 1.9.1

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Re: [alsa-devel] [Question][ASoC][Topology] Question about ASoC Topology
by Mengdong Lin 20 Jul '16

20 Jul '16

Hi Takashi, Thanks for your explanation on how topology works! I cc the ML since potential users may also benefit from this info. Hi Lee, Now Intel SKL (Skylake) audio driver is using topology, which might be a reference for you. For the text conf file in user space, please see alsa-lib/src/conf/topology/sklrt286/skl_i2s.conf. The topology user space tool alsatplg will generate the SKL topology binary file for kernel. The doxygen doc also introduces the syntax. Besides the text conf file, you may also define topology objects by C API, declared in alsa-lib/include/topology.h For how the SKL platform driver loads the toplogy binary file, please see the kernel src code: sound/soc/intel/skylake/skl-topology.c, where snd_soc_tplg_component_load() is called to load the topology info and trigger ops binding in kernel. Now the upstreamed topology code support: controls, widgets, routes and PCM (Front-ends DAI & DAI links). And now we're upstreaming support for Back-end DAIs and BE links. This may be bring some ABI update and last for several weeks. You may try using topology atm, but you'll need to rebuild your topology binary info after the ABI update. I'll keep you updated when it's done. If you have any further questions, please let us know. Thanks Mengdong On 07/18/2016 03:19 PM, Takashi Sakamoto wrote: > Hi, > > (C.C.ed to Intel developers, who committed to ALSA topology feature.) > > On Jul 18 2016 14:07, 이경택 wrote: >> At first, sorry about interrupting you and thank you for giving your >> attention to an email from stranger. > > No worry. > >> I'm a software engineer in Samsung and writing a device driver for >> our audio. >> >> In recent kernel source code(version 4.4), I found out there is a new >> feature called topology. >> >> I thought that it is a framework for seperating data and logic of audio >> drivers, >> >> so I wanted to write my driver on that framework. >> >> But, it is hard to find out a starting point. (example, documentation, >> ......) >> >> If you give me an starting point, it would be very helpful to me. > > The feature of 'topology' was originally developed by Intel developers. > So I think it better to ask your questions to them. > > > As long as I know, the original goal of this feature is to write > parsable blobs for new HDA driver to PCH in Skylake architecture > (Skylake, Kabylake, etc). > > The new HDA driver (snd-soc-skl) doesn't directly enumerate widgets in > HDA codecs at all. Instead, the parsable blob represents the codec > internal. As a result of parsing the blob, some control elements, some > DAI links and so on are added. When you're interested in the functions > to add them, please start at soc_tplg_load_header(). > http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/soc/… > > In kernel driver side, the blobs are loaded by usual way; > request_firmware(). > https://www.kernel.org/doc/Documentation/firmware_class/README > > Then, each kernel drivers parse the image by snd_soc_tplg_component_load(). > http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/soc/… > > Of cource, you need to put the blobs to typical file path; /lib/firmware. > > > When writing files to generate the blob, you need to check two items; > one is UAPI of ASoC (include/uapi/sound/asoc.h) and another is runtime > configuration of ALSA userspace library. > > What you can do via topology feature is defined in the UAPI header. The > way to describe them to files is the same way as runtime configuration > of ALSA userspace library. > http://www.alsa-project.org/alsa-doc/alsa-lib/conf.html > > When you write the files, you can generate the blobs by 'alsatplg' utils > in alsa-utils.git. I believe it can detect parse errors. > http://git.alsa-project.org/?p=alsa-utils.git;a=tree;f=topology > > That's all of what I know. I hope this will help you or you can get > enough support about the new feature from Intel developers. > > p.s. I'm committing to ALSA in my private time, so I sometimes need to > stop discussions because of my limited time. I'm not a developer belongs > to semi-conductors or hardware vendors and have paid work for my life. > > > Regards > > Takashi Sakamoto >

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Re: [alsa-devel] [PATCH 3/3] topology: Check address and length before string copy
by Lin, Mengdong 19 Jul '16

19 Jul '16

> -----Original Message----- > From: Takashi Sakamoto [mailto:o-takashi@sakamocchi.jp] > Sent: Tuesday, July 19, 2016 5:45 PM > To: mengdong.lin(a)linux.intel.com; alsa-devel(a)alsa-project.org; > broonie(a)kernel.org > Cc: tiwai(a)suse.de; Girdwood, Liam R; Lin, Mengdong > Subject: Re: [PATCH 3/3] topology: Check address and length before string > copy > > Hi, > > On Jul 19 2016 17:52, mengdong.lin(a)linux.intel.com wrote: > > From: Mengdong Lin <mengdong.lin(a)linux.intel.com> > > > > Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> > > The cover letter addresses this patch fixes a compiler warning. In this case, > it's better to write the message in commit message, I think. Like: > http://mailman.alsa-project.org/pipermail/alsa-devel/2016-July/110097.ht > ml > > This is not just from my taste. Maintainers prefer. At least, blank commit > messages are not preferrable in general. Yes, that would be better. I'll fix this in v2 tomorrow. Thanks Mengdong > > > diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h > > index 4d79aa7..cfde4cc 100644 > > --- a/src/topology/tplg_local.h > > +++ b/src/topology/tplg_local.h > > @@ -253,6 +253,9 @@ struct tplg_elem* > tplg_elem_new_common(snd_tplg_t > > *tplg, > > > > static inline void elem_copy_text(char *dest, const char *src, int len) > > { > > + if (!dest || !src || !len) > > + return; > > + > > strncpy(dest, src, len); > > dest[len - 1] = 0; > > } > > > Regards > > Takashi Sakamoto

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[alsa-devel] [PATCH 0/3] Remove support for SNDRV_CTL_ELEM_ACCESS_USER & minor fix
by mengdong.lin＠linux.intel.com 19 Jul '16

19 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. Remove support for control flag SNDRV_CTL_ELEM_ACCESS_USER as Takashi Sakamoto suggested. Also fix 2 compiler warnings and add check before string cpy. Mengdong Lin (3): topology: Remove support for control flag SNDRV_CTL_ELEM_ACCESS_USER topology: Fix compiler warnings topology: Check address and length before string copy src/topology/ctl.c | 1 - src/topology/pcm.c | 8 ++++---- src/topology/tplg_local.h | 3 +++ 3 files changed, 7 insertions(+), 5 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v2 0/5] topology: Enhance support for private data
by mengdong.lin＠linux.intel.com 19 Jul '16

19 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. We finished verification on latest Intel platforms. There will be 4 or 5 series. Later series will come with some ABI update for topology. The size check on ABI objects in current ASoC kernel will detect possible ABI version mismatch between user space and kernel History: v2: fix compiler warnings. There are 2 warnings left in topology/pcm.c that are not caused by this series and will be fixed in later patches. Mengdong Lin (5): topology: An element can refer to multipe data sections in text conf file topology: Merge lookup for data reference into tplg_copy_data() topology: Change uuid value to 16 separate characters in text conf file topology: Parse vendor private data for manifest topology: Tuple type can have an extenstion include/topology.h | 69 ++++++++++- src/topology/ctl.c | 52 +++----- src/topology/dapm.c | 17 ++- src/topology/data.c | 307 ++++++++++++++++++++++++++++++++++++++-------- src/topology/elem.c | 4 + src/topology/parser.c | 28 ++++- src/topology/tplg_local.h | 15 ++- 7 files changed, 395 insertions(+), 97 deletions(-) -- 2.5.0

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[alsa-devel] Alsa Soc details
by Chemsi Mehdi 19 Jul '16

19 Jul '16

Dear expert, I have some ALSA questions please: - Why codec driver of ASOC are always I2C based ?, is codec driver role is only control of audio hardware(set/reset registers only)? - There should be three drivers : machine driver, platform driver and codec driver , all of them have .probe and . remove , what is relation between all the probes? removes ? - As far I know McBSP is source of audio coming from host and I2C/ McBSP have no direct relation so why we connect platform driver (McBSP) with codec driver(I2C) Via DAI ? - When calling snd_pcm_open from userspace which kernel API is called ? snd_pcm_lib_ioctl or snd_stm_spdif_player_open ? if snd_stm_spdif_player_open so when called snd_pcm_lib_ioctl ? - How ALSA KERNELAPI is connected to ALSA SOC CORE, via which calls/mechanism ? Thanks q lot for your great help LINI

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix quirks code is not called
by Kazuki Oikawa 18 Jul '16

18 Jul '16

snd_usb_{set_interface,ctl_msg}_quirk checks chip->usb_id to need calling a quirks code. But existed code path that not calling dev_set_drvdata in usb_audio_probe. Signed-off-by: Kazuki Oikawa <k(a)oikw.org> Fixes: 79289e24194a ("ALSA: usb-audio: Refer to chip->usb_id for quirks and MIDI creation") --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 3fc63583a537..2d493501b7f6 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -552,7 +552,6 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } chip = usb_chip[i]; - dev_set_drvdata(&dev->dev, chip); atomic_inc(&chip->active); /* avoid autopm */ break; } @@ -578,6 +577,7 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } } + dev_set_drvdata(&dev->dev, chip); /* * For devices with more than one control interface, we assume the -- 2.9.2

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[alsa-devel] [PATCH v2] ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members
by Takashi Sakamoto 17 Jul '16

17 Jul '16

Currently, sound device drivers for PCI cards produced by Echo Audio support dimension parameter of element information. But the information has contradictions to the number of members of each element. I guess that this comes from the assumption that these sound cards are used only by 'echomixer' in userspace. But ideally, they should be used with usual ALSA control applications. This commit removes the contradiction. As a result, 'Monitor Mixer Volume' and 'VMixer Volume' elements are shown in usual ALSA control applications such as 'amixer' and 'alsamixer' in series. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/echoaudio/echoaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1cb85ae..3a8e8d5 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_busses_in(chip); + uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_pipes_out(chip); + uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1728,7 +1728,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = 0; #ifdef ECHOCARD_HAS_VMIXER @@ -1738,6 +1737,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, #endif uinfo->dimen.d[1] = 16; /* 16 channels */ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */ + uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2]; return 0; } -- 2.7.4

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[alsa-devel] [alsa-lib][PATCH] ctl: rename APIs to add an element set so that their names are appropriate
by Takashi Sakamoto 17 Jul '16

17 Jul '16

In this development period, some APIs have already been added. Most of them are to add an element set, while their names are not so appropriate against their feature. This is due to following to names of old APIs. This commit renames them. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/control.h | 10 +++++----- src/control/control.c | 34 +++++++++++++++++----------------- src/pcm/pcm_softvol.c | 4 ++-- test/user-ctl-element-set.c | 10 +++++----- 4 files changed, 29 insertions(+), 29 deletions(-) diff --git a/include/control.h b/include/control.h index cd60d90..1d4bfcb 100644 --- a/include/control.h +++ b/include/control.h @@ -425,24 +425,24 @@ void snd_ctl_elem_info_set_subdevice(snd_ctl_elem_info_t *obj, unsigned int val) void snd_ctl_elem_info_set_name(snd_ctl_elem_info_t *obj, const char *val); void snd_ctl_elem_info_set_index(snd_ctl_elem_info_t *obj, unsigned int val); -int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_integer_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count, long min, long max, long step); -int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_integer64_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count, long long min, long long max, long long step); -int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_boolean_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count); -int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_enumerated_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count, unsigned int items, const char *const labels[]); -int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_bytes_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count); diff --git a/src/control/control.c b/src/control/control.c index a48ba9d..aa05130 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -371,7 +371,7 @@ static bool validate_element_member_dimension(snd_ctl_elem_info_t *info) * \par Compatibility: * This function is added in version 1.1.2. */ -int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_integer_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count, long min, long max, long step) @@ -460,7 +460,7 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * \par Compatibility: * This function is added in version 1.1.2. */ -int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_integer64_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count, long long min, long long max, long long step) @@ -545,7 +545,7 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * \par Compatibility: * This function is added in version 1.1.2. */ -int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_boolean_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count) { @@ -609,7 +609,7 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * \par Compatibility: * This function is added in version 1.1.2. */ -int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_enumerated_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count, unsigned int items, @@ -694,7 +694,7 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * \par Compatibility: * This function is added in version 1.1.2. */ -int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, +int snd_ctl_add_bytes_elem_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, unsigned int element_count, unsigned int member_count) { @@ -716,9 +716,9 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, /** * \brief Create and add an user-defined control element of integer type. * - * This is a wrapper function to snd_ctl_elem_add_integer_set() for a control + * This is a wrapper function to snd_ctl_add_integer_elem_set() for a control * element. This doesn't fill the id data with full information, thus it's - * recommended to use snd_ctl_elem_add_integer_set(), instead. + * recommended to use snd_ctl_add_integer_elem_set(), instead. */ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, @@ -728,16 +728,16 @@ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, info.id = *id; - return snd_ctl_elem_add_integer_set(ctl, &info, 1, member_count, + return snd_ctl_add_integer_elem_set(ctl, &info, 1, member_count, min, max, step); } /** * \brief Create and add an user-defined control element of integer64 type. * - * This is a wrapper function to snd_ctl_elem_add_integer64_set() for a single + * This is a wrapper function to snd_ctl_add_integer64_elem_set() for a single * control element. This doesn't fill the id data with full information, thus - * it's recommended to use snd_ctl_elem_add_integer64_set(), instead. + * it's recommended to use snd_ctl_add_integer64_elem_set(), instead. */ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, @@ -747,16 +747,16 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, info.id = *id; - return snd_ctl_elem_add_integer64_set(ctl, &info, 1, member_count, + return snd_ctl_add_integer64_elem_set(ctl, &info, 1, member_count, min, max, step); } /** * \brief Create and add an user-defined control element of boolean type. * - * This is a wrapper function to snd_ctl_elem_add_boolean_set() for a single + * This is a wrapper function to snd_ctl_add_boolean_elem_set() for a single * control element. This doesn't fill the id data with full information, thus - * it's recommended to use snd_ctl_elem_add_boolean_set(), instead. + * it's recommended to use snd_ctl_add_boolean_elem_set(), instead. */ int snd_ctl_elem_add_boolean(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count) @@ -765,15 +765,15 @@ int snd_ctl_elem_add_boolean(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, info.id = *id; - return snd_ctl_elem_add_boolean_set(ctl, &info, 1, member_count); + return snd_ctl_add_boolean_elem_set(ctl, &info, 1, member_count); } /** * \brief Create and add a user-defined control element of enumerated type. * - * This is a wrapper function to snd_ctl_elem_add_enumerated_set() for a single + * This is a wrapper function to snd_ctl_add_enumerated_elem_set() for a single * control element. This doesn't fill the id data with full information, thus - * it's recommended to use snd_ctl_elem_add_enumerated_set(), instead. + * it's recommended to use snd_ctl_add_enumerated_elem_set(), instead. * * This function is added in version 1.0.25. */ @@ -785,7 +785,7 @@ int snd_ctl_elem_add_enumerated(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, info.id = *id; - return snd_ctl_elem_add_enumerated_set(ctl, &info, 1, member_count, + return snd_ctl_add_enumerated_elem_set(ctl, &info, 1, member_count, items, labels); } diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 6135778..1fe5784 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -671,9 +671,9 @@ static int add_user_ctl(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo, unsigned int def_val; if (svol->max_val == 1) - err = snd_ctl_elem_add_boolean_set(svol->ctl, cinfo, 1, count); + err = snd_ctl_add_boolean_elem_set(svol->ctl, cinfo, 1, count); else - err = snd_ctl_elem_add_integer_set(svol->ctl, cinfo, 1, count, + err = snd_ctl_add_integer_elem_set(svol->ctl, cinfo, 1, count, 0, svol->max_val, 0); if (err < 0) return err; diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 0481300..9b9dc59 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -31,7 +31,7 @@ struct elem_set_trial { static int add_bool_elem_set(struct elem_set_trial *trial, snd_ctl_elem_info_t *info) { - return snd_ctl_elem_add_boolean_set(trial->handle, info, + return snd_ctl_add_boolean_elem_set(trial->handle, info, trial->element_count, trial->member_count); } @@ -51,7 +51,7 @@ static void change_bool_elem_members(struct elem_set_trial *trial, static int add_int_elem_set(struct elem_set_trial *trial, snd_ctl_elem_info_t *info) { - return snd_ctl_elem_add_integer_set(trial->handle, info, + return snd_ctl_add_integer_elem_set(trial->handle, info, trial->element_count, trial->member_count, 0, 99, 1); } @@ -93,7 +93,7 @@ static const char *const labels[] = { static int add_enum_elem_set(struct elem_set_trial *trial, snd_ctl_elem_info_t *info) { - return snd_ctl_elem_add_enumerated_set(trial->handle, info, + return snd_ctl_add_enumerated_elem_set(trial->handle, info, trial->element_count, trial->member_count, sizeof(labels) / sizeof(labels[0]), labels); @@ -142,7 +142,7 @@ static void change_enum_elem_members(struct elem_set_trial *trial, static int add_bytes_elem_set(struct elem_set_trial *trial, snd_ctl_elem_info_t *info) { - return snd_ctl_elem_add_bytes_set(trial->handle, info, + return snd_ctl_add_bytes_elem_set(trial->handle, info, trial->element_count, trial->member_count); } @@ -195,7 +195,7 @@ static void change_iec958_elem_members(struct elem_set_trial *trial, static int add_int64_elem_set(struct elem_set_trial *trial, snd_ctl_elem_info_t *info) { - return snd_ctl_elem_add_integer64_set(trial->handle, info, + return snd_ctl_add_integer64_elem_set(trial->handle, info, trial->element_count, trial->member_count, 100, 10000, 30); } -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] ASoC: add driver for Atmel I2S controller
by Cyrille Pitchen 17 Jul '16

17 Jul '16

This series of patches adds support to the new Atmel I2S controller embedded on sama5d2 SoCs. ChangeLog v2 -> v3 - fix the coding style, add some more comments and add a section dedicated to sama5d2 SoCs in the DT binding documentation as suggested by Mark Brown. v1 -> v2 - initialize dev->dev before calling dev->caps->mck_init() Cyrille Pitchen (2): ASoC: atmel-i2s: add DT bindings for I2S controller ASoC: atmel-i2s: add driver for the new Atmel I2S controller .../devicetree/bindings/sound/atmel-i2s.txt | 87 +++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-i2s.c | 770 +++++++++++++++++++++ 4 files changed, 868 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt create mode 100644 sound/soc/atmel/atmel-i2s.c -- 1.8.2.2

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[alsa-devel] [PATCH 0/7] ASoC: simple: dailink_name and card_name
by Kuninori Morimoto 16 Jul '16

16 Jul '16

Hi Mark These are for dailink_name and card_name cleanup for simple-card. 1) - 3) : for dailink_name 4) : tidyup 5) - 7) : for card_name Kuninori Morimoto (7): 1) ASoC: simple-card-utils: add asoc_simple_card_set_dailink_name() 2) ASoC: simple-card: use asoc_simple_card_parse_dailink_name() 3) ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() 4) ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai 5) ASoC: simple-card-utils: add asoc_simple_card_parse_card_name() 6) ASoC: simple-card: use asoc_simple_card_parse_card_name() 7) ASoC: rsrc-card: use asoc_simple_card_parse_card_name() include/sound/simple_card_utils.h | 5 +++ sound/soc/generic/simple-card-utils.c | 43 +++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 28 +++++------------ sound/soc/sh/rcar/rsrc-card.c | 59 ++++++++++++++--------------------- 4 files changed, 80 insertions(+), 55 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/7 v2] ASoC: simple: dailink_name and card_name
by Kuninori Morimoto 16 Jul '16

16 Jul '16

Hi Mark These are v2 of dailink_name and card_name cleanup patches for simple-card. 1) - 3) : for dailink_name 4) : tidyup 5) - 7) : for card_name Kuninori Morimoto (7): 1) ASoC: simple-card-utils: add asoc_simple_card_set_dailink_name() 2) ASoC: simple-card: use asoc_simple_card_parse_dailink_name() 3) ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() 4) ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai 5) ASoC: simple-card-utils: add asoc_simple_card_parse_card_name() 6) ASoC: simple-card: use asoc_simple_card_parse_card_name() 7) ASoC: rsrc-card: use asoc_simple_card_parse_card_name() include/sound/simple_card_utils.h | 5 +++ sound/soc/generic/simple-card-utils.c | 43 +++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 28 +++++------------ sound/soc/sh/rcar/rsrc-card.c | 59 ++++++++++++++--------------------- 4 files changed, 80 insertions(+), 55 deletions(-) -- 1.9.1

2 8

[alsa-devel] [PATCH 00/39 v2] ASoC: add simple-card-core and standardize "simple" card
by Kuninori Morimoto 16 Jul '16

16 Jul '16

Hi Mark These are v2 of simple-card cleanup patches. Main purpose of these patches are - cleanup simple-card - rename rsrc-card to simple-dpcm-card Main diff v1 <-> v2 are - remove duplicate header patch - rename simple-card-core.c -> simple-card-utils.c - remove dpcm function from simple-card-utils. - remove jack/pin/gpio/ function from simple-card-utils. - remove widgets/route function from simple-card-utils. I removed widgets/routing related function. it was very thin cosmetic functions. About jack/pin/gpio on simple-card, it is still have same method, but keeping it as static function. dpcm related function too, it is keeping static. 1) - 3) cleaup patch as prepare 4) - 14) simple-card-utils 15) - 24) simple-card 25) - 36) rsrc-card 37) - 39) rsrc-card -> simple-dpcm-card Kuninori Morimoto (39): 1) ASoC: rsrc-card: remove unused dai_num 2) ASoC: simple-card: use common PREFIX for each DT property 3) ASoC: simple-card: add new asoc_simple_jack and use it 4) ASoC: add new simple-card-utils.c 5) ASoC: simple-card-utils: add asoc_simple_card_parse_tdm() 6) ASoC: simple-card-utils: add asoc_simple_card_parse_dailink_name() 7) ASoC: simple-card-utils: add asoc_simple_card_parse_card_name() 8) ASoC: simple-card-utils: add asoc_simple_card_parse_card_prefix() 9) ASoC: simple-card-utils: add asoc_simple_card_parse_clk() 10) ASoC: simple-card-utils: add asoc_simple_card_parse_endpoint() 11) ASoC: simple-card-utils: add asoc_simple_card_init_dai() 12) ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() 13) ASoC: simple-card-utils: add asoc_simple_card_canonicalize_cpu() 14) ASoC: simple-card-utils: add asoc_simple_card_clean_reference() 15) ASoC: simple-card: use asoc_simple_card_parse_daifmt() 16) ASoC: simple-card: use asoc_simple_card_parse_clk() 17) ASoC: simple-card: use asoc_simple_card_parse_endpoint() 18) ASoC: simple-card: use asoc_simple_card_parse_tdm() 19) ASoC: simple-card: use asoc_simple_card_parse_card_name() 20) ASoC: simple-card: use asoc_simple_card_parse_dailink_name() 21) ASoC: simple-card: use asoc_simple_card_init_dai() 22) ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() 23) ASoC: simple-card: use asoc_simple_card_canonicalize_cpu() 24) ASoC: simple-card: use asoc_simple_card_clean_reference() 25) ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() 26) ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() 27) ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai 28) ASoC: rsrc-card: use asoc_simple_card_parse_clk() 29) ASoC: rsrc-card: use asoc_simple_card_parse_endpoint() 30) ASoC: rsrc-card: use asoc_simple_card_parse_card_name() 31) ASoC: rsrc-card: use asoc_simple_card_parse_card_prefix() 32) ASoC: rsrc-card: use asoc_simple_card_parse_tdm() 33) ASoC: rsrc-card: use asoc_simple_card_init_dai() 34) ASoC: rsrc-card: use asoc_simple_card_canonicalize_cpu() 35) ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() 36) ASoC: rsrc-card: use asoc_simple_card_clean_reference() 37) ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase1 38) ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase2 39) ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase3 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 75 -------------- Documentation/devicetree/bindings/sound/simple-dpcm-card.txt | 79 +++++++++++++++ include/sound/simple_card.h | 11 +-- include/sound/simple_card_utils.h | 77 +++++++++++++++ sound/soc/generic/Kconfig | 10 ++ sound/soc/generic/Makefile | 4 + sound/soc/generic/simple-card-utils.c | 288 +++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 424 ++++++++++++++++++++++++------------------------------------------------------ sound/soc/generic/simple-dpcm-card.c | 351 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/Kconfig | 5 - sound/soc/sh/rcar/Makefile | 3 - sound/soc/sh/rcar/rsrc-card.c | 529 -------------------------------------------------------------------------------------------------- 12 files changed, 937 insertions(+), 919 deletions(-)

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[alsa-devel] USB audio (AOAv2) - retire_capture_urb messages
by arv 16 Jul '16

16 Jul '16

Hi All, I am trying to playback audio from Android phone using USB AOAv2. (PCM audio, 16bit, 44.1 KHz, stereo) on a Wandboard Quad (i.MX6) board running Ubuntu 14.04. The detection of phone is fine and playback via alsa commands/gstreamer is successful. But during playback, there are distortions in the sound. dmesg prints as follows, retire_capture_urb: x callbacks suppressed ----->[x is around 41 to 300] retire_capture_urb: x callbacks suppressed retire_capture_urb: x callbacks suppressed retire_capture_urb: x callbacks suppressed arecord command prints underrun occurred. So each time, retire_capture_urb is printed, there is a distortion. I tried the same program on different machines. Following are the results, On Wandboard, the distortions happen each 40 seconds or so. On a Linux server PC running Ubuntu 14.04 the audio breaks are more frequent. On a Linux PC running Ubuntu 12.02 the audio breaks are less frequent. Could anyone guide me on how retire_capture_urb be prevented or how can a smooth playback be achieved ? Should there be any special configurations done at USB driver ? lsusb -v print is as follows, The synch type for the endpoint appears to be enumerated as Synchronous. Could the synch type be modified using libusb, is this the cause of underrun? Bus 001 Device 004: ID 18d1:2d02 Google Inc. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x18d1 Google Inc. idProduct 0x2d02 bcdDevice ff.ff iManufacturer 1 motorola iProduct 2 MotoG3-TE iSerial 3 ZY222WB7TT bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 110 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 40 bInCollection 2 baInterfaceNr( 0) 0 baInterfaceNr( 1) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 2 bSourceID 2 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 1 bControlSize 2 bmaControls( 0) 0x00 bmaControls( 0) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) Thanks for the help..

1 0

[alsa-devel] [PATCH 0/4] Qualcomm IPQ4019 Audio driver addition
by njaigane＠codeaurora.org 15 Jul '16

15 Jul '16

From: Jaiganesh Narayanan <njaigane(a)codeaurora.org> These patches add the support for Qualcomm IPQ4019 ASoC with the ALSA based audio drivers. The patches are broken 1. Device Tree support 2. Audio clock driver support 3. TLMM / Pinctrl support 4. ALSA based audio drivers Jaiganesh Narayanan (4): qcom: ipq4019: Add ipq4019 ASoC device tree changes qcom: ipq4019: ASoC clock driver support qcom: ipq4019: ASoC tlmm/pinctrl support qcom: ipq4019: Add ASoC driver modules .../bindings/sound/qca,ipq4019-audio-adss.txt | 20 + .../bindings/sound/qca,ipq4019-audio.txt | 13 + .../bindings/sound/qca,ipq4019-codec.txt | 15 + .../devicetree/bindings/sound/qca,ipq4019-i2s.txt | 33 + .../devicetree/bindings/sound/qca,ipq4019-i2s1.txt | 29 + .../devicetree/bindings/sound/qca,ipq4019-i2s2.txt | 29 + .../devicetree/bindings/sound/qca,ipq4019-mbox.txt | 23 + .../bindings/sound/qca,ipq4019-pcm-i2s.txt | 15 + .../bindings/sound/qca,ipq4019-pcm-i2s1.txt | 15 + .../bindings/sound/qca,ipq4019-pcm-i2s2.txt | 15 + .../bindings/sound/qca,ipq4019-pcm-spdif.txt | 15 + .../bindings/sound/qca,ipq4019-pcm-tdm.txt | 15 + .../bindings/sound/qca,ipq4019-spdif.txt | 35 + .../bindings/sound/qca,ipq4019-stereo.txt | 17 + .../devicetree/bindings/sound/qca,ipq4019-tdm.txt | 33 + arch/arm/boot/dts/qcom-ipq4019-ap.dk01.1.dtsi | 195 ++++- arch/arm/boot/dts/qcom-ipq4019.dtsi | 247 +++++- drivers/clk/qcom/Kconfig | 8 + drivers/clk/qcom/Makefile | 2 + drivers/clk/qcom/adcc-ipq4019.c | 700 +++++++++++++++++ drivers/clk/qcom/clk-qcapll.c | 197 +++++ drivers/clk/qcom/clk-qcapll.h | 60 ++ drivers/clk/qcom/clk-rcg.h | 68 +- drivers/clk/qcom/clk-rcg2.c | 699 ++++++++++++++++- drivers/clk/qcom/common.c | 9 +- drivers/clk/qcom/common.h | 3 +- drivers/pinctrl/qcom/pinctrl-ipq4019.c | 116 ++- include/dt-bindings/clock/qca,adcc-ipq4019.h | 45 ++ include/dt-bindings/clock/qcom,gcc-ipq4019.h | 4 +- include/dt-bindings/sound/ipq4019-audio.h | 37 + sound/soc/qcom/Kconfig | 47 ++ sound/soc/qcom/Makefile | 1 + sound/soc/qcom/ipq4019/Makefile | 16 + sound/soc/qcom/ipq4019/ipq4019-adss.c | 407 ++++++++++ sound/soc/qcom/ipq4019/ipq4019-adss.h | 432 +++++++++++ sound/soc/qcom/ipq4019/ipq4019-codec.c | 475 ++++++++++++ sound/soc/qcom/ipq4019/ipq4019-codec.h | 91 +++ sound/soc/qcom/ipq4019/ipq4019-cpu-dai.c | 687 +++++++++++++++++ sound/soc/qcom/ipq4019/ipq4019-mbox.c | 825 +++++++++++++++++++++ sound/soc/qcom/ipq4019/ipq4019-mbox.h | 146 ++++ sound/soc/qcom/ipq4019/ipq4019-pcm-i2s.c | 609 +++++++++++++++ sound/soc/qcom/ipq4019/ipq4019-pcm-spdif.c | 664 +++++++++++++++++ sound/soc/qcom/ipq4019/ipq4019-pcm-tdm.c | 609 +++++++++++++++ sound/soc/qcom/ipq4019/ipq4019-pcm.h | 37 + sound/soc/qcom/ipq4019/ipq4019-stereo.c | 313 ++++++++ sound/soc/qcom/ipq4019/ipq4019.c | 121 +++ 46 files changed, 8135 insertions(+), 57 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-audio-adss.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-i2s1.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-i2s2.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-mbox.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-pcm-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-pcm-i2s1.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-pcm-i2s2.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-pcm-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-pcm-tdm.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-stereo.txt create mode 100644 Documentation/devicetree/bindings/sound/qca,ipq4019-tdm.txt create mode 100644 drivers/clk/qcom/adcc-ipq4019.c create mode 100644 drivers/clk/qcom/clk-qcapll.c create mode 100644 drivers/clk/qcom/clk-qcapll.h create mode 100644 include/dt-bindings/clock/qca,adcc-ipq4019.h create mode 100644 include/dt-bindings/sound/ipq4019-audio.h create mode 100644 sound/soc/qcom/ipq4019/Makefile create mode 100644 sound/soc/qcom/ipq4019/ipq4019-adss.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-adss.h create mode 100644 sound/soc/qcom/ipq4019/ipq4019-codec.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-codec.h create mode 100644 sound/soc/qcom/ipq4019/ipq4019-cpu-dai.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-mbox.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-mbox.h create mode 100644 sound/soc/qcom/ipq4019/ipq4019-pcm-i2s.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-pcm-spdif.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-pcm-tdm.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019-pcm.h create mode 100644 sound/soc/qcom/ipq4019/ipq4019-stereo.c create mode 100644 sound/soc/qcom/ipq4019/ipq4019.c -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: nau8825: drop redundant idiom when converting integer to boolean
by John Hsu 15 Jul '16

15 Jul '16

Thanks Mark and Anatol for the discussion. According to the result, the standard C will translate any non-zero value into true, or false otherwise. QUOTE: "6.3.1.2 Boolean type When any scalar value is converted to _Bool, the result is 0 if the value compares equal to 0; otherwise, the result is 1 " Thus, the "!!" idiom is removed. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index a97418d..5c9707a 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1349,9 +1349,9 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) int status, jkdet; regmap_read(regmap, NAU8825_REG_JACK_DET_CTRL, &jkdet); - active_high = !!(jkdet & NAU8825_JACK_POLARITY); + active_high = jkdet & NAU8825_JACK_POLARITY; regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); - is_high = !!(status & NAU8825_GPIO2JD1); + is_high = status & NAU8825_GPIO2JD1; /* return jack connection status according to jack insertion logic * active high or active low. */ -- 2.6.4

2 1

[alsa-devel] [alsa-lib][PATCH 00/29] mixer: replace usage of alloca() with automatic variables
by Takashi Sakamoto 15 Jul '16

15 Jul '16

Hi, A batch of this patchset is to apply the same idea to mixer feature as I posted in this thread. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-July/110196.html In short, replacement of alloca() with automatic variables brings us some advantages; better execution performance and saving local storages just for a little. I can see below effect on my environment (Ubuntu 16.04, amd64) $ git checkout master $ git show HEAD commit 33e946fdd32da2b918e88750fcfd78014ae3e079 ... $ ./gitcompile --with-plugindir=/usr/lib/x86_64-linux-gnu/alsa-lib/ ... $ ls -l src/.libs/libasound.so.2.0.0 -rwxrwxr-x 1 mocchi mocchi 4783080 7月 15 09:08 src/.libs/libasound.so.2.0.0 $ git checkout remove-alloca-mixer $ make ... $ ls -l src/.libs/libasound.so.2.0.0 -rwxrwxr-x 1 mocchi mocchi 4780152 7月 15 09:09 src/.libs/libasound.so.2.0.0 $ gcc -v ... gcc version 5.4.0 20160609 (Ubuntu 5.4.0-6ubuntu1~16.04.1) Takashi Sakamoto (29): mixer: include library local header to find layout of structures mixer: change code formatting for elem_read_volume() mixer: remove alloca() from elem_read_volume() mixer: change code formatting for elem_read_route() mixer: remove alloca() from elem_read_route() mixer: change code formatting for elem_read_enum() mixer: remove alloca() from elem_read_enum() mixer: change code formatting for selem_read() mixer: remove alloca() from selem_read() mixer: change code formatting for elem_write_volume() mixer: remove alloca() from elem_write_volume() mixer: change code formatting for elem_write_switch() mixer: remove alloca() from elem_write_switch() mixer: remove alloca() from elem_write_switch_constant() mixer: change code formatting for elem_write_enum() mixer: remove alloca() from elem_write_enum() mixer: change code formatting for elem_write_route() mixer: remove alloca() from elem_write_route() mixer: change code formatting for selem_write_main() mixer: remove alloca() from selem_write_main() mixer: change code formatting for init_db_range() mixer: remove alloca() from init_db_range() mixer: remove alloca() from enum_item_name_ops() mixer: remove alloca() from get_enum_item_ops() mixer: remove alloca() from set_enum_item_ops() mixer: change code formatting for simple_add1() mixer: remove alloca() from simple_add1() mixer: change code formatting for simple_event_add() mixer: remove alloca() from simple_event_add() src/mixer/simple_none.c | 208 ++++++++++++++++++++++++------------------------ 1 file changed, 106 insertions(+), 102 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH 2/3] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 15 Jul '16

15 Jul '16

On Mon, Jul 11, 2016 at 07:43:25PM +0800, kbuild test robot wrote: > Hi, > > [auto build test ERROR on sound/for-next] > [also build test ERROR on v4.7-rc7 next-20160711] > [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] > > url: https://github.com/0day-ci/linux/commits/Vinod-Koul/ALSA-hda-Add-support-fo… > base: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next > config: arm-multi_v7_defconfig (attached as .config) > compiler: arm-linux-gnueabi-gcc (Debian 5.3.1-8) 5.3.1 20160205 > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=arm > > All errors (new ones prefixed by >>): > > >> sound/pci/hda/hda_controller.c:30:21: fatal error: asm/tsc.h: No such file or directory > compilation terminated. Okay i think I need to move this bit into the intel code. Takashi, I didnt see any X86 depends on SND_HDA_INTEL. I think we should add this now. Are you okay with that? Thanks > 24 #include <linux/delay.h> > 25 #include <linux/interrupt.h> > 26 #include <linux/kernel.h> > 27 #include <linux/module.h> > 28 #include <linux/pm_runtime.h> > 29 #include <linux/slab.h> > > 30 #include <asm/tsc.h> > 31 #include <sound/core.h> > 32 #include <sound/initval.h> > 33 #include "hda_controller.h" -- ~Vinod

2 7

[alsa-devel] [PATCH] ASoC: nau8825: jack connection decision with different insertion logic
by John Hsu 14 Jul '16

14 Jul '16

The original design only covers the jack insertion logic is active low. Add more condition to cover no matter the logic is active low and high. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3f30e6e..a97418d 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1345,10 +1345,17 @@ EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); static bool nau8825_is_jack_inserted(struct regmap *regmap) { - int status; + bool active_high, is_high; + int status, jkdet; + regmap_read(regmap, NAU8825_REG_JACK_DET_CTRL, &jkdet); + active_high = !!(jkdet & NAU8825_JACK_POLARITY); regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); - return !(status & NAU8825_GPIO2JD1); + is_high = !!(status & NAU8825_GPIO2JD1); + /* return jack connection status according to jack insertion logic + * active high or active low. + */ + return active_high == is_high; } static void nau8825_restart_jack_detection(struct regmap *regmap) -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: rt5514-spi: Convert to use devm_* API
by Axel Lin 14 Jul '16

14 Jul '16

Use devm_* API to simplify the code. This patch also fixes the return value in probe error paths. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5514-spi.c | 25 ++++++------------------- 1 file changed, 6 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 8a9382e..e1b326d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -409,33 +409,21 @@ static int rt5514_spi_probe(struct spi_device *spi) rt5514_spi = spi; - ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + ret = devm_snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); if (ret < 0) { dev_err(&spi->dev, "Failed to register platform.\n"); - goto err_plat; + return ret; } - ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, - &rt5514_spi_dai, 1); + ret = devm_snd_soc_register_component(&spi->dev, + &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); if (ret < 0) { dev_err(&spi->dev, "Failed to register component.\n"); - goto err_comp; + return ret; } return 0; -err_comp: - snd_soc_unregister_platform(&spi->dev); -err_plat: - - return 0; -} - -static int rt5514_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_component(&spi->dev); - snd_soc_unregister_platform(&spi->dev); - - return 0; } static const struct of_device_id rt5514_of_match[] = { @@ -450,7 +438,6 @@ static struct spi_driver rt5514_spi_driver = { .of_match_table = of_match_ptr(rt5514_of_match), }, .probe = rt5514_spi_probe, - .remove = rt5514_spi_remove, }; module_spi_driver(rt5514_spi_driver); -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix fw reload failure
by Vinod Koul 14 Jul '16

14 Jul '16

From: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> FW reload had two issues: - We need to disable the core 0 on when fw fails - Before loading firmware mark boot flag as false This patch fixes these two Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 9c3750f49c21..16e2ed97d71a 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -139,7 +139,7 @@ static int sst_bxt_prepare_fw(struct sst_dsp *ctx, base_fw_load_failed: ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag); skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); - skl_dsp_disable_core(ctx, SKL_DSP_CORE_MASK(1)); + skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); return ret; } @@ -232,6 +232,7 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); if (skl->fw_loaded == false) { + skl->boot_complete = false; ret = bxt_load_base_firmware(ctx); if (ret < 0) dev_err(ctx->dev, "reload fw failed: %d\n", ret); -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ASoC: Intel: More Kabylake machine support
by Vinod Koul 14 Jul '16

14 Jul '16

We have two more kbl machines, one with nau88l25_ssm4567 and second one with nau88l25_max98357a. This is similar to skylake which had the same combinations. To add kbl IDs to existing machines we need to add names in id table. But current names are more than 20 characters which is limit for platform device id name names. So first truncate skl machines to less than 20 chars and then add new kbl names. Vinod Koul (5): ASoC: Intel: Skylake: reduce machine name for skl_nau88l25_ssm4567 ASoC: Intel: Skylake: reduce machine name for skl_nau88l25_max98357a ASoC: Intel: Kbl: add kabylake additional machine entries ASoC: Intel: board: add kabylake nau88l25_max98357a machine id ASoC: Intel: board: add kabylake nau88l25_ssm4567 machine id sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 ++++++++++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 12 ++++++++++-- sound/soc/intel/skylake/skl.c | 6 ++++-- 3 files changed, 24 insertions(+), 6 deletions(-) Thanks -- ~Vinod

2 10

[alsa-devel] [PATCH] ASoC: Intel: Fix conflicting pcm dev drvdata on haswell
by Ben Zhang 14 Jul '16

14 Jul '16

soc-core sets the snd_soc_pcm_runtime->dev drvdata to snd_soc_pcm_runtime in soc_post_component_init, and access it in places like codec_reg_show. hsw_pcm_open overwrites the drvdata to point to hsw_pcm_data, confusing soc-core, and causing crashes when cat /sys/devices/pci0000:00/INT3438:00/.../System PCM/codec_reg This patch removes the set in hsw_pcm_open since it's no longer used. commit 7ff9d6714a5c ("ASoC: Intel: Split hsw_pcm_data for playback and capture") already removed all calls to snd_soc_pcm_get_drvdata(rtd). Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 994256b..3154525 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -819,7 +819,6 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); - snd_soc_pcm_set_drvdata(rtd, pcm_data); pcm_data->substream = substream; snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware); -- 2.8.0.rc3.226.g39d4020

2 1

[alsa-devel] [PATCH -next] ASoC: mediatek: mt2701: cHECK-fix non static symbol warning
by weiyj_lk＠163.com 14 Jul '16

14 Jul '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warning: sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:72:5: warning: symbol 'mt2701_dai_num_to_i2s' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 15522c0..34a6123 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -69,7 +69,7 @@ static const struct mt2701_afe_rate mt2701_afe_i2s_rates[] = { { .rate = 352800, .regvalue = 24 }, }; -int mt2701_dai_num_to_i2s(struct mtk_base_afe *afe, int num) +static int mt2701_dai_num_to_i2s(struct mtk_base_afe *afe, int num) { int val = num - MT2701_IO_I2S;

3 3

[alsa-devel] [PATCH v8] ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver
by Philipp Zabel 14 Jul '16

14 Jul '16

From: PC Liao <pc.liao(a)mediatek.com> This patch adds HDMI audio output support to the MT8173 RT5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- Changes since v7: - Rebased onto asoc/for-next --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +++-- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173/mt8173-rt5650.c | 26 ++++++++++++++++++++++ 3 files changed, 30 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index f250fc7..29dce2a 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -1,8 +1,9 @@ -MT8173 with RT5650 CODECS +MT8173 with RT5650 CODECS and HDMI via I2S Required properties: - compatible : "mediatek,mt8173-rt5650" - mediatek,audio-codec: the phandles of rt5650 codecs + and of the hdmi encoder node - mediatek,platform: the phandle of MT8173 ASoC platform Optional subnodes: @@ -20,7 +21,7 @@ Example: sound { compatible = "mediatek,mt8173-rt5650"; - mediatek,audio-codec = <&rt5650>; + mediatek,audio-codec = <&rt5650 &hdmi0>; mediatek,platform = <&afe>; mediatek,mclk = <0>; codec-capture { diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 2fbe543..05cf809 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -46,6 +46,7 @@ config SND_SOC_MT8173_RT5650 tristate "ASoC Audio driver for MT8173 with RT5650 codec" depends on SND_SOC_MT8173 && I2C select SND_SOC_RT5645 + select SND_SOC_HDMI_CODEC help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 audio codec. diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index d478976..ba65f41 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -169,7 +169,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, + DAI_LINK_HDMI, DAI_LINK_CODEC_I2S, + DAI_LINK_HDMI_I2S, }; /* Digital audio interface glue - connects codec <---> CPU */ @@ -195,6 +197,16 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .dynamic = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI] = { + .name = "HDMI", + .stream_name = "HDMI PCM", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, /* Back End DAI links */ [DAI_LINK_CODEC_I2S] = { .name = "Codec", @@ -210,6 +222,13 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI_I2S] = { + .name = "HDMI BE", + .cpu_dai_name = "HDMIO", + .no_pcm = 1, + .codec_dai_name = "i2s-hifi", + .dpcm_playback = 1, + }, }; static struct snd_soc_card mt8173_rt5650_card = { @@ -284,6 +303,13 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) } } + mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); + if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codec_of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 2.8.1

2 1

[alsa-devel] [alsa-lib][PATCH 00/34] pcm/conf/alisp: replace usage of alloca() with automatic variables
by Takashi Sakamoto 14 Jul '16

14 Jul '16

Hi, A batch of this patchset is to save storage consumption and to improve execution performance, just for a little. Currently, ALSA userspace library doesn't disclose layouts of major structures to applications. The applications call some APIs in the library to get memory objects for the structures, then operates to corresponding opaque pointers. In most cases, the objects are kept on stack by calls of alloca(3). On the other hand, inner the library, the layouts are decided in pre-compile or compile time. Therefore, there's little needs to call of alloca(3). However, inner the library, we can see some usages of alloca(3). This brings two disadvantages to system. One is to execute more instructions to use stack than automatic variables, for the same task. Another is to enlarge size of shared object. Although both disadvantages are not so remarkable, it's better to improve them. This patchset replaces usages of alloca() with automatic variables just to raw structures. Additionally, this commit improves code formats in related functions. I can see below effect on my environment (Ubuntu 16.04, amd64, gcc) $ git checkout master $ git show HEAD commit 941bd150bef2560f3e38a3bf74d546447e3126d1 ... $ ./gitcompile --with-plugindir=/usr/lib/x86_64-linux-gnu/alsa-lib/ ... $ ls -l src/.libs/libasound.so.2.0.0 -rwxrwxr-x 1 mocchi mocchi 4794528 7月 14 22:22 src/.libs/libasound.so.2.0.0 $ git checkout remove-alloca-asound $ make ... $ ls -l src/.libs/libasound.so.2.0.0 -rwxrwxr-x 1 mocchi mocchi 4783712 7月 14 22:21 src/.libs/libasound.so.2.0.0 $ gcc -v ... gcc version 5.4.0 20160609 (Ubuntu 5.4.0-6ubuntu1~16.04.1) Takashi Sakamoto (34): pcm: change code formatting for snd_pcm_set_params() pcm: remove alloca() from snd_pcm_set_params() pcm: change code formatting for snd_pcm_get_params() pcm: remove alloca() from snd_pcm_get_params pcm: change code formatting for snd_pcm_direct_initialize_slave() pcm: remove alloca() from snd_pcm_direct_initialize_slave pcm: change code formatting for snd_pcm_direct_initialize_poll_fd() pcm: remove alloca() from snd_pcm_direct_initialize_poll_fd() pcm: change code formatting for snd_pcm_direct_set_timer_params() pcm: remove alloca() from snd_pcm_direct_set_timer_params pcm: remove alloca() from _snd_pcm_hook_ctl_elems_install() pcm: change code formatting for snd_pcm_hw_change_timer() pcm: remove alloca() from snd_pcm_hw_change_timer() pcm: remove alloca() from snd_pcm_query_chmaps_from_hw() pcm: remove alloca() from snd_pcm_hw_get_chmap() pcm: remove alloca() from snd_pcm_hw_set_chmap() pcm: remove alloca() from snd_spcm_init() pcm: remove alloca() from snd_spcm_init_duplex() pcm: change code formatting for softvol_load_control() pcm: remove alloca() from softvol_load_control() pcm: change code formatting for _snd_pcm_softvol_open() pcm: remove alloca() from _snd_pcm_softvol_open() conf: remove alloca() from snd_determine_driver() conf: remove alloca() from snd_func_card_id() conf: remove alloca() from snd_func_card_name() conf: remove alloca() from snd_func_pcm_id() conf: remove alloca() from snd_func_pcm_args_by_class() conf: remove alloca() from snd_func_private_pcm_subdevice() alisp: remove alloca() from FA_card_info() alisp: remove alloca() from FA_hctl_find_elem() alisp: remove alloca() from FA_hctl_elem_info() alisp: remove alloca() from FA_hctl_elem_read() alisp: remove alloca() from FA_hctl_elem_write() alisp: remove alloca() from FA_pcm_info() src/alisp/alisp_snd.c | 153 ++++++++++++++++----------------- src/confmisc.c | 51 +++++------ src/pcm/pcm.c | 232 ++++++++++++++++++++++++++++---------------------- src/pcm/pcm_direct.c | 132 ++++++++++++++++------------ src/pcm/pcm_hooks.c | 8 +- src/pcm/pcm_hw.c | 74 +++++++++------- src/pcm/pcm_simple.c | 22 ++--- src/pcm/pcm_softvol.c | 69 ++++++++------- 8 files changed, 393 insertions(+), 348 deletions(-) -- 2.7.4

2 35

[alsa-devel] [PATCH RESEND] ASoC: rt5514-spi: Convert to use devm_* API
by Axel Lin 14 Jul '16

14 Jul '16

Use devm_* API to simplify the code. This patch also fixes the return value in probe error paths. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5514-spi.c | 25 ++++++------------------- 1 file changed, 6 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 8a9382e..e1b326d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -409,33 +409,21 @@ static int rt5514_spi_probe(struct spi_device *spi) rt5514_spi = spi; - ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + ret = devm_snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); if (ret < 0) { dev_err(&spi->dev, "Failed to register platform.\n"); - goto err_plat; + return ret; } - ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, - &rt5514_spi_dai, 1); + ret = devm_snd_soc_register_component(&spi->dev, + &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); if (ret < 0) { dev_err(&spi->dev, "Failed to register component.\n"); - goto err_comp; + return ret; } return 0; -err_comp: - snd_soc_unregister_platform(&spi->dev); -err_plat: - - return 0; -} - -static int rt5514_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_component(&spi->dev); - snd_soc_unregister_platform(&spi->dev); - - return 0; } static const struct of_device_id rt5514_of_match[] = { @@ -450,7 +438,6 @@ static struct spi_driver rt5514_spi_driver = { .of_match_table = of_match_ptr(rt5514_of_match), }, .probe = rt5514_spi_probe, - .remove = rt5514_spi_remove, }; module_spi_driver(rt5514_spi_driver); -- 2.5.0

1 0

[alsa-devel] [PATCH] pcm: fix return value of snd_pcm_ioplug_sw_params()
by Takashi Sakamoto 13 Jul '16

13 Jul '16

In former commits for thread-safe PCM APIs, snd_pcm_ioplug_sw_params() got 0 as its return value, against the original implementation. This commit fixes it. Fixes: 54931e5a5455('pcm: Add thread-safety to PCM API') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/pcm/pcm_ioplug.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index 115d89b..1dc198e 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -434,14 +434,16 @@ static int snd_pcm_ioplug_hw_free(snd_pcm_t *pcm) static int snd_pcm_ioplug_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) { ioplug_priv_t *io = pcm->private_data; - int err = 0; + int err; - if (io->data->callback->sw_params) { - snd_pcm_unlock(pcm); /* to avoid deadlock */ - err = io->data->callback->sw_params(io->data, params); - snd_pcm_lock(pcm); - } - return 0; + if (!io->data->callback->sw_params) + return 0; + + snd_pcm_unlock(pcm); /* to avoid deadlock */ + err = io->data->callback->sw_params(io->data, params); + snd_pcm_lock(pcm); + + return err; } -- 2.7.4

2 1

[alsa-devel] [PATCH] pcm: fix missing return value of snd_pcm_status()
by Takashi Sakamoto 13 Jul '16

13 Jul '16

In former commits for thread-safe PCM APIs, snd_pcm_status() lost its return value. This commit regain it. Fixes: 54931e5a5455('pcm: Add thread-safety to PCM API') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/pcm/pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index a66e8af..6c34719 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -953,6 +953,8 @@ int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status) snd_pcm_lock(pcm); err = pcm->fast_ops->status(pcm->fast_op_arg, status); snd_pcm_unlock(pcm); + + return err; } /** -- 2.7.4

2 1

[alsa-devel] [patch] ALSA: mixart: don't print an unintialized variable on error
by Dan Carpenter 13 Jul '16

13 Jul '16

My static checker complains that "resp" could be unitialized on error when we print its value. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 58fd79eb..51e5349 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -965,7 +965,7 @@ static int mixart_update_monitoring(struct snd_mixart* chip, int channel) int err; struct mixart_msg request; struct mixart_set_out_audio_level audio_level; - u32 resp; + u32 resp = 0; if(chip->pipe_out_ana.status == PIPE_UNDEFINED) return -EINVAL; /* no pipe defined */

3 3

[alsa-devel] [PATCH 0/5] topology: Enhance support for private data
by mengdong.lin＠linux.intel.com 13 Jul '16

13 Jul '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. We finished verification on latest Intel platforms. There will be 4 or 5 series. Later series will come with some ABI update for topology. The size check on ABI objects in current ASoC kernel will detect possible ABI version mismatch between user space and kernel. Mengdong Lin (5): topology: An element can refer to multipe data sections in text conf file topology: Merge lookup for data reference into tplg_copy_data() topology: Change uuid value to 16 separate characters in text conf file topology: Parse vendor private data for manifest topology: Tuple type can have an extenstion include/topology.h | 69 ++++++++++- src/topology/ctl.c | 52 ++++---- src/topology/dapm.c | 17 ++- src/topology/data.c | 296 +++++++++++++++++++++++++++++++++++++++------- src/topology/elem.c | 4 + src/topology/parser.c | 28 ++++- src/topology/tplg_local.h | 14 ++- 7 files changed, 389 insertions(+), 91 deletions(-) -- 2.5.0

2 8

[alsa-devel] [alsa-lib][PATCH] ctl: add compatibility information for snd_ctl_elem_info_set_dimension()
by Takashi Sakamoto 13 Jul '16

13 Jul '16

I forgot it when committing to the API. Fixes: 16812b70daeb('ctl: add an API to set dimension levels to element information') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/src/control/control.c b/src/control/control.c index a88f7a1..aa05130 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2515,6 +2515,9 @@ use_default_symbol_version(__snd_ctl_elem_info_get_dimension, snd_ctl_elem_info_ * <dt>-EINVAL * <dd>Invalid arguments are given as parameters. * </dl> + * + * \par Compatibility: + * This function is added in version 1.1.2. */ int snd_ctl_elem_info_set_dimension(snd_ctl_elem_info_t *info, const int dimension[4]) -- 2.7.4

2 1

[alsa-devel] [GIT PULL] Add Kabylake audio firmware
by Vinod Koul 12 Jul '16

12 Jul '16

Hello, Please PULL to recive the audio firmware for Kabylake Intel SoC. The following changes since commit 1dc99be22222fbf88adc98d85f8ffb7529c6c4e3: amdgpu: update vce firmware for VI asics (2016-07-11 15:18:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git kbl for you to fetch changes up to 11e3390b4fac69b50a5f88861d295fd94e51ba6c: linux-firmware: intel: Add Kabylake audio firmware (2016-07-12 14:50:59 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Add Kabylake audio firmware WHENCE | 6 ++++++ intel/dsp_fw_kbl.bin | 1 + intel/dsp_fw_kbl_v701.bin | Bin 0 -> 238920 bytes 3 files changed, 7 insertions(+) create mode 120000 intel/dsp_fw_kbl.bin create mode 100644 intel/dsp_fw_kbl_v701.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] ALSA: ak4114: remove redundant check on err being < 0
by Colin King 12 Jul '16

12 Jul '16

From: Colin Ian King <colin.king(a)canonical.com> snd_ak4114_create checks if the error return err is less than zero or not. This is a redundant check, err can only be < 0 to get to the __fail label, in which case just return err and remove the redundant check (since we never return -EIO). Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/other/ak4114.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 5a4cf3f..d53c9bb 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -121,7 +121,7 @@ int snd_ak4114_create(struct snd_card *card, __fail: snd_ak4114_free(chip); - return err < 0 ? err : -EIO; + return err; } EXPORT_SYMBOL(snd_ak4114_create); -- 2.8.1

2 1

[alsa-devel] [PATCH] ALSA: ak4117: remove redundant check on err being < 0
by Colin King 12 Jul '16

12 Jul '16

From: Colin Ian King <colin.king(a)canonical.com> snd_ak4117_create checks if the error return err is less than zero or not. This is a redundant check, err can only be < 0 to get to the __fail label, in which case just return err and remove the redundant check (since we never return -EIO). Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/i2c/other/ak4117.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 48848909..0702f05 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -110,7 +110,7 @@ int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t __fail: snd_ak4117_free(chip); - return err < 0 ? err : -EIO; + return err; } void snd_ak4117_reg_write(struct ak4117 *chip, unsigned char reg, unsigned char mask, unsigned char val) -- 2.8.1

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[alsa-devel] [PATCH] ALSA: hda: add AMD Stoney PCI ID with proper driver caps
by Awais Belal 12 Jul '16

12 Jul '16

This allows the device to correctly show up as ATI HDMI rather than a generic one and allows the driver to use the available caps. Signed-off-by: Awais Belal <awais_belal(a)mentor.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 09920ba..4b80b16 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2106,6 +2106,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0x15b3), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), -- 1.9.1

2 1

[alsa-devel] Problem getting USB DAC to work
by John Westwood 12 Jul '16

12 Jul '16

Dear ALSA Developers I have a problem getting my obscure USB DAC (Starting Point Systems DAC3) to work in Linux. I believe the problem may be similar to this one, but am not sure: https://groups.google.com/forum/#!topic/linux.debian.kernel/HwwkRkbL5aY If I try to use aplay to play a wav I receive the following error: aplay -D default test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo aplay: pcm_write:1939: write error: No such device syslog shows the following: Jul 9 15:38:31 atlas kernel: [ 4805.148059] usb 7-2: USB disconnect, device number 19 Jul 9 15:38:32 atlas kernel: [ 4805.532022] usb 7-2: new full-speed USB device number 20 using uhci_hcd Jul 9 15:38:32 atlas kernel: [ 4805.725024] usb 7-2: New USB device found, idVendor=2504, idProduct=0201 Jul 9 15:38:32 atlas kernel: [ 4805.725029] usb 7-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 Jul 9 15:38:32 atlas kernel: [ 4805.725032] usb 7-2: Product: TDA 1543 DAC 3 v1.4 Jul 9 15:38:32 atlas kernel: [ 4805.725034] usb 7-2: Manufacturer: STARTING POINT SYSTEMS Jul 9 15:38:32 atlas kernel: [ 4805.725035] usb 7-2: SerialNumber: Jul 9 15:38:32 atlas kernel: [ 4805.745018] usb 7-2: Warning! Unlikely big volume range (=65535), cval->res is probably wrong. Jul 9 15:38:32 atlas kernel: [ 4805.745023] usb 7-2: [2] FU [PCM Playback Volume] ch = 1, val = -32768/32767/1 Jul 9 15:38:32 atlas mtp-probe: checking bus 7, device 20: "/sys/devices/pci0000:00/0000:00:1d.1/usb7/7-2" Jul 9 15:38:32 atlas mtp-probe: bus: 7, device: 20 was not an MTP device I have another USB DAC that works fine, in syslog the working DAC has the following lines which are not present with the non-working DAC: Jul 11 19:56:35 atlas kernel: [83940.049124] input: Lakewest Audio Audiolab M-DAC as /devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.0/0003:0451:ADAC.0002/input/input12 Jul 11 19:56:35 atlas kernel: [83940.104367] hid-generic 0003:0451:ADAC.0002:input,hiddev0,hidraw1: USB HID v1.11 Device [Lakewest Audio Audiolab M-DAC] on usb-0000:00:1d.1-2/input0 With the non-working DAC I get these lines instead: Jul 11 19:56:25 atlas kernel: [83930.079063] usb 7-2: Warning! Unlikely big volume range (=65535), cval->res is probably wrong. Jul 11 19:56:25 atlas kernel: [83930.079067] usb 7-2: [2] FU [PCM Playback Volume] ch = 1, val = -32768/32767/1 I am using Linux Mint 17.3 64bit, kernel 3.19.0-32. I have also tried kernel 4.4.0-28 but to no avail. The ALSA version is 1.0.27.2 . Any help or suggestions would be greatly appreciated. Thanks very much. Best regards John Westwood

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[alsa-devel] Fwd: Audio capture on OMAP3530/DM3730 (Yocto Poky Linux 3.18+) occasionally swaps channels
by Akram Hameed 12 Jul '16

12 Jul '16

Hello there, (Apologies for cross-posting from user list if you are seeing this twice). I have a fairly serious problem using ALSA for capture on an Gumstix Overo based system. The audio device in question is a TWL4030 device. It is connected to a TI DM3730 (or in some cases, an OMAP3530). Previously I was using a 3.0 kernel and my audio capture worked just fine with no extra tweaks to ALSA. Now for other reasons I have to move to a more recent kernel, and have tried with 3.18 and 4.4 stable releases. Using these kernels, at random times, the audio capture channels will swap. They will remain swapped for an unpredictable period of time, and then they will swap back. This will continue forever, again, with the swapping occurring at random times. Our system totally relies on predictable audio capture - so for this random behaviour to occur means we must seek a different platform unless we can either: a) Find a solution for 3.18 or newer kernel (the problem is still occurring with ALSA 1.1 on kernel 4.4) or, b) Make some code change to solve the issue. I found this alsa-devel thread regarding the 2.6.32 kernel that seems to be almost exactly what I am experiencing: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-February/036895.h… I also found this one which is probably more closely related: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-March/060557.html My current thought is that it is indeed a DMA issue, but I am not familiar with that area of the kernel source and therefore hoping someone might have some suggestions about what to try next. Best regards, Akram

1 0

[alsa-devel] [PATCH v2] sound: oss: Remove useless initialisation
by Amitoj Kaur Chawla 12 Jul '16

12 Jul '16

Remove useless initialisation of variable whose value is reinitialised later. The Coccinelle semantic patch used to make this change is as follows: @@ type T; identifier x; constant C; expression e; @@ T x - = C ; x = e; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- Changes in v2: -Modify commit message sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6368e5c 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -254,7 +254,7 @@ static void ad_write(ad1848_info * devc, int reg, int data) static void wait_for_calibration(ad1848_info * devc) { - int timeout = 0; + int timeout; /* * Wait until the auto calibration process has finished. -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: intel: Fix sst-dsp dependency on dw stuff
by Takashi Iwai 12 Jul '16

12 Jul '16

The recent commit [a92ea59b74e2: ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in] introduced more strict kconfig dependency (depends on DW_DMAC_CORE=y) for avoiding the build failures due to dependency messes in intel-sst. This makes, however, it impossible to use this driver with the modularized systems, i.e. typically on Linux distros. The problem addressed in the commit above is that sst_dsp_new() and sst_dsp_free() includes the firmware init / finish that call dw_*() functions. Thus building it as built-in with DW_DMAC_CORE module results in the missing symbols. However, these sst_dsp functions are basically called only from the drivers that depend on DW_DMAC_CORE already. That is, once when these functions are split out, the rest can be independent from dw stuff. This patch attempts to solve the issue by the following: - Split sst-dsp stuff into two modules: snd-soc-sst-dsp and snd-soc-sst-firmware. - Move sst_dsp_new() and sst_dsp_free() to the latter module so that the former module can be independent from DW_DMAC_CORE. - Add a new kconfig SND_SOC_INTEL_SST_FIRMWARE to select the latter module by machine drivers. One only remaining pitfall is that each machine driver has to select SND_SOC_INTEL_SST_FIRMWARE carefully depending on DW_DMAC_CORE. This can't be done cleanly due to the restriction of the current kbuild. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=988117 Fixes: a92ea59b74e2 ('ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/Kconfig | 18 +++++++--- sound/soc/intel/common/Makefile | 4 +-- sound/soc/intel/common/sst-dsp-priv.h | 4 --- sound/soc/intel/common/sst-dsp.c | 67 ---------------------------------- sound/soc/intel/common/sst-dsp.h | 2 +- sound/soc/intel/common/sst-firmware.c | 68 +++++++++++++++++++++++++++++++++++ 6 files changed, 85 insertions(+), 78 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 9c86459d0fc3..a20c3dfbcb5d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -32,6 +32,12 @@ config SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_MATCH if ACPI depends on (X86 || COMPILE_TEST) +# firmware stuff depends DW_DMAC_CORE; since there is no depends-on from +# the reverse selection, each machine driver needs to select +# SND_SOC_INTEL_SST_FIRMWARE carefully depending on DW_DMAC_CORE +config SND_SOC_INTEL_SST_FIRMWARE + tristate + config SND_SOC_INTEL_SST_ACPI tristate @@ -47,8 +53,9 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE=y + depends on DW_DMAC_CORE select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -91,8 +98,9 @@ config SND_SOC_INTEL_BXT_RT298_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n) + depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -103,8 +111,9 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n) + depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -115,8 +124,9 @@ config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE=y + depends on DW_DMAC_CORE select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index fbbb25c2ceed..1a35149bcad7 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -2,9 +2,9 @@ snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-match-objs := sst-match-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o - -snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o +snd-soc-sst-firmware-objs := sst-firmware.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o +obj-$(CONFIG_SND_SOC_INTEL_SST_FIRMWARE) += snd-soc-sst-firmware.o diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h index 97dc1ae05e69..d13c84364c3c 100644 --- a/sound/soc/intel/common/sst-dsp-priv.h +++ b/sound/soc/intel/common/sst-dsp-priv.h @@ -383,10 +383,6 @@ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, u32 index, void *private); void sst_mem_block_unregister_all(struct sst_dsp *dsp); -/* Create/Free DMA resources */ -int sst_dma_new(struct sst_dsp *sst); -void sst_dma_free(struct sst_dma *dma); - u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset, enum sst_mem_type type); #endif diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index ff2196ef359f..c00ede4ea4d7 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -420,73 +420,6 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) } EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); -#ifdef CONFIG_DW_DMAC_CORE -struct sst_dsp *sst_dsp_new(struct device *dev, - struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) -{ - struct sst_dsp *sst; - int err; - - dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); - - sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); - if (sst == NULL) - return NULL; - - spin_lock_init(&sst->spinlock); - mutex_init(&sst->mutex); - sst->dev = dev; - sst->dma_dev = pdata->dma_dev; - sst->thread_context = sst_dev->thread_context; - sst->sst_dev = sst_dev; - sst->id = pdata->id; - sst->irq = pdata->irq; - sst->ops = sst_dev->ops; - sst->pdata = pdata; - INIT_LIST_HEAD(&sst->used_block_list); - INIT_LIST_HEAD(&sst->free_block_list); - INIT_LIST_HEAD(&sst->module_list); - INIT_LIST_HEAD(&sst->fw_list); - INIT_LIST_HEAD(&sst->scratch_block_list); - - /* Initialise SST Audio DSP */ - if (sst->ops->init) { - err = sst->ops->init(sst, pdata); - if (err < 0) - return NULL; - } - - /* Register the ISR */ - err = request_threaded_irq(sst->irq, sst->ops->irq_handler, - sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); - if (err) - goto irq_err; - - err = sst_dma_new(sst); - if (err) - dev_warn(dev, "sst_dma_new failed %d\n", err); - - return sst; - -irq_err: - if (sst->ops->free) - sst->ops->free(sst); - - return NULL; -} -EXPORT_SYMBOL_GPL(sst_dsp_new); - -void sst_dsp_free(struct sst_dsp *sst) -{ - free_irq(sst->irq, sst); - if (sst->ops->free) - sst->ops->free(sst); - - sst_dma_free(sst->dma); -} -EXPORT_SYMBOL_GPL(sst_dsp_free); -#endif - /* Module information */ MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("Intel SST Core"); diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 0b84c719ec48..859f0de00339 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -216,7 +216,7 @@ struct sst_pdata { void *dsp; }; -#ifdef CONFIG_DW_DMAC_CORE +#if IS_ENABLED(CONFIG_DW_DMAC_CORE) /* Initialization */ struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 25993527370b..a086c35f91bb 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1211,3 +1211,71 @@ u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset, } } EXPORT_SYMBOL_GPL(sst_dsp_get_offset); + +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) +{ + struct sst_dsp *sst; + int err; + + dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->dma_dev = pdata->dma_dev; + sst->thread_context = sst_dev->thread_context; + sst->sst_dev = sst_dev; + sst->id = pdata->id; + sst->irq = pdata->irq; + sst->ops = sst_dev->ops; + sst->pdata = pdata; + INIT_LIST_HEAD(&sst->used_block_list); + INIT_LIST_HEAD(&sst->free_block_list); + INIT_LIST_HEAD(&sst->module_list); + INIT_LIST_HEAD(&sst->fw_list); + INIT_LIST_HEAD(&sst->scratch_block_list); + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + err = sst->ops->init(sst, pdata); + if (err < 0) + return NULL; + } + + /* Register the ISR */ + err = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (err) + goto irq_err; + + err = sst_dma_new(sst); + if (err) + dev_warn(dev, "sst_dma_new failed %d\n", err); + + return sst; + +irq_err: + if (sst->ops->free) + sst->ops->free(sst); + + return NULL; +} +EXPORT_SYMBOL_GPL(sst_dsp_new); + +void sst_dsp_free(struct sst_dsp *sst) +{ + free_irq(sst->irq, sst); + if (sst->ops->free) + sst->ops->free(sst); + + sst_dma_free(sst->dma); +} +EXPORT_SYMBOL_GPL(sst_dsp_free); + +MODULE_DESCRIPTION("Intel SST Firmware Loader"); +MODULE_LICENSE("GPL v2"); -- 2.9.0

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[alsa-devel] [PATCH] sound: oss: Remove useless initialisation
by Amitoj Kaur Chawla 12 Jul '16

12 Jul '16

@@ type T; identifier x; constant C; expression e; @@ T x - = C ; x = e; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6368e5c 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -254,7 +254,7 @@ static void ad_write(ad1848_info * devc, int reg, int data) static void wait_for_calibration(ad1848_info * devc) { - int timeout = 0; + int timeout; /* * Wait until the auto calibration process has finished. -- 1.9.1

1 0

[alsa-devel] Verification of settings
by Rob Nertney 12 Jul '16

12 Jul '16

Hi all, Could I please get a sanity check on my hw_params? I have a DMA which is providing between 1-16 channels of 4 bytes/ch worth of data as a frame. I get an interrupt to my driver every frame's worth of data (64 Bytes). The data is S32_LE, 16000Hz. My DMA has 2 buffers, PING and PONG. Each receives an IRQ on a frame length, and these local buffers are the size of a frame length (64Bytes). #define MAX_BUFFER (64 * 2) static struct snd_pcm_hardware my_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_16000, .rate_min = 16000, .rate_max = 16000, .channels_min = 1, .channels_max = NUM_CHANNELS, .buffer_bytes_max = MAX_BUFFER, .period_bytes_min = 4, .period_bytes_max = 64, .periods_min = 2, .periods_max = 2, }; My understanding is that the MAX_BUFFER needs to be at least twice the size of a period so I don't underrun. .periods_max means the maximum number of periods in a the alsa dma_area buffer, right? So when my DMA fires its ISR, I copy from its local PING buffer to the dma_area at offset 0, increment the buf_pos by the frame_length (64Bytes), and call snd_pcm_period_elapsed. My DMA fires its ISR for its local PONG buffer, copies to the dma_area+buf_pos, increments buf_pos (now back to 0, since buffer only holds 2 frames/periods), and I call snd_pcm_period_elapsed again, correct? Thanks, Rob

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[alsa-devel] [PATCH 0/4] ASoC: Intel: Add support for KBL
by Vinod Koul 12 Jul '16

12 Jul '16

Kabylake is next generation Intel SoC which also sports a DSP. This controller is quite similar to Skylake. This patch series add KBL IDs to SKL driver, hdmi driver, sets combo jack for rt286 codec and finally adds entry to skl machine driver Shreyas NC (1): ASoC: hdac_hdmi: Add device id for Kabylake Vinod Koul (3): ASoC: Intel: add kablake device IDs ASoC: rt286: set combo jack for Kabylake ASoC: Intel: board: add kabylake machine id sound/soc/codecs/hdac_hdmi.c | 1 + sound/soc/codecs/rt286.c | 7 +++++++ sound/soc/intel/boards/skl_rt286.c | 9 +++++++++ sound/soc/intel/skylake/skl-messages.c | 6 ++++++ sound/soc/intel/skylake/skl.c | 8 ++++++++ 5 files changed, 31 insertions(+) -- 1.9.1

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Re: [alsa-devel] Problem getting USB DAC to work
by Alan Stern 11 Jul '16

11 Jul '16

On Mon, 11 Jul 2016, John Westwood wrote: > I think the MTP stuff is a red herring. I checked syslog when I use a DAC that > works (Audiolab MDAC) and that has the MTP lines too, but it doesn't have the > warning line about Unlikely big volume range. > > In addition, the working DAC has the following lines which are not present the > non-working DAC: > > Jul 11 19:56:35 atlas kernel: [83940.049124] input: Lakewest Audio Audiolab > M-DAC as /devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.0/0003:0451: > ADAC.0002/input/input12 > Jul 11 19:56:35 atlas kernel: [83940.104367] hid-generic 0003:0451:ADAC.0002: > input,hiddev0,hidraw1: USB HID v1.11 Device [Lakewest Audio Audiolab M-DAC] on > usb-0000:00:1d.1-2/input0 > > With the non-working DAC I get these lines instead: > > Jul 11 19:56:25 atlas kernel: [83930.079063] usb 7-2: Warning! Unlikely big > volume range (=65535), cval->res is probably wrong. > Jul 11 19:56:25 atlas kernel: [83930.079067] usb 7-2: [2] FU [PCM Playback > Volume] ch = 1, val = -32768/32767/1 > > > I think maybe the problem is similar to this one: > > https://groups.google.com/forum/#!topic/linux.debian.kernel/HwwkRkbL5aY > > Is my problem a bug in snd-usb-audio? Or am I doing something wrong? I would > really appreciate some thoughts on this, I feel like I'm talking to myself :-( Considering that you're not really doing anything, you're probably not doing something wrong. :-) On the other hand, you're likely to get more of a response if you direct your questions to the person who solved the problem mentioned in that Debian bug report, and if you CC: the sound developers' mailing list rather than the general USB list. Alan Stern

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Re: [alsa-devel] [PATCH] snd/hda: fix use-after-free after module unload
by Takashi Iwai 11 Jul '16

11 Jul '16

On Sat, 09 Jul 2016 16:38:57 +0200, Peter Wu wrote: > > register_vga_switcheroo() sets the PM ops from the hda structure which > is freed later in azx_free. Make sure that these ops are cleared. > > Caught by KASAN. > > Fixes: 246efa4a072f ("snd/hda: add runtime suspend/resume on optimus support (v4)") > Signed-off-by: Peter Wu <peter(a)lekensteyn.nl> > --- > sound/pci/hda/hda_intel.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 94089fc..a339066 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1219,6 +1219,7 @@ static int azx_free(struct azx *chip) > snd_hda_unlock_devices(&chip->bus); > if (hda->vga_switcheroo_registered) > vga_switcheroo_unregister_client(chip->pci); > + vga_switcheroo_fini_domain_pm_ops(chip->card->dev); The domain pm ops is set only when hda->vga_switcheroo_registered flag is set. So the call should be in the previous if block. Also, the indentation looks wrong. Please use the correct indentation. Could you resubmit with these fixes? thanks, Takashi

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Fix potential NULL dereference
by Vinod Koul 11 Jul '16

11 Jul '16

Static checker warns: Pointer 'hlink' returned from call to function 'snd_hdac_ext_bus_get_link' at line may be NULL and will be dereferenced" So we should always check the return of snd_hdac_ext_bus_get_link() before referencing the link pointer Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 13002f33384e..cc7bb77a28f1 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1388,6 +1388,11 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) * exit, we call pm_runtime_suspend() so that will do for us */ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + if (!hlink) { + dev_err(&edev->hdac.dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(edev->ebus, hlink); ret = create_fill_widget_route_map(dapm); @@ -1494,6 +1499,11 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + if (!hlink) { + dev_err(&edev->hdac.dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(edev->ebus, hlink); hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL); @@ -1600,6 +1610,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) } hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + if (!hlink) { + dev_err(dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_put(ebus, hlink); return 0; @@ -1621,6 +1636,11 @@ static int hdac_hdmi_runtime_resume(struct device *dev) return 0; hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + if (!hlink) { + dev_err(dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(ebus, hlink); err = snd_hdac_display_power(bus, true); -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Initialize module list for Broxton
by Vinod Koul 11 Jul '16

11 Jul '16

The module list was not initialized for Broxton DSP code, so initialize it. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 622da5d3e3b3..9d36440c802a 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -325,6 +325,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + INIT_LIST_HEAD(&sst->module_list); ret = skl_ipc_init(dev, skl); if (ret) return ret; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: atom: fix missing breaks that would cause the wrong operation to execute
by Vinod Koul 11 Jul '16

11 Jul '16

From: Alan Cox <alan(a)linux.intel.com> Now we correctly error an attempt to execute an unsupported operation. Signed-off-by: Alan Cox <alan(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst-mfld-platform-compress.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c index 395168986462..1bead81bb510 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-compress.c +++ b/sound/soc/intel/atom/sst-mfld-platform-compress.c @@ -182,24 +182,29 @@ static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) case SNDRV_PCM_TRIGGER_START: if (stream->compr_ops->stream_start) return stream->compr_ops->stream_start(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_STOP: if (stream->compr_ops->stream_drop) return stream->compr_ops->stream_drop(sst->dev, stream->id); + break; case SND_COMPR_TRIGGER_DRAIN: if (stream->compr_ops->stream_drain) return stream->compr_ops->stream_drain(sst->dev, stream->id); + break; case SND_COMPR_TRIGGER_PARTIAL_DRAIN: if (stream->compr_ops->stream_partial_drain) return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (stream->compr_ops->stream_pause) return stream->compr_ops->stream_pause(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (stream->compr_ops->stream_pause_release) return stream->compr_ops->stream_pause_release(sst->dev, stream->id); - default: - return -EINVAL; + break; } + return -EINVAL; } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, -- 1.9.1

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[alsa-devel] [PATCH alsa-lib v2 0/3] PCM thread-safety patchset
by Takashi Iwai 11 Jul '16

11 Jul '16

Hi, here is a v2 patchset for supporting multi thread safety to alsa-lib PCM API. Now it's out from RFC and should be reviewed as an official patchset. Basically ALSA PCM functions are thread-unsafe, and applications are supposed to do the proper protection against racy accesses. The reality is, however, that application developers don't care such, as alsa-lib works in most cases. Of course, users still get occasionally mysterious crashes. As a workaround, this patchset adds the pthread mutex protection to most of exported PCM functions. This is slightly an overkill, but the biggest merit is that it's easy to implement; I just wrapped the functions and replaced the internal ones with unlocked versions. To be noted, there is an optimization for the direct hw PCM access. Performance-sensitive applications like JACK should work like before without any overhead. Another bonus by this addition is that we can finally get rid of home brew (and deadly smelling) atomic macros from the tree, since it's now more widely protected. I lightly tested on my local machines (dmix, PA, jack and plugins) and all seem working well, so far. Takashi === v1->v2: - Drop locking for some API functions like hw_params - Lock some functions even for hw plugin to keep PCM internals consistent - Fix deadlock with jack ioplug - More documentations and comments - Add debugging option via $LIBASOUND_THREAD_SAFE=0 - Reordered patches Takashi Iwai (3): pcm: Add thread-safety to PCM API pcm: Remove home brew atomic operations pcm: Add LIBASOUND_THREAD_SAFE env variable check INSTALL | 9 ++ configure.ac | 15 ++ include/Makefile.am | 2 +- include/iatomic.h | 170 -------------------- include/pcm_ioplug.h | 10 +- src/pcm/Makefile.am | 2 +- src/pcm/atomic.c | 43 ----- src/pcm/pcm.c | 433 +++++++++++++++++++++++++++++++++++++++++--------- src/pcm/pcm_direct.c | 4 +- src/pcm/pcm_dmix.c | 13 +- src/pcm/pcm_dshare.c | 13 +- src/pcm/pcm_dsnoop.c | 15 +- src/pcm/pcm_file.c | 21 ++- src/pcm/pcm_generic.c | 10 +- src/pcm/pcm_hw.c | 3 + src/pcm/pcm_ioplug.c | 70 +++++--- src/pcm/pcm_local.h | 131 +++++++++++---- src/pcm/pcm_mmap.c | 18 ++- src/pcm/pcm_plugin.c | 72 ++------- src/pcm/pcm_plugin.h | 2 - src/pcm/pcm_rate.c | 41 ++--- src/pcm/pcm_route.c | 2 +- 22 files changed, 648 insertions(+), 451 deletions(-) delete mode 100644 include/iatomic.h delete mode 100644 src/pcm/atomic.c -- 2.9.0

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[alsa-devel] [PATCH 0/3] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 11 Jul '16

11 Jul '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. While at it, fix the timestamping Documentation. Guneshwor Singh (2): ALSA - hda: Add support for GTS capability ALSA - hda: Add support for link audio time reporting Vinod Koul (1): ALSA - hda: Fix timestamping documentation Documentation/sound/alsa/timestamping.txt | 12 +- include/sound/hda_register.h | 23 ++++ sound/pci/hda/hda_controller.c | 192 +++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 27 +++++ sound/pci/hda/hda_intel.c | 1 + 5 files changed, 248 insertions(+), 7 deletions(-) -- 1.9.1

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[alsa-devel] [alsa-lib][PATCH 0/7] fix compiler warnings
by Takashi Sakamoto 11 Jul '16

11 Jul '16

Hi, This patchset is just for fixes of compiler warnings. Nothing others. Takashi Sakamoto (7): topology: remove unused variable in tplg_parse_fe_dai() topology: remove unused variable in get_tokens() topology: apply the same type to compared two variables topology: initialize variable for error output pcm: set error code at failure path mixer: remove assignment to unused variable pcm: add 'default' case for switch statement modules/mixer/simple/sbase.c | 2 -- src/pcm/pcm_direct.c | 2 ++ src/pcm/pcm_hw.c | 1 + src/topology/data.c | 10 +++++----- src/topology/pcm.c | 1 - 5 files changed, 8 insertions(+), 8 deletions(-) -- 2.7.4

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Re: [alsa-devel] ALSA sound core device deinit crash
by Takashi Iwai 11 Jul '16

11 Jul '16

(Please don't drop Cc to ML) On Fri, 08 Jul 2016 07:35:37 +0200, b_lkasam(a)codeaurora.org wrote: > > On 2016-07-08 10:58, Takashi Iwai wrote: > > On Fri, 08 Jul 2016 07:19:05 +0200, > > b_lkasam(a)codeaurora.org wrote: > >> > >> Hi Alsa team, > >> There is kernel crash observed when soundcard register failure case as > >> below -> > >> > >> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c old mode > >> 100644 new mode 100755 index 0495890..60a1eb0 > >> --- a/sound/soc/soc-core.c > >> +++ b/sound/soc/soc-core.c > >> @@ -1382,6 +1382,7 @@ static int soc_probe_link_dais(struct > >> snd_soc_card *card, int num, int order) > >> /* do machine specific initialization */ if (dai_link->init) { ret = > >> dai_link->init(rtd); > >> + ret = -ENODEV; // -> we can force error here to reproduce crash > >> easily. > >> if (ret < 0) { > >> dev_err(card->dev, "ASoC: failed to init %s: %d\n", dai_link->name, > >> ret); > >> > >> If sound card fails at initialize at above, it is crashing in > >> pcm_chmap_ctl_private_free(). > >> > >> <1>[ 40.646642] [01-01-2016 00:07:16 CPU:0x3] Unable to handle > >> kernel > >> paging request at virtual address ffffffc0da644b68 > >> <1>[ 40.646664] [01-01-2016 00:07:16 CPU:0x3] pgd = ffffffc001d28000 > >> <1>[ 40.646673] [01-01-2016 00:07:16 CPU:0x3] [ffffffc0da644b68] > >> *pgd=00000000857fc003, *pud=00000000857fc003, *pmd=000000017ddd8003, > >> *pte=00c000015a644793 > >> <0>[ 40.646697] [01-01-2016 00:07:16 CPU:0x3] Internal error: Oops: > >> 9600004f [#1] PREEMPT SMP > >> <6>[ 40.646708] [01-01-2016 00:07:16 CPU:0x3] Modules linked in: > >> brcm_bt_drv fm_drv brcm_hci_ldisc texfat(PO) > >> <6>[ 40.646735] [01-01-2016 00:07:16 CPU:0x3] CPU: 3 PID: 299 Comm: > >> kworker/u8:8 Tainted: P O 3.18.24-g41b2dda2-00002-gbe25a74 > >> #1 > >> <6>[ 40.646744] [01-01-2016 00:07:16 CPU:0x3] Hardware name: > >> Qualcomm > >> Technologies, Inc. MSM 8996 v3.x + PMI8996 MTP (DT) > >> <6>[ 40.646763] [01-01-2016 00:07:16 CPU:0x3] Workqueue: deferwq > >> deferred_probe_work_func > >> <6>[ 40.646773] [01-01-2016 00:07:16 CPU:0x3] task: ffffffc0e951c880 > >> ti: ffffffc03368c000 task.ti: ffffffc03368c000 > >> <6>[ 40.646787] [01-01-2016 00:07:16 CPU:0x3] PC is at > >> pcm_chmap_ctl_private_free+0x1c/0x2c > >> <6>[ 40.646798] [01-01-2016 00:07:16 CPU:0x3] LR is at > >> snd_ctl_free_one+0x20/0x34 > >> > >> > >> FIX: > >> > >> Can you look at the change below and share your comments on this? > >> diff --git a/sound/core/device.c b/sound/core/device.c old mode 100644 > >> new mode 100755 index 41bec30..eaffde1 > >> --- a/sound/core/device.c > >> +++ b/sound/core/device.c > >> @@ -219,6 +219,7 @@ void snd_device_free_all(struct snd_card *card) > >> > >> if (snd_BUG_ON(!card)) > >> return; > >> - list_for_each_entry_safe_reverse(dev, next, &card->devices, > >> list) > >> + list_for_each_entry_safe(dev, next, &card->devices, > >> list) > >> __snd_device_free(dev); } > >> > >> > >> Since control sound device has the lowest type value > >> (SNDRV_DEV_CONTROL), it will be the first entry linked in the > >> card->devices linked list head and will be the last one to be freed. > >> > >> This issue seems to be resolved by modifying the sequence the sound > >> devices in the card->devices list are freed as shown below (from > >> “prev” > >> direction to “next” direction) but I’m not sure if this is a right > >> approach from ALSA perspective. > > > > This doesn't look correct. The strange thing is that this error > > shouldn't happen no matter which free loop direction is. The chmap > > ctl should have been already removed by the disconnection before > > freeing. It means that either the disconnection isn't done properly > > or something else is missing. > > > > Could you give the full stack trace? It's important to know which > > code path triggers it. > > > > > > thanks, > > > > Takashi > > > Hi Takashi, > > Here is full stack trace --> > > > [Callstack] > : (struct snd_pcm_chmap *)info = 0xFFFFFFC0DA6A5200 > : (struct snd_pcm *)pcm = 0xFFFFFFC0DA644A80 //part of buddy page, > read-only > : info->stream = 0 > : snd_card_free() was called > > -012|pcm_chmap_ctl_private_free() > //info->pcm->streams[info->stream].chmap_kctl = NULL > -013|snd_ctl_free_one() > -014|snd_ctl_remove() > -015|snd_ctl_dev_free() > -016|__snd_device_free() > -017|snd_device_free_all() > -018|snd_card_do_free(inline) > -018|release_card_device() > -019|device_release() > -020|kobject_cleanup(inline) > -020|kobject_release() > -021|kobject_put() > -022|put_device() > -023|snd_card_free_when_closed() > -024|snd_card_free() > -025|snd_soc_instantiate_card(inline) -> //Here instantiate card > failed for some reason, then triggers snd_card_free > -025|snd_soc_register_card() > -026|msm8996_asoc_machine_probe() > -027|platform_drv_probe() > -028|really_probe(inline) > -028|driver_probe_device() > -029|__device_attach() > -030|bus_for_each_drv() > -031|device_attach() > -032|bus_probe_device() > -033|deferred_probe_work_func() > -034|static_key_count(inline) > -034|static_key_false(inline) > -034|trace_workqueue_execute_end(inline) > -034|process_one_work() > -035|worker_thread() > -036|kthread() > -037|ret_from_fork(asm) > ---|end of frame > > > ----------------------- > trigger point in soc-core.c > in API snd_soc_instantiate_card(), > > card_probe_error: > if (card->remove) > card->remove(card); > > snd_card_free(card->snd_card); -> this is > snd_card_free which internally leads to above function call. > ------------------------------------------ OK, thanks. So this is the case where it frees before registering, and indeed there is a bug in the PCM chmap code. It's freed at disconnection but the disconnect is called only when it was registered. Below is a quick fix. Give it a try. thanks, Takashi --- diff --git a/sound/core/control.c b/sound/core/control.c index 9ff081cd03f4..fb096cb20a80 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -160,6 +160,8 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; + if (card->shutdown) + return; read_lock(&card->ctl_files_rwlock); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 308c9ecf73db..8e980aa678d0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -849,6 +849,14 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, } EXPORT_SYMBOL(snd_pcm_new_internal); +static void free_chmap(struct snd_pcm_str *pstr) +{ + if (pstr->chmap_kctl) { + snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl); + pstr->chmap_kctl = NULL; + } +} + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -871,6 +879,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) kfree(setup); } #endif + free_chmap(pstr); if (pstr->substream_count) put_device(&pstr->dev); } @@ -1135,10 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) for (cidx = 0; cidx < 2; cidx++) { if (!pcm->internal) snd_unregister_device(&pcm->streams[cidx].dev); - if (pcm->streams[cidx].chmap_kctl) { - snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); - pcm->streams[cidx].chmap_kctl = NULL; - } + free_chmap(&pcm->streams[cidx]); } mutex_unlock(&pcm->open_mutex); mutex_unlock(&register_mutex);

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[alsa-devel] [alsa-lib][PATCH] hctl: remove assertion from element event handler because removal event overwrites all of queued events
by Takashi Sakamoto 11 Jul '16

11 Jul '16

In ALSA control core, when several events occurs for an element, they're represented bit mask (struct snd_ctl_event.data.elem.mask). Thus userspace applications can handles these events separately in one I/O operation. There's an exception; removal event. This is represented by all of bits in the mask. Therefore, when a removal event occurs for an element, all of queued events for the element are overwritten. This works worse for a combination of applications which add/remove control elements and applications which keep a cache of elements. For example, let's assume a case that: 1.some elements are added by ioctl(SNDRV_CTL_IOCTL_ELEM_ADD). 2.ALSA control core handles the request and queues 'add' event. 3.applications don't read the event yet. 4.the elements are removed by ioctl(SNDRV_CTL_IOCTL_ELEM_REMOVE) 5.ALSA control core handles the request and queues 'removal' event by overwriting queued events. 6.applications read the event at last. 7.the applications cannot find removed elements in their cache data. Especially, when applications use hctl interface of ALSA userspace library, this situation occurs because the interface is designed to keep element cache implicitly inner the library. PulseAudio is such an application. When executing a test program of ALSA library (test/user-ctl-element-set.c), pulseaudio process aborts due to assertion at calls of snd_hctl_handle_events(). The rule to represent events is in ALSA kernel/userspace interfaces (UAPI header) and it's difficult to improve the rule. Therefore, developers should pay enough attention to the fact that userspace applications can receive removal events for elements which are not notified with 'add' event. For this reason, this commit removes assertion from event handler of ALSA hctl API. CC: pulseaudio-discuss(a)lists.freedesktop.org Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/hcontrol.c | 1 - 1 file changed, 1 deletion(-) diff --git a/src/control/hcontrol.c b/src/control/hcontrol.c index 7645c57..c40a807 100644 --- a/src/control/hcontrol.c +++ b/src/control/hcontrol.c @@ -742,7 +742,6 @@ static int snd_hctl_handle_event(snd_hctl_t *hctl, snd_ctl_event_t *event) if (event->data.elem.mask == SNDRV_CTL_EVENT_MASK_REMOVE) { int dir; res = _snd_hctl_find_elem(hctl, &event->data.elem.id, &dir); - assert(res >= 0 && dir == 0); if (res < 0 || dir != 0) return -ENOENT; snd_hctl_elem_remove(hctl, (unsigned int) res); -- 2.7.4

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[alsa-devel] [alsa-lib][PATCH] ctl: improve documentation of APIs to add an element set
by Takashi Sakamoto 11 Jul '16

11 Jul '16

Currently, documentation of APIs to add an element set describes causes of errors partly. But it gives an opinion to describe them thoroughly. This is not good for users. This commit changes the documentation so that it partly describes the causes. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 85485aa..9bb8021 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -358,7 +358,8 @@ static bool validate_element_member_dimension(snd_ctl_elem_info_t *info) * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some arguments include invalid value; i.e. ID field in \a info has no + * name, or the number of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -446,7 +447,8 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some arguments include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -530,7 +532,8 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 127. + * <dd>Some parameters include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 127. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO @@ -591,9 +594,10 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A control element with ID \a id already exists. * <dt>-EINVAL - * <dd>\a element_count is not between 1 to 127, or \a items is not at least - * one, or a string in \a labels is empty, or longer than 63 bytes, or total - * length of the labels requires more than 64 KiB storage. + * <dd>Some arguments include invalid value; i.e. \a element_count is not + * between 1 to 127, or \a items is not at least one, or a string in \a + * labels is empty, or longer than 63 bytes, or total length of the labels + * requires more than 64 KiB storage. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user control elements. * <dt>-ENXIO @@ -677,7 +681,8 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, * <dt>-EBUSY * <dd>A element with ID \a id already exists. * <dt>-EINVAL - * <dd>ID has no name, or the number of members is not between 1 to 511. + * <dd>Some arguments include invalid value; i.e. ID has no name, or the number + * of members is not between 1 to 511. * <dt>-ENOMEM * <dd>Out of memory, or there are too many user elements. * <dt>-ENXIO -- 2.7.4

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[alsa-devel] [PATCH v6] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 11 Jul '16

11 Jul '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v6: - remove enable/disable dapm pins, and to switch codecs via user space - fix "Speaker" dapm route for max98357a - add SND_JACK_LINEOUT flag for da7219 jack detection Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 322 ++++++++++++++++++++ 4 files changed, 350 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..43a537c --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,322 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

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[alsa-devel] [GIT PULL] skylake audio firmware update
by Vinod Koul 09 Jul '16

09 Jul '16

Hello, Please PULL to receive the updated audio firmware for SKL which contains update for supporting multiple languages. The following changes since commit 3ef7857d511ce6a91c5ce609da76c4702651cfa5: amdgpu: update polaris ucode (2016-06-28 14:31:11 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to d4274de33ed9448e91b970225ce9afdb74d5f13e: linux-firmware: intel: Update Skylake audio firmware (2016-07-01 13:10:07 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v951.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v951.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: cht: fix uninit variable warning
by Vinod Koul 08 Jul '16

08 Jul '16

Kbuild bot reports that we might use dai_index uninitialized. sound/soc/intel/boards/cht_bsw_rt5645.c:391:37: warning: 'dai_index' may be used uninitialized in this function [-Wmaybe-uninitialized] Since it is theoretically possible, set it while initializing. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index f26c7b8545ae..56056ed7fcfd 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) char codec_name[16]; struct sst_acpi_mach *mach; const char *i2c_name = NULL; - int dai_index; + int dai_index = 0; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); if (!drv) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix to use the actual size for TLV control
by Vinod Koul 08 Jul '16

08 Jul '16

From: Dharageswari R <dharageswari.r(a)intel.com> DSP expects the actual length of parameters that is set through TLV to be passed in large config set, so pass the actual size received in tlv_control_set() instead of max size. Signed-off-by: Dharageswari R <dharageswari.r(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 13 +++++++++---- sound/soc/intel/skylake/skl-topology.h | 1 + 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index b284b3cf5f94..cc0150fc2601 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -411,7 +411,7 @@ static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w, if (bc->set_params == SKL_PARAM_SET) { ret = skl_set_module_params(ctx, - (u32 *)bc->params, bc->max, + (u32 *)bc->params, bc->size, bc->param_id, mconfig); if (ret < 0) return ret; @@ -446,7 +446,7 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w) continue; mconfig->formats_config.caps = (u32 *)&bc->params; - mconfig->formats_config.caps_size = bc->max; + mconfig->formats_config.caps_size = bc->size; break; } @@ -1054,7 +1054,7 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, if (w->power) skl_get_module_params(skl->skl_sst, (u32 *)bc->params, - bc->max, bc->param_id, mconfig); + bc->size, bc->param_id, mconfig); /* decrement size for TLV header */ size -= 2 * sizeof(u32); @@ -1088,6 +1088,10 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, struct skl *skl = get_skl_ctx(w->dapm->dev); if (ac->params) { + if (size > ac->max) + return -EINVAL; + + ac->size = size; /* * if the param_is is of type Vendor, firmware expects actual * parameter id and size from the control. @@ -1103,7 +1107,7 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, if (w->power) return skl_set_module_params(skl->skl_sst, - (u32 *)ac->params, ac->max, + (u32 *)ac->params, ac->size, ac->param_id, mconfig); } @@ -1719,6 +1723,7 @@ static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, ac->max = dfw_ac->max; ac->param_id = dfw_ac->param_id; ac->set_params = dfw_ac->set_params; + ac->size = dfw_ac->max; if (ac->max) { ac->params = (char *) devm_kzalloc(dev, ac->max, GFP_KERNEL); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 170d68b4e12d..22d3ef83817d 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -321,6 +321,7 @@ struct skl_algo_data { u32 param_id; u32 set_params; u32 max; + u32 size; char *params; }; -- 1.9.1

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[alsa-devel] [PATCH 0/3] Add audio support for Microsoft Surface 3 tablet
by Stephen Just 08 Jul '16

08 Jul '16

This series adds audio support for the Surface 3 tablet. This tablet identifes an rt5645 audio codec using the ACPI identifer 10EC5640, which is already associated with the rt5640, thus a quirk is required to have the correct drivers loaded. Additionally, in order for jack-detection and the tablet's internal microphone to work, the Surface 3 must be configured in the audio codec driver similarly to other devices. Resolves: https://bugzilla.kernel.org/show_bug.cgi?id=98001 UCM configuration to be added to alsa-lib is included in patch 3, not included in the diffstat below. Stephen Just (2): ASoC: rt5645: Add support for Surface 3 tablet ASoC: Intel: Atom: add quirk for CHT w/ RT5645 sound/soc/codecs/rt5645.c | 7 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 28 +++++++++++++++++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 1 + 3 files changed, 35 insertions(+), 1 deletion(-) -- 2.7.4

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[alsa-devel] [RFC 00/11] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 08 Jul '16

08 Jul '16

HDA devices generically can be modelled with DAPM in ASoC. This series adds a framework in ASoC to model HDA devices. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Hardik T Shah (1): ASoC: Add dai_ops to set the stream tag. Subhransu S. Prusty (10): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ASoC: hdac: Create DAPM model for HDA widgets ASoC: dapm: Create API to add a single route element ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: dapm: Export snd_soc_dapm_new_control ASoC: hdac: Export API to create machine controls include/sound/hdaudio.h | 1 + include/sound/soc-dai.h | 12 + include/sound/soc-dapm.h | 5 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_codec.c | 188 +++++ sound/hda/ext/hdac_codec.h | 52 ++ sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1561 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 31 + sound/soc/soc-core.c | 35 + sound/soc/soc-dapm.c | 22 + 13 files changed, 1929 insertions(+), 1 deletion(-) create mode 100644 sound/hda/ext/hdac_codec.c create mode 100644 sound/hda/ext/hdac_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - add new pin definition in alc225 pin quirk table
by Hui Wang 08 Jul '16

08 Jul '16

We have some Dell laptops which can't detect headset mic, the machines use the codec ALC225, they have some new pin configuration values, after adding them in the alc225 pin quirk table, they work well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- comparing with the previous patch (I said the patch should be skipped), this patch adds one more group of pin definition in the pin quirk table. sound/pci/hda/patch_realtek.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 45d4dc1..ddd29b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5753,7 +5753,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {} }; #define ALC225_STANDARD_PINS \ - {0x12, 0xb7a60130}, \ {0x21, 0x04211020} #define ALC256_STANDARD_PINS \ @@ -5778,10 +5777,24 @@ static const struct hda_model_fixup alc269_fixup_models[] = { static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, {0x14, 0x901701a0}), SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, {0x14, 0x901701b0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, + {0x14, 0x901701a0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, + {0x14, 0x901701b0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, + {0x1b, 0x90170110}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x02211020}), -- 1.9.1

2 1

[alsa-devel] ALSA sound core device deinit crash
by b_lkasam＠codeaurora.org 08 Jul '16

08 Jul '16

Hi Alsa team, There is kernel crash observed when soundcard register failure case as below -> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c old mode 100644 new mode 100755 index 0495890..60a1eb0 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1382,6 +1382,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* do machine specific initialization */ if (dai_link->init) { ret = dai_link->init(rtd); + ret = -ENODEV; // -> we can force error here to reproduce crash easily. if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", dai_link->name, ret); If sound card fails at initialize at above, it is crashing in pcm_chmap_ctl_private_free(). <1>[ 40.646642] [01-01-2016 00:07:16 CPU:0x3] Unable to handle kernel paging request at virtual address ffffffc0da644b68 <1>[ 40.646664] [01-01-2016 00:07:16 CPU:0x3] pgd = ffffffc001d28000 <1>[ 40.646673] [01-01-2016 00:07:16 CPU:0x3] [ffffffc0da644b68] *pgd=00000000857fc003, *pud=00000000857fc003, *pmd=000000017ddd8003, *pte=00c000015a644793 <0>[ 40.646697] [01-01-2016 00:07:16 CPU:0x3] Internal error: Oops: 9600004f [#1] PREEMPT SMP <6>[ 40.646708] [01-01-2016 00:07:16 CPU:0x3] Modules linked in: brcm_bt_drv fm_drv brcm_hci_ldisc texfat(PO) <6>[ 40.646735] [01-01-2016 00:07:16 CPU:0x3] CPU: 3 PID: 299 Comm: kworker/u8:8 Tainted: P O 3.18.24-g41b2dda2-00002-gbe25a74 #1 <6>[ 40.646744] [01-01-2016 00:07:16 CPU:0x3] Hardware name: Qualcomm Technologies, Inc. MSM 8996 v3.x + PMI8996 MTP (DT) <6>[ 40.646763] [01-01-2016 00:07:16 CPU:0x3] Workqueue: deferwq deferred_probe_work_func <6>[ 40.646773] [01-01-2016 00:07:16 CPU:0x3] task: ffffffc0e951c880 ti: ffffffc03368c000 task.ti: ffffffc03368c000 <6>[ 40.646787] [01-01-2016 00:07:16 CPU:0x3] PC is at pcm_chmap_ctl_private_free+0x1c/0x2c <6>[ 40.646798] [01-01-2016 00:07:16 CPU:0x3] LR is at snd_ctl_free_one+0x20/0x34 FIX: Can you look at the change below and share your comments on this? diff --git a/sound/core/device.c b/sound/core/device.c old mode 100644 new mode 100755 index 41bec30..eaffde1 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -219,6 +219,7 @@ void snd_device_free_all(struct snd_card *card) if (snd_BUG_ON(!card)) return; - list_for_each_entry_safe_reverse(dev, next, &card->devices, list) + list_for_each_entry_safe(dev, next, &card->devices, list) __snd_device_free(dev); } Since control sound device has the lowest type value (SNDRV_DEV_CONTROL), it will be the first entry linked in the card->devices linked list head and will be the last one to be freed. This issue seems to be resolved by modifying the sequence the sound devices in the card->devices list are freed as shown below (from “prev” direction to “next” direction) but I’m not sure if this is a right approach from ALSA perspective. Thanks Laxminath Kasam

2 1

[alsa-devel] [alsa-lib][PATCH] test: add linker flag with pthread for pcm-multi-thread stress test
by Takashi Sakamoto 08 Jul '16

08 Jul '16

This test program is a pthread application, while Makefile has no flags for linker about pthread library. This can cause below error. $ make pcm-multi-thread CCLD pcm-multi-thread /usr/bin/ld: pcm-multi-thread.o: undefined reference to symbol 'pthread_cancel@@GLIBC_2.2.5' //lib/x86_64-linux-gnu/libpthread.so.0: error adding symbols: DSO missing from command line collect2: error: ld returned 1 exit status Makefile:523: recipe for target 'pcm-multi-thread' failed make: *** [pcm-multi-thread] Error 1 This commit fixes the bug by adding a corresponding flag. Fixes: 147093ac5b8d('test: Add pcm-multi-thread program') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/test/Makefile.am b/test/Makefile.am index 970595a..5f35159 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -24,7 +24,7 @@ code_CFLAGS=-Wall -pipe -g -O2 chmap_LDADD=../src/libasound.la audio_time_LDADD=../src/libasound.la pcm_multi_thread_LDADD=../src/libasound.la - +pcm_multi_thread_LDFLAGS=-lpthread user_ctl_element_set_LDADD=../src/libasound.la user_ctl_element_set_CFLAGS=-Wall -g -- 2.7.4

2 1

[alsa-devel] [alsa-lib][PATCH v2] ctl: add dimension validator
by Takashi Sakamoto 08 Jul '16

08 Jul '16

Linux 4.7 or former have no validator of dimension information. This can causes an issue related to user-defined element set. For example, When calculated total members in multi-dimensional matrix is larger than actual capacity of snd_ctl_elem_value_t, processes to handle the element can cause buffer-over-run. For backward portability of this userspace library, this commit adds a validator of dimension information. When userspace applications give invalid dimension information to APIs to add element set, they receive -EINVAL. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/src/control/control.c b/src/control/control.c index 2ad3e0d..85485aa 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -91,6 +91,7 @@ I/O operations. #include <fcntl.h> #include <signal.h> #include <sys/poll.h> +#include <stdbool.h> #include "control_local.h" /** @@ -302,6 +303,32 @@ int snd_ctl_elem_info(snd_ctl_t *ctl, snd_ctl_elem_info_t *info) return ctl->ops->element_info(ctl, info); } +static bool validate_element_member_dimension(snd_ctl_elem_info_t *info) +{ + unsigned int members; + unsigned int i; + + if (info->dimen.d[0] == 0) + return true; + + members = 1; + for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] == 0) + break; + members *= info->dimen.d[i]; + + if (members > info->count) + return false; + } + + for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] > 0) + return false; + } + + return members == info->count; +} + /** * \brief Create and add some user-defined control elements of integer type. * \param ctl A handle of backend module for control interface. @@ -366,6 +393,9 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.max = max; info->value.integer.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -451,6 +481,9 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer64.max = max; info->value.integer64.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -524,6 +557,9 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.min = 0; info->value.integer.max = 1; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } @@ -605,6 +641,9 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, p += strlen(labels[i]) + 1; } + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); free(buf); @@ -663,6 +702,9 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->owner = element_count; info->count = member_count; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } -- 2.7.4

2 1

[alsa-devel] [alsa-lib][PATCH] ctl: add dimension validator
by Takashi Sakamoto 08 Jul '16

08 Jul '16

Linux 4.7 or former have no validator of dimension information. This can causes an issue related to user-defined element set. For example, When calculated total members in multi-dimensional matrix is larger than actual capacity of snd_ctl_elem_value_t, processes to handle the element can cause buffer-over-run. For backward portability of this userspace library, this commit adds a validator of dimension information. When userspace applications give invalid dimension information to APIs to add element set, they receive -EINVAL. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/src/control/control.c b/src/control/control.c index 2ad3e0d..ee6b1d8 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -91,6 +91,7 @@ I/O operations. #include <fcntl.h> #include <signal.h> #include <sys/poll.h> +#include <stdbool.h> #include "control_local.h" /** @@ -302,6 +303,32 @@ int snd_ctl_elem_info(snd_ctl_t *ctl, snd_ctl_elem_info_t *info) return ctl->ops->element_info(ctl, info); } +static bool validate_element_member_dimension(struct snd_ctl_elem_info *info) +{ + unsigned int members; + unsigned int i; + + if (info->dimen.d[0] == 0) + return true; + + members = 1; + for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] == 0) + break; + members *= info->dimen.d[i]; + + if (members > info->count) + return false; + } + + for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] > 0) + return false; + } + + return members == info->count; +} + /** * \brief Create and add some user-defined control elements of integer type. * \param ctl A handle of backend module for control interface. @@ -366,6 +393,9 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.max = max; info->value.integer.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -451,6 +481,9 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer64.max = max; info->value.integer64.step = step; + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); if (err < 0) return err; @@ -524,6 +557,9 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->value.integer.min = 0; info->value.integer.max = 1; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } @@ -605,6 +641,9 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, p += strlen(labels[i]) + 1; } + if (!validate_element_member_dimension(info)) + return -EINVAL; + err = ctl->ops->element_add(ctl, info); free(buf); @@ -663,6 +702,9 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_info_t *info, info->owner = element_count; info->count = member_count; + if (!validate_element_member_dimension(info)) + return -EINVAL; + return ctl->ops->element_add(ctl, info); } -- 2.7.4

2 2

[alsa-devel] [alsa-lib][PATCH] ctl: fix a test for user-defined element set in a point of dimension
by Takashi Sakamoto 07 Jul '16

07 Jul '16

Currently, this test program has a bug related to dimension information. When testing an element set with byte type, the program adds 512 members to an element. This is a maximum number permitted for this type. On the other hand, dimension information of the elements consists of [8, 4, 8, 4]. Calculated members in the multi-dimensional matrix is 1024. This should be less than 512. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index c346e03..0481300 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -513,7 +513,7 @@ int main(void) trial.dimension[0] = 8; trial.dimension[1] = 4; trial.dimension[2] = 8; - trial.dimension[3] = 4; + trial.dimension[3] = 2; trial.add_elem_set = add_bytes_elem_set; trial.check_elem_props = NULL; trial.change_elem_members = change_bytes_elem_members; -- 2.7.4

2 1

[alsa-devel] [PATCH alsa-lib 0/3] Misc fix and pcm-multi-thread test
by Takashi Iwai 07 Jul '16

07 Jul '16

Hi, this is an excerpt from my previous RFC patchset for thread safety, containing a trivial cleanup and an addition of pcm-multi-thread test program, where both can be applied generically without thread-safe patches. It's just a resend, as these have no changes from the previous post. Takashi === Takashi Iwai (3): pcm: Remove superfluous rmb() from PCM meter plugin test: Add pcm-multi-thread program Add pcm-multi-thread to .gitignore .gitignore | 1 + src/pcm/pcm_meter.c | 1 - test/Makefile.am | 3 +- test/pcm-multi-thread.c | 263 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 test/pcm-multi-thread.c -- 2.9.0

1 3

[alsa-devel] [PATCH v3] ALSA: control: add dimension validator for userspace elements
by Takashi Sakamoto 07 Jul '16

07 Jul '16

The 'dimen' field in struct snd_ctl_elem_info is used to compose all of members in the element as multi-dimensional matrix. The field has four members. Each member represents the width in each dimension level by element member unit. For example, if the members consist of typical two dimensional matrix, the dimen[0] represents the number of rows and dimen[1] represents the number of columns (or vise-versa). The total members in the matrix should be exactly the same as the number of members in the element, while current implementation has no validator of this information. In a view of userspace applications, the information must be valid so that it cannot cause any bugs such as buffer-over-run. This commit adds a validator of dimension information for userspace applications which add new element sets. When they add the element sets with wrong dimension information, they receive -EINVAL. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index a85d455..9ff081c 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -805,6 +805,36 @@ static int snd_ctl_elem_list(struct snd_card *card, return 0; } +static bool validate_element_member_dimension(struct snd_ctl_elem_info *info) +{ + unsigned int members; + unsigned int i; + + if (info->dimen.d[0] == 0) + return true; + + members = 1; + for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] == 0) + break; + members *= info->dimen.d[i]; + + /* + * info->count should be validated in advance, to guarantee + * calculation soundness. + */ + if (members > info->count) + return false; + } + + for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) { + if (info->dimen.d[i] > 0) + return false; + } + + return members == info->count; +} + static int snd_ctl_elem_info(struct snd_ctl_file *ctl, struct snd_ctl_elem_info *info) { @@ -1272,6 +1302,8 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1 || info->count > max_value_counts[info->type]) return -EINVAL; + if (!validate_element_member_dimension(info)) + return -EINVAL; private_size = value_sizes[info->type] * info->count; /* -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: rt5645: add DAC1 soft volume func control
by Bard Liao 07 Jul '16

07 Jul '16

This patch add an alsa control for DAC1 digital volume control function selection. The options are: 0: Gain update immediately 1: Gain update when a zero crossing 2: Gain update when a zero crossing with a soft ramp Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2 * Use SOC_ENUM --- sound/soc/codecs/rt5645.c | 11 +++++++++++ sound/soc/codecs/rt5645.h | 3 +++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9251720..aefce8f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -742,6 +742,14 @@ static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol, return ret; } +static const char * const rt5645_dac1_vol_ctrl_mode_text[] = { + "immediately", "zero crossing", "soft ramp" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac1_vol_ctrl_mode, RT5645_PR_BASE, + RT5645_DA1_ZDET_SFT, rt5645_dac1_vol_ctrl_mode_text); + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -808,6 +816,9 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), RT5645_HWEQ("Speaker HWEQ"), + + /* Digital Soft Volume Control */ + SOC_ENUM("DAC1 Digital Volume Control Func", rt5645_dac1_vol_ctrl_mode), }; /** diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 205e071..cfc5f97 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2018,6 +2018,9 @@ /* Codec Private Register definition */ +/* DAC ADC Digital Volume (0x00) */ +#define RT5645_DA1_ZDET_SFT 6 + /* 3D Speaker Control (0x63) */ #define RT5645_3D_SPK_MASK (0x1 << 15) #define RT5645_3D_SPK_SFT 15 -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [patch] ASoC: mediatek: mt2701: fix some error handling in probe
by Dan Carpenter 07 Jul '16

07 Jul '16

The check for if the "afe" allocation failed was too late and there wasn't a check for "afe->platform_priv". Fixes: 43a6a7e71063 ('ASoC: mediatek: add mt2701 platform driver implementation.') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index c865ba1..fc5a87d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1350,11 +1350,13 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) ret = 0; afe = devm_kzalloc(&pdev->dev, sizeof(*afe), GFP_KERNEL); + if (!afe) + return -ENOMEM; afe->platform_priv = devm_kzalloc(&pdev->dev, sizeof(*afe_priv), GFP_KERNEL); - afe_priv = afe->platform_priv; - if (!afe) + if (!afe->platform_priv) return -ENOMEM; + afe_priv = afe->platform_priv; afe->dev = &pdev->dev; dev = afe->dev;

3 2

[alsa-devel] [PATCH 1/2] ASoC: rt5645: set RT5645_PRIV_INDEX as volatile
by Bard Liao 07 Jul '16

07 Jul '16

RT5645_PRIV_INDEX(0x6a) indicate the address of PR- registers. So, it should be volatile. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 97bf96e..9251720 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -441,6 +441,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) switch (reg) { case RT5645_RESET: + case RT5645_PRIV_INDEX: case RT5645_PRIV_DATA: case RT5645_IN1_CTRL1: case RT5645_IN1_CTRL2: -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Fix DMA params for different SSC
by Peter Meerwald 07 Jul '16

07 Jul '16

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> follow-up patch from c706f2e55f ASoC: atmel_ssc_dai: distinguish the different SSC cpu_dai id is always 0, use platform_device id to distinguish DMA parameters of SSCs Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: "Wu, Songjun" <songjun.wu(a)atmel.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..c7daea8 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -320,7 +320,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, return ret; } - dma_params = &ssc_dma_params[dai->id][dir]; + dma_params = &ssc_dma_params[pdev->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; -- 2.9.0

2 1

[alsa-devel] [PATCH 0/3 v3] ALSA: ctl: add dimension information validator
by Takashi Sakamoto 07 Jul '16

07 Jul '16

This patchset is revised version of my previous one: [alsa-devel] [PATCH 0/3 v2] ALSA: ctl: add dimension information validator http://mailman.alsa-project.org/pipermail/alsa-devel/2016-July/109830.html Changes: * Apply simpler prevention of arithmetic overflow. Takashi Sakamoto (3): ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 78 ++++++++++++++++++++++++++++++++--------- sound/pci/echoaudio/echoaudio.c | 6 ++-- 2 files changed, 65 insertions(+), 19 deletions(-) -- 2.7.4

2 10

[alsa-devel] [PATCH] ALSA: hda/realtek - add new pin definition in alc225 pin quirk table
by Hui Wang 07 Jul '16

07 Jul '16

We have some Dell laptops which can't detect headset mic, the machines use the codec ALC225, they have some new pin configuration values, after adding them in the alc225 pin quirk table, they work well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 45d4dc1..e25ef2b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5753,7 +5753,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {} }; #define ALC225_STANDARD_PINS \ - {0x12, 0xb7a60130}, \ {0x21, 0x04211020} #define ALC256_STANDARD_PINS \ @@ -5778,9 +5777,19 @@ static const struct hda_model_fixup alc269_fixup_models[] = { static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, {0x14, 0x901701a0}), SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, + {0x12, 0xb7a60130}, + {0x14, 0x901701b0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, + {0x14, 0x901701a0}), + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC225_STANDARD_PINS, + {0x12, 0xb7a60150}, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: nau8825: jack connection decision with different insertion logic
by John Hsu 06 Jul '16

06 Jul '16

The original design only covers the jack insertion logic is active low. Add more condition to cover no matter the logic is active low and high. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 3f30e6e..a2f0d03 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1345,10 +1345,18 @@ EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); static bool nau8825_is_jack_inserted(struct regmap *regmap) { - int status; + int status, jkdet, res; regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); - return !(status & NAU8825_GPIO2JD1); + regmap_read(regmap, NAU8825_REG_JACK_DET_CTRL, &jkdet); + + /* return jack connection status according to jack insertion logic + * active high or active low. + */ + res = !(status & NAU8825_GPIO2JD1) * !(jkdet & NAU8825_JACK_POLARITY) + + (status & NAU8825_GPIO2JD1) * (jkdet & NAU8825_JACK_POLARITY); + + return res ? true : false; } static void nau8825_restart_jack_detection(struct regmap *regmap) -- 2.6.4

3 5

[alsa-devel] [PATCH 0/6] ASoC: simple-card: use asoc_simple_card_parse_xxx()
by Kuninori Morimoto 06 Jul '16

06 Jul '16

Hi Mark These are v3 of [PATCH 00/39 v2] ASoC: add simple-card-core and standardize "simple" card but as new patch series. These are separated into small pieces. Basically, these patches are util.c has new function -> simple-card use it. I will post next remain patches if previous patches was accepted. It will be 6 - 9 patches / 1 series. Now, util.c on mark/topic/simple branch has daifmt() function. So, 1) - 2) : use new daifmt() 3) - 5) : add new dailink_name() and use it. 6) : tidyup rsrc-card This existing rcar-card will be renamed to simple-dpcm-card in final patch. (Maybe we need to consider more better nameing) Kuninori Morimoto (6): 1) ASoC: simple-card: use asoc_simple_card_parse_daifmt() 2) ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() 3) ASoC: simple-card-utils: add asoc_simple_card_parse_dailink_name() 4) ASoC: simple-card: use asoc_simple_card_parse_dailink_name() 5) ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() 6) ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai include/sound/simple_card.h | 11 +---- include/sound/simple_card_utils.h | 12 +++++ sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-card-utils.c | 30 +++++++++++++ sound/soc/generic/simple-card.c | 60 ++----------------------- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/rsrc-card.c | 83 ++++++++--------------------------- 7 files changed, 67 insertions(+), 131 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 1/2] sound: lpass-cpu: add module licence and description
by Srinivas Kandagatla 06 Jul '16

06 Jul '16

This patch adds module licence to lpass-cpu driver, without this patch lpass-cpu module would taint with below error: snd_soc_lpass_cpu: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint snd_soc_lpass_cpu: Unknown symbol regmap_write (err 0) snd_soc_lpass_cpu: Unknown symbol devm_kmalloc (err 0) ... Acked-by: Kenneth Westfield <kwestfie(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 3cde9fb..eff3f9a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -586,3 +586,6 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

2 2

[alsa-devel] [PATCH v4 3/4] ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem
by Sylwester Nawrocki 05 Jul '16

05 Jul '16

This patch adds DT binding documentation for Exnos5433 based TM2 and TM2E boards sound subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v2: - none. Changes since initial version: - dropped clocks, clock-names properties, instead properties from the CODEC node will be used, - property renames: 'samsung,model' -> 'model', 'samsung,i2s-controller' -> 'i2s-controller', 'samsung,speaker-amplifier' -> 'audio-amplifier', - added 'audio-codec' property. --- .../bindings/sound/samsung,tm2-audio.txt | 36 ++++++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt new file mode 100644 index 0000000..3ae56f3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt @@ -0,0 +1,36 @@ +Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +Required properties: + + - compatible : "samsung,tm2-audio" + - model : the user-visible name of this sound complex + - audio-codec: the phandle of the wm5110 audio codec node, + as described in Documentation/devicetree/bindings/mfd/arizona.txt + - i2s-controller : the phandle of the I2S controller + - audio-amplifier : the phandle of the MAX98504 amplifier + - samsung,audio-routing : a list of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the WM5110's and MAX98504's + pins and the jacks on the board: + HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator + +Example: + +sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>; + i2s-controller = <&i2s0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 0>; + model = "wm5110"; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; +}; -- 1.9.1

1 0

[alsa-devel] [PATCH v4 0/4] ASoC: samsung: Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 05 Jul '16

05 Jul '16

This patch series adds a MFD driver for Samsung Exynos5433 SoC Low Power Audio Subsystem and an ASoC machine driver for Exynos5433 based TM2/TM2E boards. The changes since v3 are listed in individual patches, the main change was rewriting LPASS driver as a MFD driver. It seems much better now as the dependencies between LPASS Top block and its IP sub-blocks are properly taken care of. Sylwester Nawrocki (4): mfd: Add DT bindings documentation for Samsung Exynos LPASS mfd: Add Samsung Exynos Low Power Audio Subsystem driver ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem ASoC: samsung: Add machine driver for Exynos5433 based TM2 board .../bindings/mfd/samsung,exynos5433-lpass.txt | 21 + .../bindings/sound/samsung,tm2-audio.txt | 36 ++ drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/exynos-lpass.c | 185 +++++++ include/linux/mfd/syscon/exynos5-pmu.h | 4 +- sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++ 9 files changed, 844 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 drivers/mfd/exynos-lpass.c create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

1 0

[alsa-devel] [PATCH RFC alsa-lib 0/5] Add thread-safety to PCM API
by Takashi Iwai 05 Jul '16

05 Jul '16

Hi, here is an experimental patchset to add multi thread safety to PCM functions. Basically ALSA PCM functions are thread-unsafe, and applications are supposed to do the proper protection against racy accesses. The reality is, however, that application developers don't care such, as alsa-lib works in most cases. Of course, users still get occasionally mysterious crashes. As a workaround, this patchset adds the pthread mutex protection to most of exported PCM functions. This is slightly an overkill, but the biggest merit is that it's easy to implement; I just wrapped the functions and replaced the internal ones with unlocked versions. To be noted, there is an optimization for the direct hw PCM access. Performance-sensitive applications like JACK should work like before without any overhead. Another bonus by this addition is that we can finally get rid of home brew (and deadly smelling) atomic macros from the tree, since it's now more widely protected. I lightly tested on my local machines and it seems working well, so far. Comments and suggestions welcome. thanks, Takashi === Takashi Iwai (5): pcm: Add thread-safety to PCM API test: Add pcm-multi-thread program Add pcm-multi-thread to .gitignore pcm: Remove superfluous rmb() from PCM meter plugin pcm: Remove home brew atomic operations .gitignore | 1 + configure.ac | 15 ++ include/Makefile.am | 2 +- include/iatomic.h | 170 ------------------ src/pcm/Makefile.am | 2 +- src/pcm/atomic.c | 43 ----- src/pcm/pcm.c | 468 +++++++++++++++++++++++++++++++++++++++++------- src/pcm/pcm_direct.c | 4 +- src/pcm/pcm_dmix.c | 13 +- src/pcm/pcm_dshare.c | 13 +- src/pcm/pcm_dsnoop.c | 15 +- src/pcm/pcm_file.c | 21 ++- src/pcm/pcm_generic.c | 10 +- src/pcm/pcm_hw.c | 3 + src/pcm/pcm_ioplug.c | 11 +- src/pcm/pcm_local.h | 62 ++++++- src/pcm/pcm_meter.c | 1 - src/pcm/pcm_mmap.c | 16 +- src/pcm/pcm_params.c | 2 +- src/pcm/pcm_plugin.c | 72 ++------ src/pcm/pcm_plugin.h | 2 - src/pcm/pcm_rate.c | 41 ++--- src/pcm/pcm_route.c | 2 +- test/Makefile.am | 3 +- test/pcm-multi-thread.c | 263 +++++++++++++++++++++++++++ 25 files changed, 851 insertions(+), 404 deletions(-) delete mode 100644 include/iatomic.h delete mode 100644 src/pcm/atomic.c create mode 100644 test/pcm-multi-thread.c -- 2.9.0

2 7

[alsa-devel] [PATCH v2 0/6] Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 05 Jul '16

05 Jul '16

This patch series adds sound supoort for the Exynos 5433 based TM2 and TM2E board. It includes minimal drivers for the Low Power Audio Subsystem (LPASS) and the MAX98504 class D speaker amplifier as well as the actual asoc machine driver. The changes are listed in individual patches, there were only v1 3/6, 4/6 patches (max98504), for others I skipped from RFC to v2. Inha Song (2): ASoC: samsung: Add Samsung Low Power Audio Subsystem driver ASoC: samsung: Add machine driver for Exynos5433 based TM2 board Sylwester Nawrocki (4): ASoC: samsung: Add DT bindings documentation for LPASS ASoC: Add DT bindings documentation for max98504 amplifier ASoC: max98504: Add max98504 speaker amplifier driver ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem .../devicetree/bindings/sound/max98504.txt | 26 + .../bindings/sound/samsung,exynos5433-lpass.txt | 8 + .../bindings/sound/samsung,tm2-audio.txt | 36 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98504.c | 356 +++++++++++++ sound/soc/codecs/max98504.h | 76 +++ sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 4 + sound/soc/samsung/lpass.c | 162 ++++++ sound/soc/samsung/lpass.h | 47 ++ sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++ 12 files changed, 1313 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/samsung/lpass.c create mode 100644 sound/soc/samsung/lpass.h create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

3 20

Re: [alsa-devel] [PATCH] hda/realtek: Add Lenovo L460 to docking unit fixup
by Takashi Iwai 05 Jul '16

05 Jul '16

On Tue, 05 Jul 2016 10:40:22 +0200, Torsten Hilbrich wrote: > > This solves the issue that a headphone is not working on the docking > unit. > > Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH 0/4] ASoC: Mediatek: add MT2701 platform and machine driver
by Garlic Tseng 05 Jul '16

05 Jul '16

Some patches of the patchset "ASoC: Mediatek: Add support for MT2701 SOC" have been applied to the broonie tree, so I send a new patchset for the remaining patches. The patchset adds mt2701 support based on Mediatek AFE common structure. Change since the previous patchset: * remove unnecessary set_clkdiv and set_fmt ops in mt2701 platform driver * add another snd_soc_dai_driver for bt-sco codec to support 8k and 16k Garlic Tseng (4): ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- sound/soc/codecs/bt-sco.c | 52 +- sound/soc/mediatek/Kconfig | 21 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 9 - sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1654 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 412 +++++ 8 files changed, 2144 insertions(+), 26 deletions(-) create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c -- 1.9.1

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Re: [alsa-devel] ASUS STRIX sound card quiet microphone (like, really quiet ...)
by Clemens Ladisch 05 Jul '16

05 Jul '16

auspicious(a)inventati.org wrote: > On 05/07/2016 07:19, Clemens Ladisch wrote: >> auspicious(a)inventati.org wrote: >>> I currently am using an ASUS STRIX RAID DLX sound card. It contains the following processor: >>> C-Media USB2.0 6632AX High-Definition Sound Processor (Max. 384KHz / 24bit) >>> >>> the microphone bit has always been troublesome. >>> it started working but it is incredibly quiet. >> >> You did not mention mixer controls. >> Does it have any? (Try alsamixer.) > > Attached image in the attachment That "Input Gain" looks OK. Does it actually work? Does it make the mic input quieter? It's possible that there is some vendor-specific stuff that the USB audio driver does not know. Does it work in Windows with Microsoft's generic USB audio driver, or does it need a custom driver from Asus? Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: rt5645: disable soft volume control
by Bard Liao 05 Jul '16

05 Jul '16

With soft volume control function, rt5645/rt5650 will start with 0dB volume and take a little time to adjust volume to given value. And it will cause a bad user experience. This patch disable it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 97bf96e..c969f48 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -63,6 +63,7 @@ static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, + {RT5645_PR_BASE + 0x00, 0x0a2a}, {RT5645_ASRC_4, 0x0120}, }; -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] ASUS STRIX sound card quiet microphone (like, really quiet ...)
by auspicious＠inventati.org 05 Jul '16

05 Jul '16

Dear mailing list individuals, I currently am using an ASUS STRIX RAID DLX sound card. It contains the following processor: C-Media USB2.0 6632AX High-Definition Sound Processor (Max. 384KHz / 24bit) More information: https://www.asus.com/Sound-Cards/STRIX-RAID-DLX/specifications It gets detected as so on Debian testing with Linux kernel 4.6: card 1: DLX [STRIX RAID DLX], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DLX [STRIX RAID DLX], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DLX [STRIX RAID DLX], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 The output of the card has been wonderfully fine ever since I have had it, but the microphone bit has always been troublesome. On Linux Mint 17.3 it would not work (kernel 4.1 I believe). On Linux Mint 18 (kernel 4.4) it started working but it is incredibly quiet. On the loudest setting, I have to put the microphone against my mouth for anyone to hear. On the same system using on-board Intel audio, this issue does not occur. I am using Debian testing to have more recent software, but nothing made a difference. What can I do about this? I do not really know whether I need to contact the Linux kernel guys or ALSA. I am rather inexperienced with Linux driver things. Sincerely yours, Desperate Individual

2 1

[alsa-devel] SND_PCM_STREAM_CAPTURE and SND_PCM_NONBLOCK producing strange artifacts
by Trent Reed 05 Jul '16

05 Jul '16

Hey All, I've been banging my head against the wall for about a week on this bug. The following gist shows my sample reproduction code: https://gist.github.com/TReed0803/985c5d5c3d295245e19006a27be447c3 I'm simply opening up a non-blocking PCM capture stream and writing the contents of the reads to stdout. (Originally, I was writing to the playback stream, but I was hearing this strange static occasionally!) It's the static I'm trying to debug. It doesn't happen every time. In fact, sometimes I'll go a few consecutive executions without hearing it. I was able to capture some of the bad data, and I loaded it up in Audacity for visualization: https://drive.google.com/file/d/0B-1aumGKQcQTcUJoZzIwRWhYSFE/view?usp=shari… It looks like the internal buffer occasionally is sending me more data than it actually captured, and I end up either reading old PCM data from the internal ring buffer, or (at the very beginning) a bunch of zeros. Can anyone help me understand what is going on? What could cause these definitely incorrect samples to be recorded? (I get the same effect regardless of hardware, but I will list hardware just in case.) I hope I have all the information you might need: Hardware: Samson Meteor Mic (USB-Audio, USB Mixer) [Though, it even happens with my built-in microphone] ALSA version: Advanced Linux Sound Architecture Driver Version k4.4.0-28-generic. apt-cache policy (installed alsa-lib version): 1.1.0-0ubuntu1 Thanks, - Trent Reed

2 2

[alsa-devel] [PATCH v2 0/2] Add audio support for Microsoft Surface 3 tablet
by Stephen Just 05 Jul '16

05 Jul '16

This series adds audio support for the Surface 3 tablet. This tablet identifes an rt5645 audio codec using the ACPI identifer 10EC5640, which is already associated with the rt5640, thus a quirk is required to have the correct drivers loaded. Additionally, in order for jack-detection and the tablet's internal microphone to work, the Surface 3 must be configured in the audio codec driver similarly to other devices. Resolves: https://bugzilla.kernel.org/show_bug.cgi?id=98001 Changes in v2: Remove alsa-lib changes from series Stephen Just (2): ASoC: rt5645: Add support for Surface 3 tablet ASoC: Intel: Atom: add quirk for CHT w/ RT5645 sound/soc/codecs/rt5645.c | 7 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 29 ++++++++++++++++++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 1 + 3 files changed, 36 insertions(+), 1 deletion(-) -- 2.7.4

3 6

[alsa-devel] [PATCH] [alsa-lib] conf/ucm: chtrt5645: Add UCM config for chtrt5645
by Stephen Just 05 Jul '16

05 Jul '16

Add a UCM configuration for the rt5645 codec on Intel's Cherry-Trail platform. Tested on the Microsoft Surface 3. Signed-off-by: Stephen Just <stephenjust(a)gmail.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/chtrt5645/HiFi.conf | 209 ++++++++++++++++++++++++++++++++++ src/conf/ucm/chtrt5645/Makefile.am | 4 + src/conf/ucm/chtrt5645/chtrt5645.conf | 5 + 5 files changed, 220 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/chtrt5645/HiFi.conf create mode 100644 src/conf/ucm/chtrt5645/Makefile.am create mode 100644 src/conf/ucm/chtrt5645/chtrt5645.conf diff --git a/configure.ac b/configure.ac index 53fca33..9eebd0a 100644 --- a/configure.ac +++ b/configure.ac @@ -659,6 +659,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ + src/conf/ucm/chtrt5645/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 3cce36a..8865770 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 diff --git a/src/conf/ucm/chtrt5645/HiFi.conf b/src/conf/ucm/chtrt5645/HiFi.conf new file mode 100644 index 0000000..0c2c83c --- /dev/null +++ b/src/conf/ucm/chtrt5645/HiFi.conf @@ -0,0 +1,209 @@ +SectionVerb { + # ALSA PCM + Value { + TQ "HiFi" + + # ALSA PCM device for HiFi + PlaybackPCM "hw:chtrt5645,0" + CapturePCM "hw:chtrt5645,0" + } + + EnableSequence [ + cdev "hw:chtrt5645" + + # Enable audio output path + cset "name='codec_out1 mix 0 pcm0_in Switch' on" + cset "name='media0_out mix 0 media1_in Switch' on" + + cset "name='media1_in Gain 0 Ramp Delay' 50" + cset "name='media1_in Gain 0 Switch' on" + cset "name='media1_in Gain 0 Volume' 80% 80%" + + cset "name='pcm0_in Gain 0 Ramp Delay' 50" + cset "name='pcm0_in Gain 0 Switch' on" + cset "name='pcm0_in Gain 0 Volume' 80% 80%" + + cset "name='codec_out1 Gain 0 Ramp Delay' 50" + cset "name='codec_out1 Gain 0 Switch' on" + cset "name='codec_out1 Gain 0 Volume' 70% 70%" + + # Enable audio input path + cset "name='pcm1_out mix 0 media_loop2_in Switch' on" + cset "name='media_loop2_out mix 0 codec_in0 Switch' on" + + cset "name='codec_in0 Gain 0 Ramp Delay' 50" + cset "name='codec_in0 Gain 0 Switch' on" + cset "name='codec_in0 Gain 0 Volume' 80% 80%" + + cset "name='media_loop2_out Gain 0 Ramp Delay' 50" + cset "name='media_loop2_out Gain 0 Switch' on" + cset "name='media_loop2_out Gain 0 Volume' 80% 80%" + + cset "name='pcm1_out Gain 0 Ramp Delay' 50" + cset "name='pcm1_out Gain 0 Switch' on" + cset "name='pcm1_out Gain 0 Volume' 80% 80%" + + # Output Configuration + cset "name='DAC L2 Mux' 'IF1 DAC'" + cset "name='DAC R2 Mux' 'IF1 DAC'" + cset "name='Mono DAC MIXL DAC L2 Switch' on" + cset "name='Mono DAC MIXR DAC R2 Switch' on" + cset "name='DAC2 Playback Switch' on" + + cset "name='HPOVOL MIXL DAC2 Switch' on" + cset "name='HPOVOL MIXR DAC2 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='HPOVOL L Switch' on" + cset "name='HPOVOL R Switch' on" + + cset "name='SPK MIXL DAC L2 Switch' on" + cset "name='SPK MIXR DAC R2 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" + cset "name='SPOR MIX SPKVOL R Switch' on" + cset "name='SPKVOL L Switch' on" + cset "name='SPKVOL R Switch' on" + + # Input Configuration + cset "name='Stereo1 DMIC Mux' 0" + cset "name='Stereo1 ADC2 Mux' 1" + cset "name='ADC Capture Switch' on" + cset "name='ADC Capture Volume' 31" + cset "name='ADC Boost Capture Volume' 3" + cset "name='Mono ADC Capture Volume' 63" + cset "name='Mono ADC Boost Capture Volume' 2" + cset "name='IN Capture Volume' 63" + cset "name='I2S2 Func Switch' on" + + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + # Disable audio output path + cset "name='codec_out1 mix 0 pcm0_in Switch' off" + cset "name='media0_out mix 0 media1_in Switch' off" + + cset "name='media1_in Gain 0 Switch' off" + cset "name='pcm0_in Gain 0 Switch' off" + cset "name='codec_out1 Gain 0 Switch' off" + + # Disable audio input path + cset "name='pcm1_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop2_out mix 0 codec_in0 Switch' off" + + cset "name='media_loop2_out Gain 0 Switch' off" + cset "name='pcm1_out Gain 0 Switch' off" + cset "name='codec_in0 Gain 0 Switch' off" + ] +} + +SectionDevice."Speaker" { + Comment "Speaker" + + Value { + PlaybackChannels "2" + } + + ConflictingDevice [ + "Headphone" + ] + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Ext Spk Switch' on" + cset "name='Speaker Channel Switch' on" + cset "name='Speaker Playback Volume' 39" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Ext Spk Switch' off" + cset "name='Speaker Channel Switch' off" + ] +} + +SectionDevice."Headphone".0 { + Comment "Headphones" + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headphone Switch' on" + cset "name='Headphone Channel Switch' on" + cset "name='Headphone Playback Volume' 39" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headphone Switch' off" + cset "name='Headphone Channel Switch' off" + ] +} + +SectionDevice."Mic".0 { + Comment "Internal Microphone" + + Value { + CaptureChannels "2" + CapturePriority "150" + } + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Int Mic Switch' on" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + cset "name='Int Mic Switch' off" + ] +} + +SectionDevice."HSMic".0 { + Comment "Headset Microphone" + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + JackHWMute "Mic" + } + + EnableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtrt5645" + + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + ] +} diff --git a/src/conf/ucm/chtrt5645/Makefile.am b/src/conf/ucm/chtrt5645/Makefile.am new file mode 100644 index 0000000..b3a82f5 --- /dev/null +++ b/src/conf/ucm/chtrt5645/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/chtrt5645 +ucm_DATA = chtrt5645.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/chtrt5645/chtrt5645.conf b/src/conf/ucm/chtrt5645/chtrt5645.conf new file mode 100644 index 0000000..dfc36c8 --- /dev/null +++ b/src/conf/ucm/chtrt5645/chtrt5645.conf @@ -0,0 +1,5 @@ +Comment "Intel SoC Audio Device" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.9.0

2 1

[alsa-devel] [PATCH] ALSA: ppc/awacs: shut up maybe-uninitialized warning
by Arnd Bergmann 05 Jul '16

05 Jul '16

The awacs sound driver produces a false-positive warning in ppc64_defconfig: sound/ppc/awacs.c: In function 'snd_pmac_awacs_init': include/sound/control.h:219:9: warning: 'master_vol' may be used uninitialized in this function [-Wmaybe-uninitialized] I haven't come up with a good way to rewrite the code to avoid the warning, so here is a bad one: I initialize the variable before the conditionall initialization so gcc no longer has to worry about it. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/ppc/awacs.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 09da7b52bc2e..1468e4b7bf93 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -991,6 +991,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) if (err < 0) return err; } + master_vol = NULL; if (pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), -- 2.9.0

2 1

[alsa-devel] [PATCH v2 0/7] of_graph: prepare for ALSA graph support
by Kuninori Morimoto 05 Jul '16

05 Jul '16

Hi Rob These are v2 of of_graph patch-set Now OF graph is mainly used by V4L2 SoC, and ALSA SoC is using different style for SoC <-> Codec binding. But, for example, HDMI case, V4L2 <-> ALSA need to collaborate, and then ALSA SoC needs to adjust to OF graph. OTOH, V4L2's "OF graph" position is same as ALSA SoC "sound card" position. And ALSA SoC side want to keep existing supported feature on new OF graph style. I'm posting this on ALSA SoC ML now. Now, current of_graph is indicating port/endpoint, but there is no way to understand that it is for video port ? or sound port ? or other device port ? For example, HDMI has video port, and sound port. Because of this reason, ALSA SoC side can't handle OF graph correctly. Thus, this patch-set tries to add new "type" on OF graph. And this patch-set includes small feature which are useful for ALSA SoC side OF graph support. Kuninori Morimoto (7): Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port_type_is() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() of_graph: add of_graph_get_endpoint_count() Documentation/devicetree/bindings/graph.txt | 21 ++++ drivers/of/base.c | 164 ++++++++++++++++++++++++++-- include/linux/of_graph.h | 61 +++++++++++ 3 files changed, 236 insertions(+), 10 deletions(-) -- 1.9.1

3 12

[alsa-devel] [PATCH] ASoC: rockchip: implement system suspend/resume for spdif
by Sugar Zhang 05 Jul '16

05 Jul '16

restore hw registers after power loss during a suspend/resume cycle. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_spdif.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 100781e..00ae0e5 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -392,9 +392,35 @@ static int rk_spdif_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int rockchip_spdif_suspend(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + + regcache_mark_dirty(spdif->regmap); + + return 0; +} + +static int rockchip_spdif_resume(struct device *dev) +{ + struct rk_spdif_dev *spdif = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(spdif->regmap); + pm_runtime_put(dev); + + return ret; +} +#endif + static const struct dev_pm_ops rk_spdif_pm_ops = { SET_RUNTIME_PM_OPS(rk_spdif_runtime_suspend, rk_spdif_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(rockchip_spdif_suspend, rockchip_spdif_resume) }; static struct platform_driver rk_spdif_driver = { -- 1.9.1

1 0

[alsa-devel] atmel_ssc_dai.c, use of dai->id
by Peter Meerwald-Stadler 05 Jul '16

05 Jul '16

Hello, I am trying to use SSC1 (not SSC0 which might work) on a SAMA5D4-Xplained board (4.1 kernel, but close to mainline) c706f2e5 ASoC: atmel_ssc_dai: distinguish the different SSC fixes an issue with cpu_dai id being always 0, while the ssc id should have been used (or the platform_device id) I think there is a remaining issue with DMA parameters in atmel_ssc_startup(): line 324: dma_params = &ssc_dma_params[dai->id][dir]; I think it should be dma_params = &ssc_dma_params[pdev->id][dir]; I am new to this platform, please comment I can submit a formal patch later-on of course thanks, regards, p. -- Peter Meerwald-Stadler +43-664-2444418 (mobile)

2 1

[alsa-devel] [PATCH] drm: sti: fix prototypes after API change
by Arnd Bergmann 05 Jul '16

05 Jul '16

Like the mediatek hdmi driver, the sti driver now suffers from an API change in linux-next: drivers/gpu/drm/sti/sti_hdmi.c:1250:15: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] .hw_params = hdmi_audio_hw_params, ^~~~~~~~~~~~~~~~~~~~ drivers/gpu/drm/sti/sti_hdmi.c:1250:15: note: (near initialization for 'audio_codec_ops.hw_params') This follows the API change, adding the extra function arguments. I suppose we want this to be part of the ASoC merge in linux-next for now too. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 2c348e505328 ("drm: sti: Add ASoC generic hdmi codec support.") Fixes: efc9194bcff8 ("ASoC: hdmi-codec: callback function will be called with private data") Link: https://lkml.org/lkml/2016/6/29/441 --- drivers/gpu/drm/sti/sti_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index 927e2b2e5af2..fedc17f98d9b 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -1169,7 +1169,7 @@ static int hdmi_audio_configure(struct sti_hdmi *hdmi, return hdmi_audio_infoframe_config(hdmi); } -static void hdmi_audio_shutdown(struct device *dev) +static void hdmi_audio_shutdown(struct device *dev, void *data) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); int audio_cfg; @@ -1186,6 +1186,7 @@ static void hdmi_audio_shutdown(struct device *dev) } static int hdmi_audio_hw_params(struct device *dev, + void *data, struct hdmi_codec_daifmt *daifmt, struct hdmi_codec_params *params) { @@ -1221,7 +1222,7 @@ static int hdmi_audio_hw_params(struct device *dev, return 0; } -static int hdmi_audio_digital_mute(struct device *dev, bool enable) +static int hdmi_audio_digital_mute(struct device *dev, void *data, bool enable) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); @@ -1235,7 +1236,7 @@ static int hdmi_audio_digital_mute(struct device *dev, bool enable) return 0; } -static int hdmi_audio_get_eld(struct device *dev, uint8_t *buf, size_t len) +static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size_t len) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); struct drm_connector *connector = hdmi->drm_connector; -- 2.9.0

5 8

[alsa-devel] Alsa driver for Tascam US-16x08
by Detlef Urban 04 Jul '16

04 Jul '16

Hi, I'm a former application developer at a big IT company. Most of my private time I spend with semi-professional studio recording using Ubuntu Studio (Ardour). I bought a nice external sound device from Tascam called US-16x08 and it works very well. This device also contains some signal processing capabilities like a compressor and a parametric equalizer. But unfortunately there was no Linux driver available from the manufacturer and so I cant use there functionalities. So I started with a nightmare of re-engineering by monitoring USB communication via usbmon and wireshark. Now after 3-4 weeks I'm able to completely control this device with a customized snd-usb-audio kernel module. Additionally I developed a tiny GUI mixer application to test my driver. But I'm not very familiar with non-coding aspects of driver deployment (documentation, deployment etc.). Can some please help me with some hints/links, so that other users can benefit of my work? regards Detlef

1 0

[alsa-devel] [PATCH -next] ASoC: cs35l33: Remove unused including <linux/version.h>
by weiyj_lk＠163.com 04 Jul '16

04 Jul '16

From: Wei Yongjun <weiyj.lk(a)gmail.com> Remove including <linux/version.h> that don't need it. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/codecs/cs35l33.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 622a111..c20edd4 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -12,7 +12,6 @@ */ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH 0/7] ASoC: rockchip: Fix audio on Veyron.
by Enric Balletbo i Serra 04 Jul '16

04 Jul '16

Hi all, This patch series adds support and fixes the audio support for Veyron devices. With these patches I'm able to playback from speakers and headphones, and record from the microphone on a Veyron Jerry Chromebook. Patches 1/7 and 2/7 were already sent by John Keeping,[1] [2], but I think it's interesting have them included here. Patches from 1/7 to 5/7 should go through the Mark Brown ASoC tree. Patches 6/7 and 7/7 should go through the Heiko Stuebner rockchip tree. [1] https://patchwork.kernel.org/patch/9017491/ [2] https://patchwork.kernel.org/patch/9017501/ Enric Balletbo i Serra (5): ASoC: rockchip-max98090: Fix NULL pointer de reference while accessing to jack ASoC: rockchip-max98090: Fix the Headset Mic route. ASoC: rockchip-max98090: Fix DAPM unknown pin for Headset Jack ARM: dts: rockchip: Add shared file for audio related nodes for veyron boards ARM: dts: rockchip: veyron: Add analog audio codecs. John Keeping (2): ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables" ASoC: rockchip: Revert "ASoC: rockchip: i2s: separate capture and playback" arch/arm/boot/dts/rk3288-veyron-analog-audio.dtsi | 102 ++++++++++++++++++++++ arch/arm/boot/dts/rk3288-veyron-chromebook.dtsi | 1 + sound/soc/rockchip/rockchip_i2s.c | 87 +++++++++++------- sound/soc/rockchip/rockchip_max98090.c | 54 ++++++------ 4 files changed, 187 insertions(+), 57 deletions(-) create mode 100644 arch/arm/boot/dts/rk3288-veyron-analog-audio.dtsi -- 2.1.0

3 14

[alsa-devel] [PATCH -next] ASoC: sunxi: remove redundant dev_err call in sun4i_i2s_probe()
by weiyj_lk＠163.com 04 Jul '16

04 Jul '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/sunxi/sun4i-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index fab5234..687a8f8 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -599,10 +599,8 @@ static int sun4i_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "Can't request IO region\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } irq = platform_get_irq(pdev, 0); if (irq < 0) {

1 0

[alsa-devel] [PATCH] ASOC: Atmel: ClassD: Simplify use of devm_ioremap_resource
by Amitoj Kaur Chawla 04 Jul '16

04 Jul '16

Remove unneeded error handling on the result of a call to platform_get_resource when the value is passed to devm_ioremap_resource. The Coccinelle semantic patch that makes this change is as follows: // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); // </smpl> Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/soc/atmel/atmel-classd.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 6107de9..6d9b8b4 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -593,11 +593,6 @@ static int atmel_classd_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "no memory resource\n"); - return -ENXIO; - } - io_base = devm_ioremap_resource(dev, res); if (IS_ERR(io_base)) { ret = PTR_ERR(io_base); -- 1.9.1

3 6

[alsa-devel] [PATCH] ASOC: atmel-pdmic: Simplify use of devm_ioremap_resource
by Amitoj Kaur Chawla 04 Jul '16

04 Jul '16

Remove unneeded error handling on the result of a call to platform_get_resource when the value is passed to devm_ioremap_resource. The Coccinelle semantic patch that makes this change is as follows: // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); // </smpl> Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/soc/atmel/atmel-pdmic.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index aee4787..5f56da6 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -624,11 +624,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "no memory resource\n"); - return -ENXIO; - } - io_base = devm_ioremap_resource(dev, res); if (IS_ERR(io_base)) { ret = PTR_ERR(io_base); -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: timer: Fix negative queue usage by racy accesses
by Takashi Iwai 04 Jul '16

04 Jul '16

The user timer tu->qused counter may go to a negative value when multiple concurrent reads are performed since both the check and the decrement of tu->qused are done in two individual locked contexts. This results in bogus read outs, and the endless loop in the user-space side. The fix is to move the decrement of the tu->qused counter into the same spinlock context as the zero-check of the counter. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index e722022d325d..9a6157ea6881 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1955,6 +1955,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, qhead = tu->qhead++; tu->qhead %= tu->queue_size; + tu->qused--; spin_unlock_irq(&tu->qlock); if (tu->tread) { @@ -1968,7 +1969,6 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, } spin_lock_irq(&tu->qlock); - tu->qused--; if (err < 0) goto _error; result += unit; -- 2.9.0

1 0

[alsa-devel] [PATCH] ASoC: codecs: Allow user to select WM9713 codec
by Alexander Shiyan 04 Jul '16

04 Jul '16

Add simple description and necessary options to allow to select Wolfson WM9713 codec. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58..6721427 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1010,8 +1010,9 @@ config SND_SOC_WM9712 tristate config SND_SOC_WM9713 - tristate + tristate "Wolfson Microelectronics WM9713 codec" select REGMAP_AC97 + select SND_SOC_AC97_BUS # Amp config SND_SOC_LM4857 -- 2.4.9

2 1

Re: [alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Mark Brown 04 Jul '16

04 Jul '16

On Mon, Jul 04, 2016 at 10:26:54AM +0800, KaiChieh Chuang wrote: > On Thu, 2016-06-30 at 03:08 +0800, Mark Brown wrote: > > On Sun, Jun 05, 2016 at 02:58:59PM +0800, Kai Chieh Chuang wrote: > > > The BE receiving SND_SOC_DAPM_STREAM_STOP event, while still used by a > > > FE who is in STREAM_START state. The DAPM will just inactive BE related > > > widget/dai without checking. This just doesn't seems right. > > Why not? The backend is still active and running, why would it expect > > to be stopped unless it's supposed to actually stop? > Yes, I don't expect it to stop. > But at this situation, the BE change to state STREAM_STOP while > still used by a FE in STREAM_START state. > Shouldn't the BE stay in the STREAM_START state? Yes, but that sounded like the situation you were describing? It's really hard to follow what you're trying to say here, perhaps just send the fix you're proposing?

1 0

[alsa-devel] [PATCH] ASoC: bt-sco: add config prompt
by Garlic Tseng 04 Jul '16

04 Jul '16

Add config prompt for bt-sco codec driver Signed-off-by: Garlic Tseng <garlic.tseng(a)mediatek.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2f8603..1cd6ab3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -383,7 +383,7 @@ config SND_SOC_ALC5632 tristate config SND_SOC_BT_SCO - tristate + tristate "Dummy BT SCO codec driver" config SND_SOC_CQ0093VC tristate -- 1.9.1

2 1

Re: [alsa-devel] [PATCH 3/3] [alsa-lib] conf/ucm: chtrt5645: Add UCM config for chtrt5645
by Mark Brown 04 Jul '16

04 Jul '16

On Sun, Jul 03, 2016 at 11:06:07PM -0600, Stephen Just wrote: > Add a UCM configuration for the rt5645 codec on Intel's Cherry-Trail > platform. Tested on the Microsoft Surface 3. This is not a kernel patch, it should be sent separately and sent to the alsa-lib maintainers (at least Taksahi).

1 0

[alsa-devel] [PATCH] clkdev: add devm_of_clk_get()
by Kuninori Morimoto 04 Jul '16

04 Jul '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This is based on devm_clk_get() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- drivers/clk/clkdev.c | 26 ++++++++++++++++++++++++++ include/linux/clk.h | 7 +++++++ 2 files changed, 33 insertions(+) diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 89cc700..93a613b 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -55,6 +55,32 @@ struct clk *of_clk_get(struct device_node *np, int index) } EXPORT_SYMBOL(of_clk_get); +static void devm_of_clk_release(struct device *dev, void *res) +{ + clk_put(*(struct clk **)res); +} + +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + struct clk **ptr, *clk; + + ptr = devres_alloc(devm_of_clk_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return ERR_PTR(-ENOMEM); + + clk = of_clk_get(np, index); + if (!IS_ERR(clk)) { + *ptr = clk; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return clk; +} +EXPORT_SYMBOL(devm_of_clk_get); + static struct clk *__of_clk_get_by_name(struct device_node *np, const char *dev_id, const char *name) diff --git a/include/linux/clk.h b/include/linux/clk.h index 834179f..01005e78 100644 --- a/include/linux/clk.h +++ b/include/linux/clk.h @@ -494,6 +494,8 @@ struct of_phandle_args; #if defined(CONFIG_OF) && defined(CONFIG_COMMON_CLK) struct clk *of_clk_get(struct device_node *np, int index); +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index); struct clk *of_clk_get_by_name(struct device_node *np, const char *name); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec); #else @@ -501,6 +503,11 @@ static inline struct clk *of_clk_get(struct device_node *np, int index) { return ERR_PTR(-ENOENT); } +struct clk *devm_of_clk_get(struct device *dev, + struct device_node *np, int index) +{ + return ERR_PTR(-ENOENT); +} static inline struct clk *of_clk_get_by_name(struct device_node *np, const char *name) { -- 1.9.1

2 2

[alsa-devel] [PATCH v5 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 04 Jul '16

04 Jul '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v4: * remove SND_SOC_MEDIATEK config prompt * fix config prompt to meaningful one for SND_SOC_MT2701_CS42448 * remove some unnecessary random space change Change since v3: * fix comments style. Change since v2: * fix comments style. * replace EXPORT_SYMBOL with EXPORT_SYMBOL_GPL * reduce debug message * move mutex lock for irq-allocated to AFE struct * remove func mtk_simple_isr and call snd_pcm_period_elapsed directly * mt2701 driver probe will fail if clock initial error * fix dapm on/off switch control naming Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 37 +- sound/soc/mediatek/Makefile | 10 +- sound/soc/mediatek/common/Makefile | 16 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 181 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1698 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 921 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4390 insertions(+), 634 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 19

[alsa-devel] [asoc:for-next 9/16] cygnus-pcm.c:undefined reference to `bad_dma_ops'
by kbuild test robot 03 Jul '16

03 Jul '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 5664c75bd47efe7c701d1d46ee90c795ff8bc150 commit: ebe0ee8d716bcb3cb890e4feb6c6e07e0de11208 [9/16] Merge remote-tracking branches 'asoc/topic/cygnus', 'asoc/topic/da7219', 'asoc/topic/davinci' and 'asoc/topic/dpcm' into asoc-next config: m32r-allyesconfig (attached as .config) compiler: m32r-linux-gcc (GCC) 4.9.0 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout ebe0ee8d716bcb3cb890e4feb6c6e07e0de11208 # save the attached .config to linux build tree make.cross ARCH=m32r All errors (new ones prefixed by >>): drivers/built-in.o: In function `zynq_fpga_ops_write': zynq-fpga.c:(.text+0xeff96c): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff974): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeffa94): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcb4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcbc): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): undefined reference to `dma_common_mmap' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): relocation truncated to fit: R_M32R_26_PCREL_RELA against undefined symbol `dma_common_mmap' sound/built-in.o: In function `cygnus_pcm_preallocate_dma_buffer': >> cygnus-pcm.c:(.text+0x10efa0): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10efa4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `cygnus_dma_free_dma_buffers': cygnus-pcm.c:(.text+0x10f15c): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10f160): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10f16c): undefined reference to `bad_dma_ops' sound/built-in.o:cygnus-pcm.c:(.text+0x10f170): more undefined references to `bad_dma_ops' follow --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v4.7
by Mark Brown 03 Jul '16

03 Jul '16

The following changes since commit 4c2e07c6a29e0129e975727b9f57eede813eea85: Linux 4.7-rc5 (2016-06-26 17:52:03 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.7-rc5 for you to fetch changes up to 7c5eec4edf4693aa61f3ee78bf28621dfe33d9f0: Merge remote-tracking branches 'asoc/fix/rcar', 'asoc/fix/rt5670' and 'asoc/fix/wm8940' into asoc-linus (2016-07-01 18:05:34 +0200) ---------------------------------------------------------------- ASoC: Fixes for v4.7 A small clutch of hardware specific fixes for various ASoC devices, all small individually and important if you have that device but not otherwise. ---------------------------------------------------------------- Ajit Pandey (1): ASoC: wm5110: Add missing route from OUT3R to SYSCLK Alan Cox (1): ASoC: Intel: atom: fix missing breaks that would cause the wrong operation to execute Alexander Shiyan (1): ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode Arnd Bergmann (1): ASoC: hdmi-codec: select CONFIG_HDMI Bard Liao (2): ASoC: rt5670: fix HP Playback Volume control ASoC: rt5645: fix reg-2f default value. Charles Keepax (1): ASoC: wm5102: Correct supported channels on trace compressed DAI Fabio Estevam (1): ASoC: ak4613: Remove owner assignment from platform_driver Geert Uytterhoeven (2): ASoC: wm8940: Enable cache usage to fix crashes on resume ASoC: ak4613: Enable cache usage to fix crashes on resume Janusz Krzysztofik (1): ASoC: cx20442: set tty->receiver_room in v253_open Kuninori Morimoto (1): ASoC: rsnd: open 31bit of SSICKR mask Mark Brown (4): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4613', 'asoc/fix/arizona', 'asoc/fix/cx20442', 'asoc/fix/davinci', 'asoc/fix/fsl-ssi' and 'asoc/fix/hdmi' into asoc-linus Merge remote-tracking branches 'asoc/fix/rcar', 'asoc/fix/rt5670' and 'asoc/fix/wm8940' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix dra7 DMA offset when using CFG port Vinod Koul (2): ASoC: hdac_hdmi: Fix potential NULL dereference ASoC: Intel: Skylake: Initialize module list for Broxton sound/soc/codecs/Kconfig | 7 +-- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/cx20442.c | 1 + sound/soc/codecs/hdac_hdmi.c | 20 ++++++++ sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 1 + sound/soc/codecs/wm8940.c | 1 + sound/soc/davinci/davinci-mcasp.c | 56 +++++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 4 +- sound/soc/fsl/fsl_ssi.c | 12 ++--- sound/soc/intel/atom/sst-mfld-platform-compress.c | 9 +++- sound/soc/intel/skylake/bxt-sst.c | 1 + sound/soc/sh/rcar/adg.c | 2 +- 15 files changed, 100 insertions(+), 22 deletions(-)

2 1

[alsa-devel] [PATCH] drm: mediatek: fix prototypes after API change
by Arnd Bergmann 02 Jul '16

02 Jul '16

The newly added mediatek HDMI driver clashes with an API change for struct hdmi_codec_ops, causing an 'allmodconfig' build to fail: drivers/gpu/drm/mediatek/mtk_hdmi.c:1653:15: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1654:19: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1655:20: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1656:18: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] drivers/gpu/drm/mediatek/mtk_hdmi.c:1657:13: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] This changes the API accordingly for the mediatek driver. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 8f83f26891e1 ("drm/mediatek: Add HDMI support") Fixes: efc9194bcff8 ("ASoC: hdmi-codec: callback function will be called with private data") --- The new driver is currently part of Dave Airlie's drm-next tree, while the API change is in Mark Brown's asoc-next tree, so my fix has to be done on top of both. I can see two ways out of this: a) Stephen Rothwell could add my fixup on top of linux-next as a merge fix, and the fix gets sent to Linus Torvalds along with the second of the two branches b) efc9194bcff8 gets merged into drm-next along with my change in the merge commit. It is currently a standalone patch in its own branch in the ASoC tree. --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index ba812ef2c9d1..334562d06731 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -1535,7 +1535,7 @@ static int mtk_hdmi_dt_parse_pdata(struct mtk_hdmi *hdmi, * HDMI audio codec callbacks */ -static int mtk_hdmi_audio_hw_params(struct device *dev, +static int mtk_hdmi_audio_hw_params(struct device *dev, void *data, struct hdmi_codec_daifmt *daifmt, struct hdmi_codec_params *params) { @@ -1604,7 +1604,7 @@ static int mtk_hdmi_audio_hw_params(struct device *dev, return 0; } -static int mtk_hdmi_audio_startup(struct device *dev) +static int mtk_hdmi_audio_startup(struct device *dev, void *data) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); @@ -1615,7 +1615,7 @@ static int mtk_hdmi_audio_startup(struct device *dev) return 0; } -static void mtk_hdmi_audio_shutdown(struct device *dev) +static void mtk_hdmi_audio_shutdown(struct device *dev, void *data) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); @@ -1624,7 +1624,7 @@ static void mtk_hdmi_audio_shutdown(struct device *dev) mtk_hdmi_audio_disable(hdmi); } -int mtk_hdmi_audio_digital_mute(struct device *dev, bool enable) +int mtk_hdmi_audio_digital_mute(struct device *dev, void *data, bool enable) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); @@ -1638,7 +1638,7 @@ int mtk_hdmi_audio_digital_mute(struct device *dev, bool enable) return 0; } -static int mtk_hdmi_audio_get_eld(struct device *dev, uint8_t *buf, size_t len) +static int mtk_hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size_t len) { struct mtk_hdmi *hdmi = dev_get_drvdata(dev); -- 2.9.0

4 3

[alsa-devel] [PATCH] ASoC: cs4270: fix DAPM stream name mismatch
by murray foster 02 Jul '16

02 Jul '16

Mismatching stream names in DAPM route and widget definitions are causing compilation errors. Fixing these names allows the cs4270 driver to compile and function. Signed-off-by: Murray Foster <mrafoster(a)gmail.com> --- sound/soc/codecs/cs4270.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e07807d..4ffdca4 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -1,4 +1,4 @@ -/* +s/* * CS4270 ALSA SoC (ASoC) codec driver * * Author: Timur Tabi <timur(a)freescale.com> @@ -148,11 +148,11 @@ SND_SOC_DAPM_OUTPUT("AOUTR"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { - { "Capture", NULL, "AINA" }, - { "Capture", NULL, "AINB" }, + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, - { "AOUTA", NULL, "Playback" }, - { "AOUTB", NULL, "Playback" }, + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, }; /** -- 2.4.9

1 0

[alsa-devel] [GIT PULL] Broxton fw update
by Vinod Koul 01 Jul '16

01 Jul '16

Please PULL to receive the latest Broxton fw. The following changes since commit 3ef7857d511ce6a91c5ce609da76c4702651cfa5: amdgpu: update polaris ucode (2016-06-28 14:31:11 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git apl for you to fetch changes up to 040c24514b94fcdbd6308d986d1bde59a704a21a: linux-firmware: intel: Update Broxton audio firmware (2016-06-29 14:20:20 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Broxton audio firmware WHENCE | 4 ++-- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v702.bin | Bin 0 -> 505608 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v702.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH 0/3 v2] ALSA: ctl: add dimension information validator
by Takashi Sakamoto 01 Jul '16

01 Jul '16

Hi, This patchset is revised version of my previous one. [alsa-devel] [PATCH 0/3] ALSA: add dimension information validator http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109811.html The main purpose of this patchset is the same, just including some changes below. * Add arithmetic overflow check * Add negative value check of first dimension level * Change function name of the validator and name of local variable so that the aim is clear. Takashi Sakamoto (3): ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 87 +++++++++++++++++++++++++++++++++-------- sound/pci/echoaudio/echoaudio.c | 6 +-- 2 files changed, 74 insertions(+), 19 deletions(-) -- 2.7.4

2 10

[alsa-devel] Assertion failure in alsa-lib crashing applications (snd_pcm_area_copy: Assertion `dst < src || dst >= src + bytes' failed)
by Baptiste Jonglez 01 Jul '16

01 Jul '16

Hi, When using Ring [1], an audio-video application, it sometimes crashes because of an assertion in libasound.so: pcm.c:2758: snd_pcm_area_copy: Assertion `dst < src || dst >= src + bytes' failed. The complete upstream bug report is at [2]. The (not-so-useful) stack trace is the following: #0 0x00007fc356e462a8 raise (libc.so.6) #1 0x00007fc356e4772a abort (libc.so.6) #2 0x00007fc356e3f1b7 __assert_fail_base (libc.so.6) #3 0x00007fc356e3f262 __assert_fail (libc.so.6) #4 0x00007fc35ba65c84 snd_pcm_area_copy (libasound.so.2) #5 0x00007fc35ba65ebc snd_pcm_areas_copy (libasound.so.2) #6 0x00007fc35baa6d95 n/a (libasound.so.2) #7 0x00007fc35ba77c1f n/a (libasound.so.2) #8 0x00007fc35ba77a59 n/a (libasound.so.2) #9 0x000000000051ba97 n/a (dring) #10 0x000000000051c118 n/a (dring) #11 0x000000000051cfad n/a (dring) #12 0x00007fc357787c40 execute_native_thread_routine (libstdc++.so.6) #13 0x00007fc359ec2424 start_thread (libpthread.so.0) #14 0x00007fc356efacbd __clone (libc.so.6) The next time it crashes, I will have debug symbols in dring. I can also recompile alsa-lib with debug symbols, what is the best way to do that? Looking around, it seems other projects have run into the same issue: https://aur.archlinux.org/packages/ultrastardx-git/?comments=all#comment-43… https://aur.archlinux.org/packages/zoom/#comment-544696 http://ubuntuforums.org/showthread.php?t=2248373 https://github.com/js-platform/node-webrtc/issues/110 https://fedorahosted.org/fldigi/ticket/70 The output of the alsa-info.sh script on my system is at [3]. What else can I provide to debug this issue further? Thanks, Baptiste [1] https://ring.cx [2] https://tuleap.ring.cx/plugins/tracker/?aid=502 [3] http://www.alsa-project.org/db/?f=0dd2ba1021b3d535f30f07c55dc18e2ef60db26d

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: patch reg-0x8a
by Bard Liao 01 Jul '16

01 Jul '16

reg-8a assign the tracking source for each ASRC tracker. The default value is 0x0000 which means all ASRC trackers will track LRCK1. But in most cases, we wish each ASRC tracker track the corresponding LRCK. i.e. ASRC1 tracks LRCK1, ASRC2 tracks LRCK2 and so on. So, we rewrite reg-8a as 0x0120. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1f457d3..c21e41c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -64,6 +64,7 @@ static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, {RT5645_PR_BASE + 0x00, 0x0a2a}, + {RT5645_ASRC_4, 0x0120}, }; static const struct reg_sequence rt5650_init_list[] = { @@ -159,7 +160,7 @@ static const struct reg_default rt5645_reg[] = { { 0x83, 0x0000 }, { 0x84, 0x0000 }, { 0x85, 0x0000 }, - { 0x8a, 0x0000 }, + { 0x8a, 0x0120 }, { 0x8e, 0x0004 }, { 0x8f, 0x1100 }, { 0x90, 0x0646 }, @@ -316,7 +317,7 @@ static const struct reg_default rt5650_reg[] = { { 0x83, 0x0000 }, { 0x84, 0x0000 }, { 0x85, 0x0000 }, - { 0x8a, 0x0000 }, + { 0x8a, 0x0120 }, { 0x8e, 0x0004 }, { 0x8f, 0x1100 }, { 0x90, 0x0646 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: fix reg-2f default value.
by Bard Liao 01 Jul '16

01 Jul '16

The default value of reg-2f in codec rt5650 is 0x5002, not 0x1002. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3c6594d..d70847c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -253,7 +253,7 @@ static const struct reg_default rt5650_reg[] = { { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, { 0x2d, 0x0000 }, - { 0x2f, 0x1002 }, + { 0x2f, 0x5002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, { 0x33, 0x0000 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [asoc:topic/sunxi 2/2] sound/soc/sunxi/sun4i-i2s.c:624:2-3: Unneeded semicolon
by kbuild test robot 01 Jul '16

01 Jul '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sunxi head: fa7c0d13cb26216f6dec5ef19e028e68b300530d commit: fa7c0d13cb26216f6dec5ef19e028e68b300530d [2/2] ASoC: sunxi: Add Allwinner A10 Digital Audio driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/sunxi/sun4i-i2s.c:624:2-3: Unneeded semicolon Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 1/2] ASoC: cs35l33: Fix testing return value of devm_gpiod_get_optional
by Axel Lin 01 Jul '16

01 Jul '16

devm_gpiod_get_optional() returns NULL when the gpio is not assigned. So the if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) test is always false. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l33.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 622a111..aa5c0c5 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1176,11 +1176,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, /* We could issue !RST or skip it based on AMP topology */ cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, "reset-gpios", GPIOD_OUT_HIGH); - - if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { - dev_warn(&i2c_client->dev, - "%s WARNING: No reset gpio assigned\n", __func__); - } else if (IS_ERR(cs35l33->reset_gpio)) { + if (IS_ERR(cs35l33->reset_gpio)) { dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", __func__); return PTR_ERR(cs35l33->reset_gpio); -- 2.5.0

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[alsa-devel] [PATCH 0/3] ASoC: Reduce audio related kernel spew.
by vedang.patel＠intel.com 01 Jul '16

01 Jul '16

From: Vedang Patel <vedang.patel(a)intel.com> There have been complaints regarding audio related kernel messages flooding dmesg. This series tries to address those complaints. Also, I have created 3 different patches for this because they touch different drivers. But, I can combine those to reduce the number of patches. Vedang Patel (3): ASoC: hdac_hdmi: Increase loglevel of hex dump printed ASoC: Intel: common: increase the loglevel of debug message. ASoC: Intel: Skylake: Increase loglevel of debug message. sound/soc/codecs/hdac_hdmi.c | 2 ++ sound/soc/intel/common/sst-dsp.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 2 ++ sound/soc/intel/skylake/skl-sst-ipc.c | 7 +++++-- 4 files changed, 10 insertions(+), 3 deletions(-) -- 2.1.2

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Re: [alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Piotr G. 01 Jul '16

01 Jul '16

Hello Guys, I made some sniffing of USB protocol with Windows10 and Windows8 drivers for Asus Xonar U7. ALSA has no Capture Volume controls for Line Input and Mic Input. Mic doesn't work. According to Windows drivers when using Mic or Line in sliders: wValue: 0x8202 [L] -> 0x82 It's not declared in UAC2 0x8201 [R] -> 0x82 It's not declared in UAC2 AudioControl Interface Descriptor: bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 4 (Mic Terminal) bmaControls( 0) 0x00000003 Mute Control (read/write) bmaControls( 1) 0x00000000 (No Volume Control?) bmaControls( 2) 0x00000000 (No Volume Control?) iFeature 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 17 bSourceID 5 (Line in Terminal) bmaControls( 0) 0x00000003 Mute Control (read/write) bmaControls( 1) 0x00000000 (No Volume Control?) bmaControls( 2) 0x00000000 (No Volume Control?) iFeature 0 It's looks like firmware doesn't declare Volume or Input Gain in FU with 0x82 CS. How can I properly add it do the source? (redefine mixer_maps?) wIndex: 0x1100 (Line In) -> 0x1100 ALSA FU16 OK 0x1000 (Mic) -> 0x1000 ALSA FU17 OK In Alsamixer in Capture section we have only Line and Mic Controls where we can toggle On or Off. This is correct when parsing upper FU only Mute Control is declared Where or how I can manual declare that this controls have Volume Control with CS 0x82 (Control Selector)? WHat is the correct way? About Ranges: Line in Data: 0x00a6 (160) - 0x00ff (255) (Mixer from 0 (-96dB) to 99 (-0.15dB) log) 0x0000 (0) (Mixer 100 (0dB) Mic Data: >From 0x00a6 (160) - 0x0000 (0) (Mixer from 0 (-96dB) to 50 (0dB) log) >From 0x000 (0) - 0x0032 (50) (Mixer from 50 (0dB) to 100 (50dB) lin) Thanks for support. Piotr W dniu 10.09.2015 o 16:47, Piotr SQ9FK pisze: > Hello Takashi,Johan > My patches are unessesary since following last patch was applied to the > git: > ALSA: usb-audio: Change internal PCM order > > Capture working but there are no volume sliders for Line-In Capture or > Mic-In Capture (I can only toggle on off). Probably the RANGES are > broken?. From ASUS Windows driver I can see following ranges: > > Master Volume: (Phisycal Knob on Asus) > from 100 to 1 correspond 0dB to -70dB, 0 is -127dB > Range from 1 to 100 -> 0dB to -70dB (logarythmic) > > Mic-In: > from 100 to 1 correspond 30dB to -10dB, 0 is -192dB > Range from 1 to 10 -> -10dB to 10dB (linear?) > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Line-In: > from 100 to 1 correspond 0dB to -40dB, 0 is -192dB > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Actual Situation: > Input Gain Pad Control - 3 steps [0 - 50 - 100] > Double Slider. In Capture and Playback Categories, both of them are > controlled in the same time. > Controls Playback Volume only, Capture not. > > "Mic" and "Line" Switches [on/off] > PCM Cature Source [Line/Mic] > Electret Mic doesn't work. With all possible configurations - Silent. > When I pluged Line level signal, I can record the signal without problems. > PCM Capture Source [Line] - Line [on] Mic [off] - Sound > PCM Capture Source [Line] - Line [on] Mic [on] - Sound > PCM Capture Source [Line] - Line [off] Mic [off] - No Sound > PCM Capture Source [Line] - Line [off] Mic [on] - No Sound > The same with selected [Mic]. There is no Capture Volume control. > > PCM 1 [mute/on] > I think this is SPDIF switch. It doesn't work. When I switch it on -> > Blue Led on Xonar doesn't light up. > > I attached screen of alsamixer, usbmix from proc and my mixer_map from > my old patch. > I think there are is still bugs in parsing. Please point me how I can > add manually Volume control for Capture? > > Piotr Gaska > > W dniu 2015-07-27 o 14:10, Takashi Iwai pisze: >> On Sun, 26 Jul 2015 10:00:12 +0200, >> Piotr G. wrote: >>> Hello, >>> This is my first patch to alsa-devel, and I'm not very skilled >>> programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS >>> in mixer_map.c I don't know why the driver by default doesn't parse all >>> info correctly. With this patch you can see Line Switch and Mic Switch >>> in alsamixer. I can record my voice in Audacity with all supported bit >>> and sampling rates in Mono or Stereo >>> This is temporary solution. Please correct me if the patch is not >>> properly made. >>> >>> What doesn't work: >>> - Cannot change Capture Volume for Line/Mic. No mixer control in >>> alsamixer >>> - Switching between Line/Mic doesn't work (Mute doesn't work) >>> - SPDIF not tested >>> As what I wrote at beginning I'm not skilled programmer so any >>> help is >>> aprreciated >>> >>> >>> proc-usbmixer -> cat /proc/asound/*/usbmix >>> usb.debug -> lsusb -vv (only Asus Xonar U7) >> The changes in mixer_maps.c look good, but what's the reason for >> uac2_audio_feature_info[]? It seems adding "Control" suffix to each >> entry, which doesn't look correct. >> >> >> thanks, >> >> Takashi >> >> >>> --- >>> sound/usb/mixer.c | 23 +++++++++++++++++++++-- >>> sound/usb/mixer_maps.c | 48 >>> ++++++++++++++++++++++++++++++++++++++++++++++++ >>> 2 files changed, 69 insertions(+), 2 deletions(-) >>> >>> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c >>> index 6b3acba..027d160 100644 >>> --- a/sound/usb/mixer.c >>> +++ b/sound/usb/mixer.c >>> @@ -818,6 +818,25 @@ static struct usb_feature_control_info >>> audio_feature_info[] = { >>> { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> }; >>> >>> +static struct usb_feature_control_info uac2_audio_feature_info[] = { >>> + /* UAC2 specific */ >>> + { "Mute Control", USB_MIXER_INV_BOOLEAN }, >>> + { "Volume Control", USB_MIXER_S16 }, >>> + { "Bass Control", USB_MIXER_S8 }, >>> + { "Mid Control", USB_MIXER_S8 }, >>> + { "Treble Control", USB_MIXER_S8 }, >>> + { "Graphic Equalizer Control", USB_MIXER_S8 }, >>> + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, >>> + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in UAC2 */ >>> + { "Bass Boost Control", USB_MIXER_BOOLEAN }, >>> + { "Loudness Control", USB_MIXER_BOOLEAN }, >>> + { "Input Gain Control", USB_MIXER_S16 }, >>> + { "Input Gain Pad Control", USB_MIXER_S16 }, >>> + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> + { "Underflow Control", USB_MIXER_BOOLEAN }, >>> + { "Overflow Control", USB_MIXER_BOOLEAN }, >>> +}; >>> + >>> /* private_free callback */ >>> void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) >>> { >>> @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); >>> cval->control = control; >>> cval->cmask = ctl_mask; >>> - cval->val_type = audio_feature_info[control-1].type; >>> + cval->val_type = uac2_audio_feature_info[control-1].type; >>> if (ctl_mask == 0) { >>> cval->channels = 1; /* master channel */ >>> cval->master_readonly = readonly_mask; >>> @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> break; >>> default: >>> if (!len) >>> - strlcpy(kctl->id.name, audio_feature_info[control-1].name, >>> + strlcpy(kctl->id.name, >>> uac2_audio_feature_info[control-1].name, >>> sizeof(kctl->id.name)); >>> break; >>> } >>> diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c >>> index e5000da..07f510e 100644 >>> --- a/sound/usb/mixer_maps.c >>> +++ b/sound/usb/mixer_maps.c >>> @@ -341,6 +341,48 @@ static const struct usbmix_name_map >>> scms_usb3318_map[] = { >>> { 0 } >>> }; >>> >>> +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC >>> Stereo CS4398 */ >>> +/* Topology: >>> + >>> +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] >>> + >>> +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] >>> + >>> +Mic_IN[4]------->FU[16]---+ >>> + SU[20]--CS[22]--->USB_OUT[11] >>> +Line_IN[5]------>FU[17]---+ >>> + >>> +*/ >>> + >>> +static const struct usbmix_name_map xonar_u7_map[] = { >>> + /* 1: IT PCM Analog (USB Streaming) */ >>> + /* 2: IT PCM2 Digital (USB Streaming) */ >>> + /* 4: IT Microphone */ >>> + /* 5: IT Line IN */ >>> + /* 7: OT Speaker */ >>> + /* 8: OT SPDIF */ >>> + /* 11 OT Capture Playback (USB Streaming) */ >>> + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume */ >>> + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ >>> + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in >>> Mute/Volume */ >>> + { 17, "Line Capture" }, /* 17 FU Capture Line-in Mute/Volume */ >>> + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ >>> + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ >>> + { 20, "Capture Source" }, /* 20 SU Capture Source Selector */ >>> + /* 22 CS Line-in and Mic-in Clock Source */ >>> + { 0 } >>> +}; >>> + >>> +static struct usbmix_selector_map xonar_u7_selectors[] = { >>> +{ >>> + .id = 20, >>> + .count = 2, >>> + .names = (const char*[]) {"Mic", "Line"} >>> + }, >>> + { 0 } /* terminator */ >>> + >>> +}; >>> + >>> /* >>> * Control map entries >>> */ >>> @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { >>> .id = USB_ID(0x25c4, 0x0003), >>> .map = scms_usb3318_map, >>> }, >>> + { >>> + /* Asus Xonar U7 */ >>> + .id = USB_ID(0x1043, 0x857c), >>> + .map = xonar_u7_map, >>> + .selector_map = xonar_u7_selectors, >>> + }, >>> { 0 } /* terminator */ >>> }; >>> >>> -- >>> 2.3.6 >>> >

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[alsa-devel] [PATCH 0/3] ALSA: add dimension information validator
by Takashi Sakamoto 01 Jul '16

01 Jul '16

Hi, This patchset is to add a validator of dimension information for drivers in kernel/userspace. The dimension information was added to ALSA control interface. With the information, members in an element compose multi-dimensional matrix. For example, this matrix can be used for row/column table with values. In this case, dimension level 1 and 2 are utilized to describe the number of rows and columns by element member unit. When dimension information has no contradictions to the number of members in an element, there's no problem. Once it has, it can cause an issue. In worst case, it can cause buffer-overrun in userspace. The aim of this patchset is to prevent this situation. Takashi Sakamoto (3): ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members ALSA: control: add dimension validator for userspace element ALSA: control: add dimension validator for kernel driver sound/core/control.c | 77 ++++++++++++++++++++++++++++++++--------- sound/pci/echoaudio/echoaudio.c | 6 ++-- 2 files changed, 64 insertions(+), 19 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 0/4] ASoC: sunxi: Add i2s controller support
by Maxime Ripard 30 Jun '16

30 Jun '16

Hi everyone, This is the second version of the I2S support for the controller found in the Allwinner A10 and later SoCs. Playback has been tested with an UDA1380 on an A20-Olinuxino. Capture is not implemented yet, but will come eventually. Let me know what you think, Maxime Changes from v1: - Fixed bogus indentation and spurious whitespaces / newlines - Moved the device initialisation in the probe to the startup function - Disabled the device in shutdown - Removed function to get wss and sr to inline them in hw_params - Changed the way to access the divider array - Checked clk_set_rate return code - Fixed condition for invalid bclk and mclk dividers - Added sound-dai-cells to the bindings documentation required properties Emilio López (1): ARM: sun7i: Add mod1 clock nodes Maxime Ripard (3): ASoC: sunxi: Add A10 I2S controller binding documentation ASoC: sunxi: Add Allwinner A10 Digital Audio driver ARM: sun7i: Add DAI nodes .../devicetree/bindings/sound/sun4i-i2s.txt | 34 + arch/arm/boot/dts/sun7i-a20.dtsi | 87 ++- sound/soc/sunxi/Kconfig | 9 + sound/soc/sunxi/Makefile | 2 +- sound/soc/sunxi/sun4i-i2s.c | 703 +++++++++++++++++++++ 5 files changed, 832 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-i2s.txt create mode 100644 sound/soc/sunxi/sun4i-i2s.c -- 2.9.0

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[alsa-devel] [alsa-lib][PATCH 0/5] ctl: support extra information for user-defined element set
by Takashi Sakamoto 30 Jun '16

30 Jun '16

Hi, This patchset adds support extra information to the type-specific for user-defined element set. Currently, 'dimension' is such an extra information. In ALSA kernel/userspace interface, information to an element is described in this structure. struct snd_ctl_elem_info { struct snd_ctl_elem_id id; unsigned int access; unsigned int count; pid_t owner; union value; union dimen; unsigned char reserved[64 - 4 * sizeof(unsigned short)]; }; This structure includes reserved fields, thus it's possible to add more fields for future extension. Currently, APIs to add user-defined element set don't support such extra fields. Meanwhile, supporting just the dimension field is not good as stable library APIs. This patchset changes prototype of the APIs with 'snd_ctl_elem_info_t' instead of adding more parameters. Callers are expected to fill the parameter with identification information and the extra information. Takashi Sakamoto (5): ctl: support extra information in user-defined element set ctl: add an API to set dimension levels to element information ctl: optimize a test for user-defined element set to older kernels ctl: optimize a test for user-defined element set to changes of APIs ctl: support dimension test for user-defined element set include/control.h | 12 ++- src/control/control.c | 219 +++++++++++++++++++++++--------------------- src/pcm/pcm_softvol.c | 10 +- test/user-ctl-element-set.c | 122 ++++++++++++++++++------ 4 files changed, 220 insertions(+), 143 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: riptide: Use DIV_ROUND_UP
by Amitoj Kaur Chawla 30 Jun '16

30 Jun '16

The kernel.h macro DIV_ROUND_UP performs the computation (((n) + (d) - 1) /(d)) but is perhaps more readable. The Coccinelle script used to make this change is as follows: @haskernel@ @@ #include <linux/kernel.h> @depends on haskernel@ expression n,d; @@ ( - (n + d - 1) / d + DIV_ROUND_UP(n,d) | - (n + (d - 1)) / d + DIV_ROUND_UP(n,d) ) Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/pci/riptide/riptide.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 94639d6..067a912 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1496,7 +1496,7 @@ static int snd_riptide_prepare(struct snd_pcm_substream *substream) f = PAGE_SIZE; while ((size + (f >> 1) - 1) <= (f << 7) && (f << 1) > period) f = f >> 1; - pages = (size + f - 1) / f; + pages = DIV_ROUND_UP(size, f); data->size = size; data->pages = pages; snd_printdd -- 1.9.1

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Re: [alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode
by Nicolin Chen 29 Jun '16

29 Jun '16

On Wed, Jun 29, 2016 at 09:50:24PM +0300, Alexander Shiyan wrote: > >Среда, 29 июня 2016, 21:26 +03:00 от Nicolin Chen <nicoleotsuka(a)gmail.com>: > > > >On Wed, Jun 29, 2016 at 10:52:01AM +0300, Alexander Shiyan wrote: > > > >> >your failed case. Could you please tell me what were the FSclk, > >> >BITclk and size of your case? > > > >> Here is my oscillograms with/without patch: > >> Bclk=12 MHz, sound is just a "cat /dev/urandom > /dev/dsp", i.e. FS=8 KHz, MONO. > >> Upper (yellow) line is TXDATA, Lower (blue) is FS. > >> > >> Measurement without patch: https://cloud.mail.ru/public/Rcsv/xr88V5Sa1 > >> Measurement with patch: https://cloud.mail.ru/public/NCwn/DQnzioQEL > > > >Thanks for the reply. So in the case without patch, it looks like > >SSI just sent the data ignoring the slot size right? > > I could be wrong, but think that WL (word length) still point to number of valid > data bits within 32bit chunk (as defined for I2S-Master mode), > but since DC=0, this cause to send next word to the channel again and again. I see. Thanks for the fix.

1 0

[alsa-devel] [PATCH v3 0/6] ASoC: samsung: Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 29 Jun '16

29 Jun '16

This patch series adds sound support for the Exynos 5433 based TM2 and TM2E board. It includes minimal drivers for the Low Power Audio Subsystem (LPASS) and the MAX98504 class D speaker amplifier as well as the actual asoc machine driver. The changes since v2 are listed in individual patches, I didn't add any Acked-by tag to the max98504 DT binding patch as there was significant change since v2. Inha Song (1): ASoC: samsung: Add Samsung Low Power Audio Subsystem driver Sylwester Nawrocki (5): ASoC: samsung: Add DT bindings documentation for LPASS ASoC: Add DT bindings documentation for max98504 amplifier ASoC: max98504: Add max98504 speaker amplifier driver ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem ASoC: samsung: Add machine driver for Exynos5433 based TM2 board .../devicetree/bindings/sound/max98504.txt | 44 ++ .../bindings/sound/samsung,exynos5433-lpass.txt | 8 + .../bindings/sound/samsung,tm2-audio.txt | 36 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98504.c | 383 ++++++++++++++ sound/soc/codecs/max98504.h | 59 +++ sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 4 + sound/soc/samsung/lpass.c | 162 ++++++ sound/soc/samsung/lpass.h | 47 ++ sound/soc/samsung/tm2_wm5110.c | 579 +++++++++++++++++++++ 12 files changed, 1341 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/samsung/lpass.c create mode 100644 sound/soc/samsung/lpass.h create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

2 8

[alsa-devel] [PATCH v4 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 29 Jun '16

29 Jun '16

Sorry for submitting new version in a short time. I find some comment style error and fix it. please ignore v3 and directly review v4. This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v3: * fix comments style. Change since v2: * fix comments style. * replace EXPORT_SYMBOL with EXPORT_SYMBOL_GPL * reduce debug message * move mutex lock for irq-allocated to AFE struct * remove func mtk_simple_isr and call snd_pcm_period_elapsed directly * mt2701 driver probe will fail if clock initial error * fix dapm on/off switch control naming Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 44 +- sound/soc/mediatek/Makefile | 11 +- sound/soc/mediatek/common/Makefile | 17 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 181 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1698 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 921 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4399 insertions(+), 634 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 14

[alsa-devel] [PATCH v2 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 29 Jun '16

29 Jun '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 44 +- sound/soc/mediatek/Makefile | 11 +- sound/soc/mediatek/common/Makefile | 17 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 387 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 47 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 103 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 357 +++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 33 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 185 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1664 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 923 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4267 insertions(+), 635 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

2 18

[alsa-devel] [PATCH v3 0/9] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 29 Jun '16

29 Jun '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since v2: * fix comments style. * replace EXPORT_SYMBOL with EXPORT_SYMBOL_GPL * reduce debug message * move mutex lock for irq-allocated to AFE struct * remove func mtk_simple_isr and call snd_pcm_period_elapsed directly * mt2701 driver probe will fail if clock initial error * fix dapm on/off switch control naming Change since v1: implement common private structure, fe dai and platform driver. let mt8173 and mt2701 use the common structure. Garlic Tseng (9): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: implement mediatek common structure ASoC: mediatek: let mt8173 use mediatek common structure ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add structure define and clock control for 2701 ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 44 +- sound/soc/mediatek/Makefile | 11 +- sound/soc/mediatek/common/Makefile | 17 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 379 +++++ sound/soc/mediatek/common/mtk-afe-fe-dai.h | 45 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 90 ++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 23 + sound/soc/mediatek/common/mtk-base-afe.h | 104 ++ sound/soc/mediatek/mt2701/Makefile | 19 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 464 ++++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 38 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 181 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1698 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 422 +++++ sound/soc/mediatek/mt2701/mt2701-reg.h | 186 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 73 + .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 921 +++++------ sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 27 files changed, 4399 insertions(+), 634 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/common/Makefile create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.c create mode 100644 sound/soc/mediatek/common/mtk-afe-fe-dai.h create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.c create mode 100644 sound/soc/mediatek/common/mtk-afe-platform-driver.h create mode 100644 sound/soc/mediatek/common/mtk-base-afe.h create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (51%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

2 10

Re: [alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Kai Chieh Chuang 29 Jun '16

29 Jun '16

> If this isn't happening that seems like a bug, I'm a bit surprised > nobody else ran into it? Shouldn't the counts of stream events that > happen be symmetric (ie, we get as many stops as starts), or are we > possibly missing some from things being switched in and out or something? This idea is same as the DPCM BE state control, in int dpcm_be_dai_hw_free(..) { ...... /* do not free hw if this BE is used by other FE */ if (be->dpcm[stream].users > 1) continue; and int dpcm_be_dai_shutdown(...) { .... if (--be->dpcm[stream].users != 0) continue; The BE receiving SND_SOC_DAPM_STREAM_STOP event, while still used by a FE who is in STREAM_START state. The DAPM will just inactive BE related widget/dai without checking. This just doesn't seems right. And the last FE using the BE calls in to dpcm_fe_dai_shutdown(), will run the process normally. Actually, i trace the log, when one of the FE sent stop event to "still in use BE", the BE power state will not be changed. Until the system goes into SNDRV_CTL_POWER_D3cold, the BE will be shutdown (i did set the ignore_suspend=1). I didn't check why the BE is not turned off at the point when receiving the stop event. I'm also curious why no one seems to have the same issue. Probably there is not much DPCM architecture system yet on the Linux?

2 1

Re: [alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Mark Brown 29 Jun '16

29 Jun '16

On Fri, Jun 24, 2016 at 02:21:34PM +0800, KaiChieh Chuang wrote: > Is there any comment on this? Please don't send content free pings and please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review. Sending content free pings just adds to the mail volume (if they are seen at all) and if something has gone wrong you'll have to resend the patches anyway.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode
by Nicolin Chen 29 Jun '16

29 Jun '16

On Tue, Jun 28, 2016 at 10:59:32PM +0300, Alexander Shiyan wrote: > >Вторник, 28 июня 2016, 21:20 +03:00 от Nicolin Chen <nicoleotsuka(a)gmail.com>: > > > >On Sat, Jun 25, 2016 at 07:59:22AM +0300, Alexander Shiyan wrote: > >> The i.MX51 datasheet says: > >> Chapter 56.1.2.4 I2S Mode > >> ... > >> When I2S modes are entered (I2S master (01) or I2S slave (10)), > >> the following settings are recommended: > >> ... > >> - TX Frame Rate should be 2 i.e. (STCCR[12:8] = 1) > >> - RX Frame Rate should be 2 i.e. (SRCCR[12:8] = 1) > >> > >> Chapter 56.3.3.12 SSI Transmit and Receive Clock Control Registers (STCCR & SRCCR) > >> ... > >> Bits 12-8 DC4-DC0 > >> Frame Rate Divider Control. These bits are used to control the divide ratio > >> for the programmable frame rate dividers. The divide ratio works on the word > >> clock. In Normal mode, this ratio determines the word transfer rate. > >> In Network mode, this ratio sets the number of words per frame. The divide > >> ratio ranges from 1 to 32 in Normal mode and from 2 to 32 in Network mode. > >> In Normal mode, a divide ratio of 1 (DC=00000) provides continuous periodic > >> data word transfer. A bit-length frame sync must be used in this case. > >> > >> Function fsl_ssi_hw_params() setup Normal mode for MONO output, > >> so with DC=0, SSI enters to continuous periodic data word transfer. > >> To fix this, setup DC for any I2S mode. > > > >The change looks fine and didn't show any problem when I tested it. > >But I am wondering what's the failed user case here so we need this > >patch to fix? (The mono playback works without this patch as well.) > > Probably this works fine without patch if FSclk == BITclk * size. I acked this patch since it's more likely a correct configuration according to the reference manual while still want to know what's your failed case. Could you please tell me what were the FSclk, BITclk and size of your case? Thanks Nicolin

2 2

[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Fix number of words per frame for I2S-slave mode
by Alexander Shiyan 29 Jun '16

29 Jun '16

The i.MX51 datasheet says: Chapter 56.1.2.4 I2S Mode ... When I2S modes are entered (I2S master (01) or I2S slave (10)), the following settings are recommended: ... - TX Frame Rate should be 2 i.e. (STCCR[12:8] = 1) - RX Frame Rate should be 2 i.e. (SRCCR[12:8] = 1) Chapter 56.3.3.12 SSI Transmit and Receive Clock Control Registers (STCCR & SRCCR) ... Bits 12-8 DC4-DC0 Frame Rate Divider Control. These bits are used to control the divide ratio for the programmable frame rate dividers. The divide ratio works on the word clock. In Normal mode, this ratio determines the word transfer rate. In Network mode, this ratio sets the number of words per frame. The divide ratio ranges from 1 to 32 in Normal mode and from 2 to 32 in Network mode. In Normal mode, a divide ratio of 1 (DC=00000) provides continuous periodic data word transfer. A bit-length frame sync must be used in this case. Function fsl_ssi_hw_params() setup Normal mode for MONO output, so with DC=0, SSI enters to continuous periodic data word transfer. To fix this, setup DC for any I2S mode. Patch has tested on custom board based on Digi CCMX-51 module (i.MX51). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 632ecc0..bedec4a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -952,16 +952,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, ssi_private->i2s_mode = CCSR_SSI_SCR_NET; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + regmap_update_bits(regs, CCSR_SSI_STCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + regmap_update_bits(regs, CCSR_SSI_SRCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; - regmap_update_bits(regs, CCSR_SSI_STCCR, - CCSR_SSI_SxCCR_DC_MASK, - CCSR_SSI_SxCCR_DC(2)); - regmap_update_bits(regs, CCSR_SSI_SRCCR, - CCSR_SSI_SxCCR_DC_MASK, - CCSR_SSI_SxCCR_DC(2)); break; case SND_SOC_DAIFMT_CBM_CFM: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE; -- 2.4.9

3 3

[alsa-devel] [PATCH v4 0/4] ASoC: MAX9860: new driver
by Peter Rosin 29 Jun '16

29 Jun '16

Hi! I kept the sidetone changes in separate patches since I don't know if I solved the DAPM-off issue cleanly enough. Also, a new driver may still have a chance to make 4.7, but maybe it's too late for the DAPM changes? Finally, I wonder if the device tree crowd prefers seeing the whole of a series such as this, or only the bindings patch? Cheers, Peter New in v4: - Split out the device tree bindings in its own patch. - Move format check from ->startup to ->set_fmt. - Don't spam the log on probe defer. - Follow-up patches for the backwards sidetone control. New in v3: - The updated bindings file went missing in v2. Sorry for the confusion. New in v2: - Add comment about fall through when Integer Clock Mode is not possible. - Drop export of max9860_probe. - Ignore clk docs and read the mclk rate w/o enabling the clock. - Manage the DVDDIO supply. Peter Rosin (4): dt-bindings: sound: add bindings for the max9860 codec ASoC: MAX9860: new driver ASoC: dapm: support mixer controls with mute at non-zero value ASoC: MAX9860: add sidetone mixer control .../devicetree/bindings/sound/max9860.txt | 28 + MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 765 +++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 +++++ sound/soc/soc-dapm.c | 38 +- 7 files changed, 1005 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h -- 2.1.4

2 10

[alsa-devel] [PATCH] ASoC: cs35l33: mark PM functions as __maybe_unused
by Arnd Bergmann 29 Jun '16

29 Jun '16

The newly added cs35l33 driver produces a harmless warning when CONFIG_PM is disabled: sound/soc/codecs/cs35l33.c:908:12: error: 'cs35l33_runtime_suspend' defined but not used [-Werror=unused-function] sound/soc/codecs/cs35l33.c:868:12: error: 'cs35l33_runtime_resume' defined but not used [-Werror=unused-function] This adds __maybe_unused annotations to shut up the warning regardless of the configuration. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l33.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 622a1111b21c..d8b5fc3fc45d 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -862,7 +862,7 @@ static const struct regmap_config cs35l33_regmap = { .use_single_rw = true, }; -static int cs35l33_runtime_resume(struct device *dev) +static int __maybe_unused cs35l33_runtime_resume(struct device *dev) { struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); int ret; @@ -902,7 +902,7 @@ err: return ret; } -static int cs35l33_runtime_suspend(struct device *dev) +static int __maybe_unused cs35l33_runtime_suspend(struct device *dev) { struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); -- 2.9.0

3 3

[alsa-devel] [PATCH v3 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 29 Jun '16

29 Jun '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 58 +++++++++++++++++++++++++++++++++++----------- sound/soc/soc-pcm.c | 3 ++- 3 files changed, 51 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..8eed5c4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..a6976d5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,9 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition && custom_stop_condition(widget, dir)) + return con; + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1113,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1126,37 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1164,24 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1201,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1300,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1960,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

2 2

[alsa-devel] old bug in au88x0_core.c
by Dan Carpenter 29 Jun '16

29 Jun '16

Hi ALSA devs, The patch 1da177e4c3f4: "Linux-2.6.12-rc2" from Apr 16, 2005, leads to the following static checker warning: sound/pci/au88x0/au88x0_core.c:1449 vortex_wtdma_bufshift() warn: mask and shift to zero sound/pci/au88x0/au88x0_core.c 1441 static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma) 1442 { 1443 stream_t *dma = &vortex->dma_wt[wtdma]; 1444 int page, p, pp, delta, i; 1445 1446 page = 1447 (hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) & 1448 WT_SUBBUF_MASK) 1449 >> WT_SUBBUF_SHIFT; This is always zero because: #define WT_SUBBUF_MASK 0x3 #define WT_SUBBUF_SHIFT 0x15 1450 if (dma->nr_periods >= 4) 1451 delta = (page - dma->period_real) & 3; 1452 else { 1453 delta = (page - dma->period_real); 1454 if (delta < 0) 1455 delta += dma->nr_periods; 1456 } 1457 if (delta == 0) 1458 return 0; 1493 #if 0 1494 static void 1495 vortex_wtdma_getposition(vortex_t * vortex, int wtdma, int *subbuf, int *pos) 1496 { 1497 int temp; 1498 temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)); 1499 *subbuf = (temp >> WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK; >From this ifdef zero code, it looks like we should shift first then do the mask. 1500 *pos = temp & POS_MASK; 1501 } 1502 1503 static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) 1504 { 1505 return ((hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) >> 1506 POS_SHIFT) & POS_MASK); 1507 } 1508 #endif regards, dan carpenter

2 1

[alsa-devel] [PATCH] ctl: use consistent name for second argument of snd_ctl_elem_read/write
by Takashi Sakamoto 29 Jun '16

29 Jun '16

Second argument of these functions has three different names in documentation/header/code (obj/control/value). This easily causes users' confusion. This commit applies consistent names for the arguments. Fixes: 90020c05f886 ('ctl: improve comments for handling element data') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/control.h | 4 ++-- src/control/control.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/control.h b/include/control.h index 0d4ae86..6931b1f 100644 --- a/include/control.h +++ b/include/control.h @@ -245,8 +245,8 @@ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe); int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info); int snd_ctl_elem_list(snd_ctl_t *ctl, snd_ctl_elem_list_t *list); int snd_ctl_elem_info(snd_ctl_t *ctl, snd_ctl_elem_info_t *info); -int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *value); -int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *value); +int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *data); +int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *data); int snd_ctl_elem_lock(snd_ctl_t *ctl, snd_ctl_elem_id_t *id); int snd_ctl_elem_unlock(snd_ctl_t *ctl, snd_ctl_elem_id_t *id); int snd_ctl_elem_tlv_read(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, diff --git a/src/control/control.c b/src/control/control.c index 6090abd..bbb5a78 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -803,27 +803,27 @@ int snd_ctl_elem_remove(snd_ctl_t *ctl, snd_ctl_elem_id_t *id) /** * \brief Get CTL element value * \param ctl CTL handle - * \param obj Data of an element. + * \param data Data of an element. * \return 0 on success otherwise a negative error code */ -int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *control) +int snd_ctl_elem_read(snd_ctl_t *ctl, snd_ctl_elem_value_t *data) { - assert(ctl && control && (control->id.name[0] || control->id.numid)); - return ctl->ops->element_read(ctl, control); + assert(ctl && data && (data->id.name[0] || data->id.numid)); + return ctl->ops->element_read(ctl, data); } /** * \brief Set CTL element value * \param ctl CTL handle - * \param obj Data of an element. + * \param data Data of an element. * \retval 0 on success * \retval >0 on success when value was changed * \retval <0 a negative error code */ -int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *control) +int snd_ctl_elem_write(snd_ctl_t *ctl, snd_ctl_elem_value_t *data) { - assert(ctl && control && (control->id.name[0] || control->id.numid)); - return ctl->ops->element_write(ctl, control); + assert(ctl && data && (data->id.name[0] || data->id.numid)); + return ctl->ops->element_write(ctl, data); } static int snd_ctl_tlv_do(snd_ctl_t *ctl, int op_flag, -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: hda - Add PCI ID for Kabylake-H
by Vinod Koul 29 Jun '16

29 Jun '16

Kabylake-H shows up as PCI ID 0xa2f0. We missed adding this earlier with other KBL IDs. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/hda_intel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 94089fc71884..e320c44714b1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -367,9 +367,10 @@ enum { #define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) #define IS_KBL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa171) #define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71) +#define IS_KBL_H(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa2f0) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \ - IS_KBL(pci) || IS_KBL_LP(pci) + IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2190,6 +2191,9 @@ static const struct pci_device_id azx_ids[] = { /* Kabylake-LP */ { PCI_DEVICE(0x8086, 0x9d71), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Kabylake-H */ + { PCI_DEVICE(0x8086, 0xa2f0), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, -- 1.9.1

2 1

[alsa-devel] [PATCH v7 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by Paul.Handrigan＠cirrus.com 29 Jun '16

29 Jun '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 48 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1315 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1591 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..b6eadce --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,48 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * 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 __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 60f8f13..02d2084 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -391,6 +392,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1264fa4..abf2126 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-arizona-objs := arizona.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -257,6 +258,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..841374a --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1315 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.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/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.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 <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_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 cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_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; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_set_bst_ipk(struct snd_soc_codec *codec, unsigned int bst) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + int ret = 0, steps = 0; + + /* Boost current in uA */ + if (bst > 3600000 || bst < 1850000) { + dev_err(codec->dev, "Invalid boost current %d\n", bst); + ret = -EINVAL; + goto err; + } + + if (bst % 15625) { + dev_err(codec->dev, "Current not a multiple of 15625uA (%d)\n", + bst); + ret = -EINVAL; + goto err; + } + + while (bst > 1850000) { + bst -= 15625; + steps++; + } + + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + steps+0x70); + +err: + return ret; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + cs35l33_set_bst_ipk(codec, cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "cirrus,hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "cirrus,mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "cirrus,release-rate", + &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "cirrus,ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "cirrus,ldo-path-disable", + &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "cirrus,ldo-entry-delay", + &val32) >= 0) + hg_config->ldo_entry_delay = val32; + + hg_config->vp_hg_auto = of_property_read_bool(hg, + "cirrus,vp-hg-auto"); + + if (of_property_read_u32(hg, "cirrus,vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + return IRQ_NONE; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "cirrus,boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "cirrus,ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "cirrus,boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "cirrus,imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l33_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l33", cs35l33); + if (ret != 0) + dev_warn(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .owner = THIS_MODULE, + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..c045737 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.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 __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

4 9

[alsa-devel] [PATCH] ASoC: cs35l33: Remove unnecessary free_irq call
by Axel Lin 28 Jun '16

28 Jun '16

Current code uses devm_request_threaded_irq() so it does not need to explicitly call free_irq() in .probe error path and .remove. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l33.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 55c1f75..90dc743 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1195,7 +1195,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "Failed to enable core supplies: %d\n", ret); - goto err_irq; + return ret; } if (cs35l33->reset_gpio) @@ -1251,7 +1251,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, if (ret < 0) { dev_err(&i2c_client->dev, "%s: Register codec failed\n", __func__); - goto err_irq; + goto err_enable; } return 0; @@ -1259,8 +1259,6 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, err_enable: regulator_bulk_disable(cs35l33->num_core_supplies, cs35l33->core_supplies); -err_irq: - free_irq(i2c_client->irq, cs35l33); return ret; } @@ -1277,7 +1275,6 @@ static int cs35l33_i2c_remove(struct i2c_client *client) pm_runtime_disable(&client->dev); regulator_bulk_disable(cs35l33->num_core_supplies, cs35l33->core_supplies); - free_irq(client->irq, cs35l33); return 0; } -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoC: cs35l33: Remove setting dapm->bias_level in cs35l33_set_bias_level
by Axel Lin 28 Jun '16

28 Jun '16

This is done by ASoC core now. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l33.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 90dc743..622a111 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -365,7 +365,6 @@ static int cs35l33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int val; - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); switch (level) { @@ -392,8 +391,6 @@ static int cs35l33_set_bias_level(struct snd_soc_codec *codec, return -EINVAL; } - dapm->bias_level = level; - return 0; } -- 2.5.0

3 2

[alsa-devel] [PATCH] ASoC: fsl_spdif: fix spelling mistake: "receivce" -> "receive"
by Colin King 28 Jun '16

28 Jun '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 151849f..beec793 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -172,7 +172,7 @@ static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name) if (*pos >= size * 2) { *pos = 0; } else if (unlikely((*pos % size) + 3 > size)) { - dev_err(&pdev->dev, "User bit receivce buffer overflow\n"); + dev_err(&pdev->dev, "User bit receive buffer overflow\n"); return; } -- 2.8.1

2 2

[alsa-devel] [PATCH] ALSA: echoaudio: Fix memory allocation
by Christophe JAILLET 28 Jun '16

28 Jun '16

'commpage_bak' is allocated with 'sizeof(struct echoaudio)' bytes. We then copy 'sizeof(struct comm_page)' bytes in it. On my system, smatch complains because one is 2960 and the other is 3072. This would result in memory corruption or a oops. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/pci/echoaudio/echoaudio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1cb85ae..286f5e3 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2200,11 +2200,11 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.7.4

3 2

[alsa-devel] [alsa-lib][PATCH] ctl: use automatic variable instead of call of alloca(3)
by Takashi Sakamoto 27 Jun '16

27 Jun '16

Inner this library, layouts of all structures are public. At a compilation time, each size of the structures can be calculated. It means that we can use automatic variable instead of calling alloca(3) to program this library because in both ways storages are kept on stack frame of process VMA. Besides, the usage of automatic variables requires less instructions than calls of alloca(3). Furthermore, alloca(3) is not described in any C language standards. This commit replaces calls of alloca(3) just for structures with automatic variables, for control features. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 201 +++++++++++++++++++++++-------------------------- src/control/ctlparse.c | 7 +- src/control/namehint.c | 40 +++++----- src/control/setup.c | 5 +- src/control/tlv.c | 15 ++-- 5 files changed, 125 insertions(+), 143 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 0cb0105..5c0efc6 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -347,8 +347,8 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int member_count, long min, long max, long step) { - snd_ctl_elem_info_t *info; - snd_ctl_elem_value_t *data; + snd_ctl_elem_info_t info = {0}; + snd_ctl_elem_value_t data = {0}; unsigned int i; unsigned int j; unsigned int numid; @@ -356,38 +356,36 @@ int snd_ctl_elem_add_integer_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_INTEGER; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.integer.min = min; - info->value.integer.max = max; - info->value.integer.step = step; - - err = ctl->ops->element_add(ctl, info); + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_INTEGER; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.integer.min = min; + info.value.integer.max = max; + info.value.integer.step = step; + + err = ctl->ops->element_add(ctl, &info); if (err < 0) return err; - numid = snd_ctl_elem_id_get_numid(&info->id); + numid = snd_ctl_elem_id_get_numid(&info.id); /* Set initial value to all of members in all of added elements. */ - snd_ctl_elem_value_alloca(&data); - data->id = info->id; + data.id = info.id; for (i = 0; i < element_count; i++) { - snd_ctl_elem_id_set_numid(&data->id, numid + i); + snd_ctl_elem_id_set_numid(&data.id, numid + i); for (j = 0; j < member_count; j++) - data->value.integer.value[j] = min; + data.value.integer.value[j] = min; - err = ctl->ops->element_write(ctl, data); + err = ctl->ops->element_write(ctl, &data); if (err < 0) return err; } - *id = info->id; + *id = info.id; return 0; } @@ -436,8 +434,8 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int member_count, long long min, long long max, long long step) { - snd_ctl_elem_info_t *info; - snd_ctl_elem_value_t *data; + snd_ctl_elem_info_t info = {0}; + snd_ctl_elem_value_t data = {0}; unsigned int i; unsigned int j; unsigned int numid; @@ -445,38 +443,36 @@ int snd_ctl_elem_add_integer64_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_INTEGER64; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.integer64.min = min; - info->value.integer64.max = max; - info->value.integer64.step = step; - - err = ctl->ops->element_add(ctl, info); + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_INTEGER64; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.integer64.min = min; + info.value.integer64.max = max; + info.value.integer64.step = step; + + err = ctl->ops->element_add(ctl, &info); if (err < 0) return err; - numid = snd_ctl_elem_id_get_numid(&info->id); + numid = snd_ctl_elem_id_get_numid(&info.id); /* Set initial value to all of members in all of added elements. */ - snd_ctl_elem_value_alloca(&data); - data->id = info->id; + data.id = info.id; for (i = 0; i < element_count; i++) { - snd_ctl_elem_id_set_numid(&data->id, numid + i); + snd_ctl_elem_id_set_numid(&data.id, numid + i); for (j = 0; j < member_count; j++) - data->value.integer64.value[j] = min; + data.value.integer64.value[j] = min; - err = ctl->ops->element_write(ctl, data); + err = ctl->ops->element_write(ctl, &data); if (err < 0) return err; } - *id = info->id; + *id = info.id; return 0; } @@ -520,25 +516,24 @@ int snd_ctl_elem_add_boolean_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int element_count, unsigned int member_count) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; int err; assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_BOOLEAN; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.integer.min = 0; - info->value.integer.max = 1; - - err = ctl->ops->element_add(ctl, info); + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_BOOLEAN; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.integer.min = 0; + info.value.integer.max = 1; + + err = ctl->ops->element_add(ctl, &info); if (err >= 0) - *id = info->id; + *id = info.id; return err; } @@ -590,22 +585,21 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int items, const char *const labels[]) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; unsigned int i, bytes; char *buf, *p; int err; assert(ctl && id && id->name[0] && labels); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_ENUMERATED; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; - info->value.enumerated.items = items; + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_ENUMERATED; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; + info.value.enumerated.items = items; bytes = 0; for (i = 0; i < items; ++i) @@ -615,17 +609,17 @@ int snd_ctl_elem_add_enumerated_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, buf = malloc(bytes); if (buf == NULL) return -ENOMEM; - info->value.enumerated.names_ptr = (uintptr_t)buf; - info->value.enumerated.names_length = bytes; + info.value.enumerated.names_ptr = (uintptr_t)buf; + info.value.enumerated.names_length = bytes; p = buf; for (i = 0; i < items; ++i) { strcpy(p, labels[i]); p += strlen(labels[i]) + 1; } - err = ctl->ops->element_add(ctl, info); + err = ctl->ops->element_add(ctl, &info); if (err >= 0) - *id = info->id; + *id = info.id; free(buf); @@ -673,23 +667,22 @@ int snd_ctl_elem_add_bytes_set(snd_ctl_t *ctl, snd_ctl_elem_id_t *id, unsigned int element_count, unsigned int member_count) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; int err; assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_BYTES; - info->access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | - SNDRV_CTL_ELEM_ACCESS_USER; - info->owner = element_count; - info->count = member_count; + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_BYTES; + info.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_USER; + info.owner = element_count; + info.count = member_count; - err = ctl->ops->element_add(ctl, info); + err = ctl->ops->element_add(ctl, &info); if (err >= 0) - *id = info->id; + *id = info.id; return err; } @@ -705,12 +698,11 @@ int snd_ctl_elem_add_integer(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, long min, long max, long step) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_integer_set(ctl, local_id, 1, member_count, + return snd_ctl_elem_add_integer_set(ctl, &local_id, 1, member_count, min, max, step); } @@ -725,12 +717,11 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, long long min, long long max, long long step) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_integer64_set(ctl, local_id, 1, member_count, + return snd_ctl_elem_add_integer64_set(ctl, &local_id, 1, member_count, min, max, step); } @@ -744,12 +735,11 @@ int snd_ctl_elem_add_integer64(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, int snd_ctl_elem_add_boolean(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_boolean_set(ctl, local_id, 1, member_count); + return snd_ctl_elem_add_boolean_set(ctl, &local_id, 1, member_count); } /** @@ -765,12 +755,11 @@ int snd_ctl_elem_add_enumerated(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, unsigned int member_count, unsigned int items, const char *const labels[]) { - snd_ctl_elem_id_t *local_id; + snd_ctl_elem_id_t local_id = {0}; - snd_ctl_elem_id_alloca(&local_id); - *local_id = *id; + local_id = *id; - return snd_ctl_elem_add_enumerated_set(ctl, local_id, 1, member_count, + return snd_ctl_elem_add_enumerated_set(ctl, &local_id, 1, member_count, items, labels); } @@ -805,15 +794,15 @@ int snd_ctl_elem_add_enumerated(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, */ int snd_ctl_elem_add_iec958(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; assert(ctl && id && id->name[0]); - snd_ctl_elem_info_alloca(&info); - info->id = *id; - info->type = SND_CTL_ELEM_TYPE_IEC958; - info->owner = 1; - info->count = 1; - return ctl->ops->element_add(ctl, info); + + info.id = *id; + info.type = SND_CTL_ELEM_TYPE_IEC958; + info.owner = 1; + info.count = 1; + return ctl->ops->element_add(ctl, &info); } /** diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index d38b44e..5c9a168 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -312,17 +312,16 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, const char *value) { const char *ptr = value; - snd_ctl_elem_id_t *myid; + snd_ctl_elem_id_t myid = {0}; snd_ctl_elem_type_t type; unsigned int idx, count; long tmp; long long tmp64; - snd_ctl_elem_id_alloca(&myid); - snd_ctl_elem_info_get_id(info, myid); + snd_ctl_elem_info_get_id(info, &myid); type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); - snd_ctl_elem_value_set_id(dst, myid); + snd_ctl_elem_value_set_id(dst, &myid); for (idx = 0; idx < count && idx < 128 && ptr && *ptr; idx++) { if (*ptr == ',') diff --git a/src/control/namehint.c b/src/control/namehint.c index ad8dda3..4dbd216 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -97,45 +97,42 @@ static int get_dev_name1(struct hint_list *list, char **res, int device, #ifdef BUILD_HWDEP case SND_CTL_ELEM_IFACE_HWDEP: { - snd_hwdep_info_t *info; - snd_hwdep_info_alloca(&info); - snd_hwdep_info_set_device(info, device); - if (snd_ctl_hwdep_info(list->ctl, info) < 0) + snd_hwdep_info_t info = {0}; + snd_hwdep_info_set_device(&info, device); + if (snd_ctl_hwdep_info(list->ctl, &info) < 0) return 0; - *res = strdup(snd_hwdep_info_get_name(info)); + *res = strdup(snd_hwdep_info_get_name(&info)); return 0; } #endif #ifdef BUILD_PCM case SND_CTL_ELEM_IFACE_PCM: { - snd_pcm_info_t *info; - snd_pcm_info_alloca(&info); - snd_pcm_info_set_device(info, device); - snd_pcm_info_set_stream(info, stream ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK); - if (snd_ctl_pcm_info(list->ctl, info) < 0) + snd_pcm_info_t info = {0}; + snd_pcm_info_set_device(&info, device); + snd_pcm_info_set_stream(&info, stream ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK); + if (snd_ctl_pcm_info(list->ctl, &info) < 0) return 0; - switch (snd_pcm_info_get_class(info)) { + switch (snd_pcm_info_get_class(&info)) { case SND_PCM_CLASS_MODEM: case SND_PCM_CLASS_DIGITIZER: return -ENODEV; default: break; } - *res = strdup(snd_pcm_info_get_name(info)); + *res = strdup(snd_pcm_info_get_name(&info)); return 0; } #endif #ifdef BUILD_RAWMIDI case SND_CTL_ELEM_IFACE_RAWMIDI: { - snd_rawmidi_info_t *info; - snd_rawmidi_info_alloca(&info); - snd_rawmidi_info_set_device(info, device); - snd_rawmidi_info_set_stream(info, stream ? SND_RAWMIDI_STREAM_INPUT : SND_RAWMIDI_STREAM_OUTPUT); - if (snd_ctl_rawmidi_info(list->ctl, info) < 0) + snd_rawmidi_info_t info = {0}; + snd_rawmidi_info_set_device(&info, device); + snd_rawmidi_info_set_stream(&info, stream ? SND_RAWMIDI_STREAM_INPUT : SND_RAWMIDI_STREAM_OUTPUT); + if (snd_ctl_rawmidi_info(list->ctl, &info) < 0) return 0; - *res = strdup(snd_rawmidi_info_get_name(info)); + *res = strdup(snd_rawmidi_info_get_name(&info)); return 0; } #endif @@ -419,11 +416,10 @@ static int add_card(snd_config_t *config, snd_config_t *rw_config, struct hint_l snd_config_iterator_t i, next; const char *str; char ctl_name[16]; - snd_ctl_card_info_t *info; + snd_ctl_card_info_t info = {0}; int device, max_device = 0; - snd_ctl_card_info_alloca(&info); - list->info = info; + list->info = &info; err = snd_config_search(config, list->siface, &conf); if (err < 0) return err; @@ -431,7 +427,7 @@ static int add_card(snd_config_t *config, snd_config_t *rw_config, struct hint_l err = snd_ctl_open(&list->ctl, ctl_name, 0); if (err < 0) return err; - err = snd_ctl_card_info(list->ctl, info); + err = snd_ctl_card_info(list->ctl, &info); if (err < 0) goto __error; snd_config_for_each(i, next, conf) { diff --git a/src/control/setup.c b/src/control/setup.c index f23bf2c..c598c8b 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -252,13 +252,12 @@ static int snd_config_get_ctl_elem_value(snd_config_t *conf, { int err; snd_config_iterator_t i, next; - snd_ctl_elem_id_t *id; + snd_ctl_elem_id_t id = {0}; snd_ctl_elem_type_t type; unsigned int count; long v; long idx; - snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_value_get_id(val, id); + snd_ctl_elem_value_get_id(val, &id); count = snd_ctl_elem_info_get_count(info); type = snd_ctl_elem_info_get_type(info); if (count == 1) { diff --git a/src/control/tlv.c b/src/control/tlv.c index b08d887..6ad64a1 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -409,20 +409,19 @@ struct tlv_info { static int get_tlv_info(snd_ctl_t *ctl, const snd_ctl_elem_id_t *id, struct tlv_info *rec) { - snd_ctl_elem_info_t *info; + snd_ctl_elem_info_t info = {0}; int err; - snd_ctl_elem_info_alloca(&info); - snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_info(ctl, info); + snd_ctl_elem_info_set_id(&info, id); + err = snd_ctl_elem_info(ctl, &info); if (err < 0) return err; - if (!snd_ctl_elem_info_is_tlv_readable(info)) + if (!snd_ctl_elem_info_is_tlv_readable(&info)) return -EINVAL; - if (snd_ctl_elem_info_get_type(info) != SND_CTL_ELEM_TYPE_INTEGER) + if (snd_ctl_elem_info_get_type(&info) != SND_CTL_ELEM_TYPE_INTEGER) return -EINVAL; - rec->minval = snd_ctl_elem_info_get_min(info); - rec->maxval = snd_ctl_elem_info_get_max(info); + rec->minval = snd_ctl_elem_info_get_min(&info); + rec->maxval = snd_ctl_elem_info_get_max(&info); err = snd_ctl_elem_tlv_read(ctl, id, rec->buf, sizeof(rec->buf)); if (err < 0) return err; -- 2.7.4

2 1

[alsa-devel] period time vs. buffer time
by Joël Krähemann 27 Jun '16

27 Jun '16

Hi all Do I need to call both functions in order to get good performance? snd_pcm_hw_params_set_period_time() snd_pcm_hw_params_set_buffer_time() Or is one enough? Since I call: snd_pcm_hw_params_set_buffer_size() I have developed an application called gsequencer I experience good throughput but playback collapses after a specified amount of time. The application is unusable after 60 seconds of playback. http://www.nongnu.org/gsequencer/ http://www.nongnu.org/gsequencer/api/ags/AgsThread.html - does the JIFFIEs http://www.nongnu.org/gsequencer/api/ags/AgsDevout.html - does setup audio HW http://www.nongnu.org/gsequencer/api/ags/AgsSoundcardThread.html - calls ags_soundcard_play() I do some kind of overclocking within thread frequency. I ceil() the frequency of the thread in Hertz jiffies and add one. Within this thread I call: snd_pcm_writei() In numbers: 2 audio channels 44100 samplerate 944 buffer size in samples = 2 * 944 frames = 2 * 944 * sizeof(signed short) 48 times per second call: snd_pcm_writei() I use the soundcard in non-blocking mode because I need the time to compute the rest of the tree within time. The question is how to handle lost precision? As I round values and do some overclocking to ensure alsa has enough frames to do the playback. What do I have to do in order to get great performance and throughput? Do I need to drop frames? Or even buffers? And what is alsa's job? So playback gets delay by nanosleep() as following: if(g_atomic_pointer_get(&(thread->parent)) == NULL){ gdouble time_spent, relative_time_spent; gdouble time_cycle; static const gdouble nsec_per_jiffie = NSEC_PER_SEC / AGS_THREAD_HERTZ_JIFFIE; static const gdouble lost_per_jiffie = nsec_per_jiffie / AGS_THREAD_MAX_PRECISION; gdouble lost_precision; if(thread->tic_delay == thread->delay){ struct timespec timed_sleep = { 0, 0, }; clock_gettime(CLOCK_MONOTONIC, &time_now); if(time_now.tv_sec == thread->computing_time->tv_sec + 1){ time_spent = (time_now.tv_nsec) + (NSEC_PER_SEC - thread->computing_time->tv_nsec); }else if(time_now.tv_sec > thread->computing_time->tv_sec + 1){ time_spent = (time_now.tv_sec - thread->computing_time->tv_sec) * NSEC_PER_SEC; time_spent += (time_now.tv_nsec - thread->computing_time->tv_nsec); }else{ time_spent = time_now.tv_nsec - thread->computing_time->tv_nsec; } //FIXME:JK: do optimization time_cycle = ((gdouble) NSEC_PER_SEC / thread->freq); relative_time_spent = time_cycle - time_spent; if(relative_time_spent > 0.0 && relative_time_spent < time_cycle){ timed_sleep.tv_nsec = (long) relative_time_spent; /* lost precision */ //NOTE:JK: this is experimental lost_precision = lost_per_jiffie / (1.0 + (1.0 / thread->freq)); if(relative_time_spent - lost_precision > 0.0){ timed_sleep.tv_nsec = (long) (relative_time_spent - lost_precision); }else{ g_warning("ags_thread-posix.c - not optimized tree\0"); } /* sleep the rest of available time of the period */ nanosleep(&timed_sleep, NULL); } clock_gettime(CLOCK_MONOTONIC, thread->computing_time); } Here's a piece of soundcard initialization: /* set the period time */ period_time = (USEC_PER_SEC / (devout->samplerate)); dir = -1; err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir); if (err < 0) { pthread_mutex_unlock(mutex); str = snd_strerror(err); g_warning("Unable to set period time %i for playback: %s\n", period_time, str); free(str); return; } My problem with alsa is that as I don't tell the exact values I get good throughput but playback collapses. When I do exact values the sound is soften and loses volume. Strange is I can sometimes hear some crackling and I'm unsure what to set as period time. Does it need the time to do periodic things or should it be reduced or overloaded to have accurate time? Bests, Joël -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

1 0

[alsa-devel] [alsa-lib][PATCH] ctl: fix returning zero for dimension level 4
by Takashi Sakamoto 27 Jun '16

27 Jun '16

In ALSA kernel/userspace interfaces, 'struct snd_ctl_elem_info' has a 'dimension' parameter. This parameter consists of an array with four elements. Each element represents the number of members in corresponding dimension level to construct matrix. In current implementation, a get function, 'snd_ctl_elem_info_get_dimension()' is hardcoded to return zero to level 4, against actual value. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index d515cff..0cb0105 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2483,7 +2483,7 @@ int snd_ctl_elem_info_get_dimension(const snd_ctl_elem_info_t *obj, unsigned int #endif { assert(obj); - if (idx >= 3) + if (idx > 3) return 0; return obj->dimen.d[idx]; } -- 2.7.4

2 1

[alsa-devel] [PATCH 00/39 v2][RESEND] ASoC: add simple-card-core and standardize "simple" card
by Kuninori Morimoto 27 Jun '16

27 Jun '16

Hi Mark These are resend of v2 of simple-card cleanup patches. My previous patch-set was sent almost 1month ago. Some of them are aleady accepted, but nothing happen for almost all other patches. Because of this stagnation, I already have almost 90 patches for simple-card + graph driver in locally... I would like to post them as soon as possible. OTOH, I know you don't like super big patchset, Thus, I will post small batch patch set only today. I will post next small batch patch set if 1st batch was accepted. I listed my local patch set below. It is for simple-card + graph support, and HDMI support for Renesas. This time I will post * mark patches. These are for simple-card cleanup. Kuninori Morimoto (90): * ASoC: add new simple-card-utils.c * ASoC: simple-card-utils: add asoc_simple_card_parse_tdm() * ASoC: simple-card-utils: add asoc_simple_card_parse_dailink_name() * ASoC: simple-card-utils: add asoc_simple_card_parse_card_name() * ASoC: simple-card-utils: add asoc_simple_card_parse_card_prefix() * ASoC: simple-card-utils: add asoc_simple_card_parse_clk() * ASoC: simple-card-utils: add asoc_simple_card_parse_endpoint() * ASoC: simple-card-utils: add asoc_simple_card_init_dai() * ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() * ASoC: simple-card-utils: add asoc_simple_card_canonicalize_cpu() * ASoC: simple-card-utils: add asoc_simple_card_clean_reference() ASoC: simple-card: use common PREFIX for each DT property ASoC: simple-card: add new asoc_simple_jack and use it ASoC: simple-card: use asoc_simple_card_parse_daifmt() ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: simple-card: use asoc_simple_card_parse_endpoint() ASoC: simple-card: use asoc_simple_card_parse_tdm() ASoC: simple-card: use asoc_simple_card_parse_card_name() ASoC: simple-card: use asoc_simple_card_parse_dailink_name() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_cpu() ASoC: simple-card: use asoc_simple_card_clean_reference() ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_endpoint() ASoC: rsrc-card: use asoc_simple_card_parse_card_name() ASoC: rsrc-card: use asoc_simple_card_parse_card_prefix() ASoC: rsrc-card: use asoc_simple_card_parse_tdm() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_canonicalize_cpu() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_clean_reference() ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase3 Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port/endpoint_type_is() of_graph: add of_graph_get_endpoint_count() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() drm: bridge: add DesignWare HDMI I2S audio support ASoC: hdmi-codec: callback function will be called with private data ASoC: hdmi-codec: enable multi probe for same device ASoC: soc-core: snd_soc_get_dai_name() become non static ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix ASoC: simple-card-utils: rename asoc_simple_card_parse_endpoint to dai ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_try_to_probe_graph_card() ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_prefix ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: use kzalloc() for dai_props / dai_link ASoC: simple-dpcm-card: tidyup codec daifmt handling method ASoC: simple-dpcm-card: tidyup asoc_simple_card_parse_links() method ASoC: simple-dpcm-card: depends on CONFIG_OF ASoC: simple-dpcm-card: add 2 CPU 1 Codec example ASoC: simple-dpcm-card: code sync: follow to simple family style ASoC: simple-dpcm-card: is GPL v2 ASoC: simple-dpcm-card: code sync: move asoc_simple_card_of_match ASoC: simple-dpcm-card: remove unused definitions ASoC: simple-dpcm-card: code sync: rename asoc_simple_card_priv ASoC: simple-dpcm-card: code sync: tidyup props/link naming ASoC: add simple-graph-card document ASoC: add simple-graph-card support ASoC: add simple-graph-dpcm-card document ASoC: add simple-graph-dpcm-card support ASoC: rsnd: remove duplicate define of rsnd_dvc_of_node() ASoC: rsnd: amend .probe/.remove call for DPCM ASoC: rsnd: add rsnd_mod_next() for for_each_rsnd_mod_xxx() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_dai_call() ASoC: rsnd: use for_each_rsnd_mod_xxx() on rsnd_rdai_continuance_probe() ASoC: rsnd: add rsnd_parse_of_node() and integrate rsnd_xxx_of_node ASoC: rsnd: add support graph base DT phase 1 ASoC: rsnd: add support graph base DT phase 2 ASoC: rsnd: depends on OF ASoC: rsnd: use simple-graph-card if possible ASoC: rsnd: add HDMI output support Documentation/devicetree/bindings/graph.txt | 26 + .../devicetree/bindings/sound/renesas,rsnd.txt | 2 + .../bindings/sound/renesas,rsrc-card.txt | 75 --- .../devicetree/bindings/sound/simple-dpcm-card.txt | 109 +++++ .../bindings/sound/simple-graph-card.txt | 65 +++ .../bindings/sound/simple-graph-dpcm-card.txt | 53 +++ drivers/gpu/drm/bridge/Kconfig | 8 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 + drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 123 +++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 +- drivers/gpu/drm/bridge/dw-hdmi.h | 21 + drivers/of/base.c | 197 +++++++- include/linux/of_graph.h | 67 +++ include/sound/hdmi-codec.h | 13 +- include/sound/simple_card.h | 11 +- include/sound/simple_card_utils.h | 92 ++++ include/sound/soc.h | 31 +- sound/soc/codecs/hdmi-codec.c | 81 +++- sound/soc/generic/Kconfig | 25 + sound/soc/generic/Makefile | 12 +- sound/soc/generic/simple-card-utils.c | 387 +++++++++++++++ sound/soc/generic/simple-card.c | 501 +++++++------------- sound/soc/generic/simple-dpcm-card.c | 323 +++++++++++++ sound/soc/generic/simple-graph-card.c | 452 ++++++++++++++++++ sound/soc/generic/simple-graph-dpcm-card.c | 409 ++++++++++++++++ sound/soc/sh/Kconfig | 8 +- sound/soc/sh/rcar/Makefile | 3 - sound/soc/sh/rcar/core.c | 318 ++++++++----- sound/soc/sh/rcar/dma.c | 11 + sound/soc/sh/rcar/dvc.c | 2 - sound/soc/sh/rcar/gen.c | 2 + sound/soc/sh/rcar/rsnd.h | 62 ++- sound/soc/sh/rcar/rsrc-card.c | 527 --------------------- sound/soc/sh/rcar/ssi.c | 27 +- sound/soc/sh/rcar/ssiu.c | 37 ++ sound/soc/soc-core.c | 39 +- 37 files changed, 3003 insertions(+), 1146 deletions(-)

3 16

[alsa-devel] [PATCH] ASOC: wm8753: Remove unneeded header file
by Amitoj Kaur Chawla 26 Jun '16

26 Jun '16

Drop redundant include of moduleparam.h The Coccinelle semantic patch used to make this change is as follows: @ includesmodule @ @@ #include <linux/module.h> @ depends on includesmodule @ @@ - #include <linux/moduleparam.h> Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/soc/codecs/wm8753.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6f1024f..b4e6893 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -32,7 +32,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.9.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: codecs: wm8753: Replace magic number
by Alexander Shiyan 26 Jun '16

26 Jun '16

Use SND_SOC_NOPM constant, instead of -1. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6f1024f..9f851b8 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -486,7 +486,7 @@ SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0), SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls), SND_SOC_DAPM_OUTPUT("MONO2"), -SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Out3 Left + Right", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls), SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("OUT3"), -- 2.4.9

2 2

[alsa-devel] [PATCH] SoC: dwc: trivial fix of spelling mistake "unsuppted" -> "unsupported"
by Colin King 26 Jun '16

26 Jun '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/dwc/designware_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 0db69b7..1223ba6 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -278,7 +278,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, break; default: - dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt"); + dev_err(dev->dev, "designware-i2s: unsupported PCM fmt"); return -EINVAL; } -- 2.8.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: tpa6130a2: Add support for multiple instances
by Helen Koike 26 Jun '16

26 Jun '16

The current tpa6130a2 driver supports only a single instance. This patch series add support for multiple instances by removing the global variable that holds the instance. This is performed by using the component API, regmap, the snd_soc_{info,put,get}_volsw API and DAPM. This patch series also touches code from the Nokia RX51 which I didn't tested (as I am testing the tpa6130a2 in another board that is not upstream). I would appreciate is if someone who possesses the Nokia RX51 (n900) could please test the code. This patch series is based on git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Available at https://git.collabora.com/cgit/user/koike/linux.git/log/?h=sound/review/tpa… Helen Koike (4): ASoC: tpa6130a2: Register component ASoC: tap6130a2: Use regmap ASoC: tpa6130a2: Use snd soc volsw functions ASoC: tpa6130a2: Add DAPM support sound/soc/codecs/tpa6130a2.c | 382 ++++++++++++------------------------------- sound/soc/codecs/tpa6130a2.h | 14 +- sound/soc/omap/rx51.c | 45 ++--- 3 files changed, 130 insertions(+), 311 deletions(-) -- 1.9.1

5 35

[alsa-devel] [PATCH] ASoC: Samsung: s3c-i2s-v2: fix spelling mistake: "unknwon" -> "unknown"
by Colin King 26 Jun '16

26 Jun '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in pr_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- 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 b6ab3fc..bf8ae79 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -268,7 +268,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod &= ~S3C2412_IISMOD_SLAVE; break; default: - pr_err("unknwon master/slave format\n"); + pr_err("unknown master/slave format\n"); return -EINVAL; } -- 2.8.1

2 1

[alsa-devel] [PATCH v4 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 26 Jun '16

26 Jun '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.7-rc4 kernel. Changes in v4: - Rebase to v4.7-rc4 - Use strcmp() in acpi_data_node_match() for matching ACPI data nodes. Changes in v3: - Use of_node_cmp() in device_get_named_child_node() to match DT node. Changes in v2: - Rebase to v4.7-rc1 - Small updates to codec patch based on previous reviewer comments Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/of.h | 14 +++--- include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 7 files changed, 119 insertions(+), 76 deletions(-) -- 1.9.3

5 9

[alsa-devel] [PATCH] ALSA: hda - fix read before array start
by Bob Copeland 26 Jun '16

26 Jun '16

UBSAN reports the following warning from accessing path->path[-1] in set_path_power(): [ 16.078040] ================================================================================ [ 16.078124] UBSAN: Undefined behaviour in sound/pci/hda/hda_generic.c:3981:17 [ 16.078198] index -1 is out of range for type 'hda_nid_t [10]' [ 16.078270] CPU: 2 PID: 1738 Comm: modprobe Not tainted 4.7.0-rc1-wt+ #47 [ 16.078274] Hardware name: LENOVO 3443CTO/3443CTO, BIOS G6ET23WW (1.02 ) 08/14/2012 [ 16.078278] ffff8800cb246000 ffff8800cb3638b8 ffffffff815c4fe3 0000000000000032 [ 16.078286] ffff8800cb3638e0 ffffffffffffffff ffff8800cb3638d0 ffffffff8162443d [ 16.078294] ffffffffa0894200 ffff8800cb363920 ffffffff81624af7 0000000000000292 [ 16.078302] Call Trace: [ 16.078311] [<ffffffff815c4fe3>] dump_stack+0x86/0xd3 [ 16.078317] [<ffffffff8162443d>] ubsan_epilogue+0xd/0x40 [ 16.078324] [<ffffffff81624af7>] __ubsan_handle_out_of_bounds+0x67/0x70 [ 16.078335] [<ffffffffa087665f>] set_path_power+0x1bf/0x230 [snd_hda_codec_generic] [ 16.078344] [<ffffffffa087880d>] add_pin_power_ctls+0x8d/0xc0 [snd_hda_codec_generic] [ 16.078352] [<ffffffffa087f190>] ? pin_power_down_callback+0x20/0x20 [snd_hda_codec_generic] [ 16.078360] [<ffffffffa0878947>] add_all_pin_power_ctls+0x107/0x150 [snd_hda_codec_generic] [ 16.078370] [<ffffffffa08842b3>] snd_hda_gen_parse_auto_config+0x2d73/0x49e0 [snd_hda_codec_generic] [ 16.078376] [<ffffffff81173360>] ? trace_hardirqs_on_caller+0x1b0/0x2c0 [ 16.078390] [<ffffffffa089df27>] alc_parse_auto_config+0x147/0x310 [snd_hda_codec_realtek] [ 16.078402] [<ffffffffa08a332a>] patch_alc269+0x23a/0x560 [snd_hda_codec_realtek] [ 16.078417] [<ffffffffa0838644>] hda_codec_driver_probe+0xa4/0x1a0 [snd_hda_codec] [ 16.078424] [<ffffffff817bbac1>] driver_probe_device+0x101/0x380 [ 16.078430] [<ffffffff817bbdf9>] __driver_attach+0xb9/0x100 [ 16.078438] [<ffffffff817bbd40>] ? driver_probe_device+0x380/0x380 [ 16.078444] [<ffffffff817b8d20>] bus_for_each_dev+0x70/0xc0 [ 16.078449] [<ffffffff817bb087>] driver_attach+0x27/0x50 [ 16.078454] [<ffffffff817ba956>] bus_add_driver+0x166/0x2c0 [ 16.078460] [<ffffffffa0369000>] ? 0xffffffffa0369000 [ 16.078465] [<ffffffff817bd13d>] driver_register+0x7d/0x130 [ 16.078477] [<ffffffffa083816f>] __hda_codec_driver_register+0x6f/0x90 [snd_hda_codec] [ 16.078488] [<ffffffffa036901e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek] [ 16.078493] [<ffffffff8100215e>] do_one_initcall+0x4e/0x1d0 [ 16.078499] [<ffffffff8119f54d>] ? rcu_read_lock_sched_held+0x6d/0x80 [ 16.078504] [<ffffffff813701b1>] ? kmem_cache_alloc_trace+0x391/0x560 [ 16.078510] [<ffffffff812bb314>] ? do_init_module+0x28/0x273 [ 16.078515] [<ffffffff812bb387>] do_init_module+0x9b/0x273 [ 16.078522] [<ffffffff811e3782>] load_module+0x20b2/0x3410 [ 16.078527] [<ffffffff811df140>] ? m_show+0x210/0x210 [ 16.078533] [<ffffffff813b2b26>] ? kernel_read+0x66/0xe0 [ 16.078541] [<ffffffff811e4cfa>] SYSC_finit_module+0xba/0xc0 [ 16.078547] [<ffffffff811e4d1e>] SyS_finit_module+0xe/0x10 [ 16.078552] [<ffffffff81a860fc>] entry_SYSCALL_64_fastpath+0x1f/0xbd [ 16.078556] ================================================================================ Fix by checking path->depth before use. Signed-off-by: Bob Copeland <me(a)bobcopeland.com> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 320445f3bf73..79c7b340acc2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3977,6 +3977,8 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, for (n = 0; n < spec->paths.used; n++) { path = snd_array_elem(&spec->paths, n); + if (!path->depth) + continue; if (path->path[0] == nid || path->path[path->depth - 1] == nid) { bool pin_old = path->pin_enabled; -- 2.6.1

2 1

[alsa-devel] UBSAN bug in hda_generic.c
by Bob Copeland 25 Jun '16

25 Jun '16

Hi, I have UBSAN reporting an out-of-bounds array access (see below) on my machine. The following patch fixes the warning for me, but not sure if that is just papering over some other bug. Thanks in advance for looking! >From 551e904f7a7aea9e9c03c439e554100643239c5c Mon Sep 17 00:00:00 2001 From: Bob Copeland <me(a)bobcopeland.com> Date: Sat, 25 Jun 2016 06:53:44 -0400 Subject: [PATCH] ALSA: hda - fix read before array start UBSAN reports the following warning from accessing path->path[-1] in set_path_power(): [ 16.078040] ================================================================================ [ 16.078124] UBSAN: Undefined behaviour in sound/pci/hda/hda_generic.c:3981:17 [ 16.078198] index -1 is out of range for type 'hda_nid_t [10]' [ 16.078270] CPU: 2 PID: 1738 Comm: modprobe Not tainted 4.7.0-rc1-wt+ #47 [ 16.078274] Hardware name: LENOVO 3443CTO/3443CTO, BIOS G6ET23WW (1.02 ) 08/14/2012 [ 16.078278] ffff8800cb246000 ffff8800cb3638b8 ffffffff815c4fe3 0000000000000032 [ 16.078286] ffff8800cb3638e0 ffffffffffffffff ffff8800cb3638d0 ffffffff8162443d [ 16.078294] ffffffffa0894200 ffff8800cb363920 ffffffff81624af7 0000000000000292 [ 16.078302] Call Trace: [ 16.078311] [<ffffffff815c4fe3>] dump_stack+0x86/0xd3 [ 16.078317] [<ffffffff8162443d>] ubsan_epilogue+0xd/0x40 [ 16.078324] [<ffffffff81624af7>] __ubsan_handle_out_of_bounds+0x67/0x70 [ 16.078335] [<ffffffffa087665f>] set_path_power+0x1bf/0x230 [snd_hda_codec_generic] [ 16.078344] [<ffffffffa087880d>] add_pin_power_ctls+0x8d/0xc0 [snd_hda_codec_generic] [ 16.078352] [<ffffffffa087f190>] ? pin_power_down_callback+0x20/0x20 [snd_hda_codec_generic] [ 16.078360] [<ffffffffa0878947>] add_all_pin_power_ctls+0x107/0x150 [snd_hda_codec_generic] [ 16.078370] [<ffffffffa08842b3>] snd_hda_gen_parse_auto_config+0x2d73/0x49e0 [snd_hda_codec_generic] [ 16.078376] [<ffffffff81173360>] ? trace_hardirqs_on_caller+0x1b0/0x2c0 [ 16.078390] [<ffffffffa089df27>] alc_parse_auto_config+0x147/0x310 [snd_hda_codec_realtek] [ 16.078402] [<ffffffffa08a332a>] patch_alc269+0x23a/0x560 [snd_hda_codec_realtek] [ 16.078417] [<ffffffffa0838644>] hda_codec_driver_probe+0xa4/0x1a0 [snd_hda_codec] [ 16.078424] [<ffffffff817bbac1>] driver_probe_device+0x101/0x380 [ 16.078430] [<ffffffff817bbdf9>] __driver_attach+0xb9/0x100 [ 16.078438] [<ffffffff817bbd40>] ? driver_probe_device+0x380/0x380 [ 16.078444] [<ffffffff817b8d20>] bus_for_each_dev+0x70/0xc0 [ 16.078449] [<ffffffff817bb087>] driver_attach+0x27/0x50 [ 16.078454] [<ffffffff817ba956>] bus_add_driver+0x166/0x2c0 [ 16.078460] [<ffffffffa0369000>] ? 0xffffffffa0369000 [ 16.078465] [<ffffffff817bd13d>] driver_register+0x7d/0x130 [ 16.078477] [<ffffffffa083816f>] __hda_codec_driver_register+0x6f/0x90 [snd_hda_codec] [ 16.078488] [<ffffffffa036901e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek] [ 16.078493] [<ffffffff8100215e>] do_one_initcall+0x4e/0x1d0 [ 16.078499] [<ffffffff8119f54d>] ? rcu_read_lock_sched_held+0x6d/0x80 [ 16.078504] [<ffffffff813701b1>] ? kmem_cache_alloc_trace+0x391/0x560 [ 16.078510] [<ffffffff812bb314>] ? do_init_module+0x28/0x273 [ 16.078515] [<ffffffff812bb387>] do_init_module+0x9b/0x273 [ 16.078522] [<ffffffff811e3782>] load_module+0x20b2/0x3410 [ 16.078527] [<ffffffff811df140>] ? m_show+0x210/0x210 [ 16.078533] [<ffffffff813b2b26>] ? kernel_read+0x66/0xe0 [ 16.078541] [<ffffffff811e4cfa>] SYSC_finit_module+0xba/0xc0 [ 16.078547] [<ffffffff811e4d1e>] SyS_finit_module+0xe/0x10 [ 16.078552] [<ffffffff81a860fc>] entry_SYSCALL_64_fastpath+0x1f/0xbd [ 16.078556] ================================================================================ Fix by checking path->depth before use. Signed-off-by: Bob Copeland <me(a)bobcopeland.com> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 320445f3bf73..bc23a9d8e7b3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3978,7 +3978,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, for (n = 0; n < spec->paths.used; n++) { path = snd_array_elem(&spec->paths, n); if (path->path[0] == nid || - path->path[path->depth - 1] == nid) { + (path->depth > 0 && path->path[path->depth - 1] == nid)) { bool pin_old = path->pin_enabled; bool stream_old = path->stream_enabled; -- 2.6.1 -- Bob Copeland %% http://bobcopeland.com/

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[alsa-devel] snd_pcm_open() leaks error
by Joël Krähemann 25 Jun '16

25 Jun '16

Hi all I have some issues as opening soundcard multiple times the error is handled but it leaks memory: http://pastebin.com/aQfAQV2q The problematic function is snd_pcm_open() but I think it doesn't have any performance impact on ̀gsequencer` http://www.nongnu.org/gsequencer/ Bests, Joël -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

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