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[alsa-devel] [patch] ASoC: cs35l35: returning uninitialized in probe()
by Dan Carpenter 13 Mar '17

13 Mar '17

If cs35l35->pdata.stereo is false then "ret" isn't initialized. Fixes: 6387f866a2cc ("ASoC: Add support for Cirrus Logic CS35L35 Amplifier") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 260ed42c71e9..142ae642eaa1 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -964,7 +964,7 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) } } - return ret; + return 0; } static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = {

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[alsa-devel] [PATCH v2 1/2] ASoC: rt5665: move set_pll to codec level
by Bard Liao 13 Mar '17

13 Mar '17

Move set_pll function to codec level and people can use it at both codec and dai level. Also, lower case "source" to keep it consistent. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 1baa9ce..b36e217 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4222,15 +4222,15 @@ static int rt5665_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, - unsigned int freq_in, unsigned int freq_out) +static int rt5665_set_codec_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) { - struct snd_soc_codec *codec = dai->codec; struct rt5665_priv *rt5665 = snd_soc_codec_get_drvdata(codec); struct rl6231_pll_code pll_code; int ret; - if (Source == rt5665->pll_src && freq_in == rt5665->pll_in && + if (source == rt5665->pll_src && freq_in == rt5665->pll_in && freq_out == rt5665->pll_out) return 0; @@ -4244,7 +4244,7 @@ static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, return 0; } - switch (Source) { + switch (source) { case RT5665_PLL1_S_MCLK: snd_soc_update_bits(codec, RT5665_GLB_CLK, RT5665_PLL1_SRC_MASK, RT5665_PLL1_SRC_MCLK); @@ -4262,7 +4262,7 @@ static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, RT5665_PLL1_SRC_MASK, RT5665_PLL1_SRC_BCLK3); break; default: - dev_err(codec->dev, "Unknown PLL Source %d\n", Source); + dev_err(codec->dev, "Unknown PLL Source %d\n", source); return -EINVAL; } @@ -4284,7 +4284,7 @@ static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, rt5665->pll_in = freq_in; rt5665->pll_out = freq_out; - rt5665->pll_src = Source; + rt5665->pll_src = source; return 0; } @@ -4403,7 +4403,6 @@ static const struct snd_soc_dai_ops rt5665_aif_dai_ops = { .set_fmt = rt5665_set_dai_fmt, .set_sysclk = rt5665_set_dai_sysclk, .set_tdm_slot = rt5665_set_tdm_slot, - .set_pll = rt5665_set_dai_pll, .set_bclk_ratio = rt5665_set_bclk_ratio, }; @@ -4512,7 +4511,8 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5665 = { .num_dapm_widgets = ARRAY_SIZE(rt5665_dapm_widgets), .dapm_routes = rt5665_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt5665_dapm_routes), - } + }, + .set_pll = rt5665_set_codec_pll, }; -- 2.7.4

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[alsa-devel] [PATCH v3 1/2] ASoC: rt5665: move set_pll to codec level
by Bard Liao 13 Mar '17

13 Mar '17

Move set_pll function to codec level and people can use it at both codec and dai level. Also, lower case "source" to keep it consistent. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 1baa9ce..b36e217 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4222,15 +4222,15 @@ static int rt5665_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } -static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, - unsigned int freq_in, unsigned int freq_out) +static int rt5665_set_codec_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) { - struct snd_soc_codec *codec = dai->codec; struct rt5665_priv *rt5665 = snd_soc_codec_get_drvdata(codec); struct rl6231_pll_code pll_code; int ret; - if (Source == rt5665->pll_src && freq_in == rt5665->pll_in && + if (source == rt5665->pll_src && freq_in == rt5665->pll_in && freq_out == rt5665->pll_out) return 0; @@ -4244,7 +4244,7 @@ static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, return 0; } - switch (Source) { + switch (source) { case RT5665_PLL1_S_MCLK: snd_soc_update_bits(codec, RT5665_GLB_CLK, RT5665_PLL1_SRC_MASK, RT5665_PLL1_SRC_MCLK); @@ -4262,7 +4262,7 @@ static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, RT5665_PLL1_SRC_MASK, RT5665_PLL1_SRC_BCLK3); break; default: - dev_err(codec->dev, "Unknown PLL Source %d\n", Source); + dev_err(codec->dev, "Unknown PLL Source %d\n", source); return -EINVAL; } @@ -4284,7 +4284,7 @@ static int rt5665_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int Source, rt5665->pll_in = freq_in; rt5665->pll_out = freq_out; - rt5665->pll_src = Source; + rt5665->pll_src = source; return 0; } @@ -4403,7 +4403,6 @@ static const struct snd_soc_dai_ops rt5665_aif_dai_ops = { .set_fmt = rt5665_set_dai_fmt, .set_sysclk = rt5665_set_dai_sysclk, .set_tdm_slot = rt5665_set_tdm_slot, - .set_pll = rt5665_set_dai_pll, .set_bclk_ratio = rt5665_set_bclk_ratio, }; @@ -4512,7 +4511,8 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5665 = { .num_dapm_widgets = ARRAY_SIZE(rt5665_dapm_widgets), .dapm_routes = rt5665_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt5665_dapm_routes), - } + }, + .set_pll = rt5665_set_codec_pll, }; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: core: remove redundunt address operation for array type variable
by Takashi Sakamoto 13 Mar '17

13 Mar '17

The type of 'struct snd_ctl_elem_value.value.bytes.data' is 'unsigned char[512]'. In C language specification, address operation (&) for this type is just ignored, therefore: struct snd_ctl_elem_value elem_value; printf("%p\n", elem_value.value.bytes.data); printf("%p\n", &elem_value.value.bytes.data); These two evaluation have the same value, a pointer to the first element of the array. This kind of address operation is not good for readers, because it easily brings confusions as an address operations to pointers. For example, struct snd_kcontrol_new template; printf("%p\n", template.name); printf("%p\n", &template.name); These two evaluation have the different value. The former is 'unsigned char *', while the latter is 'unsigned char **'. This commit removes redundant address operation for array type variable to avoid misreading in reader side. Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Fixes: f831b055ecec ("ASoC: core: Add support for masking out parts of coefficient blocks") Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 500f98c..76b199b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -658,11 +658,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, ucontrol->value.bytes.data[0] &= ~params->mask; break; case 2: - ((u16 *)(&ucontrol->value.bytes.data))[0] + ((u16 *)ucontrol->value.bytes.data)[0] &= cpu_to_be16(~params->mask); break; case 4: - ((u32 *)(&ucontrol->value.bytes.data))[0] + ((u32 *)ucontrol->value.bytes.data)[0] &= cpu_to_be32(~params->mask); break; default: -- 2.9.3

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[alsa-devel] [PATCH] ASoC: cs35l35: Fix display revision id
by Axel Lin 13 Mar '17

13 Mar '17

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 260ed42..9c82e5e 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1468,7 +1468,7 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "Cirrus Logic CS35L35 (%x), Revision: %02X\n", devid, - ret & 0xFF); + reg & 0xFF); /* Set the INT Masks for critical errors */ regmap_write(cs35l35->regmap, CS35L35_INT_MASK_1, -- 2.9.3

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[alsa-devel] [PATCH] ASoC: rt286: fix headphone click/crack noise on Dell XPS 9343 I2S mode
by Kai-Heng Feng 13 Mar '17

13 Mar '17

HDA mode does not have these issues because necessary workarounds in linux/sound/pci/hda/patch_realtek.c are already applied. So we can leverage these workaournds into rt286. When jack is plugged, it rapidly generates I2C interrupts, which triggers rt286_irq() and rt286_jack_detect(), which causes the noise. alc_fixup_dell_xps13() in patch_realtek.c sets up a pin that can stop the frantic interrupts, hence fixes the click noise. When rt286 is under powersaving state, play a sound with headphone or plug a headphone in will produce a loud crack sound. alc_shutup_dell_xps13() patch_realtek.c mutes the headphone at plugging. Apply the same workaround to LDO2 power event can make the loud crack sound to a more tolerable pop sound. I found that unconditionally apply the workaround to all related power event can help a little further. Link: https://bugzilla.kernel.org/show_bug.cgi?id=112611 Link: https://bugzilla.redhat.com/show_bug.cgi?id=1313434 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/soc/codecs/rt286.c | 67 +++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 63 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 74c0e4e..d041937 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,6 +36,8 @@ #define RT286_VENDOR_ID 0x10ec0286 #define RT288_VENDOR_ID 0x10ec0288 +#define AMP_OUT_MUTE 0xb080 + struct rt286_priv { struct reg_default *index_cache; int index_cache_size; @@ -47,6 +49,7 @@ struct rt286_priv { struct delayed_work jack_detect_work; int sys_clk; int clk_id; + void (*mute_hpo_func)(struct snd_soc_codec *codec); }; static const struct reg_default rt286_index_def[] = { @@ -472,10 +475,51 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } +/* Add HV, VREF and LDO1 event functions to workaround headphone crack noise */ +static int rt286_hv_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); + + return 0; +} + +static int rt286_vref_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); + + return 0; +} + +static int rt286_ldo1_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); + + return 0; +} + static int rt286_ldo2_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->mute_hpo_func) + rt286->mute_hpo_func(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -516,16 +560,24 @@ static int rt286_mic1_event(struct snd_soc_dapm_widget *w, return 0; } +static void dell_dino_mute_hpo(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, RT286_SET_AMP_GAIN_HPO, AMP_OUT_MUTE); +} + static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1, - 12, 1, NULL, 0), + 12, 1, rt286_hv_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1, - 0, 1, NULL, 0), + 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2, - 2, 0, NULL, 0), + 2, 0, rt286_ldo1_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1, 13, 1, rt286_ldo2_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MCLK MODE", RT286_PLL_CTRL1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, @@ -1190,6 +1242,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { + /* Fix headphone click noise */ + if (dmi_check_system(dmi_dell_dino)) + regmap_write(rt286->regmap, + RT286_MIC1_DET_CTRL, 0x0020); + regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0x5000); } @@ -1204,6 +1261,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + /* Power down LDO, VREF */ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0); regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); @@ -1222,6 +1280,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, RT286_SET_GPIO_DATA, 0x40, 0x40); regmap_update_bits(rt286->regmap, RT286_GPIO_CTRL, 0xc, 0x8); + rt286->mute_hpo_func = dell_dino_mute_hpo; } if (rt286->i2c->irq) { -- 2.10.0

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[alsa-devel] Support for hardware amplifier
by Felix Bruns 13 Mar '17

13 Mar '17

Hi, I am working on embedded device (using an i.MX6 MCU) which has an external amplifier. Unfortunately I could not find much information if and how the gain of hardware amplifiers are controlled by alsa. (The only example for hardware amplification I found is the arcam-av plugin...) Is creating a plugin the recommended way of supporting (external) hardware amplification in alsa? Can somebody clarify or point me to another example? Thanks and kind regards, Felix Bruns ps: Sorry for posting this twice.

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Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Mark Brown 13 Mar '17

13 Mar '17

On Fri, Mar 10, 2017 at 04:24:07PM -0800, Brian Norris wrote: > On Thu, Mar 09, 2017 at 12:09:18PM +0100, Mark Brown wrote: > > Please think hard before including complete backtraces in upstream > > reports, they are very large and contain almost no useful information > > relative to their size so often obscure the relevant content in your > > message. If part of the backtrace is usefully illustrative then it's > > usually better to pull out the relevant sections. > I did think, and I did trim the trace significantly. I believe (or > believed; maybe incorrectly) that much of the remaining context *was* > useful, and that it could have answered some of Kuninori's questions, if > he chose to dig himself. As admitted in the commit message, I don't > really understand all of the relationships here, so it's hard to > highlight "only the relevant sections". Any editing you'd done is really not obvious looking at the trace (the inclusion of the top of the trace with the pgd and so on which really need the image to be useful is a bit of a flag here, as is the bit at the end after the end of the trace). It's certainly not mentioned or referenced. At the very least for probe errors it's generally safe to cut down to just the driver probe function is being called rather than showing the entire device model call stack, it's vanishingly rare for that to be adding anything and it's generally several times deeper than what the driver is doing. > I'm sorry that my post did not meet your standards though. Next time > I'll just post a revert with little context and no attempt to understand > what actually went wrong. Providing changes with little context and no conveyed understanding of exactly what went wrong is a very big part of the reason for complaining about people just splatting backtraces in and certainly not what I was asking for, it's sadly common to see a backtrace provided in lieu of any analysis and it does make the whole thing harder to read. That would be even more problematic than what you sent, I wouldn't have applied such a change at all unless I somehow reverse engineered the analysis. You do have a reasonable analysis of your change in the first couple of lines ("Not all platform drivers have pcm_{new,free} callbacks...") which is why I applied it, it's at least not going to do any harm to check that there's a function being provided.

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[alsa-devel] Applied "ASoC: don't dereference NULL pcm_{new, free}" to the asoc tree
by Mark Brown 13 Mar '17

13 Mar '17

The patch ASoC: don't dereference NULL pcm_{new,free} has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d6c098a1db468b7fd4635e831f276851dfd8852c Mon Sep 17 00:00:00 2001 From: Brian Norris <briannorris(a)chromium.org> Date: Wed, 8 Mar 2017 15:18:54 -0800 Subject: [PATCH] ASoC: don't dereference NULL pcm_{new,free} Not all platform drivers have pcm_{new,free} callbacks. Seen with a "snd-soc-dummy" codec from sound/soc/rockchip/rk3399_gru_sound.c. Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") Signed-off-by: Brian Norris <briannorris(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6dca408faae3..2722bb0c5573 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3326,7 +3326,10 @@ static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_platform *platform = rtd->platform; - return platform->driver->pcm_new(rtd); + if (platform->driver->pcm_new) + return platform->driver->pcm_new(rtd); + else + return 0; } static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) @@ -3334,7 +3337,8 @@ static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_soc_platform *platform = rtd->platform; - platform->driver->pcm_free(pcm); + if (platform->driver->pcm_free) + platform->driver->pcm_free(pcm); } /** -- 2.11.0

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[alsa-devel] Is snd_pcm_hw_constraint_mask64 really needed ?
by Chemsi Mehdi 13 Mar '17

13 Mar '17

Dear ALSA expert , I am working with an alsa codec driver where the driver is calling err = snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, formats); my driver is already setting format before that so I wonder why this function is called? Looks like it is needed to add audio constraints to substreams at open/startup time, but why ? what can be the issue if I do not call it ? Please explain to me Takashi. Thanks a lot in advance.

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[alsa-devel] Help debugging a sound problem
by Jorge Fernandez Monteagudo 13 Mar '17

13 Mar '17

Hi all, This is my first email to the list. I've joined to it because I don't know where to look for help. We have a iMX6 (ARM) board with a STA559BW audio codec connected by the I2S interface. We are using a 3.14.52 kernel with a custom sta559 codec driver based on the TAS5709 codec driver. Our system has the alsa library 1.0.28. Our application uses SDL 1.2.15 to handle the sound system. When we run the application the sound system works ok, but when we leave the application running without playing any sound for one night then next morning we found the sound is delayed more or less one second from when the sound is played until the sound it's heard. Traces in the SDL audiocallback are ok, the same as when the sound is played ok. Another clue. An 'aplay' executed in another console is played ok with no delay. Another one. If the application process is attached with gdb and then deattached then the sounds works ok again! Any help is welcome in order to begin to debug this situation. Thanks! Jorge

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[alsa-devel] [PATCH v3 0/7] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 13 Mar '17

13 Mar '17

Hi Mark, Rob These are v3 of OF-graph base audio card patch-set. Main difference from v2 is that its name is Audio Graph Card instead of Simple Graph Card. I added 3 OF-graph helper functions for it. 1) - 3) : new OF-graph helper functions 4) - 7) : OF-graph base simple-card Kuninori Morimoto (7): 1) of_graph: add of_graph_get_remote_endpoint() 2) of_graph: add of_graph_get_port_parent() 3) of_graph: add of_graph_get_endpoint_count() 4) ASoC: add snd_soc_get_dai_id() 5) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 6) ASoC: add audio-graph-card document 7) ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 126 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 303 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 36 +++ sound/soc/soc-core.c | 23 ++ 10 files changed, 580 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

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Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Kuninori Morimoto 13 Mar '17

13 Mar '17

Hi Brian Thank you for your feedback > There are 4 drivers calling that: > > snd_soc_dummy_probe > rt5514_spi_probe > 2 instances of snd_dmaengine_pcm_register, via rockchip_i2s_probe > > Only the latter two seem to run the assignment here: > > if (platform_drv->pcm_new) > platform->component.pcm_new = snd_soc_platform_drv_pcm_new; > > Both snd_soc_dummy_probe and rt5514_spi_probe find ->pcm_new NULL here. Hmm... The crasher was snd_dmaengine_pcm_register's platform ? This means, in your current kernel, dmaengine platform dosn't call its .pcm_new (= dmaengine_pcm_new) somehow ? I'm wondering why ->pcm_new became NULL which exists on probe timing ? Can you check component and driver by this patch ? This is very rough but enough for debug --------------------- diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5933851..43da1ec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3322,6 +3322,10 @@ static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_platform *platform = rtd->platform; + printk("-------use name: %s, %p\n", + platform->component.name, + platform->driver); + return platform->driver->pcm_new(rtd); } @@ -3356,8 +3360,12 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->pcm_new) + if (platform_drv->pcm_new) { + printk("-------add name: %s, %p\n", + platform->component.name, + platform->driver); platform->component.pcm_new = snd_soc_platform_drv_pcm_new; + } if (platform_drv->pcm_free) platform->component.pcm_free = snd_soc_platform_drv_pcm_free; ---------------------

1 0

[alsa-devel] [PATCH] ASoC: cs35l35: trivial fix to indentation
by Colin King 12 Mar '17

12 Mar '17

From: Colin Ian King <colin.king(a)canonical.com> Remove extraneous tab to correct the nesting level indentation Detected by CoverityScan, CID#1416584 ("Nesting level does not match indentation") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/cs35l35.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 260ed42..43d33f4 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1370,7 +1370,8 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) cs35l35->supplies[i].supply = cs35l35_supplies[i]; - cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); + + cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); ret = devm_regulator_bulk_get(&i2c_client->dev, cs35l35->num_supplies, -- 2.10.2

2 1

Re: [alsa-devel] [PATCH 01/10] ALSA SoC MAX98927 driver - Initial release
by Rob Herring 12 Mar '17

12 Mar '17

On Fri, Mar 03, 2017 at 11:52:39PM +0900, Ryan Lee wrote: > From: Ryan Lee <RyanS.Lee(a)maximintegrated.com> > > Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> > --- > > Initial release of ALSA SoC MAX98927 driver. > > .../devicetree/bindings/sound/max98927.txt | 32 + In addition to what Mark said, please split bindings to a separate patch. > sound/soc/codecs/Kconfig | 5 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/max98927.c | 954 +++++++++++++++ > sound/soc/codecs/max98927.h | 1253 ++++++++++++++++++++ > 5 files changed, 2246 insertions(+) > create mode 100755 Documentation/devicetree/bindings/sound/max98927.txt > mode change 100644 => 100755 sound/soc/codecs/Kconfig > mode change 100644 => 100755 sound/soc/codecs/Makefile > create mode 100755 sound/soc/codecs/max98927.c > create mode 100755 sound/soc/codecs/max98927.h These files should not be executable. Rob

1 0

[alsa-devel] bcm2835-analog-audio: Is my driver PWM, ALSA or neither
by Michael Zoran 12 Mar '17

12 Mar '17

I have submitted to ALSA a driver that uses the PWM hardware of the Raspbery PI(bcm2835) as an audio driver. http://mailman.alsa-project.org/pipermail/alsa-devel/2017-March/118652. html The driver is currently mutually exclusive with the standard PWM driver, mostly because the standard PWM interface is not adequate to have an audio driver written on top of it. But as I think more, I see no reason the user needs to chose at boot time if they want traditional PWM or AUDIO. So I'm thinking of some how combining the two drivers. My question: 1. Is my driver a PWM driver that exposes an ALSA interface in additon to PWM? 2. Is my driver a ALSA driver that also exposes PWM. 3. Could my driver potentially go under misc or something new and expose both a PWM and ALSA interface. 4. Perhaps I should have both a PWM driver and a ALSA driver that both talk through a totally new but very small driver that controls access to the PWM hardware. How would the function drivers get loaded in this case? And the $10,000 question of the day is who should I be e-mailing the driver to and which topic branches should the driver be merged through?

1 0

Re: [alsa-devel] [PATCH] Meter causes freeze when closing audio device
by Alistair Buxton 12 Mar '17

12 Mar '17

After adding assert(0); to snd_timer_stop I got the following backtrace: #0 0x00007ffff730d428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 #1 0x00007ffff730f02a in __GI_abort () at abort.c:89 #2 0x00007ffff7305bd7 in __assert_fail_base (fmt=<optimised out>, assertion=assertion@entry=0x7ffff7b9439e "0", file=file@entry=0x7ffff7b9e8b9 "timer.c", line=line@entry=918, function=function@entry=0x7ffff7b9e9c8 <__PRETTY_FUNCTION__.10913> "snd_timer_stop") at assert.c:92 #3 0x00007ffff7305c82 in __GI___assert_fail (assertion=assertion@entry=0x7ffff7b9439e "0", file=file@entry=0x7ffff7b9e8b9 "timer.c", line=line@entry=918, function=function@entry=0x7ffff7b9e9c8 <__PRETTY_FUNCTION__.10913> "snd_timer_stop") at assert.c:101 #4 0x00007ffff7b64503 in snd_timer_stop (timer=<optimised out>) at timer.c:918 #5 0x00007ffff7b5158d in snd_pcm_dmix_sync_ptr0 (pcm=pcm@entry=0x622de0, slave_hw_ptr=<optimised out>) at pcm_dmix.c:421 #6 0x00007ffff7b51aac in snd_pcm_dmix_status (pcm=0x622de0, status=0x7ffff6af1e80) at pcm_dmix.c:496 #7 0x00007ffff7b18f11 in snd_pcm_status (pcm=pcm@entry=0x622de0, status=status@entry=0x7ffff6af1e80) at pcm.c:977 #8 0x00007ffff7b48bc0 in snd_pcm_meter_thread (data=0x622940) at pcm_meter.c:200 #9 0x00007ffff76a86ba in start_thread (arg=0x7ffff6af2700) at pthread_create.c:333 #10 0x00007ffff73de82d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:109 So the meter thread is indirectly stopping the timer while the main thread is trying to poll it. The following patch avoids this issue. Is it the right way to fix it? >From f21cd1ec8245198279cc2ad606041f7f59e8341d Mon Sep 17 00:00:00 2001 From: Alistair Buxton <a.j.buxton(a)gmail.com> Date: Mon, 6 Mar 2017 02:50:35 +0000 Subject: [PATCH] pcm: meter: Put a mutex around slave drain to prevent deadlock The meter thread calls _status on its slave. If this happens while dmix is draining, the timer will be stopped, causing the main thread to get stuck in poll(). --- src/pcm/pcm_meter.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index d71af7a..43a25fc 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -324,6 +324,18 @@ static int snd_pcm_meter_start(snd_pcm_t *pcm) return err; } +static int snd_pcm_meter_drain(snd_pcm_t *pcm) +{ + snd_pcm_meter_t *meter = pcm->private_data; + int err; + pthread_mutex_lock(&meter->running_mutex); + err = snd_pcm_drain(meter->gen.slave); + if (err >= 0) + pthread_cond_signal(&meter->running_cond); + pthread_mutex_unlock(&meter->running_mutex); + return err; +} + static snd_pcm_sframes_t snd_pcm_meter_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t frames) { snd_pcm_meter_t *meter = pcm->private_data; @@ -530,7 +542,7 @@ static const snd_pcm_fast_ops_t snd_pcm_meter_fast_ops = { .reset = snd_pcm_meter_reset, .start = snd_pcm_meter_start, .drop = snd_pcm_generic_drop, - .drain = snd_pcm_generic_drain, + .drain = snd_pcm_meter_drain, .pause = snd_pcm_generic_pause, .rewindable = snd_pcm_generic_rewindable, .rewind = snd_pcm_meter_rewind, -- 2.7.4 -- Alistair Buxton a.j.buxton(a)gmail.com

2 1

Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Mark Brown 09 Mar '17

09 Mar '17

On Wed, Mar 08, 2017 at 03:18:54PM -0800, Brian Norris wrote: > Not all platform drivers have pcm_{new,free} callbacks. Seen with a > "snd-soc-dummy" codec from sound/soc/rockchip/rk3399_gru_sound.c. > > Resolves an OOPS seen on v4.11-rc1 with Google Kevin (Samsung Chromebook > Plus): > > [ 2.863304] rk3399-gru-sound sound: HiFi <-> ff880000.i2s mapping ok > [ 2.886930] rk3399-gru-sound sound: rt5514-aif1 <-> ff880000.i2s mapping ok Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative then it's usually better to pull out the relevant sections.

1 0

[alsa-devel] (no subject)
by Felix Bruns 09 Mar '17

09 Mar '17

Hello, I am working on embedded device (using an i.MX6 MCU) which has an external amplifier for sound amplification. Unfortunately I could not find information if and how the gain of hardware amplifiers are controlled by ALSA. Can somebody clarify or point me to an example? Thanks and kind regards, Felix Bruns

1 0

Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Kuninori Morimoto 09 Mar '17

09 Mar '17

Hi Brian > > It is a littlle bit strange for me. > > Yes, and honestly I'm a little confused by the inheritance in this > framework. Yes, I agree :) This is 1st prepare for future ALSA SoC framework cleanup It is Lars-Peter's idea > I have a feeling you're checking the wrong thing below for this case. If so, we should fix this "wrong thing" ? > All I know is that I'm definitely hitting a NULL > platform->driver->pcm_new callback, and that either reverting your patch > or applying the patch I just sent fixes it. I want to know why this happen. Can you show me which driver is calling snd_soc_add_platform() in your case ?

1 0

Re: [alsa-devel] [PATCH for-4.11] ASoC: don't dereference NULL pcm_{new, free}
by Kuninori Morimoto 09 Mar '17

09 Mar '17

Hi Brian Thank you for your patch > Not all platform drivers have pcm_{new,free} callbacks. Seen with a > "snd-soc-dummy" codec from sound/soc/rockchip/rk3399_gru_sound.c. (snip) > Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") > Signed-off-by: Brian Norris <briannorris(a)chromium.org> > --- > I'm really not that familiar with this subsystem... but this does fix the > crash seen here. > > sound/soc/soc-core.c | 8 ++++++-- > 1 file changed, 6 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index 6dca408faae3..2722bb0c5573 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -3326,7 +3326,10 @@ static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_soc_platform *platform = rtd->platform; > > - return platform->driver->pcm_new(rtd); > + if (platform->driver->pcm_new) > + return platform->driver->pcm_new(rtd); > + else > + return 0; > } > > static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) > @@ -3334,7 +3337,8 @@ static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) > struct snd_soc_pcm_runtime *rtd = pcm->private_data; > struct snd_soc_platform *platform = rtd->platform; > > - platform->driver->pcm_free(pcm); > + if (platform->driver->pcm_free) > + platform->driver->pcm_free(pcm); > } It is a littlle bit strange for me. commit 99b04f4c4051 has below code. This means, if platform doesn't have pcm_new/free callback, component doesn't have snd_soc_platform_drv_pcm_new/free. But your case, platform doesn't have pcm_new/free, but component had it ? ... @@ -3181,6 +3198,10 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; + if (platform_drv->pcm_new) + platform->component.pcm_new = snd_soc_platform_drv_pcm_new; + if (platform_drv->pcm_free) + platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform";

1 0

Re: [alsa-devel] [RFC PATCH] sound: bcm2835-analog-audio: Add driver for bcm2835(Raspberry PI) headphone jack.
by Eric Anholt 09 Mar '17

09 Mar '17

Michael Zoran <mzoran(a)crowfest.net> writes: > Someone on the internet mentioned that if we have a HDMI Audio driver, why not > have a driver for the headphone/analog jack on the Raspberry PI that doesn't > use the firmware. So I quickly make up a prototype that isn't submit quality > yet, but is very functional. The idea is to like the firmware, just use the > PWM of the bcm2835 to generate the audio. > > Questions: > > 1. Is this something that people would want in the upstream kernel and > would be seriously considered for submission? I'm certainly interested -- the more we can take over from the closed source stack, the better Linux on the Raspberry Pi will be. Make sure you use standard DT properties for describing the DMA channel, add your new compatible string to pwm-bcm2835.txt, and document that that compat string requires the DMA properties. > 2. Is it a complete deal breaker that the driver is a regular ALSA driver > instead of a ASoC driver? I don't think so. I think the goal of ASoC is to provide utilities that make your SoC ALSA drivers smaller.

1 0

[alsa-devel] [PLUGINS] Multichannel over I2S : extra cost of user space plugins
by Matt Flax 08 Mar '17

08 Mar '17

Recently I was advised to channel multichannel audio over stereo I2S and unpack in user space with plugins. I have questions about this approach. Right now, with a hacked machine driver, I get DMA unpacking to multichannel for free. It works with all sample rates in silicon and it also works with all word sizes. If I switch to user space ALSA plugin unpacking, as advised, can anyone answer the following questions please : If I want to handle all word sizes which are currently implemented S16 -> S32, do I need to use different plugins to handle different word sizes ? Regarding the actual channel unpacking, will it cost a lot more to do this in user space, rather then using the DMA silicon ? The bcm2835 is not a very powerful processor and this is a concern. Does anyone have an example plugin which will handle all word sizes through one plugin device ? thanks Matt

2 1

[alsa-devel] [PATCH] ASoC: imx-pcm-dma: simplify pcm_config
by Lucas Stach 08 Mar '17

08 Mar '17

The generic snd_dmaengine_pcm is able to retrieve all the needed information from the attached dmaengine and is in fact able to provide much more accurate flags to userspace, like the SDMA engine being only able to operate in batch mode. To avoid any future inconsistencies between the dmaengine and the pcm_config, rip out the fixed config and rely on the core to fill in the right flags derived from the dmaengine information. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/fsl/imx-pcm-dma.c | 28 ---------------------------- 1 file changed, 28 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f3d3d1ffa84e..314814ddd2b0 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -33,48 +33,20 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static const struct snd_pcm_hardware imx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE, - .period_bytes_min = 128, - .period_bytes_max = 65535, /* Limited by SDMA engine */ - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { - .pcm_hardware = &imx_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .compat_filter_fn = filter, - .prealloc_buffer_size = IMX_DEFAULT_DMABUF_SIZE, }; int imx_pcm_dma_init(struct platform_device *pdev, size_t size) { struct snd_dmaengine_pcm_config *config; - struct snd_pcm_hardware *pcm_hardware; config = devm_kzalloc(&pdev->dev, sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL); if (!config) return -ENOMEM; *config = imx_dmaengine_pcm_config; - if (size) - config->prealloc_buffer_size = size; - - pcm_hardware = devm_kzalloc(&pdev->dev, - sizeof(struct snd_pcm_hardware), GFP_KERNEL); - *pcm_hardware = imx_pcm_hardware; - if (size) - pcm_hardware->buffer_bytes_max = size; - - config->pcm_hardware = pcm_hardware; return devm_snd_dmaengine_pcm_register(&pdev->dev, config, -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: use a helper macro to rounding-up calculation
by Takashi Sakamoto 08 Mar '17

08 Mar '17

In 'include/linux/kernel.h', there's a helper macro to round numerical value. Let's use it. Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> --- sound/soc/intel/skylake/skl-topology.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2dbfb1b..f38f8e6 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -299,8 +299,6 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, { int multiplier = 1; struct skl_module_fmt *in_fmt, *out_fmt; - int in_rate, out_rate; - /* Since fixups is applied to pin 0 only, ibs, obs needs * change for pin 0 only @@ -311,21 +309,13 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) multiplier = 5; - if (in_fmt->s_freq % 1000) - in_rate = (in_fmt->s_freq / 1000) + 1; - else - in_rate = (in_fmt->s_freq / 1000); - - mcfg->ibs = in_rate * (mcfg->in_fmt->channels) * + mcfg->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) * + (mcfg->in_fmt->channels) * (mcfg->in_fmt->bit_depth >> 3) * multiplier; - if (mcfg->out_fmt->s_freq % 1000) - out_rate = (mcfg->out_fmt->s_freq / 1000) + 1; - else - out_rate = (mcfg->out_fmt->s_freq / 1000); - - mcfg->obs = out_rate * (mcfg->out_fmt->channels) * + mcfg->obs = DIV_ROUND_UP(mcfg->out_fmt->s_freq, 1000) * + (mcfg->out_fmt->channels) * (mcfg->out_fmt->bit_depth >> 3) * multiplier; } -- 2.9.3

3 2

[alsa-devel] [PATCH] ASoC: fsl: Remove unneeded init of static variable
by Alin Grigorean 08 Mar '17

08 Mar '17

This was reported by checkpatch.pl Signed-off-by: Alin Grigorean <alin.grig93(a)gmail.com> --- sound/soc/fsl/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index dac6688540dc..92410f7ca1fa 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -282,7 +282,7 @@ static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static int ssi_irq = 0; +static int ssi_irq; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { -- 2.11.1

3 2

[alsa-devel] [PATCH] ASoC: rt5665: fix getting wrong work handler container
by Bard Liao 08 Mar '17

08 Mar '17

We got rt5665 private data from wrong work. It will result in kernel panic. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index c866502..a0c1d9a 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1241,7 +1241,7 @@ static irqreturn_t rt5665_irq(int irq, void *data) static void rt5665_jd_check_handler(struct work_struct *work) { struct rt5665_priv *rt5665 = container_of(work, struct rt5665_priv, - calibrate_work.work); + jd_check_work.work); if (snd_soc_read(rt5665->codec, RT5665_AJD1_CTRL) & 0x0010) { /* jack out */ -- 2.7.4

2 1

[alsa-devel] [PATCH 1/8] ASoC: rt5665: increase LDO level
by Bard Liao 08 Mar '17

08 Mar '17

Too low LDO level will cause a few functions unstable. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index fe2cf1e..a4c07c2 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4798,7 +4798,7 @@ static int rt5665_i2c_probe(struct i2c_client *i2c, /* Enhance performance*/ regmap_update_bits(rt5665->regmap, RT5665_PWR_ANLG_1, RT5665_HP_DRIVER_MASK | RT5665_LDO1_DVO_MASK, - RT5665_HP_DRIVER_5X | RT5665_LDO1_DVO_09); + RT5665_HP_DRIVER_5X | RT5665_LDO1_DVO_12); INIT_DELAYED_WORK(&rt5665->jack_detect_work, rt5665_jack_detect_handler); -- 2.7.4

2 13

Re: [alsa-devel] [PATCH 6/8] ASoC: rt5665: move set_pll to codec level
by Mark Brown 08 Mar '17

08 Mar '17

On Wed, Mar 08, 2017 at 11:19:52AM +0000, Bard Liao wrote: > This patch doesn't use lower case of 's'ource in the function body. > Could we apply the first 5 patches if there is no comment? > Or ignore the error and I will send a patch to lower case in the function > body? I'll skip this patch, please send a new version.

1 0

Re: [alsa-devel] [RFC] Merge DSD_U32_BE devices list
by Jurgen Kramer 08 Mar '17

08 Mar '17

Hello Naoki, On Tue, 2017-03-07 at 15:53 +0900, Naoki MATSUMOTO wrote: > Dear Mr.lintweaker > CC:alsa-ml > > Hello nice to meet you. > I know that you work for xmos-native-dsd. > You has contributed linux, alsa-lib and mpd. > You have patches of xmos-native-dsd device list on github. > https://github.com/lintweaker/xmos-native-dsd/tree/master/SRPMS/patch > es/kernel > so, I want to merge upstreame kernel. because, MPD 0.20 over support > DSD_U32_BE. > > Current kernel has few support devices. > https://patchwork.kernel.org/project/alsa-devel/list/?q=dsd > > I want to create patch. Could you please tell me anything problems? > I am planning on sending in a patch(set) adding all remaining devices not in the kernel yet. Time frame: +/- 2 weeks. Jurgen

2 1

[alsa-devel] [PATCH] ASoC: bcm_2835: Simplify mono guard by returning error on mono
by Matt Flax 08 Mar '17

08 Mar '17

This patch simplifies the bcm2835_i2s.c code when the driver returns error on mono channel requests. It does this by testing for mono instead of using a switch statement to test for stereo. The bcm2835_i2s driver can only work in stereo mode. Whilst leaving the driver entirel compliant with the I2S protocol, it simplifies the code by removing a few redundant lines of code. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..1e6a1da 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -310,15 +310,16 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, ch1pos = data_delay; ch2pos = bclk_ratio / 2 + data_delay; - switch (params_channels(params)) { - case 2: - format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); - format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); - format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); - break; - default: + /* C + * Check we aren't setting channel 2 on a mono stream. + * We currently only support stereo + */ + if (params_channels(params) == 1) return -EINVAL; - } + + format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); + format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); + format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); /* * Set format for both streams. -- 2.7.4

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[alsa-devel] DSD native format on SoC
by Michael Nazzareno Trimarchi 08 Mar '17

08 Mar '17

Hi DSD is a continuous stream of bits that flows over two separate channels, left and right, synchronized by a clock, while I2S is a single data wire and an additional wire that states if the incoming sample refers to the left channel or the right channel. There is no way to get DSD data other than a circuit which decode DSD streams coming directly from a conventional source, like, for examples, a SACD. The Volta dac has an internal decoding circuitry that still employs the I2S standard as a usual 44100Hz/16 bits, but sourcing the shift clock at twice the DSD standard frequency of 2.8224MHz, since it has to split two the16 bits words received on a single wire into the two DSD channels, as the standard wants. When DSD is enabled, the control signal PCD/DSD must be high, CK3..CK0 must be all low. The I2S interface works as a standard 44100Hz/16 bit and the DSD streaming must be packed into 16 bits lenght words left/right as per LRCK logic. The BCLK frequency supplied from the interface is 5644800Hz in case of DSD and 11289600 for DSD2. Can you explain how now DSD can be pass to audio card? The idea is to declare it on the audio card and then configure it as a PCM card but I don't know how data are suppose to arrive from alsa userspace. Michael

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[alsa-devel] [RESEND RFC PATCH] sound: bcm2835-analog-audio: Add driver for bcm2835(Raspberry PI) headphone jack.
by Michael Zoran 08 Mar '17

08 Mar '17

I tried sending this a few times, but I'm not sure it made it through to the e-mail list. Someone on the internet mentioned that if we have a HDMI Audio driver, why not have a driver for the headphone/analog jack on the Raspberry PI that doesn't use the firmware. So I quickly make up a prototype that isn't submit quality yet, but is very functional. The idea is to like the firmware, just use the PWM of the bcm2835 to generate the audio. Questions: 1. Is this something that people would want in the upstream kernel and would be seriously considered for submission? 2. Is it a complete deal breaker that the driver is a regular ALSA driver instead of a ASoC driver? TODO(Things I will do before submitting for real): 1. I found several people on the Internet that have written bare metal applications to do the same thing as this driver. One thing they all typically have is more signal conditioning(higher hardware bitrates, noise shaping filters, support For multiple software formats). 2. The DMA engine doesn't have support for bus priorities even though the hardware supports it. It might be good to put support for this in the DMA engine driver since it might be required for HDMI audio as well. 3. Checkpatch.pl/Formatting issues. --- arch/arm/boot/dts/bcm283x.dtsi | 19 ++ drivers/dma/bcm2835-dma.c | 33 ++- sound/arm/Kconfig | 10 +- sound/arm/Makefile | 3 + sound/arm/bcm2835-analog-audio.c | 585 +++++++++++++++++++++++++++++++++++++++ 5 files changed, 641 insertions(+), 9 deletions(-) create mode 100644 sound/arm/bcm2835-analog-audio.c diff --git a/arch/arm/boot/dts/bcm283x.dtsi b/arch/arm/boot/dts/bcm283x.dtsi index 9798bc9293d8..0810931addd8 100644 --- a/arch/arm/boot/dts/bcm283x.dtsi +++ b/arch/arm/boot/dts/bcm283x.dtsi @@ -338,6 +338,12 @@ brcm,pins = <42 43>; brcm,function = <BCM2835_FSEL_ALT5>; }; + + analog_audio_pins: analog_audio_pins { + brcm,pins = <40 45>; + brcm,function = <4 4>; /* Alt0 */ + }; + }; uart0: serial@7e201000 { @@ -464,6 +470,19 @@ status = "disabled"; }; + analogaudio: analogaudio@7e20c000 { + compatible = "brcm,bcm2835-analog-audio"; + reg = <0x7e20c000 0x28>; + clocks = <&clocks BCM2835_CLOCK_PWM>; + assigned-clocks = <&clocks BCM2835_CLOCK_PWM>; + pinctrl-names = "default"; + pinctrl-0 = <&analog_audio_pins>; + assigned-clock-rates = <250000000>; + + dmas = <&dma 5>; + dma-names = "tx"; + }; + sdhci: sdhci@7e300000 { compatible = "brcm,bcm2835-sdhci"; reg = <0x7e300000 0x100>; diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c index 6204cc32d09c..73b50009ea4b 100644 --- a/drivers/dma/bcm2835-dma.c +++ b/drivers/dma/bcm2835-dma.c @@ -179,6 +179,17 @@ struct bcm2835_desc { #define MAX_DMA_LEN SZ_1G #define MAX_LITE_DMA_LEN (SZ_64K - 4) +static inline u32 bcm2835_get_priority(struct bcm2835_chan *c) +{ + if (c->dreq != 5) + return BCM2835_DMA_PRIORITY(0) | BCM2835_DMA_PANIC_PRIORITY(0); + + else { +// printk( KERN_ERR "Setting priority to 15 for DRAQ 5"); + return BCM2835_DMA_PRIORITY(15) | BCM2835_DMA_PANIC_PRIORITY(15); + } +} + static inline size_t bcm2835_dma_max_frame_length(struct bcm2835_chan *c) { /* lite and normal channels have different max frame length */ @@ -465,7 +476,7 @@ static void bcm2835_dma_start_desc(struct bcm2835_chan *c) c->desc = d = to_bcm2835_dma_desc(&vd->tx); writel(d->cb_list[0].paddr, c->chan_base + BCM2835_DMA_ADDR); - writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); + writel(BCM2835_DMA_ACTIVE | bcm2835_get_priority(c), c->chan_base + BCM2835_DMA_CS); } static irqreturn_t bcm2835_dma_callback(int irq, void *data) @@ -485,24 +496,30 @@ static irqreturn_t bcm2835_dma_callback(int irq, void *data) spin_lock_irqsave(&c->vc.lock, flags); - /* Acknowledge interrupt */ - writel(BCM2835_DMA_INT, c->chan_base + BCM2835_DMA_CS); - d = c->desc; if (d) { if (d->cyclic) { + + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT | BCM2835_DMA_ACTIVE | bcm2835_get_priority(c), c->chan_base + BCM2835_DMA_CS); + /* call the cyclic callback */ vchan_cyclic_callback(&d->vd); - - /* Keep the DMA engine running */ - writel(BCM2835_DMA_ACTIVE, - c->chan_base + BCM2835_DMA_CS); } else { + + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT | bcm2835_get_priority(c), c->chan_base + BCM2835_DMA_CS); + vchan_cookie_complete(&c->desc->vd); bcm2835_dma_start_desc(c); } } + else + { + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT, c->chan_base + BCM2835_DMA_CS); + } spin_unlock_irqrestore(&c->vc.lock, flags); diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..71aef1dae6b8 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -2,7 +2,7 @@ menuconfig SND_ARM bool "ARM sound devices" - depends on ARM + depends on ARM || ARM64 default y help Support for sound devices specific to ARM architectures. @@ -11,6 +11,14 @@ menuconfig SND_ARM if SND_ARM +config SND_BCM2835_ANALOG_AUDIO + tristate "Analog/Headphone jack support for BCM2835" + depends on ARCH_BCM2835 && HAS_DMA + select SND_PCM + help + Say y or m to support the analog/headphone jack + for the BCM2835 as used by the Raspberry PI. + config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 8c0c851d4641..744ff77e580a 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -14,3 +14,6 @@ snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o snd-pxa2xx-ac97-objs := pxa2xx-ac97.o + +obj-$(CONFIG_SND_BCM2835_ANALOG_AUDIO) += bcm2835-analog-audio.o + diff --git a/sound/arm/bcm2835-analog-audio.c b/sound/arm/bcm2835-analog-audio.c new file mode 100644 index 000000000000..b04dbe068ecd --- /dev/null +++ b/sound/arm/bcm2835-analog-audio.c @@ -0,0 +1,585 @@ +/* + * Michael Zoran <mzoran(a)crowfest.net> + * + * 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; version 2. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/err.h> +#include <linux/io.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <linux/of_address.h> +#include <linux/dma-mapping.h> + +/* + * The channel order that needs to be passed to the PWM FIFO is opposite the + * order that is passed by the application. So the order needs to be flipped + * in software. + */ + + +struct bcm2835_hardware_frame +{ + u32 right; + u32 left; +}; + +struct bcm2835_software_frame +{ + s16 left; + s16 right; +}; + +#define HARDWARE_BUFFER_FRAMES_PER_PERIOD 720 +#define HARDWARE_BUFFER_PERIODS_PER_BUFFER 2 +#define HARDWARE_BUFFER_PERIOD_BYTES (sizeof(struct bcm2835_hardware_frame) * \ + HARDWARE_BUFFER_FRAMES_PER_PERIOD) +#define HARDWARE_BUFFER_BYTES (HARDWARE_BUFFER_PERIOD_BYTES * \ + HARDWARE_BUFFER_PERIODS_PER_BUFFER ) + +struct bcm2835_chip; + +struct bcm2835_chip_runtime { + struct bcm2835_chip *chip; + struct snd_pcm_substream *substream; + struct dma_slave_config dma_slave_config; + struct dma_async_tx_descriptor *dma_desc; + dma_cookie_t dma_cookie; + struct bcm2835_hardware_frame *hardware_buffer; + dma_addr_t hardware_buffer_dma; + int hardware_period_number; + struct bcm2835_software_frame *playback_src_buffer; + snd_pcm_uframes_t playback_src_pos; + snd_pcm_uframes_t playback_src_frames_this_period; +}; + +struct bcm2835_chip { + struct platform_device *pdev; + struct device *dev; + spinlock_t spinlock; + void __iomem *base; + struct clk *clk; + int clk_enabled; + struct dma_chan *dma_channel; + struct snd_card *card; + struct snd_pcm *pcm; + struct bcm2835_chip_runtime *runtime; +}; + +static u32 convert_audio_data(s16 input) +{ + s32 output; + output = ((s32) input * (s32) (5120 / 2)) / (s32) 32762; + + return(u32) (output + (5120 / 2)); +} + +static void convert_dma_buffer(struct bcm2835_chip_runtime *chip_runtime) +{ + snd_pcm_uframes_t i; + snd_pcm_uframes_t hardware_start_pos = + chip_runtime->hardware_period_number * + HARDWARE_BUFFER_FRAMES_PER_PERIOD; + snd_pcm_uframes_t buffer_size = + chip_runtime->substream->runtime->buffer_size; + + for (i = 0; i < HARDWARE_BUFFER_FRAMES_PER_PERIOD; i++) { + volatile struct bcm2835_software_frame *soft_frame; + volatile struct bcm2835_hardware_frame *hard_frame; + + hard_frame = chip_runtime->hardware_buffer + hardware_start_pos + i; + soft_frame = chip_runtime->playback_src_buffer + + chip_runtime->playback_src_pos; + + hard_frame->left = convert_audio_data(soft_frame->left); + hard_frame->right = convert_audio_data(soft_frame->right); + + if (chip_runtime->playback_src_pos >= buffer_size - 1) + chip_runtime->playback_src_pos = 0; + else + chip_runtime->playback_src_pos++; + } +} + +static void dma_complete(void *arg) +{ + struct bcm2835_chip_runtime *chip_runtime = arg; + struct bcm2835_chip *chip = chip_runtime->chip; + unsigned long flags; + bool period_elapsed = false; + + spin_lock_irqsave(&chip->spinlock, flags); + + chip_runtime->hardware_period_number++; + if (chip_runtime->hardware_period_number >= + HARDWARE_BUFFER_PERIODS_PER_BUFFER) + chip_runtime->hardware_period_number = 0; + + chip_runtime->playback_src_frames_this_period += + HARDWARE_BUFFER_FRAMES_PER_PERIOD; + if (chip_runtime->playback_src_frames_this_period >= + chip_runtime->substream->runtime->period_size) { + chip_runtime->playback_src_frames_this_period = 0; + period_elapsed = true; + } + + convert_dma_buffer(chip_runtime); + + spin_unlock_irqrestore(&chip->spinlock, flags); + + if (period_elapsed) + snd_pcm_period_elapsed(chip_runtime->substream); +} + +static struct snd_pcm_hardware snd_bcm2835_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 4 * 1024, + .period_bytes_max = 128 * 1024, + .periods_min = 1, + .periods_max = 128 / 4, + .fifo_size = 0, + +}; + +static void snd_bcm2835_playback_free(struct snd_pcm_runtime *runtime) +{ + kfree(runtime->private_data); + runtime->private_data = NULL; +} + +static int snd_bcm2835_playback_open(struct snd_pcm_substream *substream) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm2835_chip_runtime *chip_runtime; + int err; + + chip_runtime = kzalloc(sizeof(*chip_runtime), GFP_KERNEL); + + runtime->hw = snd_bcm2835_playback_hw; + runtime->private_data = chip_runtime; + runtime->private_free = snd_bcm2835_playback_free; + + chip->runtime = chip_runtime; + + chip_runtime->chip = chip; + chip_runtime->substream = substream; + + if (!chip->clk_enabled) { + + clk_set_rate(chip->clk, 250000000); + err = clk_prepare_enable(chip->clk); + if (err) { + kfree(chip_runtime); + return err; + } + } + + chip->clk_enabled++; + return 0; +} + +static int snd_bcm2835_playback_close(struct snd_pcm_substream *substream) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + + writel(0x00, chip->base + 0x00); + writel(0x00, chip->base + 0x08); + + chip->runtime = NULL; + + if (chip->clk_enabled) { + clk_disable_unprepare(chip->clk); + chip->clk_enabled--; + } + + return 0; +} + +static int snd_bcm2835_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + struct bcm2835_chip_runtime *chip_runtime = chip->runtime; + int ret; + + if (chip_runtime->dma_cookie) { + dmaengine_terminate_async(chip->dma_channel); + chip_runtime->dma_cookie = 0; + } + + writel(0x00, chip->base + 0x00); + writel(0x00, chip->base + 0x08); + + if (chip_runtime->hardware_buffer) { + dma_free_coherent(chip->dma_channel->device->dev, + HARDWARE_BUFFER_BYTES, + chip_runtime->hardware_buffer, + chip_runtime->hardware_buffer_dma); + } + + memset(chip_runtime, 0, sizeof(*chip_runtime)); + + chip_runtime->chip = chip; + chip_runtime->substream = substream; + + ret = snd_pcm_lib_free_pages(substream); + return ret; +} + +static int snd_bcm2835_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + struct bcm2835_chip_runtime *chip_runtime = chip->runtime; + snd_pcm_uframes_t playback_src_buffer_frames; + snd_pcm_uframes_t playback_src_period_frames; + int err = 0; + + snd_bcm2835_pcm_hw_free(substream); + + playback_src_buffer_frames = params_buffer_bytes(params) / 4; + playback_src_period_frames = params_period_bytes(params) / 4; + + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (err < 0) { + snd_bcm2835_pcm_hw_free(substream); + return err; + } + + chip_runtime->playback_src_buffer = + (struct bcm2835_software_frame *) (substream->runtime->dma_area); + chip_runtime->playback_src_pos = 0; + chip_runtime->playback_src_frames_this_period = 0; + + chip_runtime->hardware_buffer = + dma_alloc_coherent(chip->dma_channel->device->dev, + HARDWARE_BUFFER_BYTES, + &chip_runtime->hardware_buffer_dma, + GFP_KERNEL); + + if (!chip_runtime->hardware_buffer) { + snd_bcm2835_pcm_hw_free(substream); + return -ENOMEM; + } + + memset(chip_runtime->hardware_buffer, 0, HARDWARE_BUFFER_BYTES); + chip_runtime->hardware_period_number = 0; + + chip_runtime->dma_slave_config.direction = DMA_MEM_TO_DEV; + chip_runtime->dma_slave_config.dst_addr = 0x7e20c000 + 0x18; + chip_runtime->dma_slave_config.dst_maxburst = 2; + chip_runtime->dma_slave_config.dst_addr_width = 4; + chip_runtime->dma_slave_config.src_addr = + chip_runtime->hardware_buffer_dma; + chip_runtime->dma_slave_config.src_maxburst = 2; + chip_runtime->dma_slave_config.src_addr_width = 4; + + err = dmaengine_slave_config(chip->dma_channel, + &chip_runtime->dma_slave_config); + + if (err < 0) { + snd_bcm2835_pcm_hw_free(substream); + return err; + } + + chip_runtime->dma_desc = + dmaengine_prep_dma_cyclic(chip->dma_channel, + chip_runtime->hardware_buffer_dma, + HARDWARE_BUFFER_BYTES, + HARDWARE_BUFFER_PERIOD_BYTES, + DMA_MEM_TO_DEV, + DMA_CTRL_ACK | DMA_PREP_INTERRUPT); + + if (!chip_runtime->dma_desc) { + snd_bcm2835_pcm_hw_free(substream); + return -ENOMEM; + } + + chip_runtime->dma_desc->callback = dma_complete; + chip_runtime->dma_desc->callback_param = chip_runtime; + + return 0; +} + +static int snd_bcm2835_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + struct bcm2835_chip_runtime *chip_runtime = chip->runtime; + + memset(chip_runtime->hardware_buffer, 0x8000, HARDWARE_BUFFER_BYTES); + chip_runtime->hardware_period_number = + (HARDWARE_BUFFER_PERIODS_PER_BUFFER - 1); + + chip_runtime->playback_src_buffer = + (struct bcm2835_software_frame *) (substream->runtime->dma_area); + chip_runtime->playback_src_pos = 0; + chip_runtime->playback_src_frames_this_period = 0; + + memset(chip_runtime->playback_src_buffer, 0, + substream->runtime->buffer_size * + sizeof(*(chip_runtime->playback_src_buffer))); + + return 0; +} + +static int snd_bcm2835_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + struct bcm2835_chip_runtime *chip_runtime = chip->runtime; + int ret = 0; + unsigned long flags; + + spin_lock_irqsave(&chip->spinlock, flags); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!chip_runtime->dma_cookie) { + convert_dma_buffer(chip_runtime); + chip_runtime->playback_src_frames_this_period = + HARDWARE_BUFFER_FRAMES_PER_PERIOD; + + writel(5120, chip->base + 0x10); + writel(5120, chip->base + 0x20); + writel(0xa1e1, chip->base + 0x00); + writel(0x80000606, chip->base + 0x08); + + chip_runtime->dma_cookie = + dmaengine_submit(chip_runtime->dma_desc); + dma_async_issue_pending(chip->dma_channel); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + if (chip_runtime->dma_cookie) { + dmaengine_terminate_async(chip->dma_channel); + chip_runtime->dma_cookie = 0; + + writel(0x00, chip->base + 0x00); + writel(0x00, chip->base + 0x08); + } + break; + default: + ret = -EINVAL; + } + + spin_unlock_irqrestore(&chip->spinlock, flags); + + return ret; +} + +static snd_pcm_uframes_t +snd_bcm2835_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct bcm2835_chip *chip = snd_pcm_substream_chip(substream); + struct bcm2835_chip_runtime *chip_runtime = chip->runtime; + snd_pcm_uframes_t ret; + unsigned long flags; + + spin_lock_irqsave(&chip->spinlock, flags); + ret = chip_runtime->playback_src_pos; + spin_unlock_irqrestore(&chip->spinlock, flags); + + return ret; +} + +static struct snd_pcm_ops snd_bcm2835_playback_ops = { + .open = snd_bcm2835_playback_open, + .close = snd_bcm2835_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_bcm2835_pcm_hw_params, + .hw_free = snd_bcm2835_pcm_hw_free, + .prepare = snd_bcm2835_pcm_prepare, + .trigger = snd_bcm2835_pcm_trigger, + .pointer = snd_bcm2835_pcm_pointer, +}; + +static int snd_bcm2835_new_pcm(struct bcm2835_chip * chip) +{ + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "BCM2835 Analog", 0, 1, 0, &pcm); + + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "BCM2835 Analog"); + chip->pcm = pcm; + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_bcm2835_playback_ops); + + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + snd_bcm2835_playback_hw.buffer_bytes_max, + snd_bcm2835_playback_hw.buffer_bytes_max); + + return 0; +} + +static int snd_bcm2835_free(struct bcm2835_chip * chip) +{ + if (chip->dma_channel) + dma_release_channel(chip->dma_channel); + + return 0; +} + +static int snd_bcm2835_dev_free(struct snd_device *device) +{ + return snd_bcm2835_free(device->device_data); +} + +static struct snd_device_ops snd_bcm2835_dev_ops = { + .dev_free = snd_bcm2835_dev_free, +}; + +static int snd_bcm2835_create(struct snd_card *card, + struct platform_device *pdev, + struct bcm2835_chip ** rchip) +{ + struct bcm2835_chip *chip; + struct resource *res; + int err; + + *rchip = NULL; + + chip = devm_kzalloc(&pdev->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + + chip->pdev = pdev; + chip->dev = &pdev->dev; + chip->card = card; + + spin_lock_init(&chip->spinlock); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + chip->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(chip->base)) + return PTR_ERR(chip->base); + + chip->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(chip->clk)) { + dev_err(&pdev->dev, "clock not found: %ld\n", + PTR_ERR(chip->clk)); + return PTR_ERR(chip->clk); + } + + chip->dma_channel = dma_request_slave_channel(&pdev->dev, "tx"); + + if (!chip->dma_channel) + return -ENOMEM; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, + &snd_bcm2835_dev_ops); + + if (err < 0) { + snd_bcm2835_free(chip); + return err; + } + + *rchip = chip; + return 0; +} + +static int bcm2835_analog_audio_probe(struct platform_device *pdev) +{ + struct snd_card *card; + + int ret; + struct device *dev = &pdev->dev; + struct bcm2835_chip * chip; + + ret = snd_card_new(&pdev->dev, -1, NULL, THIS_MODULE, 0, &card); + if (ret) { + dev_err(dev, "Failed to create sound card structure\n"); + return ret; + } + + ret = snd_bcm2835_create(card, pdev, &chip); + if (ret < 0) { + dev_err(dev, "Failed to create bcm2835 chip\n"); + return ret; + } + + snd_card_set_dev(card, dev); + strcpy(card->driver, "BCM2835 Analog"); + strcpy(card->shortname, "BCM2835 Analog"); + sprintf(card->longname, "%s", card->shortname); + + ret = snd_bcm2835_new_pcm(chip); + if (ret < 0) { + snd_card_free(card); + return ret; + } + + ret = snd_card_register(card); + if (ret < 0) { + snd_card_free(card); + return ret; + } + + platform_set_drvdata(pdev, card); + + dev_info(dev, "BCM2835 Analog Audio Initialized\n"); + + return 0; +} + +static int bcm2835_analog_audio_remove(struct platform_device *pdev) +{ + struct snd_card *card; + + card = platform_get_drvdata(pdev); + + if (card) + snd_card_free(card); + + return 0; +} + +static const struct of_device_id bcm2835_analog_audio_of_match[] = { + { .compatible = "brcm,bcm2835-analog-audio",}, + { /* sentinel */} +}; +MODULE_DEVICE_TABLE(of, bcm2835_analog_audio_of_match); + +static struct platform_driver bcm2835_analog_audio_driver = { + .driver = + { + .owner = THIS_MODULE, + .name = "bcm2835-analog-audio", + .of_match_table = bcm2835_analog_audio_of_match, + }, + .probe = bcm2835_analog_audio_probe, + .remove = bcm2835_analog_audio_remove, +}; +module_platform_driver(bcm2835_analog_audio_driver); + +MODULE_AUTHOR("Michael Zoran"); +MODULE_DESCRIPTION("Audio driver for analog output on the BCM2835 chip"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bcm2835-analog-audio"); -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: dio2125: use gpiod_set_value_cansleep
by Jerome Brunet 07 Mar '17

07 Mar '17

Use the "cansleep" variant of gpiod_set_value so the driver can be used with slow gpio controllers as well. Fixes: 85825d5e8869 ("ASoC: dio2125: add dio2125 amp driver") Reported-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/codecs/dio2125.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/dio2125.c index 015e310556d3..09451cd44f9b 100644 --- a/sound/soc/codecs/dio2125.c +++ b/sound/soc/codecs/dio2125.c @@ -46,7 +46,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, return -EINVAL; } - gpiod_set_value(priv->gpiod_enable, val); + gpiod_set_value_cansleep(priv->gpiod_enable, val); return 0; } -- 2.9.3

2 1

[alsa-devel] [PATCH] ASoC: samsung: Remove extra blank lines
by Calin Cruceru 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Calin Cruceru <calin(a)rosedu.org> --- sound/soc/samsung/bells.c | 1 - sound/soc/samsung/i2s-regs.h | 2 -- sound/soc/samsung/i2s.c | 1 - sound/soc/samsung/s3c-i2s-v2.c | 1 - 4 files changed, 5 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 3dd246fa0059..34deba461ae1 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -446,7 +446,6 @@ static struct snd_soc_card bells_cards[] = { }, }; - static int bells_probe(struct platform_device *pdev) { int ret; diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index 9170c311d66e..fe6914005494 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -160,5 +160,3 @@ #define I2SSIZE_SHIFT (16) #endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */ - - diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 52a47ed292a4..af3ba4d4ccc5 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1242,7 +1242,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) i2s_dai_data = (struct samsung_i2s_dai_data *) platform_get_device_id(pdev)->driver_data; - pri_dai = i2s_alloc_dai(pdev, false); if (!pri_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_pri\n"); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 644f186fd35c..8f42deaa184b 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -72,7 +72,6 @@ static inline void dbg_showcon(const char *fn, u32 con) } #endif - /* Turn on or off the transmission path. */ static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) { -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: txx9: Added requiered spaces.
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/txx9/txx9aclc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index a8f705b..98fb989 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -206,7 +206,7 @@ static void txx9aclc_dma_tasklet(unsigned long data) static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; - struct txx9aclc_plat_drvdata *drvdata =txx9aclc_drvdata; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_drvdata; void __iomem *base = drvdata->base; unsigned long flags; int ret = 0; @@ -340,7 +340,7 @@ static bool filter(struct dma_chan *chan, void *param) static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, struct txx9aclc_dmadata *dmadata) { - struct txx9aclc_plat_drvdata *drvdata =txx9aclc_drvdata; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_drvdata; struct txx9dmac_slave *ds = &dmadata->dma_slave; dma_cap_mask_t mask; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix invalid memory access due to wrong reference of pointer
by Takashi Sakamoto 07 Mar '17

07 Mar '17

In 'skl_tplg_set_module_init_data()', a pointer to 'params' member of 'struct skl_algo_data' is calculated, then casted to (u32 *) and assigned to a member of configuration data. The configuration data is passed to the other functions and used to process intel IPC. In this processing, the value of member is used to get message data, however this can bring invalid memory access in 'skl_set_module_params()' as a result of calculation of a pointer for actual message data. (sound/soc/intel/skylake/skl-topology.c) skl_tplg_init_pipe_modules() ->skl_tplg_set_module_init_data() (has this bug) ->skl_tplg_set_module_params() (sound/soc/intel/skylake/skl-messages.c) ->skl_set_module_params() ((char *)param) + data_offset This commit fixes the bug. Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Cc: <stable(a)vger.kernel.org> # v4.5+ Fixes: abb740033b56 ("ASoC: Intel: Skylake: Add support to configure module params") Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ed58b5b..2dbfb1b 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -512,7 +512,7 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w) if (bc->set_params != SKL_PARAM_INIT) continue; - mconfig->formats_config.caps = (u32 *)&bc->params; + mconfig->formats_config.caps = (u32 *)bc->params; mconfig->formats_config.caps_size = bc->size; break; -- 2.9.3

3 2

[alsa-devel] [PATCH] ASoC: atmel-classd: fix audio clock rate
by Songjun Wu 07 Mar '17

07 Mar '17

Fix the audio clock rate according to the datasheet. Reported-by: Dushara Jayasinghe <dushara(a)successful.com.au> Signed-off-by: Songjun Wu <songjun.wu(a)microchip.com> --- sound/soc/atmel/atmel-classd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 89ac5f5a93eb..7ae46c2647d4 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -349,7 +349,7 @@ static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, } #define CLASSD_ACLK_RATE_11M2896_MPY_8 (112896 * 100 * 8) -#define CLASSD_ACLK_RATE_12M288_MPY_8 (12228 * 1000 * 8) +#define CLASSD_ACLK_RATE_12M288_MPY_8 (12288 * 1000 * 8) static struct { int rate; -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: pxa-ssp: Added blank line after declarations
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3bdace6..72a5e87 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -355,6 +355,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, if (ssp->type == PXA3xx_SSP) { u32 val; u64 tmp = 19968; + tmp *= 1000000; do_div(tmp, freq_out); val = tmp; -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: pxa: Remove space before semicolon" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa: Remove space before semicolon has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3e22e9d702ee719c8106fd44578b12ce27999ae6 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 22:33:00 +0200 Subject: [PATCH] ASoC: pxa: Remove space before semicolon This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/spitz.c | 4 ++-- sound/soc/pxa/z2.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 07d77cddac60..d38a2b519c52 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -230,8 +230,8 @@ static const struct snd_soc_dapm_route spitz_audio_map[] = { {"Headset Jack", NULL, "ROUT1"}, /* ext speaker connected to LOUT2, ROUT2 */ - {"Ext Spk", NULL , "ROUT2"}, - {"Ext Spk", NULL , "LOUT2"}, + {"Ext Spk", NULL, "ROUT2"}, + {"Ext Spk", NULL, "LOUT2"}, /* mic is connected to input 1 - with bias */ {"LINPUT1", NULL, "Mic Bias"}, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 990b1aa6d7f6..6d88d9acc666 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -119,8 +119,8 @@ static const struct snd_soc_dapm_route z2_audio_map[] = { {"Headphone Jack", NULL, "ROUT1"}, /* ext speaker connected to LOUT2, ROUT2 */ - {"Ext Spk", NULL , "ROUT2"}, - {"Ext Spk", NULL , "LOUT2"}, + {"Ext Spk", NULL, "ROUT2"}, + {"Ext Spk", NULL, "LOUT2"}, /* mic is connected to R input 2 - with bias */ {"RINPUT2", NULL, "Mic Bias"}, -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: pxa: Remove unneeded else after return statement" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa: Remove unneeded else after return statement has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 291aaff0c6f2c56645061b9c8afadc0bb1251676 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 23:26:56 +0200 Subject: [PATCH] ASoC: pxa: Remove unneeded else after return statement This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa2xx-ac97.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 2e2fb1838ec2..f49bf02e5ec2 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -140,9 +140,8 @@ static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, - &pxa2xx_ac97_pcm_mic_mono_in); + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); return 0; } -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: pxa: The open brace is placed with the previous line" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa: The open brace is placed with the previous line has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 05af0917aaf8a0755de8fbbf72153ad6ca1f5443 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 23:54:16 +0200 Subject: [PATCH] ASoC: pxa: The open brace is placed with the previous line This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/raumfeld.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 552b763005ed..47c91730e93c 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -228,14 +228,12 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .codec_name = "spi0.0", \ } -static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = -{ +static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { DAI_LINK_CS4270, DAI_LINK_AK4104, }; -static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = -{ +static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { DAI_LINK_CS4270, }; -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: pxa: Add space around ':' and '('" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa: Add space around ':' and '(' has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ca87cfad69e6b657a4f5fa665b67aee275792af4 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 22:42:33 +0200 Subject: [PATCH] ASoC: pxa: Add space around ':' and '(' This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/tosa.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 0389cf7b4b1e..27aa0e4341bc 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -119,7 +119,7 @@ static int pxa_i2s_wait(void) int i; /* flush the Rx FIFO */ - for(i = 0; i < 16; i++) + for (i = 0; i < 16; i++) SADR; return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index e022b2a777f6..1812748b75bf 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -133,7 +133,7 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol, static int tosa_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0); + gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); return 0; } -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: pxa: Remove spaces before tabs" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa: Remove spaces before tabs has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 672e3cbe783e2f3f40bdc19e5cb09accdd3e4b6b Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 22:50:29 +0200 Subject: [PATCH] ASoC: pxa: Remove spaces before tabs This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa2xx-i2s.c | 6 +++--- sound/soc/pxa/pxa2xx-pcm.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 27aa0e4341bc..3fb60baf6eab 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -46,10 +46,10 @@ #define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */ #define SACR0_EFWR (1 << 4) /* Enable EFWR Function */ #define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */ -#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ +#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ #define SACR0_ENB (1 << 0) /* Enable I2S Link */ #define SACR1_ENLBF (1 << 5) /* Enable Loopback */ -#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ +#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ #define SACR1_DREC (1 << 3) /* Disable Recording Function */ #define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */ @@ -60,7 +60,7 @@ #define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */ #define SASR0_BSY (1 << 2) /* I2S Busy */ #define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */ -#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ +#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ #define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */ #define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */ diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 410d48b93031..b51d7a0755d5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -85,7 +85,7 @@ static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) } static struct snd_soc_platform_driver pxa2xx_soc_platform = { - .ops = &pxa2xx_pcm_ops, + .ops = &pxa2xx_pcm_ops, .pcm_new = pxa2xx_soc_pcm_new, .pcm_free = pxa2xx_pcm_free_dma_buffers, }; -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: pxa: Remove unneeded return statement in void function" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa: Remove unneeded return statement in void function has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 17339f6099154678485244cc17d700f19ece0dc0 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 23:06:36 +0200 Subject: [PATCH] ASoC: pxa: Remove unneeded return statement in void function This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/mmp-pcm.c | 1 - sound/soc/pxa/mmp-sspa.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 96df9b2d8fc4..5b5f1a442891 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -166,7 +166,6 @@ static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) buf->area = NULL; } - return; } static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index ca8b23f8c525..9cc35012e6e5 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -119,7 +119,6 @@ static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, clk_disable(priv->sspa->clk); clk_disable(priv->sysclk); - return; } /* -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: pxa-ssp: Line up *s in block comments" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: pxa-ssp: Line up *s in block comments has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 34e82433c26ea6908fa19783fee2c4bba5bad9c8 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 23:40:31 +0200 Subject: [PATCH] ASoC: pxa-ssp: Line up *s in block comments This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa-ssp.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e9f6df38b152..0291c7cb64eb 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -591,13 +591,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ if (ssp->type != PXA3xx_SSP) return -EINVAL; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: omap: Remove unnecessary 'out of memory' message
by Adriana Constantinescu 07 Mar '17

07 Mar '17

Out of memory message detected using checkpatch.pl Signed-off-by: Adriana Constantinescu <constantinescu33(a)gmail.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- sound/soc/omap/rx51.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index a768457..086b59b 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -433,10 +433,9 @@ static int rx51_soc_probe(struct platform_device *pdev) } pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); - if (pdata == NULL) { - dev_err(card->dev, "failed to create private data\n"); + if (pdata == NULL) return -ENOMEM; - } + snd_soc_card_set_drvdata(card, pdata); pdata->tvout_selection_gpio = devm_gpiod_get(card->dev, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tegra: Add blank line after declarations
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/tegra/tegra20_das.c | 1 + sound/soc/tegra/tegra30_ahub.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 6d4a277..4024e3a 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -41,6 +41,7 @@ static inline void tegra20_das_write(u32 reg, u32 val) static inline u32 tegra20_das_read(u32 reg) { u32 val; + regmap_read(das->regmap, reg, &val); return val; } diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d7494c0..8c10ae7 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -41,6 +41,7 @@ static inline void tegra30_apbif_write(u32 reg, u32 val) static inline u32 tegra30_apbif_read(u32 reg) { u32 val; + regmap_read(ahub->regmap_apbif, reg, &val); return val; } 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: txx9: Added blank line after declarations
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/txx9/txx9aclc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 98fb989..7912bf0 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -392,6 +392,7 @@ static int txx9aclc_pcm_remove(struct snd_soc_platform *platform) for (i = 0; i < 2; i++) { struct txx9aclc_dmadata *dmadata = &dev->dmadata[i]; struct dma_chan *chan = dmadata->dma_chan; + if (chan) { dmadata->frag_count = -1; dmaengine_terminate_all(chan); 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ux500: Added */ to the next line
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/ux500/ux500_msp_dai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index b343efd..36e9bb2 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -482,7 +482,8 @@ static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream, if ((drvdata->fmt & SND_SOC_DAIFMT_MASTER_MASK) && (drvdata->msp->f_bitclk > 19200000)) { /* If the bit-clock is higher than 19.2MHz, Vape should be - * run in 100% OPP. Only when bit-clock is used (MSP master) */ + * run in 100% OPP. Only when bit-clock is used (MSP master) + */ prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, "ux500-msp-i2s", 100); drvdata->vape_opp_constraint = 1; 1.9.1

2 1

[alsa-devel] Applied "ASoC: ux500: Remove unuseful break after return" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: ux500: Remove unuseful break after return has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b1c5d923720a066b95249f3702a94d27b5714ee9 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 21:41:25 +0200 Subject: [PATCH] ASoC: ux500: Remove unuseful break after return This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/ux500/ux500_msp_i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 959d7b4edf56..bd5266aca0f1 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -604,7 +604,6 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) break; default: return -EINVAL; - break; } return 0; -- 2.11.0

1 0

[alsa-devel] Applied "ASoC: ux500: Added blank line after declarations" to the asoc tree
by Mark Brown 07 Mar '17

07 Mar '17

The patch ASoC: ux500: Added blank line after declarations has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fe3a980cd39125efb7e9ddd67dd0f1a9ef237171 Mon Sep 17 00:00:00 2001 From: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> Date: Sat, 25 Feb 2017 21:50:21 +0200 Subject: [PATCH] ASoC: ux500: Added blank line after declarations This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/ux500/ux500_msp_dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 36e9bb2bd634..ec5152aa3f6e 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -133,6 +133,7 @@ static int setup_pcm_framing(struct snd_soc_dai *dai, unsigned int rate, struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); u32 frame_length = MSP_FRAME_LEN_1; + prot_desc->frame_width = 0; switch (drvdata->slots) { -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: sirf: Added blank line after declarations
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/sirf/sirf-audio-port.c | 1 + sound/soc/sirf/sirf-audio.c | 1 + sound/soc/sirf/sirf-usp.c | 3 +++ 3 files changed, 5 insertions(+) diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c index 3f2cce0..be066de 100644 --- a/sound/soc/sirf/sirf-audio-port.c +++ b/sound/soc/sirf/sirf-audio-port.c @@ -19,6 +19,7 @@ struct sirf_audio_port { static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai) { struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_init_dma_data(dai, &port->playback_dma_data, &port->capture_dma_data); return 0; diff --git a/sound/soc/sirf/sirf-audio.c b/sound/soc/sirf/sirf-audio.c index 94ea152..f2bc507 100644 --- a/sound/soc/sirf/sirf-audio.c +++ b/sound/soc/sirf/sirf-audio.c @@ -27,6 +27,7 @@ static int sirf_audio_hp_event(struct snd_soc_dapm_widget *w, struct snd_soc_card *card = dapm->card; struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card); int on = !SND_SOC_DAPM_EVENT_OFF(event); + if (gpio_is_valid(sirf_audio_card->gpio_hp_pa)) gpio_set_value(sirf_audio_card->gpio_hp_pa, on); return 0; diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 45fc06c..77e7dcf 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -71,6 +71,7 @@ static void sirf_usp_rx_disable(struct sirf_usp *usp) static int sirf_usp_pcm_dai_probe(struct snd_soc_dai *dai) { struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_init_dma_data(dai, &usp->playback_dma_data, &usp->capture_dma_data); return 0; @@ -294,6 +295,7 @@ static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd, static int sirf_usp_pcm_runtime_suspend(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); + clk_disable_unprepare(usp->clk); return 0; } @@ -302,6 +304,7 @@ static int sirf_usp_pcm_runtime_resume(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); int ret; + ret = clk_prepare_enable(usp->clk); if (ret) { dev_err(dev, "clk_enable failed: %d\n", ret); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: pxa: Add space around '='
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/e750_wm9705.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index fdcd94a..82bcbbb 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -81,7 +81,7 @@ static int e750_hp_amp_event(struct snd_soc_dapm_widget *w, .name = "AC97 Aux", .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_dai_name ="wm9705-aux", + .codec_dai_name = "wm9705-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 2df714f..1ed8aa2 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -81,7 +81,7 @@ static int e800_hp_amp_event(struct snd_soc_dapm_widget *w, .name = "AC97 Aux", .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_dai_name ="wm9712-aux", + .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 6f2020f..e046770 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -43,7 +43,7 @@ .name = "AC97 Aux", .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_dai_name ="wm9712-aux", + .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 8760a66..c4c6fbe 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -157,7 +157,7 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) .name = "AC97 Aux", .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_dai_name ="wm9713-aux", + .codec_dai_name = "wm9713-aux", .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, -- 1.9.1

3 2

[alsa-devel] [v5 2/2] dt: bindings: sound: Add bindings file for CS35L35
by Brian Austin 07 Mar '17

07 Mar '17

Add bindings documentation for CS35L35 Amplifier Changes for v5 Make IPeak and BST parameters consistent with previous devices that have similar properties Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l35.txt | 171 +++++++++++++++++++++ 1 file changed, 171 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l35.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 0000000..07d9c0f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,171 @@ +CS35L35 Boosted Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L35 are delivered to. + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + +Optional properties: + - reset-gpios : gpio used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2600mV to 9000mV with + increments of 100mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1680mA and goes to a maximum of 4480mA with increments of + 110mA. + (Default) 2.46 Amps + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 3 8 bit values to set the depth, location, and frame + of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: cs35l35@20 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl-millivolt = <9000>; + + cirrus,stereo-config; + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix spelling mistake: "allocationf" -> "allocation"
by Colin King 07 Mar '17

07 Mar '17

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/intel/skylake/skl-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index e12520e..635cbb1 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1165,7 +1165,7 @@ static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_pci_data(skl->pci), size, MAX_PREALLOC_SIZE); if (retval) { - dev_err(dai->dev, "dma buffer allocationf fail\n"); + dev_err(dai->dev, "dma buffer allocation fail\n"); return retval; } } -- 2.10.2

3 2

[alsa-devel] [PATCH v1] ASoC: Intel: bdw-rt5677: Use devm_gpiod_get()
by Andy Shevchenko 07 Mar '17

07 Mar '17

Since index is always 0 replace devm_gpiod_get_index() by devm_gpiod_get() and apply proper flags. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index ab383af14ca2..8284a40829be 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -205,13 +205,12 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) RT5677_CLK_SEL_I2S1_ASRC); /* Request rt5677 GPIO for headphone amp control */ - bdw_rt5677->gpio_hp_en = devm_gpiod_get_index(codec->dev, - "headphone-enable", 0, 0); + bdw_rt5677->gpio_hp_en = devm_gpiod_get(codec->dev, "headphone-enable", + GPIOD_OUT_LOW); if (IS_ERR(bdw_rt5677->gpio_hp_en)) { dev_err(codec->dev, "Can't find HP_AMP_SHDN_L gpio\n"); return PTR_ERR(bdw_rt5677->gpio_hp_en); } - gpiod_direction_output(bdw_rt5677->gpio_hp_en, 0); /* Create and initialize headphone jack */ if (!snd_soc_card_jack_new(rtd->card, "Headphone Jack", -- 2.11.0

3 2

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Driver updates on DSP firmware
by jeeja.kp＠intel.com 07 Mar '17

07 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides update on DSP firmware download by optimizing o updating the dsp register poll method to use accurate timeout. o optimizing ROM init retries. o saving the firmware/library context at boot and use this ctx for downloading the firmware to DSP memory. o cleanup to store the library manifest data o release the firmware in cleanup routine. Jeeja KP (6): ASoC: Intel: Common: Update dsp register poll implementation ASoC: Intel: bxtn: Use DSP poll API to poll FW status ASoC: Intel: Skylake: Clean up manifest info ASoC: Intel: Skylake: Release FW ctx in cleanup ASoC: Intel: bxtn: Fix to store the FW/Library context at boot ASoC: Intel: bxtn: optimize ROM init retries sound/soc/intel/common/sst-dsp.c | 52 +++++----- sound/soc/intel/skylake/bxt-sst.c | 142 ++++++++++++++------------- sound/soc/intel/skylake/skl-sst-dsp.h | 4 +- sound/soc/intel/skylake/skl-sst-ipc.h | 5 +- sound/soc/intel/skylake/skl-sst.c | 3 + sound/soc/intel/skylake/skl-topology.c | 41 ++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++ sound/soc/intel/skylake/skl-tplg-interface.h | 12 --- 8 files changed, 137 insertions(+), 135 deletions(-) -- 2.5.0

4 15

[alsa-devel] [PATCH] ASoC: tegra: Remove unnecessary 'out of memory' message
by Codrut Grosu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/tegra/tegra20_ac97.c | 1 - sound/soc/tegra/tegra20_das.c | 1 - sound/soc/tegra/tegra20_i2s.c | 1 - sound/soc/tegra/tegra20_spdif.c | 5 ++--- sound/soc/tegra/tegra30_ahub.c | 4 +--- sound/soc/tegra/tegra30_i2s.c | 1 - sound/soc/tegra/tegra_alc5632.c | 4 +--- sound/soc/tegra/tegra_max98090.c | 4 +--- sound/soc/tegra/tegra_rt5640.c | 4 +--- sound/soc/tegra/tegra_sgtl5000.c | 4 +--- sound/soc/tegra/tegra_wm8753.c | 4 +--- sound/soc/tegra/tegra_wm8903.c | 4 +--- sound/soc/tegra/tegra_wm9712.c | 4 +--- sound/soc/tegra/trimslice.c | 4 +--- 14 files changed, 11 insertions(+), 34 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index a68368e..affad46 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -318,7 +318,6 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97 = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_ac97), GFP_KERNEL); if (!ac97) { - dev_err(&pdev->dev, "Can't allocate tegra20_ac97\n"); ret = -ENOMEM; goto err; } diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 89add13..6d4a277 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -142,7 +142,6 @@ static int tegra20_das_probe(struct platform_device *pdev) das = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_das), GFP_KERNEL); if (!das) { - dev_err(&pdev->dev, "Can't allocate tegra20_das\n"); ret = -ENOMEM; goto err; } diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 14106fa..26253c2 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -345,7 +345,6 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev) i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_i2s), GFP_KERNEL); if (!i2s) { - dev_err(&pdev->dev, "Can't allocate tegra20_i2s\n"); ret = -ENOMEM; goto err; } diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index a0c3640..767c049 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -271,10 +271,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif), GFP_KERNEL); - if (!spdif) { - dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); + if (!spdif) return -ENOMEM; - } + dev_set_drvdata(&pdev->dev, spdif); spdif->clk_spdif_out = devm_clk_get(&pdev->dev, "spdif_out"); diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index fef3b9a..d7494c0 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -560,10 +560,8 @@ static int tegra30_ahub_probe(struct platform_device *pdev) ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), GFP_KERNEL); - if (!ahub) { - dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); + if (!ahub) return -ENOMEM; - } dev_set_drvdata(&pdev->dev, ahub); ahub->soc_data = soc_data; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 8e55583..b2b279c 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -385,7 +385,6 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) i2s = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_i2s), GFP_KERNEL); if (!i2s) { - dev_err(&pdev->dev, "Can't allocate tegra30_i2s\n"); ret = -ENOMEM; goto err; } diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index eead6e7..0509902 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -169,10 +169,8 @@ static int tegra_alc5632_probe(struct platform_device *pdev) alc5632 = devm_kzalloc(&pdev->dev, sizeof(struct tegra_alc5632), GFP_KERNEL); - if (!alc5632) { - dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + if (!alc5632) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index a403db6..c34a54d 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -225,10 +225,8 @@ static int tegra_max98090_probe(struct platform_device *pdev) machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_max98090), GFP_KERNEL); - if (!machine) { - dev_err(&pdev->dev, "Can't allocate tegra_max98090\n"); + if (!machine) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 25b9fc0..93a3568 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -170,10 +170,8 @@ static int tegra_rt5640_probe(struct platform_device *pdev) machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_rt5640), GFP_KERNEL); - if (!machine) { - dev_err(&pdev->dev, "Can't allocate tegra_rt5640\n"); + if (!machine) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 4bbab09..6dda01f 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -120,10 +120,8 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_sgtl5000), GFP_KERNEL); - if (!machine) { - dev_err(&pdev->dev, "Can't allocate tegra_sgtl5000 struct\n"); + if (!machine) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index bdedd10..d0ab0026 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -128,10 +128,8 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8753), GFP_KERNEL); - if (!machine) { - dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n"); + if (!machine) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2013e9c..dbfb492 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -248,10 +248,8 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8903), GFP_KERNEL); - if (!machine) { - dev_err(&pdev->dev, "Can't allocate tegra_wm8903 struct\n"); + if (!machine) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 6492f81..c9cd224 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -77,10 +77,8 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm9712), GFP_KERNEL); - if (!machine) { - dev_err(&pdev->dev, "Can't allocate tegra_wm9712 struct\n"); + if (!machine) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 870f84a..c9dcad9 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -123,10 +123,8 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev) trimslice = devm_kzalloc(&pdev->dev, sizeof(struct tegra_trimslice), GFP_KERNEL); - if (!trimslice) { - dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n"); + if (!trimslice) return -ENOMEM; - } card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: Add space around '='
by Adrian Dinu 07 Mar '17

07 Mar '17

This was reported by checkpatch.pl Signed-off-by: Adrian Dinu <adrian.dinu95(a)gmail.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a110d39..70905bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3139,7 +3139,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->suspend = component->driver->suspend; component->resume = component->driver->resume; component->pcm_new = component->driver->pcm_new; - component->pcm_free= component->driver->pcm_free; + component->pcm_free = component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dwc: remove 'out of memory' message
by Mihai Burduselu 07 Mar '17

07 Mar '17

Reported by checkpatch.pl Signed-off-by: Mihai Burduselu <michelcatalin(a)gmail.com> --- sound/soc/dwc/designware_i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 9c46e4112026..6479768cc6a1 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -611,10 +611,8 @@ static int dw_i2s_probe(struct platform_device *pdev) const char *clk_id; dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); - if (!dev) { - dev_warn(&pdev->dev, "kzalloc fail\n"); + if (!dev) return -ENOMEM; - } dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); if (!dw_i2s_dai) -- 2.11.0

4 3

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Return an error on write to a disabled volatile control
by Charles Keepax 07 Mar '17

07 Mar '17

Volatile controls should only be accessed when the firmware is active, currently however writes to these controls will succeed, but the data will be lost, if the firmware is powered down. Update this behaviour such that an error is returned the same as it is for reads. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d151224..6313b3d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -899,7 +899,10 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - memcpy(ctl->cache, p, ctl->len); + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) + ret = -EPERM; + else + memcpy(ctl->cache, p, ctl->len); ctl->set = 1; if (ctl->enabled && ctl->dsp->running) @@ -926,6 +929,8 @@ static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, ctl->set = 1; if (ctl->enabled && ctl->dsp->running) ret = wm_coeff_write_control(ctl, ctl->cache, size); + else if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) + ret = -EPERM; } mutex_unlock(&ctl->dsp->pwr_lock); -- 2.1.4

2 3

Re: [alsa-devel] [PATCH] ALSA: ASoC: TLV320AIC23: Unquote NULL from control name
by Mark Brown 07 Mar '17

07 Mar '17

On Tue, Mar 07, 2017 at 12:54:08AM +0000, Liviu Dudau wrote: > On Mon, Mar 06, 2017 at 11:41:18AM +0100, Mark Brown wrote: > > Please use subject lines matching the style for the subsystem. This > > makes it easier for people to identify relevant patches. > Other than the additional ALSA: prefix I have matched the format used > by previous commits touching the file. I'm not clear where I have made the > mistake and what would have been the better subject line. It's the extra ALSA prefix, the start of the subject line is the most important bit to get right since it's that that people are most likely to pattern match on. You also used caps for the name of the CODEC, normally they use lower case, though that is less important.

1 0

[alsa-devel] [PATCH 18/35] perf thread_map: Convert thread_map.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-10-git-send-email-elena.reshetova… [ Did missing tests/thread-map.c conversion ] Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/tests/thread-map.c | 6 +++--- tools/perf/util/thread_map.c | 20 ++++++++++---------- tools/perf/util/thread_map.h | 4 ++-- 3 files changed, 15 insertions(+), 15 deletions(-) diff --git a/tools/perf/tests/thread-map.c b/tools/perf/tests/thread-map.c index f2d2e542d0ee..a63d6945807b 100644 --- a/tools/perf/tests/thread-map.c +++ b/tools/perf/tests/thread-map.c @@ -29,7 +29,7 @@ int test__thread_map(int subtest __maybe_unused) thread_map__comm(map, 0) && !strcmp(thread_map__comm(map, 0), NAME)); TEST_ASSERT_VAL("wrong refcnt", - atomic_read(&map->refcnt) == 1); + refcount_read(&map->refcnt) == 1); thread_map__put(map); /* test dummy pid */ @@ -44,7 +44,7 @@ int test__thread_map(int subtest __maybe_unused) thread_map__comm(map, 0) && !strcmp(thread_map__comm(map, 0), "dummy")); TEST_ASSERT_VAL("wrong refcnt", - atomic_read(&map->refcnt) == 1); + refcount_read(&map->refcnt) == 1); thread_map__put(map); return 0; } @@ -71,7 +71,7 @@ static int process_event(struct perf_tool *tool __maybe_unused, thread_map__comm(threads, 0) && !strcmp(thread_map__comm(threads, 0), NAME)); TEST_ASSERT_VAL("wrong refcnt", - atomic_read(&threads->refcnt) == 1); + refcount_read(&threads->refcnt) == 1); thread_map__put(threads); return 0; } diff --git a/tools/perf/util/thread_map.c b/tools/perf/util/thread_map.c index 7c3fcc538a70..9026408ea55b 100644 --- a/tools/perf/util/thread_map.c +++ b/tools/perf/util/thread_map.c @@ -66,7 +66,7 @@ struct thread_map *thread_map__new_by_pid(pid_t pid) for (i = 0; i < items; i++) thread_map__set_pid(threads, i, atoi(namelist[i]->d_name)); threads->nr = items; - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); } for (i=0; i<items; i++) @@ -83,7 +83,7 @@ struct thread_map *thread_map__new_by_tid(pid_t tid) if (threads != NULL) { thread_map__set_pid(threads, 0, tid); threads->nr = 1; - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); } return threads; @@ -105,7 +105,7 @@ struct thread_map *thread_map__new_by_uid(uid_t uid) goto out_free_threads; threads->nr = 0; - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); while ((dirent = readdir(proc)) != NULL) { char *end; @@ -235,7 +235,7 @@ static struct thread_map *thread_map__new_by_pid_str(const char *pid_str) out: strlist__delete(slist); if (threads) - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); return threads; out_free_namelist: @@ -255,7 +255,7 @@ struct thread_map *thread_map__new_dummy(void) if (threads != NULL) { thread_map__set_pid(threads, 0, -1); threads->nr = 1; - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); } return threads; } @@ -300,7 +300,7 @@ struct thread_map *thread_map__new_by_tid_str(const char *tid_str) } out: if (threads) - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); return threads; out_free_threads: @@ -326,7 +326,7 @@ static void thread_map__delete(struct thread_map *threads) if (threads) { int i; - WARN_ONCE(atomic_read(&threads->refcnt) != 0, + WARN_ONCE(refcount_read(&threads->refcnt) != 0, "thread map refcnt unbalanced\n"); for (i = 0; i < threads->nr; i++) free(thread_map__comm(threads, i)); @@ -337,13 +337,13 @@ static void thread_map__delete(struct thread_map *threads) struct thread_map *thread_map__get(struct thread_map *map) { if (map) - atomic_inc(&map->refcnt); + refcount_inc(&map->refcnt); return map; } void thread_map__put(struct thread_map *map) { - if (map && atomic_dec_and_test(&map->refcnt)) + if (map && refcount_dec_and_test(&map->refcnt)) thread_map__delete(map); } @@ -423,7 +423,7 @@ static void thread_map__copy_event(struct thread_map *threads, threads->map[i].comm = strndup(event->entries[i].comm, 16); } - atomic_set(&threads->refcnt, 1); + refcount_set(&threads->refcnt, 1); } struct thread_map *thread_map__new_event(struct thread_map_event *event) diff --git a/tools/perf/util/thread_map.h b/tools/perf/util/thread_map.h index ea0ef08c6303..bd34d7a0b9fa 100644 --- a/tools/perf/util/thread_map.h +++ b/tools/perf/util/thread_map.h @@ -3,7 +3,7 @@ #include <sys/types.h> #include <stdio.h> -#include <linux/atomic.h> +#include <linux/refcount.h> struct thread_map_data { pid_t pid; @@ -11,7 +11,7 @@ struct thread_map_data { }; struct thread_map { - atomic_t refcnt; + refcount_t refcnt; int nr; struct thread_map_data map[]; }; -- 2.9.3

1 0

[alsa-devel] [PATCH 17/35] perf thread: convert thread.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-9-git-send-email-elena.reshetova@… [ Did missing conversion in __machine__remove_thread() ] Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/util/machine.c | 2 +- tools/perf/util/thread.c | 6 +++--- tools/perf/util/thread.h | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/tools/perf/util/machine.c b/tools/perf/util/machine.c index 71c9720d4973..b9974fe41bc1 100644 --- a/tools/perf/util/machine.c +++ b/tools/perf/util/machine.c @@ -1439,7 +1439,7 @@ static void __machine__remove_thread(struct machine *machine, struct thread *th, if (machine->last_match == th) machine->last_match = NULL; - BUG_ON(atomic_read(&th->refcnt) == 0); + BUG_ON(refcount_read(&th->refcnt) == 0); if (lock) pthread_rwlock_wrlock(&machine->threads_lock); rb_erase_init(&th->rb_node, &machine->threads); diff --git a/tools/perf/util/thread.c b/tools/perf/util/thread.c index f5af87f66663..74e79d26b421 100644 --- a/tools/perf/util/thread.c +++ b/tools/perf/util/thread.c @@ -53,7 +53,7 @@ struct thread *thread__new(pid_t pid, pid_t tid) goto err_thread; list_add(&comm->list, &thread->comm_list); - atomic_set(&thread->refcnt, 1); + refcount_set(&thread->refcnt, 1); RB_CLEAR_NODE(&thread->rb_node); } @@ -88,13 +88,13 @@ void thread__delete(struct thread *thread) struct thread *thread__get(struct thread *thread) { if (thread) - atomic_inc(&thread->refcnt); + refcount_inc(&thread->refcnt); return thread; } void thread__put(struct thread *thread) { - if (thread && atomic_dec_and_test(&thread->refcnt)) { + if (thread && refcount_dec_and_test(&thread->refcnt)) { /* * Remove it from the dead_threads list, as last reference * is gone. diff --git a/tools/perf/util/thread.h b/tools/perf/util/thread.h index 99263cb6e6b6..e57188546465 100644 --- a/tools/perf/util/thread.h +++ b/tools/perf/util/thread.h @@ -1,7 +1,7 @@ #ifndef __PERF_THREAD_H #define __PERF_THREAD_H -#include <linux/atomic.h> +#include <linux/refcount.h> #include <linux/rbtree.h> #include <linux/list.h> #include <unistd.h> @@ -23,7 +23,7 @@ struct thread { pid_t tid; pid_t ppid; int cpu; - atomic_t refcnt; + refcount_t refcnt; char shortname[3]; bool comm_set; int comm_len; -- 2.9.3

1 0

[alsa-devel] [PATCH 16/35] perf evlist: Convert perf_map.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-8-git-send-email-elena.reshetova@… Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/util/evlist.c | 18 +++++++++--------- tools/perf/util/evlist.h | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/tools/perf/util/evlist.c b/tools/perf/util/evlist.c index b601f2814a30..564b924fb48a 100644 --- a/tools/perf/util/evlist.c +++ b/tools/perf/util/evlist.c @@ -777,7 +777,7 @@ union perf_event *perf_mmap__read_forward(struct perf_mmap *md, bool check_messu /* * Check if event was unmapped due to a POLLHUP/POLLERR. */ - if (!atomic_read(&md->refcnt)) + if (!refcount_read(&md->refcnt)) return NULL; head = perf_mmap__read_head(md); @@ -794,7 +794,7 @@ perf_mmap__read_backward(struct perf_mmap *md) /* * Check if event was unmapped due to a POLLHUP/POLLERR. */ - if (!atomic_read(&md->refcnt)) + if (!refcount_read(&md->refcnt)) return NULL; head = perf_mmap__read_head(md); @@ -856,7 +856,7 @@ void perf_mmap__read_catchup(struct perf_mmap *md) { u64 head; - if (!atomic_read(&md->refcnt)) + if (!refcount_read(&md->refcnt)) return; head = perf_mmap__read_head(md); @@ -875,14 +875,14 @@ static bool perf_mmap__empty(struct perf_mmap *md) static void perf_mmap__get(struct perf_mmap *map) { - atomic_inc(&map->refcnt); + refcount_inc(&map->refcnt); } static void perf_mmap__put(struct perf_mmap *md) { - BUG_ON(md->base && atomic_read(&md->refcnt) == 0); + BUG_ON(md->base && refcount_read(&md->refcnt) == 0); - if (atomic_dec_and_test(&md->refcnt)) + if (refcount_dec_and_test(&md->refcnt)) perf_mmap__munmap(md); } @@ -894,7 +894,7 @@ void perf_mmap__consume(struct perf_mmap *md, bool overwrite) perf_mmap__write_tail(md, old); } - if (atomic_read(&md->refcnt) == 1 && perf_mmap__empty(md)) + if (refcount_read(&md->refcnt) == 1 && perf_mmap__empty(md)) perf_mmap__put(md); } @@ -937,7 +937,7 @@ static void perf_mmap__munmap(struct perf_mmap *map) munmap(map->base, perf_mmap__mmap_len(map)); map->base = NULL; map->fd = -1; - atomic_set(&map->refcnt, 0); + refcount_set(&map->refcnt, 0); } auxtrace_mmap__munmap(&map->auxtrace_mmap); } @@ -1001,7 +1001,7 @@ static int perf_mmap__mmap(struct perf_mmap *map, * evlist layer can't just drop it when filtering events in * perf_evlist__filter_pollfd(). */ - atomic_set(&map->refcnt, 2); + refcount_set(&map->refcnt, 2); map->prev = 0; map->mask = mp->mask; map->base = mmap(NULL, perf_mmap__mmap_len(map), mp->prot, diff --git a/tools/perf/util/evlist.h b/tools/perf/util/evlist.h index 389b9ccdf8c7..39942995f537 100644 --- a/tools/perf/util/evlist.h +++ b/tools/perf/util/evlist.h @@ -1,7 +1,7 @@ #ifndef __PERF_EVLIST_H #define __PERF_EVLIST_H 1 -#include <linux/atomic.h> +#include <linux/refcount.h> #include <linux/list.h> #include <api/fd/array.h> #include <stdio.h> @@ -29,7 +29,7 @@ struct perf_mmap { void *base; int mask; int fd; - atomic_t refcnt; + refcount_t refcnt; u64 prev; struct auxtrace_mmap auxtrace_mmap; char event_copy[PERF_SAMPLE_MAX_SIZE] __attribute__((aligned(8))); -- 2.9.3

1 0

[alsa-devel] [PATCH 15/35] perf map: Convert map_groups.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-7-git-send-email-elena.reshetova@… [ Did the missing conversion of tests/thread-mg-share.c too ] Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/tests/thread-mg-share.c | 12 ++++++------ tools/perf/util/map.c | 4 ++-- tools/perf/util/map.h | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/tools/perf/tests/thread-mg-share.c b/tools/perf/tests/thread-mg-share.c index 188b63140fc8..76686dd6f5ec 100644 --- a/tools/perf/tests/thread-mg-share.c +++ b/tools/perf/tests/thread-mg-share.c @@ -43,7 +43,7 @@ int test__thread_mg_share(int subtest __maybe_unused) leader && t1 && t2 && t3 && other); mg = leader->mg; - TEST_ASSERT_EQUAL("wrong refcnt", atomic_read(&mg->refcnt), 4); + TEST_ASSERT_EQUAL("wrong refcnt", refcount_read(&mg->refcnt), 4); /* test the map groups pointer is shared */ TEST_ASSERT_VAL("map groups don't match", mg == t1->mg); @@ -71,25 +71,25 @@ int test__thread_mg_share(int subtest __maybe_unused) machine__remove_thread(machine, other_leader); other_mg = other->mg; - TEST_ASSERT_EQUAL("wrong refcnt", atomic_read(&other_mg->refcnt), 2); + TEST_ASSERT_EQUAL("wrong refcnt", refcount_read(&other_mg->refcnt), 2); TEST_ASSERT_VAL("map groups don't match", other_mg == other_leader->mg); /* release thread group */ thread__put(leader); - TEST_ASSERT_EQUAL("wrong refcnt", atomic_read(&mg->refcnt), 3); + TEST_ASSERT_EQUAL("wrong refcnt", refcount_read(&mg->refcnt), 3); thread__put(t1); - TEST_ASSERT_EQUAL("wrong refcnt", atomic_read(&mg->refcnt), 2); + TEST_ASSERT_EQUAL("wrong refcnt", refcount_read(&mg->refcnt), 2); thread__put(t2); - TEST_ASSERT_EQUAL("wrong refcnt", atomic_read(&mg->refcnt), 1); + TEST_ASSERT_EQUAL("wrong refcnt", refcount_read(&mg->refcnt), 1); thread__put(t3); /* release other group */ thread__put(other_leader); - TEST_ASSERT_EQUAL("wrong refcnt", atomic_read(&other_mg->refcnt), 1); + TEST_ASSERT_EQUAL("wrong refcnt", refcount_read(&other_mg->refcnt), 1); thread__put(other); diff --git a/tools/perf/util/map.c b/tools/perf/util/map.c index f0e2428efd0b..1d9ebcf9e38e 100644 --- a/tools/perf/util/map.c +++ b/tools/perf/util/map.c @@ -485,7 +485,7 @@ void map_groups__init(struct map_groups *mg, struct machine *machine) maps__init(&mg->maps[i]); } mg->machine = machine; - atomic_set(&mg->refcnt, 1); + refcount_set(&mg->refcnt, 1); } static void __maps__purge(struct maps *maps) @@ -547,7 +547,7 @@ void map_groups__delete(struct map_groups *mg) void map_groups__put(struct map_groups *mg) { - if (mg && atomic_dec_and_test(&mg->refcnt)) + if (mg && refcount_dec_and_test(&mg->refcnt)) map_groups__delete(mg); } diff --git a/tools/perf/util/map.h b/tools/perf/util/map.h index 9545ff343ec5..c8a5a644c0a9 100644 --- a/tools/perf/util/map.h +++ b/tools/perf/util/map.h @@ -67,7 +67,7 @@ struct maps { struct map_groups { struct maps maps[MAP__NR_TYPES]; struct machine *machine; - atomic_t refcnt; + refcount_t refcnt; }; struct map_groups *map_groups__new(struct machine *machine); @@ -77,7 +77,7 @@ bool map_groups__empty(struct map_groups *mg); static inline struct map_groups *map_groups__get(struct map_groups *mg) { if (mg) - atomic_inc(&mg->refcnt); + refcount_inc(&mg->refcnt); return mg; } -- 2.9.3

1 0

[alsa-devel] [PATCH 14/35] perf map: Convert map.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-6-git-send-email-elena.reshetova@… Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/util/map.c | 6 +++--- tools/perf/util/map.h | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/tools/perf/util/map.c b/tools/perf/util/map.c index 0a943e7b1ea7..f0e2428efd0b 100644 --- a/tools/perf/util/map.c +++ b/tools/perf/util/map.c @@ -141,7 +141,7 @@ void map__init(struct map *map, enum map_type type, RB_CLEAR_NODE(&map->rb_node); map->groups = NULL; map->erange_warned = false; - atomic_set(&map->refcnt, 1); + refcount_set(&map->refcnt, 1); } struct map *map__new(struct machine *machine, u64 start, u64 len, @@ -255,7 +255,7 @@ void map__delete(struct map *map) void map__put(struct map *map) { - if (map && atomic_dec_and_test(&map->refcnt)) + if (map && refcount_dec_and_test(&map->refcnt)) map__delete(map); } @@ -354,7 +354,7 @@ struct map *map__clone(struct map *from) struct map *map = memdup(from, sizeof(*map)); if (map != NULL) { - atomic_set(&map->refcnt, 1); + refcount_set(&map->refcnt, 1); RB_CLEAR_NODE(&map->rb_node); dso__get(map->dso); map->groups = NULL; diff --git a/tools/perf/util/map.h b/tools/perf/util/map.h index abdacf800c98..9545ff343ec5 100644 --- a/tools/perf/util/map.h +++ b/tools/perf/util/map.h @@ -1,7 +1,7 @@ #ifndef __PERF_MAP_H #define __PERF_MAP_H -#include <linux/atomic.h> +#include <linux/refcount.h> #include <linux/compiler.h> #include <linux/list.h> #include <linux/rbtree.h> @@ -51,7 +51,7 @@ struct map { struct dso *dso; struct map_groups *groups; - atomic_t refcnt; + refcount_t refcnt; }; struct kmap { @@ -150,7 +150,7 @@ struct map *map__clone(struct map *map); static inline struct map *map__get(struct map *map) { if (map) - atomic_inc(&map->refcnt); + refcount_inc(&map->refcnt); return map; } -- 2.9.3

1 0

[alsa-devel] [PATCH 13/35] perf dso: Convert dso.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-5-git-send-email-elena.reshetova@… Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/util/dso.c | 6 +++--- tools/perf/util/dso.h | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/tools/perf/util/dso.c b/tools/perf/util/dso.c index d38b62a700ca..42db00d78573 100644 --- a/tools/perf/util/dso.c +++ b/tools/perf/util/dso.c @@ -1109,7 +1109,7 @@ struct dso *dso__new(const char *name) INIT_LIST_HEAD(&dso->node); INIT_LIST_HEAD(&dso->data.open_entry); pthread_mutex_init(&dso->lock, NULL); - atomic_set(&dso->refcnt, 1); + refcount_set(&dso->refcnt, 1); } return dso; @@ -1147,13 +1147,13 @@ void dso__delete(struct dso *dso) struct dso *dso__get(struct dso *dso) { if (dso) - atomic_inc(&dso->refcnt); + refcount_inc(&dso->refcnt); return dso; } void dso__put(struct dso *dso) { - if (dso && atomic_dec_and_test(&dso->refcnt)) + if (dso && refcount_dec_and_test(&dso->refcnt)) dso__delete(dso); } diff --git a/tools/perf/util/dso.h b/tools/perf/util/dso.h index ecc4bbd3f82e..12350b171727 100644 --- a/tools/perf/util/dso.h +++ b/tools/perf/util/dso.h @@ -1,7 +1,7 @@ #ifndef __PERF_DSO #define __PERF_DSO -#include <linux/atomic.h> +#include <linux/refcount.h> #include <linux/types.h> #include <linux/rbtree.h> #include <sys/types.h> @@ -187,7 +187,7 @@ struct dso { void *priv; u64 db_id; }; - atomic_t refcnt; + refcount_t refcnt; char name[0]; }; -- 2.9.3

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[alsa-devel] [PATCH 12/35] perf comm: Convert comm_str.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-4-git-send-email-elena.reshetova@… [ Reinstated comm_str__get() function, needed when reusing entries in the rbtree ] Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/util/comm.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/tools/perf/util/comm.c b/tools/perf/util/comm.c index 21b7ff382c3f..32837b6f7879 100644 --- a/tools/perf/util/comm.c +++ b/tools/perf/util/comm.c @@ -2,12 +2,12 @@ #include "util.h" #include <stdlib.h> #include <stdio.h> -#include <linux/atomic.h> +#include <linux/refcount.h> struct comm_str { char *str; struct rb_node rb_node; - atomic_t refcnt; + refcount_t refcnt; }; /* Should perhaps be moved to struct machine */ @@ -16,13 +16,13 @@ static struct rb_root comm_str_root; static struct comm_str *comm_str__get(struct comm_str *cs) { if (cs) - atomic_inc(&cs->refcnt); + refcount_inc(&cs->refcnt); return cs; } static void comm_str__put(struct comm_str *cs) { - if (cs && atomic_dec_and_test(&cs->refcnt)) { + if (cs && refcount_dec_and_test(&cs->refcnt)) { rb_erase(&cs->rb_node, &comm_str_root); zfree(&cs->str); free(cs); @@ -43,7 +43,7 @@ static struct comm_str *comm_str__alloc(const char *str) return NULL; } - atomic_set(&cs->refcnt, 0); + refcount_set(&cs->refcnt, 1); return cs; } @@ -61,7 +61,7 @@ static struct comm_str *comm_str__findnew(const char *str, struct rb_root *root) cmp = strcmp(str, iter->str); if (!cmp) - return iter; + return comm_str__get(iter); if (cmp < 0) p = &(*p)->rb_left; @@ -95,8 +95,6 @@ struct comm *comm__new(const char *str, u64 timestamp, bool exec) return NULL; } - comm_str__get(comm->comm_str); - return comm; } @@ -108,7 +106,6 @@ int comm__override(struct comm *comm, const char *str, u64 timestamp, bool exec) if (!new) return -ENOMEM; - comm_str__get(new); comm_str__put(old); comm->comm_str = new; comm->start = timestamp; -- 2.9.3

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[alsa-devel] [PATCH 11/35] perf cpumap: Convert cpu_map.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: alsa-devel(a)alsa-project.org Link: http://lkml.kernel.org/r/1487691303-31858-3-git-send-email-elena.reshetova@… [ fixed mixed conversion to refcount in tests/cpumap.c ] Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/tests/cpumap.c | 2 +- tools/perf/util/cpumap.c | 16 ++++++++-------- tools/perf/util/cpumap.h | 4 ++-- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/tools/perf/tests/cpumap.c b/tools/perf/tests/cpumap.c index f168a85992d0..4478773cdb97 100644 --- a/tools/perf/tests/cpumap.c +++ b/tools/perf/tests/cpumap.c @@ -66,7 +66,7 @@ static int process_event_cpus(struct perf_tool *tool __maybe_unused, TEST_ASSERT_VAL("wrong nr", map->nr == 2); TEST_ASSERT_VAL("wrong cpu", map->map[0] == 1); TEST_ASSERT_VAL("wrong cpu", map->map[1] == 256); - TEST_ASSERT_VAL("wrong refcnt", atomic_read(&map->refcnt) == 1); + TEST_ASSERT_VAL("wrong refcnt", refcount_read(&map->refcnt) == 1); cpu_map__put(map); return 0; } diff --git a/tools/perf/util/cpumap.c b/tools/perf/util/cpumap.c index 8c7504939113..39ad2caccf56 100644 --- a/tools/perf/util/cpumap.c +++ b/tools/perf/util/cpumap.c @@ -29,7 +29,7 @@ static struct cpu_map *cpu_map__default_new(void) cpus->map[i] = i; cpus->nr = nr_cpus; - atomic_set(&cpus->refcnt, 1); + refcount_set(&cpus->refcnt, 1); } return cpus; @@ -43,7 +43,7 @@ static struct cpu_map *cpu_map__trim_new(int nr_cpus, int *tmp_cpus) if (cpus != NULL) { cpus->nr = nr_cpus; memcpy(cpus->map, tmp_cpus, payload_size); - atomic_set(&cpus->refcnt, 1); + refcount_set(&cpus->refcnt, 1); } return cpus; @@ -252,7 +252,7 @@ struct cpu_map *cpu_map__dummy_new(void) if (cpus != NULL) { cpus->nr = 1; cpus->map[0] = -1; - atomic_set(&cpus->refcnt, 1); + refcount_set(&cpus->refcnt, 1); } return cpus; @@ -269,7 +269,7 @@ struct cpu_map *cpu_map__empty_new(int nr) for (i = 0; i < nr; i++) cpus->map[i] = -1; - atomic_set(&cpus->refcnt, 1); + refcount_set(&cpus->refcnt, 1); } return cpus; @@ -278,7 +278,7 @@ struct cpu_map *cpu_map__empty_new(int nr) static void cpu_map__delete(struct cpu_map *map) { if (map) { - WARN_ONCE(atomic_read(&map->refcnt) != 0, + WARN_ONCE(refcount_read(&map->refcnt) != 0, "cpu_map refcnt unbalanced\n"); free(map); } @@ -287,13 +287,13 @@ static void cpu_map__delete(struct cpu_map *map) struct cpu_map *cpu_map__get(struct cpu_map *map) { if (map) - atomic_inc(&map->refcnt); + refcount_inc(&map->refcnt); return map; } void cpu_map__put(struct cpu_map *map) { - if (map && atomic_dec_and_test(&map->refcnt)) + if (map && refcount_dec_and_test(&map->refcnt)) cpu_map__delete(map); } @@ -357,7 +357,7 @@ int cpu_map__build_map(struct cpu_map *cpus, struct cpu_map **res, /* ensure we process id in increasing order */ qsort(c->map, c->nr, sizeof(int), cmp_ids); - atomic_set(&c->refcnt, 1); + refcount_set(&c->refcnt, 1); *res = c; return 0; } diff --git a/tools/perf/util/cpumap.h b/tools/perf/util/cpumap.h index 1a0549af8f5c..e84491636c1b 100644 --- a/tools/perf/util/cpumap.h +++ b/tools/perf/util/cpumap.h @@ -3,13 +3,13 @@ #include <stdio.h> #include <stdbool.h> -#include <linux/atomic.h> +#include <linux/refcount.h> #include "perf.h" #include "util/debug.h" struct cpu_map { - atomic_t refcnt; + refcount_t refcnt; int nr; int map[]; }; -- 2.9.3

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[alsa-devel] [PATCH 10/35] perf cgroup: Convert cgroup_sel.refcnt from atomic_t to refcount_t
by Arnaldo Carvalho de Melo 06 Mar '17

06 Mar '17

From: Elena Reshetova <elena.reshetova(a)intel.com> The refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: David Windsor <dwindsor(a)gmail.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Kook <keescook(a)chromium.org> Tested-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> Cc: Alexander Shishkin <alexander.shishkin(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: David Windsor <dwindsor(a)gmail.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Hans Liljestrand <ishkamiel(a)gmail.com> Cc: Jiri Olsa <jolsa(a)kernel.org> Cc: Kees Kook <keescook(a)chromium.org> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Matija Glavinic Pecotic <matija.glavinic-pecotic.ext(a)nokia.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Link: http://lkml.kernel.org/r/1487691303-31858-2-git-send-email-elena.reshetova@… Signed-off-by: Arnaldo Carvalho de Melo <acme(a)redhat.com> --- tools/perf/util/cgroup.c | 6 +++--- tools/perf/util/cgroup.h | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/tools/perf/util/cgroup.c b/tools/perf/util/cgroup.c index eafbf11442b2..86399eda3684 100644 --- a/tools/perf/util/cgroup.c +++ b/tools/perf/util/cgroup.c @@ -127,19 +127,19 @@ static int add_cgroup(struct perf_evlist *evlist, char *str) goto found; n++; } - if (atomic_read(&cgrp->refcnt) == 0) + if (refcount_read(&cgrp->refcnt) == 0) free(cgrp); return -1; found: - atomic_inc(&cgrp->refcnt); + refcount_inc(&cgrp->refcnt); counter->cgrp = cgrp; return 0; } void close_cgroup(struct cgroup_sel *cgrp) { - if (cgrp && atomic_dec_and_test(&cgrp->refcnt)) { + if (cgrp && refcount_dec_and_test(&cgrp->refcnt)) { close(cgrp->fd); zfree(&cgrp->name); free(cgrp); diff --git a/tools/perf/util/cgroup.h b/tools/perf/util/cgroup.h index 31f8dcdbd7ef..d91966b97cbd 100644 --- a/tools/perf/util/cgroup.h +++ b/tools/perf/util/cgroup.h @@ -1,14 +1,14 @@ #ifndef __CGROUP_H__ #define __CGROUP_H__ -#include <linux/atomic.h> +#include <linux/refcount.h> struct option; struct cgroup_sel { char *name; int fd; - atomic_t refcnt; + refcount_t refcnt; }; -- 2.9.3

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[alsa-devel] [Satellite Pro P200, Realtek ALC268, Black Headphone Out, Front] No sound at all
by Raúl Vidal 06 Mar '17

06 Mar '17

I do not have sound in the headphone jack of my laptop. Ubuntu bug report: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1670394 Attached to this e-mail you will find the alsa-info.sh output.

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[alsa-devel] [PATCH v6 2/2] ASoC: cs35l35: Add bindings file for CS35L35
by Brian Austin 06 Mar '17

06 Mar '17

Add bindings documentation for CS35L35 Amplifier Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- v5: Make IPeak and BST parameters consistent with previous devices that have similar properties. v6: Use proper subject for subsystem. .../devicetree/bindings/sound/cs35l35.txt | 171 +++++++++++++++++++++ 1 file changed, 171 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l35.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l35.txt b/Documentation/devicetree/bindings/sound/cs35l35.txt new file mode 100644 index 0000000..07d9c0f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l35.txt @@ -0,0 +1,171 @@ +CS35L35 Boosted Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupt-parent : Specifies the phandle of the interrupt controller to + which the IRQs from CS35L35 are delivered to. + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + +Optional properties: + - reset-gpios : gpio used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2600mV to 9000mV with + increments of 100mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1680mA and goes to a maximum of 4480mA with increments of + 110mA. + (Default) 2.46 Amps + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 3 8 bit values to set the depth, location, and frame + of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: cs35l35@20 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl-millivolt = <9000>; + + cirrus,stereo-config; + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; -- 1.9.1

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[alsa-devel] [PATCH v6 1/2] ASoC: Add support for Cirrus Logic CS35L35 Amplifier
by Brian Austin 06 Mar '17

06 Mar '17

This patch adds support for the Cirrus Logic CS35L35 9V Boosted Amplifier Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- v5: Update bindings documentation and parameters to be inline with previous Cirrus AMP's. Shared /RST config for mulit-device style changes. v6: Coding style. Put device in reset if err. Change gain to Volume include/sound/cs35l35.h | 103 +++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l35.c | 1553 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l35.h | 284 ++++++++ 5 files changed, 1947 insertions(+) create mode 100644 include/sound/cs35l35.h create mode 100644 sound/soc/codecs/cs35l35.c create mode 100644 sound/soc/codecs/cs35l35.h diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h new file mode 100644 index 0000000..de92e45 --- /dev/null +++ b/include/sound/cs35l35.h @@ -0,0 +1,103 @@ +/* + * linux/sound/cs35l35.h -- Platform data for CS35l35 + * + * 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 __CS35L35_H +#define __CS35L35_H + +struct classh_cfg { + /* + * Class H Algorithm Control Variables + * You can either have it done + * automatically or you can adjust + * these variables for tuning + * + * if you do not enable the internal algorithm + * you will get a set of mixer controls for + * Class H tuning + * + * Section 4.3 of the datasheet + */ + bool classh_bst_override; + bool classh_algo_enable; + int classh_bst_max_limit; + int classh_mem_depth; + int classh_release_rate; + int classh_headroom; + int classh_wk_fet_disable; + int classh_wk_fet_delay; + int classh_wk_fet_thld; + int classh_vpch_auto; + int classh_vpch_rate; + int classh_vpch_man; +}; + +struct monitor_cfg { + /* + * Signal Monitor Data + * highly configurable signal monitoring + * data positioning and different types of + * monitoring data. + * + * Section 4.8.2 - 4.8.4 of the datasheet + */ + bool is_present; + bool imon_specs; + bool vmon_specs; + bool vpmon_specs; + bool vbstmon_specs; + bool vpbrstat_specs; + bool zerofill_specs; + u8 imon_dpth; + u8 imon_loc; + u8 imon_frm; + u8 vmon_dpth; + u8 vmon_loc; + u8 vmon_frm; + u8 vpmon_dpth; + u8 vpmon_loc; + u8 vpmon_frm; + u8 vbstmon_dpth; + u8 vbstmon_loc; + u8 vbstmon_frm; + u8 vpbrstat_dpth; + u8 vpbrstat_loc; + u8 vpbrstat_frm; + u8 zerofill_dpth; + u8 zerofill_loc; + u8 zerofill_frm; +}; + +struct cs35l35_platform_data { + + /* Stereo (2 Device) */ + bool stereo; + /* serial port drive strength */ + int sp_drv_str; + /* Boost Power Down with FET */ + bool bst_pdn_fet_on; + /* Boost Voltage : used if ClassH Algo Enabled */ + int bst_vctl; + /* Boost Converter Peak Current CTRL */ + int bst_ipk; + /* Amp Gain Zero Cross */ + bool gain_zc; + /* Audio Input Location */ + int aud_channel; + /* Advisory Input Location */ + int adv_channel; + /* Shared Boost for stereo */ + bool shared_bst; + /* ClassH Algorithm */ + struct classh_cfg classh_algo; + /* Monitor Config */ + struct monitor_cfg mon_cfg; +}; + +#endif /* __CS35L35_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e49e9da..00885c2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C + select SND_SOC_CS35L35 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -408,6 +409,10 @@ config SND_SOC_CS35L34 tristate "Cirrus Logic CS35L34 CODEC" depends on I2C +config SND_SOC_CS35L35 + tristate "Cirrus Logic CS35L35 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1796cb9..97b36e4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o +snd-soc-cs35l35-objs := cs35l35.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -269,6 +270,7 @@ 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_CS35L34) += snd-soc-cs35l34.o +obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c new file mode 100644 index 0000000..260ed42 --- /dev/null +++ b/sound/soc/codecs/cs35l35.c @@ -0,0 +1,1553 @@ +/* + * cs35l35.c -- CS35L35 ALSA SoC audio driver + * + * Copyright 2017 Cirrus Logic, Inc. + * + * Author: Brian Austin <brian.austin(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/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l35.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l35.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L35_VALID_PDATA 0x80000000 + +static const struct reg_default cs35l35_reg[] = { + {CS35L35_PWRCTL1, 0x01}, + {CS35L35_PWRCTL2, 0x11}, + {CS35L35_PWRCTL3, 0x00}, + {CS35L35_CLK_CTL1, 0x04}, + {CS35L35_CLK_CTL2, 0x10}, + {CS35L35_CLK_CTL3, 0xCF}, + {CS35L35_SP_FMT_CTL1, 0x20}, + {CS35L35_SP_FMT_CTL2, 0x00}, + {CS35L35_SP_FMT_CTL3, 0x02}, + {CS35L35_MAG_COMP_CTL, 0x00}, + {CS35L35_AMP_INP_DRV_CTL, 0x01}, + {CS35L35_AMP_DIG_VOL_CTL, 0x12}, + {CS35L35_AMP_DIG_VOL, 0x00}, + {CS35L35_ADV_DIG_VOL, 0x00}, + {CS35L35_PROTECT_CTL, 0x06}, + {CS35L35_AMP_GAIN_AUD_CTL, 0x13}, + {CS35L35_AMP_GAIN_PDM_CTL, 0x00}, + {CS35L35_AMP_GAIN_ADV_CTL, 0x00}, + {CS35L35_GPI_CTL, 0x00}, + {CS35L35_BST_CVTR_V_CTL, 0x00}, + {CS35L35_BST_PEAK_I, 0x07}, + {CS35L35_BST_RAMP_CTL, 0x85}, + {CS35L35_BST_CONV_COEF_1, 0x24}, + {CS35L35_BST_CONV_COEF_2, 0x24}, + {CS35L35_BST_CONV_SLOPE_COMP, 0x47}, + {CS35L35_BST_CONV_SW_FREQ, 0x04}, + {CS35L35_CLASS_H_CTL, 0x0B}, + {CS35L35_CLASS_H_HEADRM_CTL, 0x0B}, + {CS35L35_CLASS_H_RELEASE_RATE, 0x08}, + {CS35L35_CLASS_H_FET_DRIVE_CTL, 0x41}, + {CS35L35_CLASS_H_VP_CTL, 0xC5}, + {CS35L35_VPBR_CTL, 0x0A}, + {CS35L35_VPBR_VOL_CTL, 0x09}, + {CS35L35_VPBR_TIMING_CTL, 0x6A}, + {CS35L35_VPBR_MODE_VOL_CTL, 0x40}, + {CS35L35_SPKR_MON_CTL, 0xC0}, + {CS35L35_IMON_SCALE_CTL, 0x30}, + {CS35L35_AUDIN_RXLOC_CTL, 0x00}, + {CS35L35_ADVIN_RXLOC_CTL, 0x80}, + {CS35L35_VMON_TXLOC_CTL, 0x00}, + {CS35L35_IMON_TXLOC_CTL, 0x80}, + {CS35L35_VPMON_TXLOC_CTL, 0x04}, + {CS35L35_VBSTMON_TXLOC_CTL, 0x84}, + {CS35L35_VPBR_STATUS_TXLOC_CTL, 0x04}, + {CS35L35_ZERO_FILL_LOC_CTL, 0x00}, + {CS35L35_AUDIN_DEPTH_CTL, 0x0F}, + {CS35L35_SPKMON_DEPTH_CTL, 0x0F}, + {CS35L35_SUPMON_DEPTH_CTL, 0x0F}, + {CS35L35_ZEROFILL_DEPTH_CTL, 0x00}, + {CS35L35_MULT_DEV_SYNCH1, 0x02}, + {CS35L35_MULT_DEV_SYNCH2, 0x80}, + {CS35L35_PROT_RELEASE_CTL, 0x00}, + {CS35L35_DIAG_MODE_REG_LOCK, 0x00}, + {CS35L35_DIAG_MODE_CTL_1, 0x40}, + {CS35L35_DIAG_MODE_CTL_2, 0x00}, + {CS35L35_INT_MASK_1, 0xFF}, + {CS35L35_INT_MASK_2, 0xFF}, + {CS35L35_INT_MASK_3, 0xFF}, + {CS35L35_INT_MASK_4, 0xFF}, + +}; + +static bool cs35l35_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_DEVID_AB ... CS35L35_PWRCTL3: + case CS35L35_CLK_CTL1 ... CS35L35_SP_FMT_CTL3: + case CS35L35_MAG_COMP_CTL ... CS35L35_AMP_GAIN_AUD_CTL: + case CS35L35_AMP_GAIN_PDM_CTL ... CS35L35_BST_PEAK_I: + case CS35L35_BST_RAMP_CTL ... CS35L35_BST_CONV_SW_FREQ: + case CS35L35_CLASS_H_CTL ... CS35L35_CLASS_H_VP_CTL: + case CS35L35_CLASS_H_STATUS: + case CS35L35_VPBR_CTL ... CS35L35_VPBR_MODE_VOL_CTL: + case CS35L35_VPBR_ATTEN_STATUS: + case CS35L35_SPKR_MON_CTL: + case CS35L35_IMON_SCALE_CTL ... CS35L35_ZEROFILL_DEPTH_CTL: + case CS35L35_MULT_DEV_SYNCH1 ... CS35L35_PROT_RELEASE_CTL: + case CS35L35_DIAG_MODE_REG_LOCK ... CS35L35_DIAG_MODE_CTL_2: + case CS35L35_INT_MASK_1 ... CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static int cs35l35_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 cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, + 0 << CS35L35_MCLK_DIS_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, + 0 << CS35L35_DISCHG_FILT_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, + 1 << CS35L35_DISCHG_FILT_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 1); + + reinit_completion(&cs35l35->pdn_done); + + ret = wait_for_completion_timeout(&cs35l35->pdn_done, + msecs_to_jiffies(100)); + if (ret == 0) { + dev_err(codec->dev, "TIMEOUT PDN_DONE did not complete in 100ms\n"); + ret = -ETIMEDOUT; + } + + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, + 1 << CS35L35_MCLK_DIS_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + ret = -EINVAL; + } + return ret; +} + +static int cs35l35_main_amp_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 cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + unsigned int reg[4]; + int i; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 0 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 0 << CS35L35_PDN_BST_FETOFF_SHIFT); + break; + case SND_SOC_DAPM_POST_PMU: + usleep_range(5000, 5100); + /* If in PDM mode we must use VP for Voltage control */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, + 0 << CS35L35_BST_CTL_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 0); + + for (i = 0; i < 2; i++) + regmap_bulk_read(cs35l35->regmap, CS35L35_INT_STATUS_1, + &reg, ARRAY_SIZE(reg)); + + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, + 1 << CS35L35_AMP_MUTE_SHIFT); + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + usleep_range(5000, 5100); + /* + * If PDM mode we should switch back to pdata value + * for Voltage control when we go down + */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, + cs35l35->pdata.bst_vctl + << CS35L35_BST_CTL_SHIFT); + + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + } + return 0; +} + +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l35_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("Analog Audio Volume", CS35L35_AMP_GAIN_AUD_CTL, 0, 19, 0, + amp_gain_tlv), + SOC_SINGLE_TLV("PDM Volume", CS35L35_AMP_GAIN_PDM_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_kcontrol_new cs35l35_adv_controls[] = { + SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("Analog Advisory Volume", CS35L35_AMP_GAIN_ADV_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_soc_dapm_widget cs35l35_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L35_PWRCTL3, 1, 1, + cs35l35_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L35_PWRCTL3, 2, 1), + + SND_SOC_DAPM_OUTPUT("SPK"), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L35_PWRCTL2, 7, 1), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L35_PWRCTL2, 6, 1), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L35_PWRCTL3, 3, 1), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L35_PWRCTL3, 4, 1), + SND_SOC_DAPM_ADC("CLASS H", NULL, CS35L35_PWRCTL2, 5, 1), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L35_PWRCTL2, 0, 1, NULL, 0, + cs35l35_main_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l35_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + {"SDOUT", NULL, "IMON ADC"}, + {"SDOUT", NULL, "VMON ADC"}, + {"SDOUT", NULL, "VBSTMON ADC"}, + {"SDOUT", NULL, "VPMON ADC"}, + {"AMP Capture", NULL, "SDOUT"}, + + {"SDIN", NULL, "AMP Playback"}, + {"CLASS H", NULL, "SDIN"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = false; + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs35l35->i2s_mode = true; + cs35l35->pdm_mode = false; + break; + case SND_SOC_DAIFMT_PDM: + cs35l35->pdm_mode = true; + cs35l35->i2s_mode = false; + break; + default: + return -EINVAL; + } + + return 0; +} + +struct cs35l35_sysclk_config { + int sysclk; + int srate; + u8 clk_cfg; +}; + +static struct cs35l35_sysclk_config cs35l35_clk_ctl[] = { + + /* SYSCLK, Sample Rate, Serial Port Cfg */ + {5644800, 44100, 0x00}, + {5644800, 88200, 0x40}, + {6144000, 48000, 0x10}, + {6144000, 96000, 0x50}, + {11289600, 44100, 0x01}, + {11289600, 88200, 0x41}, + {11289600, 176400, 0x81}, + {12000000, 44100, 0x03}, + {12000000, 48000, 0x13}, + {12000000, 88200, 0x43}, + {12000000, 96000, 0x53}, + {12000000, 176400, 0x83}, + {12000000, 192000, 0x93}, + {12288000, 48000, 0x11}, + {12288000, 96000, 0x51}, + {12288000, 192000, 0x91}, + {13000000, 44100, 0x07}, + {13000000, 48000, 0x17}, + {13000000, 88200, 0x47}, + {13000000, 96000, 0x57}, + {13000000, 176400, 0x87}, + {13000000, 192000, 0x97}, + {22579200, 44100, 0x02}, + {22579200, 88200, 0x42}, + {22579200, 176400, 0x82}, + {24000000, 44100, 0x0B}, + {24000000, 48000, 0x1B}, + {24000000, 88200, 0x4B}, + {24000000, 96000, 0x5B}, + {24000000, 176400, 0x8B}, + {24000000, 192000, 0x9B}, + {24576000, 48000, 0x12}, + {24576000, 96000, 0x52}, + {24576000, 192000, 0x92}, + {26000000, 44100, 0x0F}, + {26000000, 48000, 0x1F}, + {26000000, 88200, 0x4F}, + {26000000, 96000, 0x5F}, + {26000000, 176400, 0x8F}, + {26000000, 192000, 0x9F}, +}; + +static int cs35l35_get_clk_config(int sysclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l35_clk_ctl); i++) { + if (cs35l35_clk_ctl[i].sysclk == sysclk && + cs35l35_clk_ctl[i].srate == srate) + return cs35l35_clk_ctl[i].clk_cfg; + } + return -EINVAL; +} + +static int cs35l35_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 cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + int srate = params_rate(params); + int ret = 0; + u8 sp_sclks; + int audin_format; + int errata_chk; + + int clk_ctl = cs35l35_get_clk_config(cs35l35->sysclk, srate); + + if (clk_ctl < 0) { + dev_err(codec->dev, "Invalid CLK:Rate %d:%d\n", + cs35l35->sysclk, srate); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL2, + CS35L35_CLK_CTL2_MASK, clk_ctl); + if (ret != 0) { + dev_err(codec->dev, "Failed to set port config %d\n", ret); + return ret; + } + + /* + * Rev A0 Errata + * When configured for the weak-drive detection path (CH_WKFET_DIS = 0) + * the Class H algorithm does not enable weak-drive operation for + * nonzero values of CH_WKFET_DELAY if SP_RATE = 01 or 10 + */ + errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; + + if (classh->classh_wk_fet_disable == 0x00 && + (errata_chk == 0x01 || errata_chk == 0x03)) { + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_DEL_MASK, + 0 << CS35L35_CH_WKFET_DEL_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fet config %d\n", + ret); + return ret; + } + } + + /* + * You can pull more Monitor data from the SDOUT pin than going to SDIN + * Just make sure your SCLK is fast enough to fill the frame + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (params_width(params)) { + case 8: + audin_format = CS35L35_SDIN_DEPTH_8; + break; + case 16: + audin_format = CS35L35_SDIN_DEPTH_16; + break; + case 24: + audin_format = CS35L35_SDIN_DEPTH_24; + break; + default: + dev_err(codec->dev, "Unsupported Width %d\n", + params_width(params)); + return -EINVAL; + } + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, + CS35L35_AUDIN_DEPTH_MASK, + audin_format << + CS35L35_AUDIN_DEPTH_SHIFT); + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, + CS35L35_ADVIN_DEPTH_MASK, + audin_format << + CS35L35_ADVIN_DEPTH_SHIFT); + } + } + + if (cs35l35->i2s_mode) { + /* We have to take the SCLK to derive num sclks + * to configure the CLOCK_CTL3 register correctly + */ + if ((cs35l35->sclk / srate) % 4) { + dev_err(codec->dev, "Unsupported sclk/fs ratio %d:%d\n", + cs35l35->sclk, srate); + return -EINVAL; + } + sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; + + /* Only certain ratios are supported in I2S Slave Mode */ + if (cs35l35->slave_mode) { + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_48FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } else { + /* Only certain ratios supported in I2S MASTER Mode */ + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLK_CTL3, + CS35L35_SP_SCLKS_MASK, sp_sclks << + CS35L35_SP_SCLKS_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fsclk %d\n", ret); + return ret; + } + } + + return ret; +} + +static const unsigned int cs35l35_src_rates[] = { + 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_constraints = { + .count = ARRAY_SIZE(cs35l35_src_rates), + .list = cs35l35_src_rates, +}; + +static int cs35l35_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + if (!substream->runtime) + return 0; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l35_constraints); + + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, + 0 << CS35L35_PDM_MODE_SHIFT); + + return 0; +} + +static const unsigned int cs35l35_pdm_rates[] = { + 44100, 48000, 88200, 96000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_pdm_constraints = { + .count = ARRAY_SIZE(cs35l35_pdm_rates), + .list = cs35l35_pdm_rates, +}; + +static int cs35l35_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + if (!substream->runtime) + return 0; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l35_pdm_constraints); + + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, + 1 << CS35L35_PDM_MODE_SHIFT); + + return 0; +} + +static int cs35l35_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + /* Need the SCLK Frequency regardless of sysclk source for I2S */ + cs35l35->sclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops cs35l35_ops = { + .startup = cs35l35_pcm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_hw_params, + .set_sysclk = cs35l35_dai_set_sysclk, +}; + +static const struct snd_soc_dai_ops cs35l35_pdm_ops = { + .startup = cs35l35_pdm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_hw_params, +}; + +static struct snd_soc_dai_driver cs35l35_dai[] = { + { + .name = "cs35l35-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_ops, + .symmetric_rates = 1, + }, + { + .name = "cs35l35-pdm", + .id = 1, + .playback = { + .stream_name = "PDM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_pdm_ops, + }, +}; + +static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int clksrc; + int ret = 0; + + switch (clk_id) { + case 0: + clksrc = CS35L35_CLK_SOURCE_MCLK; + break; + case 1: + clksrc = CS35L35_CLK_SOURCE_SCLK; + break; + case 2: + clksrc = CS35L35_CLK_SOURCE_PDM; + break; + default: + dev_err(codec->dev, "Invalid CLK Source\n"); + return -EINVAL; + }; + + switch (freq) { + case 5644800: + case 6144000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs35l35->sysclk = freq; + break; + default: + dev_err(codec->dev, "Invalid CLK Frequency Input : %d\n", freq); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_CLK_SOURCE_MASK, + clksrc << CS35L35_CLK_SOURCE_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set sysclk %d\n", ret); + return ret; + } + + return ret; +} + +static int cs35l35_codec_probe(struct snd_soc_codec *codec) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + struct monitor_cfg *monitor_config = &cs35l35->pdata.mon_cfg; + int ret; + + cs35l35->codec = codec; + + /* Set Platform Data */ + if (cs35l35->pdata.bst_vctl) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, + cs35l35->pdata.bst_vctl); + + if (cs35l35->pdata.bst_ipk) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_PEAK_I, + CS35L35_BST_IPK_MASK, + cs35l35->pdata.bst_ipk << + CS35L35_BST_IPK_SHIFT); + + if (cs35l35->pdata.gain_zc) + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_GAIN_ZC_MASK, + cs35l35->pdata.gain_zc << + CS35L35_AMP_GAIN_ZC_SHIFT); + + if (cs35l35->pdata.aud_channel) + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_RXLOC_CTL, + CS35L35_AUD_IN_LR_MASK, + cs35l35->pdata.aud_channel << + CS35L35_AUD_IN_LR_SHIFT); + + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_ADVIN_RXLOC_CTL, + CS35L35_ADV_IN_LR_MASK, + cs35l35->pdata.adv_channel << + CS35L35_ADV_IN_LR_SHIFT); + if (cs35l35->pdata.shared_bst) + regmap_update_bits(cs35l35->regmap, CS35L35_CLASS_H_CTL, + CS35L35_CH_STEREO_MASK, + 1 << CS35L35_CH_STEREO_SHIFT); + ret = snd_soc_add_codec_controls(codec, cs35l35_adv_controls, + ARRAY_SIZE(cs35l35_adv_controls)); + if (ret) + return ret; + } + + if (cs35l35->pdata.sp_drv_str) + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_SP_DRV_MASK, + cs35l35->pdata.sp_drv_str << + CS35L35_SP_DRV_SHIFT); + + if (classh->classh_algo_enable) { + if (classh->classh_bst_override) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, + CS35L35_CH_BST_OVR_MASK, + classh->classh_bst_override << + CS35L35_CH_BST_OVR_SHIFT); + if (classh->classh_bst_max_limit) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, + CS35L35_CH_BST_LIM_MASK, + classh->classh_bst_max_limit << + CS35L35_CH_BST_LIM_SHIFT); + if (classh->classh_mem_depth) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, + CS35L35_CH_MEM_DEPTH_MASK, + classh->classh_mem_depth << + CS35L35_CH_MEM_DEPTH_SHIFT); + if (classh->classh_headroom) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_HEADRM_CTL, + CS35L35_CH_HDRM_CTL_MASK, + classh->classh_headroom << + CS35L35_CH_HDRM_CTL_SHIFT); + if (classh->classh_release_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_RELEASE_RATE, + CS35L35_CH_REL_RATE_MASK, + classh->classh_release_rate << + CS35L35_CH_REL_RATE_SHIFT); + if (classh->classh_wk_fet_disable) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_DIS_MASK, + classh->classh_wk_fet_disable << + CS35L35_CH_WKFET_DIS_SHIFT); + if (classh->classh_wk_fet_delay) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_DEL_MASK, + classh->classh_wk_fet_delay << + CS35L35_CH_WKFET_DEL_SHIFT); + if (classh->classh_wk_fet_thld) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_THLD_MASK, + classh->classh_wk_fet_thld << + CS35L35_CH_WKFET_THLD_SHIFT); + if (classh->classh_vpch_auto) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, + CS35L35_CH_VP_AUTO_MASK, + classh->classh_vpch_auto << + CS35L35_CH_VP_AUTO_SHIFT); + if (classh->classh_vpch_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, + CS35L35_CH_VP_RATE_MASK, + classh->classh_vpch_rate << + CS35L35_CH_VP_RATE_SHIFT); + if (classh->classh_vpch_man) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, + CS35L35_CH_VP_MAN_MASK, + classh->classh_vpch_man << + CS35L35_CH_VP_MAN_SHIFT); + } + + if (monitor_config->is_present) { + if (monitor_config->vmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, + CS35L35_VMON_DEPTH_MASK, + monitor_config->vmon_dpth << + CS35L35_VMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vmon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vmon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->imon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, + CS35L35_IMON_DEPTH_MASK, + monitor_config->imon_dpth << + CS35L35_IMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->imon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->imon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_VPMON_DEPTH_MASK, + monitor_config->vpmon_dpth << + CS35L35_VPMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vpmon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vpmon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vbstmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_VBSTMON_DEPTH_MASK, + monitor_config->vpmon_dpth << + CS35L35_VBSTMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vbstmon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vbstmon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpbrstat_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_VPBRSTAT_DEPTH_MASK, + monitor_config->vpbrstat_dpth << + CS35L35_VPBRSTAT_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vpbrstat_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vpbrstat_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->zerofill_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_ZEROFILL_DEPTH_MASK, + monitor_config->zerofill_dpth << + CS35L35_ZEROFILL_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->zerofill_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->zerofill_frm << + CS35L35_MON_FRM_SHIFT); + } + } + + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = { + .probe = cs35l35_codec_probe, + .set_sysclk = cs35l35_codec_set_sysclk, + .component_driver = { + .dapm_widgets = cs35l35_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l35_dapm_widgets), + + .dapm_routes = cs35l35_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l35_audio_map), + + .controls = cs35l35_aud_controls, + .num_controls = ARRAY_SIZE(cs35l35_aud_controls), + }, + +}; + +static struct regmap_config cs35l35_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L35_MAX_REGISTER, + .reg_defaults = cs35l35_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l35_reg), + .volatile_reg = cs35l35_volatile_register, + .readable_reg = cs35l35_readable_register, + .precious_reg = cs35l35_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l35_irq(int irq, void *data) +{ + struct cs35l35_private *cs35l35 = data; + struct snd_soc_codec *codec = cs35l35->codec; + unsigned int sticky1, sticky2, sticky3, sticky4; + unsigned int mask1, mask2, mask3, mask4, current1; + + /* ack the irq by reading all status registers */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_4, &sticky4); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_3, &sticky3); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_2, &sticky2); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &sticky1); + + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_4, &mask4); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_3, &mask3); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_2, &mask2); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_1, &mask1); + + /* Check to see if unmasked bits are active */ + if (!(sticky1 & ~mask1) && !(sticky2 & ~mask2) && !(sticky3 & ~mask3) + && !(sticky4 & ~mask4)) + return IRQ_NONE; + + if (sticky2 & CS35L35_PDN_DONE) + complete(&cs35l35->pdn_done); + + /* read the current values */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &current1); + + /* handle the interrupts */ + if (sticky1 & CS35L35_CAL_ERR) { + dev_crit(codec->dev, "Calibration Error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_CAL_ERR)) { + pr_debug("%s : Cal error release\n", __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_CAL_ERR_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_CAL_ERR_RLS, + CS35L35_CAL_ERR_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_CAL_ERR_RLS, 0); + } + } + + if (sticky1 & CS35L35_AMP_SHORT) { + dev_crit(codec->dev, "AMP Short Error\n"); + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_AMP_SHORT)) { + dev_dbg(codec->dev, "Amp short error release\n"); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_SHORT_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_SHORT_RLS, + CS35L35_SHORT_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_SHORT_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTW) { + dev_warn(codec->dev, "Over temperature warning\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTW)) { + dev_dbg(codec->dev, "Over temperature warn release\n"); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTW_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTW_RLS, + CS35L35_OTW_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTW_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTE)) { + dev_dbg(codec->dev, "Over temperature error release\n"); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTE_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTE_RLS, + CS35L35_OTE_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTE_RLS, 0); + } + } + + if (sticky3 & CS35L35_BST_HIGH) { + dev_crit(codec->dev, "VBST error: powering off!\n"); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky3 & CS35L35_LBST_SHORT) { + dev_crit(codec->dev, "LBST error: powering off!\n"); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky2 & CS35L35_VPBR_ERR) + dev_dbg(codec->dev, "Error: Reactive Brownout\n"); + + if (sticky4 & CS35L35_VMON_OVFL) + dev_dbg(codec->dev, "Error: VMON overflow\n"); + + if (sticky4 & CS35L35_IMON_OVFL) + dev_dbg(codec->dev, "Error: IMON overflow\n"); + + return IRQ_HANDLED; +} + + +static int cs35l35_handle_of_data(struct i2c_client *i2c_client, + struct cs35l35_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct device_node *classh, *signal_format; + struct classh_cfg *classh_config = &pdata->classh_algo; + struct monitor_cfg *monitor_config = &pdata->mon_cfg; + unsigned int val32 = 0; + u8 monitor_array[3]; + int ret = 0; + + if (!np) + return 0; + + pdata->bst_pdn_fet_on = of_property_read_bool(np, + "cirrus,boost-pdn-fet-on"); + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val32); + if (ret >= 0) { + if (val32 < 2600 || val32 > 9000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val32); + return -EINVAL; + } + pdata->bst_vctl = ((val32 - 2600) / 100) + 1; + } + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val32); + if (ret >= 0) { + if (val32 < 1680 || val32 > 4480) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val32); + return -EINVAL; + } + + pdata->bst_ipk = (val32 - 1680) / 110; + } + + if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) + pdata->sp_drv_str = val32; + + pdata->stereo = of_property_read_bool(np, "cirrus,stereo-config"); + + if (pdata->stereo) { + ret = of_property_read_u32(np, "cirrus,audio-channel", &val32); + if (ret >= 0) + pdata->aud_channel = val32; + + ret = of_property_read_u32(np, "cirrus,advisory-channel", + &val32); + if (ret >= 0) + pdata->adv_channel = val32; + + pdata->shared_bst = of_property_read_bool(np, + "cirrus,shared-boost"); + } + + pdata->gain_zc = of_property_read_bool(np, "cirrus,amp-gain-zc"); + + classh = of_get_child_by_name(np, "cirrus,classh-internal-algo"); + classh_config->classh_algo_enable = classh ? true : false; + + if (classh_config->classh_algo_enable) { + classh_config->classh_bst_override = + of_property_read_bool(np, "cirrus,classh-bst-overide"); + + ret = of_property_read_u32(classh, + "cirrus,classh-bst-max-limit", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_bst_max_limit = val32; + } + + ret = of_property_read_u32(classh, + "cirrus,classh-bst-max-limit", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_bst_max_limit = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-mem-depth", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_mem_depth = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-release-rate", + &val32); + if (ret >= 0) + classh_config->classh_release_rate = val32; + + ret = of_property_read_u32(classh, "cirrus,classh-headroom", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_headroom = val32; + } + + ret = of_property_read_u32(classh, + "cirrus,classh-wk-fet-disable", + &val32); + if (ret >= 0) + classh_config->classh_wk_fet_disable = val32; + + ret = of_property_read_u32(classh, "cirrus,classh-wk-fet-delay", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_wk_fet_delay = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-wk-fet-thld", + &val32); + if (ret >= 0) + classh_config->classh_wk_fet_thld = val32; + + ret = of_property_read_u32(classh, "cirrus,classh-vpch-auto", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_vpch_auto = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-vpch-rate", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_vpch_rate = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-vpch-man", + &val32); + if (ret >= 0) + classh_config->classh_vpch_man = val32; + } + of_node_put(classh); + + /* frame depth location */ + signal_format = of_get_child_by_name(np, "cirrus,monitor-signal-format"); + monitor_config->is_present = signal_format ? true : false; + if (monitor_config->is_present) { + ret = of_property_read_u8_array(signal_format, "cirrus,imon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->imon_specs = true; + monitor_config->imon_dpth = monitor_array[0]; + monitor_config->imon_loc = monitor_array[1]; + monitor_config->imon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vmon_specs = true; + monitor_config->vmon_dpth = monitor_array[0]; + monitor_config->vmon_loc = monitor_array[1]; + monitor_config->vmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpmon_specs = true; + monitor_config->vpmon_dpth = monitor_array[0]; + monitor_config->vpmon_loc = monitor_array[1]; + monitor_config->vpmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vbstmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vbstmon_specs = true; + monitor_config->vbstmon_dpth = monitor_array[0]; + monitor_config->vbstmon_loc = monitor_array[1]; + monitor_config->vbstmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpbrstat", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpbrstat_specs = true; + monitor_config->vpbrstat_dpth = monitor_array[0]; + monitor_config->vpbrstat_loc = monitor_array[1]; + monitor_config->vpbrstat_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,zerofill", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->zerofill_specs = true; + monitor_config->zerofill_dpth = monitor_array[0]; + monitor_config->zerofill_loc = monitor_array[1]; + monitor_config->zerofill_frm = monitor_array[2]; + } + } + of_node_put(signal_format); + + return 0; +} + +/* Errata Rev A0 */ +static const struct reg_sequence cs35l35_errata_patch[] = { + + { 0x7F, 0x99 }, + { 0x00, 0x99 }, + { 0x52, 0x22 }, + { 0x04, 0x14 }, + { 0x6D, 0x44 }, + { 0x24, 0x10 }, + { 0x58, 0xC4 }, + { 0x00, 0x98 }, + { 0x18, 0x08 }, + { 0x00, 0x00 }, + { 0x7F, 0x00 }, +}; + +static int cs35l35_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l35_private *cs35l35; + struct cs35l35_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int i; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs35l35 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_private), + GFP_KERNEL); + if (!cs35l35) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l35); + cs35l35->regmap = devm_regmap_init_i2c(i2c_client, &cs35l35_regmap); + if (IS_ERR(cs35l35->regmap)) { + ret = PTR_ERR(cs35l35->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) + cs35l35->supplies[i].supply = cs35l35_supplies[i]; + cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l35->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + if (i2c_client->dev.of_node) { + ret = cs35l35_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l35->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + return ret; + } + + /* returning NULL can be valid if in stereo mode */ + cs35l35->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l35->reset_gpio)) { + ret = PTR_ERR(cs35l35->reset_gpio); + if (ret == -EBUSY) { + dev_info(&i2c_client->dev, + "Reset line busy, assuming shared reset\n"); + cs35l35->reset_gpio = NULL; + } else { + dev_err(&i2c_client->dev, + "Failed to get reset GPIO: %d\n", ret); + goto err; + } + } + + gpiod_set_value_cansleep(cs35l35->reset_gpio, 1); + + init_completion(&cs35l35->pdn_done); + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l35_irq, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l35", cs35l35); + if (ret != 0) { + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L35_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L35 Device ID (%X). Expected ID %X\n", + devid, CS35L35_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l35->regmap, CS35L35_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed: %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l35->regmap, cs35l35_errata_patch, + ARRAY_SIZE(cs35l35_errata_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch: %d\n", + ret); + goto err; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L35 (%x), Revision: %02X\n", devid, + ret & 0xFF); + + /* Set the INT Masks for critical errors */ + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_1, + CS35L35_INT1_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_2, + CS35L35_INT2_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_3, + CS35L35_INT3_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_4, + CS35L35_INT4_CRIT_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PWR2_PDN_MASK, + CS35L35_PWR2_PDN_MASK); + + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL3, + CS35L35_PWR3_PDN_MASK, + CS35L35_PWR3_PDN_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 1 << CS35L35_AMP_MUTE_SHIFT); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l35, cs35l35_dai, + ARRAY_SIZE(cs35l35_dai)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Failed to register codec: %d\n", ret); + goto err; + } + +err: + regulator_bulk_disable(cs35l35->num_supplies, + cs35l35->supplies); + gpiod_set_value_cansleep(cs35l35->reset_gpio, 0); + + return ret; +} + +static int cs35l35_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id cs35l35_of_match[] = { + {.compatible = "cirrus,cs35l35"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l35_of_match); + +static const struct i2c_device_id cs35l35_id[] = { + {"cs35l35", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l35_id); + +static struct i2c_driver cs35l35_i2c_driver = { + .driver = { + .name = "cs35l35", + .of_match_table = cs35l35_of_match, + }, + .id_table = cs35l35_id, + .probe = cs35l35_i2c_probe, + .remove = cs35l35_i2c_remove, +}; + +module_i2c_driver(cs35l35_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L35 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h new file mode 100644 index 0000000..e27a7fe --- /dev/null +++ b/sound/soc/codecs/cs35l35.h @@ -0,0 +1,284 @@ +/* + * cs35l35.h -- CS35L35 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Brian Austin <brian.austin(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 __CS35L35_H__ +#define __CS35L35_H__ + +#define CS35L35_FIRSTREG 0x01 +#define CS35L35_LASTREG 0x7E +#define CS35L35_CHIP_ID 0x00035A35 +#define CS35L35_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L35_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L35_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L35_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L35_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L35_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L35_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L35_PWRCTL3 0x08 /* Power Ctl 3 */ +#define CS35L35_CLK_CTL1 0x0A /* Clocking Ctl 1 */ +#define CS35L35_CLK_CTL2 0x0B /* Clocking Ctl 2 */ +#define CS35L35_CLK_CTL3 0x0C /* Clocking Ctl 3 */ +#define CS35L35_SP_FMT_CTL1 0x0D /* Serial Port Format CTL1 */ +#define CS35L35_SP_FMT_CTL2 0x0E /* Serial Port Format CTL2 */ +#define CS35L35_SP_FMT_CTL3 0x0F /* Serial Port Format CTL3 */ +#define CS35L35_MAG_COMP_CTL 0x13 /* Magnitude Comp CTL */ +#define CS35L35_AMP_INP_DRV_CTL 0x14 /* Amp Input Drive Ctl */ +#define CS35L35_AMP_DIG_VOL_CTL 0x15 /* Amplifier Dig Volume Ctl */ +#define CS35L35_AMP_DIG_VOL 0x16 /* Amplifier Dig Volume */ +#define CS35L35_ADV_DIG_VOL 0x17 /* Advisory Digital Volume */ +#define CS35L35_PROTECT_CTL 0x18 /* Amp Gain - Prot Ctl Param */ +#define CS35L35_AMP_GAIN_AUD_CTL 0x19 /* Amp Serial Port Gain Ctl */ +#define CS35L35_AMP_GAIN_PDM_CTL 0x1A /* Amplifier Gain PDM Ctl */ +#define CS35L35_AMP_GAIN_ADV_CTL 0x1B /* Amplifier Gain Ctl */ +#define CS35L35_GPI_CTL 0x1C /* GPI Ctl */ +#define CS35L35_BST_CVTR_V_CTL 0x1D /* Boost Conv Voltage Ctl */ +#define CS35L35_BST_PEAK_I 0x1E /* Boost Conv Peak Current */ +#define CS35L35_BST_RAMP_CTL 0x20 /* Boost Conv Soft Ramp Ctl */ +#define CS35L35_BST_CONV_COEF_1 0x21 /* Boost Conv Coefficients 1 */ +#define CS35L35_BST_CONV_COEF_2 0x22 /* Boost Conv Coefficients 2 */ +#define CS35L35_BST_CONV_SLOPE_COMP 0x23 /* Boost Conv Slope Comp */ +#define CS35L35_BST_CONV_SW_FREQ 0x24 /* Boost Conv L BST SW Freq */ +#define CS35L35_CLASS_H_CTL 0x30 /* CLS H Control */ +#define CS35L35_CLASS_H_HEADRM_CTL 0x31 /* CLS H Headroom Ctl */ +#define CS35L35_CLASS_H_RELEASE_RATE 0x32 /* CLS H Release Rate */ +#define CS35L35_CLASS_H_FET_DRIVE_CTL 0x33 /* CLS H Weak FET Drive Ctl */ +#define CS35L35_CLASS_H_VP_CTL 0x34 /* CLS H VP Ctl */ +#define CS35L35_CLASS_H_STATUS 0x38 /* CLS H Status */ +#define CS35L35_VPBR_CTL 0x3A /* VPBR Ctl */ +#define CS35L35_VPBR_VOL_CTL 0x3B /* VPBR Volume Ctl */ +#define CS35L35_VPBR_TIMING_CTL 0x3C /* VPBR Timing Ctl */ +#define CS35L35_VPBR_MODE_VOL_CTL 0x3D /* VPBR Mode/Attack Vol Ctl */ +#define CS35L35_VPBR_ATTEN_STATUS 0x4B /* VPBR Attenuation Status */ +#define CS35L35_SPKR_MON_CTL 0x4E /* Speaker Monitoring Ctl */ +#define CS35L35_IMON_SCALE_CTL 0x51 /* IMON Scale Ctl */ +#define CS35L35_AUDIN_RXLOC_CTL 0x52 /* Audio Input RX Loc Ctl */ +#define CS35L35_ADVIN_RXLOC_CTL 0x53 /* Advisory Input RX Loc Ctl */ +#define CS35L35_VMON_TXLOC_CTL 0x54 /* VMON TX Loc Ctl */ +#define CS35L35_IMON_TXLOC_CTL 0x55 /* IMON TX Loc Ctl */ +#define CS35L35_VPMON_TXLOC_CTL 0x56 /* VPMON TX Loc Ctl */ +#define CS35L35_VBSTMON_TXLOC_CTL 0x57 /* VBSTMON TX Loc Ctl */ +#define CS35L35_VPBR_STATUS_TXLOC_CTL 0x58 /* VPBR Status TX Loc Ctl */ +#define CS35L35_ZERO_FILL_LOC_CTL 0x59 /* Zero Fill Loc Ctl */ +#define CS35L35_AUDIN_DEPTH_CTL 0x5A /* Audio Input Depth Ctl */ +#define CS35L35_SPKMON_DEPTH_CTL 0x5B /* SPK Mon Output Depth Ctl */ +#define CS35L35_SUPMON_DEPTH_CTL 0x5C /* Supply Mon Out Depth Ctl */ +#define CS35L35_ZEROFILL_DEPTH_CTL 0x5D /* Zero Fill Mon Output Ctl */ +#define CS35L35_MULT_DEV_SYNCH1 0x62 /* Multidevice Synch */ +#define CS35L35_MULT_DEV_SYNCH2 0x63 /* Multidevice Synch 2 */ +#define CS35L35_PROT_RELEASE_CTL 0x64 /* Protection Release Ctl */ +#define CS35L35_DIAG_MODE_REG_LOCK 0x68 /* Diagnostic Mode Reg Lock */ +#define CS35L35_DIAG_MODE_CTL_1 0x69 /* Diagnostic Mode Ctl 1 */ +#define CS35L35_DIAG_MODE_CTL_2 0x6A /* Diagnostic Mode Ctl 2 */ +#define CS35L35_INT_MASK_1 0x70 /* Interrupt Mask 1 */ +#define CS35L35_INT_MASK_2 0x71 /* Interrupt Mask 2 */ +#define CS35L35_INT_MASK_3 0x72 /* Interrupt Mask 3 */ +#define CS35L35_INT_MASK_4 0x73 /* Interrupt Mask 4 */ +#define CS35L35_INT_STATUS_1 0x74 /* Interrupt Status 1 */ +#define CS35L35_INT_STATUS_2 0x75 /* Interrupt Status 2 */ +#define CS35L35_INT_STATUS_3 0x76 /* Interrupt Status 3 */ +#define CS35L35_INT_STATUS_4 0x77 /* Interrupt Status 4 */ +#define CS35L35_PLL_STATUS 0x78 /* PLL Status */ +#define CS35L35_OTP_TRIM_STATUS 0x7E /* OTP Trim Status */ + +#define CS35L35_MAX_REGISTER 0x7F + +/* CS35L35_PWRCTL1 */ +#define CS35L35_SFT_RST 0x80 +#define CS35L35_DISCHG_FLT 0x02 +#define CS35L35_PDN_ALL 0x01 + +/* CS35L35_PWRCTL2 */ +#define CS35L35_PDN_VMON 0x80 +#define CS35L35_PDN_IMON 0x40 +#define CS35L35_PDN_CLASSH 0x20 +#define CS35L35_PDN_VPBR 0x10 +#define CS35L35_PDN_BST 0x04 +#define CS35L35_PDN_AMP 0x01 + +/* CS35L35_PWRCTL3 */ +#define CS35L35_PDN_VBSTMON_OUT 0x10 +#define CS35L35_PDN_VMON_OUT 0x08 + +#define CS35L35_AUDIN_DEPTH_MASK 0x03 +#define CS35L35_AUDIN_DEPTH_SHIFT 0 +#define CS35L35_ADVIN_DEPTH_MASK 0x0C +#define CS35L35_ADVIN_DEPTH_SHIFT 2 +#define CS35L35_SDIN_DEPTH_8 0x01 +#define CS35L35_SDIN_DEPTH_16 0x02 +#define CS35L35_SDIN_DEPTH_24 0x03 + +#define CS35L35_SDOUT_DEPTH_8 0x01 +#define CS35L35_SDOUT_DEPTH_12 0x02 +#define CS35L35_SDOUT_DEPTH_16 0x03 + +#define CS35L35_AUD_IN_LR_MASK 0x80 +#define CS35L35_AUD_IN_LR_SHIFT 7 +#define CS35L35_ADV_IN_LR_MASK 0x80 +#define CS35L35_ADV_IN_LR_SHIFT 7 +#define CS35L35_AUD_IN_LOC_MASK 0x0F +#define CS35L35_AUD_IN_LOC_SHIFT 0 +#define CS35L35_ADV_IN_LOC_MASK 0x0F +#define CS35L35_ADV_IN_LOC_SHIFT 0 + +#define CS35L35_IMON_DEPTH_MASK 0x03 +#define CS35L35_IMON_DEPTH_SHIFT 0 +#define CS35L35_VMON_DEPTH_MASK 0x0C +#define CS35L35_VMON_DEPTH_SHIFT 2 +#define CS35L35_VBSTMON_DEPTH_MASK 0x03 +#define CS35L35_VBSTMON_DEPTH_SHIFT 0 +#define CS35L35_VPMON_DEPTH_MASK 0x0C +#define CS35L35_VPMON_DEPTH_SHIFT 2 +#define CS35L35_VPBRSTAT_DEPTH_MASK 0x30 +#define CS35L35_VPBRSTAT_DEPTH_SHIFT 4 +#define CS35L35_ZEROFILL_DEPTH_MASK 0x03 +#define CS35L35_ZEROFILL_DEPTH_SHIFT 0x00 + +#define CS35L35_MON_TXLOC_MASK 0x3F +#define CS35L35_MON_TXLOC_SHIFT 0 +#define CS35L35_MON_FRM_MASK 0x80 +#define CS35L35_MON_FRM_SHIFT 7 + +#define CS35L35_MS_MASK 0x80 +#define CS35L35_MS_SHIFT 7 +#define CS35L35_SPMODE_MASK 0x40 +#define CS35L35_SP_DRV_MASK 0x10 +#define CS35L35_SP_DRV_SHIFT 4 +#define CS35L35_CLK_CTL2_MASK 0xFF +#define CS35L35_PDM_MODE_MASK 0x40 +#define CS35L35_PDM_MODE_SHIFT 6 +#define CS35L35_CLK_SOURCE_MASK 0x03 +#define CS35L35_CLK_SOURCE_SHIFT 0 +#define CS35L35_CLK_SOURCE_MCLK 0 +#define CS35L35_CLK_SOURCE_SCLK 1 +#define CS35L35_CLK_SOURCE_PDM 2 + +#define CS35L35_SP_SCLKS_MASK 0x0F +#define CS35L35_SP_SCLKS_SHIFT 0x00 +#define CS35L35_SP_SCLKS_16FS 0x03 +#define CS35L35_SP_SCLKS_32FS 0x07 +#define CS35L35_SP_SCLKS_48FS 0x0B +#define CS35L35_SP_SCLKS_64FS 0x0F +#define CS35L35_SP_RATE_MASK 0xC0 + +#define CS35L35_PDN_BST_MASK 0x06 +#define CS35L35_PDN_BST_FETON_SHIFT 1 +#define CS35L35_PDN_BST_FETOFF_SHIFT 2 +#define CS35L35_PWR2_PDN_MASK 0xE0 +#define CS35L35_PWR3_PDN_MASK 0x1E +#define CS35L35_PDN_ALL_MASK 0x01 +#define CS35L35_DISCHG_FILT_MASK 0x02 +#define CS35L35_DISCHG_FILT_SHIFT 1 +#define CS35L35_MCLK_DIS_MASK 0x04 +#define CS35L35_MCLK_DIS_SHIFT 2 + +#define CS35L35_BST_CTL_MASK 0x7F +#define CS35L35_BST_CTL_SHIFT 0 +#define CS35L35_BST_IPK_MASK 0x1F +#define CS35L35_BST_IPK_SHIFT 0 +#define CS35L35_AMP_MUTE_MASK 0x20 +#define CS35L35_AMP_MUTE_SHIFT 5 +#define CS35L35_AMP_GAIN_ZC_MASK 0x10 +#define CS35L35_AMP_GAIN_ZC_SHIFT 4 + +/* Class H Algorithm Control */ +#define CS35L35_CH_STEREO_MASK 0x40 +#define CS35L35_CH_STEREO_SHIFT 6 +#define CS35L35_CH_BST_OVR_MASK 0x04 +#define CS35L35_CH_BST_OVR_SHIFT 2 +#define CS35L35_CH_BST_LIM_MASK 0x08 +#define CS35L35_CH_BST_LIM_SHIFT 3 +#define CS35L35_CH_MEM_DEPTH_MASK 0x01 +#define CS35L35_CH_MEM_DEPTH_SHIFT 0 +#define CS35L35_CH_HDRM_CTL_MASK 0x3F +#define CS35L35_CH_HDRM_CTL_SHIFT 0 +#define CS35L35_CH_REL_RATE_MASK 0xFF +#define CS35L35_CH_REL_RATE_SHIFT 0 +#define CS35L35_CH_WKFET_DIS_MASK 0x80 +#define CS35L35_CH_WKFET_DIS_SHIFT 7 +#define CS35L35_CH_WKFET_DEL_MASK 0x70 +#define CS35L35_CH_WKFET_DEL_SHIFT 4 +#define CS35L35_CH_WKFET_THLD_MASK 0x0F +#define CS35L35_CH_WKFET_THLD_SHIFT 0 +#define CS35L35_CH_VP_AUTO_MASK 0x80 +#define CS35L35_CH_VP_AUTO_SHIFT 7 +#define CS35L35_CH_VP_RATE_MASK 0x60 +#define CS35L35_CH_VP_RATE_SHIFT 5 +#define CS35L35_CH_VP_MAN_MASK 0x1F +#define CS35L35_CH_VP_MAN_SHIFT 0 + +/* CS35L35_PROT_RELEASE_CTL */ +#define CS35L35_CAL_ERR_RLS 0x80 +#define CS35L35_SHORT_RLS 0x04 +#define CS35L35_OTW_RLS 0x02 +#define CS35L35_OTE_RLS 0x01 + +/* INT Mask Registers */ +#define CS35L35_INT1_CRIT_MASK 0x38 +#define CS35L35_INT2_CRIT_MASK 0xEF +#define CS35L35_INT3_CRIT_MASK 0xEE +#define CS35L35_INT4_CRIT_MASK 0xFF + +/* PDN DONE Masks */ +#define CS35L35_M_PDN_DONE_SHIFT 4 +#define CS35L35_M_PDN_DONE_MASK 0x10 + +/* CS35L35_INT_1 */ +#define CS35L35_CAL_ERR 0x80 +#define CS35L35_OTP_ERR 0x40 +#define CS35L35_LRCLK_ERR 0x20 +#define CS35L35_SPCLK_ERR 0x10 +#define CS35L35_MCLK_ERR 0x08 +#define CS35L35_AMP_SHORT 0x04 +#define CS35L35_OTW 0x02 +#define CS35L35_OTE 0x01 + +/* CS35L35_INT_2 */ +#define CS35L35_PDN_DONE 0x10 +#define CS35L35_VPBR_ERR 0x02 +#define CS35L35_VPBR_CLR 0x01 + +/* CS35L35_INT_3 */ +#define CS35L35_BST_HIGH 0x10 +#define CS35L35_BST_HIGH_FLAG 0x08 +#define CS35L35_BST_IPK_FLAG 0x04 +#define CS35L35_LBST_SHORT 0x01 + +/* CS35L35_INT_4 */ +#define CS35L35_VMON_OVFL 0x08 +#define CS35L35_IMON_OVFL 0x04 + +#define CS35L35_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct cs35l35_private { + struct snd_soc_codec *codec; + struct cs35l35_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int sysclk; + int sclk; + bool pdm_mode; + bool i2s_mode; + bool slave_mode; + /* GPIO for /RST */ + struct gpio_desc *reset_gpio; + struct completion pdn_done; +}; + +static const char * const cs35l35_supplies[] = { + "VA", + "VP", +}; + +#endif -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: rcar: avoid SSI_MODEx settings for SSI8
by Kuninori Morimoto 06 Mar '17

06 Mar '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSI8 is is sharing pin with SSI7, and nothing to do for SSI_MODEx. It is special pin and it needs special settings whole system, but we can't confirm it, because we never have SSI8 available board. This patch fixup SSI_MODEx settings error for SSI8 on connection test, but should be confirmed behavior on real board in the future. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/ssiu.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 54b9323..13d648e 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -64,7 +64,11 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, mask1 = (1 << 4) | (1 << 20); /* mask sync bit */ mask2 = (1 << 4); /* mask sync bit */ val1 = val2 = 0; - if (rsnd_ssi_is_pin_sharing(io)) { + if (id == 8) { + /* + * SSI8 pin is sharing with SSI7, nothing to do. + */ + } else if (rsnd_ssi_is_pin_sharing(io)) { int shift = -1; switch (id) { -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: es828: various improvements
by Romain Perier 06 Mar '17

06 Mar '17

This set of patches fixes clock handling in slave mode, enables sampling frequency of 192khz in slave mode and adds support for this frame rate in the driver rk3288_hdmi_analog.c. Romain Perier (4): ASoC: es8328: Simplify rates definition ASoC: es8328: Let device auto detect ratios in slave mode ASoC: es8328: Enabling support for 192k ASoC: rockchip: Enable 192khz in hw_params operation sound/soc/codecs/es8328.c | 51 +++++++++++++++++++-------------- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 ++ 2 files changed, 32 insertions(+), 22 deletions(-) -- 2.9.3

2 9

[alsa-devel] Applied "ASoC: TLV320AIC23: Unquote NULL from control name" to the asoc tree
by Mark Brown 06 Mar '17

06 Mar '17

The patch ASoC: TLV320AIC23: Unquote NULL from control name has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a03faba972cb0f9b3a46d8054e674d5492e06c38 Mon Sep 17 00:00:00 2001 From: Liviu Dudau <liviu(a)dudau.co.uk> Date: Wed, 1 Mar 2017 12:26:28 +0000 Subject: [PATCH] ASoC: TLV320AIC23: Unquote NULL from control name Without this I am getting the following messages at boot on my Trimslice: tlv320aic23-codec 2-001a: Control not supported for path LLINEIN -> [NULL] -> Line Input tlv320aic23-codec 2-001a: ASoC: no dapm match for LLINEIN --> NULL --> Line Input tlv320aic23-codec 2-001a: ASoC: Failed to add route LLINEIN -> NULL -> Line Input tlv320aic23-codec 2-001a: Control not supported for path RLINEIN -> [NULL] -> Line Input tlv320aic23-codec 2-001a: ASoC: no dapm match for RLINEIN --> NULL --> Line Input tlv320aic23-codec 2-001a: ASoC: Failed to add route RLINEIN -> NULL -> Line Input tlv320aic23-codec 2-001a: Control not supported for path MICIN -> [NULL] -> Mic Input tlv320aic23-codec 2-001a: ASoC: no dapm match for MICIN --> NULL --> Mic Input tlv320aic23-codec 2-001a: ASoC: Failed to add route MICIN -> NULL -> Mic Input tegra-snd-trimslice sound: tlv320aic23-hifi <-> 70002800.i2s mapping ok Signed-off-by: Liviu Dudau <liviu(a)dudau.co.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic23.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 410cae0f2060..628a8eeaab68 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -174,10 +174,9 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { {"ROUT", NULL, "Output Mixer"}, /* Inputs */ - {"Line Input", "NULL", "LLINEIN"}, - {"Line Input", "NULL", "RLINEIN"}, - - {"Mic Input", "NULL", "MICIN"}, + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, + {"Mic Input", NULL, "MICIN"}, /* input mux */ {"Capture Source", "Line", "Line Input"}, -- 2.11.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: hdac_hdmi: avoid reference to invalid variable of the pin list
by jeeja.kp＠intel.com 06 Mar '17

06 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> Using pin list array iterator outside the iteration of the list can point to dummy element, which can be invalid. So don't use pin variable outside the pin list iteration. This fixes the following coccinelle warning: sound/soc/codecs/hdac_hdmi.c:1419:5-8: ERROR: invalid reference to the index variable of the iterator Fixes: 2acd8309a3a4('ASoC: hdac_hdmi: Add support to handle MST capable pin') Reported-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 78fca8a..bb40569 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1534,21 +1534,20 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) pin->mst_capable = false; /* if not MST, default is port[0] */ hport = &pin->ports[0]; - goto out; } else { for (i = 0; i < pin->num_ports; i++) { pin->mst_capable = true; if (pin->ports[i].id == pipe) { hport = &pin->ports[i]; - goto out; + break; } } } + + if (hport) + hdac_hdmi_present_sense(pin, hport); } -out: - if (pin && hport) - hdac_hdmi_present_sense(pin, hport); } static struct i915_audio_component_audio_ops aops = { -- 2.5.0

3 4

[alsa-devel] [PATCH] ASoC: ux500: drop platform DAI assignments
by Linus Walleij 06 Mar '17

06 Mar '17

This platform is completely probed by device tree nowadays, so we need to do a bigger cleanup removing all the non-DT codepaths. This cleanup must however go in as a fix since it fixes a regression. Currently when Ux500 audio is enabled, dmesg complains like this: entry->name == "prealloc" entry->name == "prealloc_max" entry->name == "prealloc" ------------[ cut here ]------------ WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 proc_dir_entry 'sub0/prealloc' already registered (...) entry->name == "prealloc_max" ------------[ cut here ]------------ WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 proc_dir_entry 'sub0/prealloc_max' already registered (...) snd-soc-mop500 soc:sound: ab8500-codec-dai.0 <-> 80124000.msp mapping ok entry->name == "prealloc" entry->name == "prealloc_max" entry->name == "prealloc" ------------[ cut here ]------------ WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 proc_dir_entry 'sub0/prealloc' already registered (...) entry->name == "prealloc_max" ------------[ cut here ]------------ WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 proc_dir_entry 'sub0/prealloc_max' already registered snd-soc-mop500 soc:sound: ab8500-codec-dai.1 <-> 80125000.msp mapping ok This is because PCMs are created twice for the same hardware, and this happens because both "platform" and "CPU" DAI links are specified. But platform/CPU is an either/or pair, not a both/and pair. This has maybe worked in the past, but it is causing trouble now, so let us begin the cleanups by removing the platform assignment and silencing the boot noise, and make a proper DT cleanup for the next kernel cycle. Cc: Ulf Hansson <ulf.hansson(a)linaro.org> Cc: Lee Jones <lee.jones(a)linaro.org> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/ux500/mop500.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index ba9fc099cf67..b50f68a439ce 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -33,7 +33,6 @@ static struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_0", .cpu_dai_name = "ux500-msp-i2s.1", .codec_dai_name = "ab8500-codec-dai.0", - .platform_name = "ux500-msp-i2s.1", .codec_name = "ab8500-codec.0", .init = mop500_ab8500_machine_init, .ops = mop500_ab8500_ops, @@ -43,7 +42,6 @@ static struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_1", .cpu_dai_name = "ux500-msp-i2s.3", .codec_dai_name = "ab8500-codec-dai.1", - .platform_name = "ux500-msp-i2s.3", .codec_name = "ab8500-codec.0", .init = NULL, .ops = mop500_ab8500_ops, @@ -87,8 +85,6 @@ static int mop500_of_probe(struct platform_device *pdev, for (i = 0; i < 2; i++) { mop500_dai_links[i].cpu_of_node = msp_np[i]; mop500_dai_links[i].cpu_dai_name = NULL; - mop500_dai_links[i].platform_of_node = msp_np[i]; - mop500_dai_links[i].platform_name = NULL; mop500_dai_links[i].codec_of_node = codec_np; mop500_dai_links[i].codec_name = NULL; } -- 2.9.3

2 1

[alsa-devel] [PATCH 0/2] Add es7134 DAC driver support
by Jerome Brunet 06 Mar '17

06 Mar '17

This patchset adds the support for the es7134 from Everest Semiconductor. The es7134 is simple i2s DAC with no configuration interface. It has been tested on Amlogic's meson-gxbb-p200 board. Jerome Brunet (2): ASoC: es7134: add es7134 DAC driver ASoC: es7134: add dt-bindings for the es7134 dac .../devicetree/bindings/sound/everest,es7134.txt | 10 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7134.c | 116 +++++++++++++++++++++ 4 files changed, 132 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7134.txt create mode 100644 sound/soc/codecs/es7134.c -- 2.9.3

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Re: [alsa-devel] [PATCH] ALSA: ASoC: TLV320AIC23: Unquote NULL from control name
by Mark Brown 06 Mar '17

06 Mar '17

On Wed, Mar 01, 2017 at 12:26:28PM +0000, Liviu Dudau wrote: > Without this I am getting the following messages at boot on my Trimslice: > tlv320aic23-codec 2-001a: Control not supported for path LLINEIN -> [NULL] -> Line Input Please use subject lines matching the style for the subsystem. This makes it easier for people to identify relevant patches.

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Re: [alsa-devel] [PATCH 01/10] ALSA SoC MAX98927 driver - Initial release
by Mark Brown 06 Mar '17

06 Mar '17

On Fri, Mar 03, 2017 at 11:52:39PM +0900, Ryan Lee wrote: > From: Ryan Lee <RyanS.Lee(a)maximintegrated.com> > > Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> > --- > > Initial release of ALSA SoC MAX98927 driver. The rest of this patch series appears to be bug fixes to this driver. Please don't submit things like this, if there are problems with code you're trying to add then squash those fixes into the initial patch. Otherwise you're making it harder for people to review the series by sending code with known problems in it.

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Re: [alsa-devel] [PATCH 02/10] Updated max98927_reg table with physical defaults. Replaced max98927.h for better legibility
by Mark Brown 06 Mar '17

06 Mar '17

On Fri, Mar 03, 2017 at 11:52:40PM +0900, Ryan Lee wrote: > Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> > --- > Some register values in 'max98927_reg_map' table was not physical default. > Random mix of strangely formatted #defines and numbers was fixed. > Reclassified register definition in 'max98927.h' for better legibility. Unused definition has been removed, too. Please place the changelog before the ---, as covered in SubmttingPatches the changelog after the --- is for inter-version changelogs and other administrative things that shouldn't end up in the git history.

1 0

[alsa-devel] [v5 1/2] ASoC: Add support for Cirrus Logic CS35L35 Amplifier
by Brian Austin 06 Mar '17

06 Mar '17

This patch adds support for the Cirrus Logic CS35L35 9V Boosted Amplifier Changes for v5 Update bindings documentation and parameters to be inline with previous Cirrus AMP's Shared /RST config for mulit-device style changes Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs35l35.h | 103 +++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l35.c | 1552 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l35.h | 284 ++++++++ 5 files changed, 1946 insertions(+) create mode 100644 include/sound/cs35l35.h create mode 100644 sound/soc/codecs/cs35l35.c create mode 100644 sound/soc/codecs/cs35l35.h diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h new file mode 100644 index 0000000..de92e45 --- /dev/null +++ b/include/sound/cs35l35.h @@ -0,0 +1,103 @@ +/* + * linux/sound/cs35l35.h -- Platform data for CS35l35 + * + * 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 __CS35L35_H +#define __CS35L35_H + +struct classh_cfg { + /* + * Class H Algorithm Control Variables + * You can either have it done + * automatically or you can adjust + * these variables for tuning + * + * if you do not enable the internal algorithm + * you will get a set of mixer controls for + * Class H tuning + * + * Section 4.3 of the datasheet + */ + bool classh_bst_override; + bool classh_algo_enable; + int classh_bst_max_limit; + int classh_mem_depth; + int classh_release_rate; + int classh_headroom; + int classh_wk_fet_disable; + int classh_wk_fet_delay; + int classh_wk_fet_thld; + int classh_vpch_auto; + int classh_vpch_rate; + int classh_vpch_man; +}; + +struct monitor_cfg { + /* + * Signal Monitor Data + * highly configurable signal monitoring + * data positioning and different types of + * monitoring data. + * + * Section 4.8.2 - 4.8.4 of the datasheet + */ + bool is_present; + bool imon_specs; + bool vmon_specs; + bool vpmon_specs; + bool vbstmon_specs; + bool vpbrstat_specs; + bool zerofill_specs; + u8 imon_dpth; + u8 imon_loc; + u8 imon_frm; + u8 vmon_dpth; + u8 vmon_loc; + u8 vmon_frm; + u8 vpmon_dpth; + u8 vpmon_loc; + u8 vpmon_frm; + u8 vbstmon_dpth; + u8 vbstmon_loc; + u8 vbstmon_frm; + u8 vpbrstat_dpth; + u8 vpbrstat_loc; + u8 vpbrstat_frm; + u8 zerofill_dpth; + u8 zerofill_loc; + u8 zerofill_frm; +}; + +struct cs35l35_platform_data { + + /* Stereo (2 Device) */ + bool stereo; + /* serial port drive strength */ + int sp_drv_str; + /* Boost Power Down with FET */ + bool bst_pdn_fet_on; + /* Boost Voltage : used if ClassH Algo Enabled */ + int bst_vctl; + /* Boost Converter Peak Current CTRL */ + int bst_ipk; + /* Amp Gain Zero Cross */ + bool gain_zc; + /* Audio Input Location */ + int aud_channel; + /* Advisory Input Location */ + int adv_channel; + /* Shared Boost for stereo */ + bool shared_bst; + /* ClassH Algorithm */ + struct classh_cfg classh_algo; + /* Monitor Config */ + struct monitor_cfg mon_cfg; +}; + +#endif /* __CS35L35_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e49e9da..00885c2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C + select SND_SOC_CS35L35 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -408,6 +409,10 @@ config SND_SOC_CS35L34 tristate "Cirrus Logic CS35L34 CODEC" depends on I2C +config SND_SOC_CS35L35 + tristate "Cirrus Logic CS35L35 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1796cb9..97b36e4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o +snd-soc-cs35l35-objs := cs35l35.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -269,6 +270,7 @@ 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_CS35L34) += snd-soc-cs35l34.o +obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c new file mode 100644 index 0000000..2204abc --- /dev/null +++ b/sound/soc/codecs/cs35l35.c @@ -0,0 +1,1552 @@ +/* + * cs35l35.c -- CS35L35 ALSA SoC audio driver + * + * Copyright 2017 Cirrus Logic, Inc. + * + * Author: Brian Austin <brian.austin(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/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l35.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l35.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L35_VALID_PDATA 0x80000000 + +static const struct reg_default cs35l35_reg[] = { + {CS35L35_PWRCTL1, 0x01}, + {CS35L35_PWRCTL2, 0x11}, + {CS35L35_PWRCTL3, 0x00}, + {CS35L35_CLK_CTL1, 0x04}, + {CS35L35_CLK_CTL2, 0x10}, + {CS35L35_CLK_CTL3, 0xCF}, + {CS35L35_SP_FMT_CTL1, 0x20}, + {CS35L35_SP_FMT_CTL2, 0x00}, + {CS35L35_SP_FMT_CTL3, 0x02}, + {CS35L35_MAG_COMP_CTL, 0x00}, + {CS35L35_AMP_INP_DRV_CTL, 0x01}, + {CS35L35_AMP_DIG_VOL_CTL, 0x12}, + {CS35L35_AMP_DIG_VOL, 0x00}, + {CS35L35_ADV_DIG_VOL, 0x00}, + {CS35L35_PROTECT_CTL, 0x06}, + {CS35L35_AMP_GAIN_AUD_CTL, 0x13}, + {CS35L35_AMP_GAIN_PDM_CTL, 0x00}, + {CS35L35_AMP_GAIN_ADV_CTL, 0x00}, + {CS35L35_GPI_CTL, 0x00}, + {CS35L35_BST_CVTR_V_CTL, 0x00}, + {CS35L35_BST_PEAK_I, 0x07}, + {CS35L35_BST_RAMP_CTL, 0x85}, + {CS35L35_BST_CONV_COEF_1, 0x24}, + {CS35L35_BST_CONV_COEF_2, 0x24}, + {CS35L35_BST_CONV_SLOPE_COMP, 0x47}, + {CS35L35_BST_CONV_SW_FREQ, 0x04}, + {CS35L35_CLASS_H_CTL, 0x0B}, + {CS35L35_CLASS_H_HEADRM_CTL, 0x0B}, + {CS35L35_CLASS_H_RELEASE_RATE, 0x08}, + {CS35L35_CLASS_H_FET_DRIVE_CTL, 0x41}, + {CS35L35_CLASS_H_VP_CTL, 0xC5}, + {CS35L35_VPBR_CTL, 0x0A}, + {CS35L35_VPBR_VOL_CTL, 0x09}, + {CS35L35_VPBR_TIMING_CTL, 0x6A}, + {CS35L35_VPBR_MODE_VOL_CTL, 0x40}, + {CS35L35_SPKR_MON_CTL, 0xC0}, + {CS35L35_IMON_SCALE_CTL, 0x30}, + {CS35L35_AUDIN_RXLOC_CTL, 0x00}, + {CS35L35_ADVIN_RXLOC_CTL, 0x80}, + {CS35L35_VMON_TXLOC_CTL, 0x00}, + {CS35L35_IMON_TXLOC_CTL, 0x80}, + {CS35L35_VPMON_TXLOC_CTL, 0x04}, + {CS35L35_VBSTMON_TXLOC_CTL, 0x84}, + {CS35L35_VPBR_STATUS_TXLOC_CTL, 0x04}, + {CS35L35_ZERO_FILL_LOC_CTL, 0x00}, + {CS35L35_AUDIN_DEPTH_CTL, 0x0F}, + {CS35L35_SPKMON_DEPTH_CTL, 0x0F}, + {CS35L35_SUPMON_DEPTH_CTL, 0x0F}, + {CS35L35_ZEROFILL_DEPTH_CTL, 0x00}, + {CS35L35_MULT_DEV_SYNCH1, 0x02}, + {CS35L35_MULT_DEV_SYNCH2, 0x80}, + {CS35L35_PROT_RELEASE_CTL, 0x00}, + {CS35L35_DIAG_MODE_REG_LOCK, 0x00}, + {CS35L35_DIAG_MODE_CTL_1, 0x40}, + {CS35L35_DIAG_MODE_CTL_2, 0x00}, + {CS35L35_INT_MASK_1, 0xFF}, + {CS35L35_INT_MASK_2, 0xFF}, + {CS35L35_INT_MASK_3, 0xFF}, + {CS35L35_INT_MASK_4, 0xFF}, + +}; + +static bool cs35l35_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_DEVID_AB ... CS35L35_PWRCTL3: + case CS35L35_CLK_CTL1 ... CS35L35_SP_FMT_CTL3: + case CS35L35_MAG_COMP_CTL ... CS35L35_AMP_GAIN_AUD_CTL: + case CS35L35_AMP_GAIN_PDM_CTL ... CS35L35_BST_PEAK_I: + case CS35L35_BST_RAMP_CTL ... CS35L35_BST_CONV_SW_FREQ: + case CS35L35_CLASS_H_CTL ... CS35L35_CLASS_H_VP_CTL: + case CS35L35_CLASS_H_STATUS: + case CS35L35_VPBR_CTL ... CS35L35_VPBR_MODE_VOL_CTL: + case CS35L35_VPBR_ATTEN_STATUS: + case CS35L35_SPKR_MON_CTL: + case CS35L35_IMON_SCALE_CTL ... CS35L35_ZEROFILL_DEPTH_CTL: + case CS35L35_MULT_DEV_SYNCH1 ... CS35L35_PROT_RELEASE_CTL: + case CS35L35_DIAG_MODE_REG_LOCK ... CS35L35_DIAG_MODE_CTL_2: + case CS35L35_INT_MASK_1 ... CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l35_precious_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L35_INT_STATUS_1: + case CS35L35_INT_STATUS_2: + case CS35L35_INT_STATUS_3: + case CS35L35_INT_STATUS_4: + case CS35L35_PLL_STATUS: + case CS35L35_OTP_TRIM_STATUS: + return true; + default: + return false; + } +} + +static int cs35l35_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 cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, + 0 << CS35L35_MCLK_DIS_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, + 0 << CS35L35_DISCHG_FILT_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_DISCHG_FILT_MASK, + 1 << CS35L35_DISCHG_FILT_SHIFT); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL_MASK, 1); + + reinit_completion(&cs35l35->pdn_done); + + ret = wait_for_completion_timeout(&cs35l35->pdn_done, + msecs_to_jiffies(100)); + if (ret == 0) { + dev_err(codec->dev, "TIMEOUT PDN_DONE did not complete in 100ms\n"); + ret = -ETIMEDOUT; + } + + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MCLK_DIS_MASK, + 1 << CS35L35_MCLK_DIS_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + ret = -EINVAL; + } + return ret; +} + +static int cs35l35_main_amp_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 cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + unsigned int reg[4]; + int i; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 0 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 0 << CS35L35_PDN_BST_FETOFF_SHIFT); + break; + case SND_SOC_DAPM_POST_PMU: + usleep_range(5000, 5100); + /* If in PDM mode we must use VP for Voltage control */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, + 0 << CS35L35_BST_CTL_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 0); + + for (i = 0; i < 2; i++) + regmap_bulk_read(cs35l35->regmap, CS35L35_INT_STATUS_1, + &reg, ARRAY_SIZE(reg)); + + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, + 1 << CS35L35_AMP_MUTE_SHIFT); + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + usleep_range(5000, 5100); + /* + * If PDM mode we should switch back to pdata value + * for Voltage control when we go down + */ + if (cs35l35->pdm_mode) + regmap_update_bits(cs35l35->regmap, + CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, + cs35l35->pdata.bst_vctl + << CS35L35_BST_CTL_SHIFT); + + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + } + return 0; +} + +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l35_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("AMP Audio Gain", CS35L35_AMP_GAIN_AUD_CTL, 0, 19, 0, + amp_gain_tlv), + SOC_SINGLE_TLV("AMP PDM Gain", CS35L35_AMP_GAIN_PDM_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_kcontrol_new cs35l35_adv_controls[] = { + SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, + 0, 0x34, 0xE4, dig_vol_tlv), + SOC_SINGLE_TLV("AMP Advisory Gain", CS35L35_AMP_GAIN_ADV_CTL, 0, 19, 0, + amp_gain_tlv), +}; + +static const struct snd_soc_dapm_widget cs35l35_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L35_PWRCTL3, 1, 1, + cs35l35_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_OUT("SDOUT", NULL, 0, CS35L35_PWRCTL3, 2, 1), + + SND_SOC_DAPM_OUTPUT("SPK"), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("ISENSE"), + SND_SOC_DAPM_INPUT("VSENSE"), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L35_PWRCTL2, 7, 1), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L35_PWRCTL2, 6, 1), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L35_PWRCTL3, 3, 1), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L35_PWRCTL3, 4, 1), + SND_SOC_DAPM_ADC("CLASS H", NULL, CS35L35_PWRCTL2, 5, 1), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L35_PWRCTL2, 0, 1, NULL, 0, + cs35l35_main_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l35_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "ISENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + {"SDOUT", NULL, "IMON ADC"}, + {"SDOUT", NULL, "VMON ADC"}, + {"SDOUT", NULL, "VBSTMON ADC"}, + {"SDOUT", NULL, "VPMON ADC"}, + {"AMP Capture", NULL, "SDOUT"}, + + {"SDIN", NULL, "AMP Playback"}, + {"CLASS H", NULL, "SDIN"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = false; + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); + cs35l35->slave_mode = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cs35l35->i2s_mode = true; + cs35l35->pdm_mode = false; + break; + case SND_SOC_DAIFMT_PDM: + cs35l35->pdm_mode = true; + cs35l35->i2s_mode = false; + break; + default: + return -EINVAL; + } + + return 0; +} + +struct cs35l35_sysclk_config { + int sysclk; + int srate; + u8 clk_cfg; +}; + +static struct cs35l35_sysclk_config cs35l35_clk_ctl[] = { + + /* SYSCLK, Sample Rate, Serial Port Cfg */ + {5644800, 44100, 0x00}, + {5644800, 88200, 0x40}, + {6144000, 48000, 0x10}, + {6144000, 96000, 0x50}, + {11289600, 44100, 0x01}, + {11289600, 88200, 0x41}, + {11289600, 176400, 0x81}, + {12000000, 44100, 0x03}, + {12000000, 48000, 0x13}, + {12000000, 88200, 0x43}, + {12000000, 96000, 0x53}, + {12000000, 176400, 0x83}, + {12000000, 192000, 0x93}, + {12288000, 48000, 0x11}, + {12288000, 96000, 0x51}, + {12288000, 192000, 0x91}, + {13000000, 44100, 0x07}, + {13000000, 48000, 0x17}, + {13000000, 88200, 0x47}, + {13000000, 96000, 0x57}, + {13000000, 176400, 0x87}, + {13000000, 192000, 0x97}, + {22579200, 44100, 0x02}, + {22579200, 88200, 0x42}, + {22579200, 176400, 0x82}, + {24000000, 44100, 0x0B}, + {24000000, 48000, 0x1B}, + {24000000, 88200, 0x4B}, + {24000000, 96000, 0x5B}, + {24000000, 176400, 0x8B}, + {24000000, 192000, 0x9B}, + {24576000, 48000, 0x12}, + {24576000, 96000, 0x52}, + {24576000, 192000, 0x92}, + {26000000, 44100, 0x0F}, + {26000000, 48000, 0x1F}, + {26000000, 88200, 0x4F}, + {26000000, 96000, 0x5F}, + {26000000, 176400, 0x8F}, + {26000000, 192000, 0x9F}, +}; + +static int cs35l35_get_clk_config(int sysclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l35_clk_ctl); i++) { + if (cs35l35_clk_ctl[i].sysclk == sysclk && + cs35l35_clk_ctl[i].srate == srate) + return cs35l35_clk_ctl[i].clk_cfg; + } + return -EINVAL; +} + +static int cs35l35_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 cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + int srate = params_rate(params); + int ret = 0; + u8 sp_sclks; + int audin_format; + int errata_chk; + + int clk_ctl = cs35l35_get_clk_config(cs35l35->sysclk, srate); + + if (clk_ctl < 0) { + dev_err(codec->dev, "Invalid CLK:Rate %d:%d\n", + cs35l35->sysclk, srate); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL2, + CS35L35_CLK_CTL2_MASK, clk_ctl); + if (ret != 0) { + dev_err(codec->dev, "Failed to set port config %d\n", ret); + return ret; + } + + /* + * Rev A0 Errata + * When configured for the weak-drive detection path (CH_WKFET_DIS = 0) + * the Class H algorithm does not enable weak-drive operation for + * nonzero values of CH_WKFET_DELAY if SP_RATE = 01 or 10 + */ + errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; + + if (classh->classh_wk_fet_disable == 0x00 && + (errata_chk == 0x01 || errata_chk == 0x03)) { + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_DEL_MASK, + 0 << CS35L35_CH_WKFET_DEL_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fet config %d\n", + ret); + return ret; + } + } + + /* + * You can pull more Monitor data from the SDOUT pin than going to SDIN + * Just make sure your SCLK is fast enough to fill the frame + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (params_width(params)) { + case 8: + audin_format = CS35L35_SDIN_DEPTH_8; + break; + case 16: + audin_format = CS35L35_SDIN_DEPTH_16; + break; + case 24: + audin_format = CS35L35_SDIN_DEPTH_24; + break; + default: + dev_err(codec->dev, "Unsupported Width %d\n", + params_width(params)); + return -EINVAL; + } + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, + CS35L35_AUDIN_DEPTH_MASK, + audin_format << + CS35L35_AUDIN_DEPTH_SHIFT); + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_DEPTH_CTL, + CS35L35_ADVIN_DEPTH_MASK, + audin_format << + CS35L35_ADVIN_DEPTH_SHIFT); + } + } + + if (cs35l35->i2s_mode) { + /* We have to take the SCLK to derive num sclks + * to configure the CLOCK_CTL3 register correctly + */ + if ((cs35l35->sclk / srate) % 4) { + dev_err(codec->dev, "Unsupported sclk/fs ratio %d:%d\n", + cs35l35->sclk, srate); + return -EINVAL; + } + sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; + + /* Only certain ratios are supported in I2S Slave Mode */ + if (cs35l35->slave_mode) { + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_48FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } else { + /* Only certain ratios supported in I2S MASTER Mode */ + switch (sp_sclks) { + case CS35L35_SP_SCLKS_32FS: + case CS35L35_SP_SCLKS_64FS: + break; + default: + dev_err(codec->dev, "ratio not supported\n"); + return -EINVAL; + }; + } + ret = regmap_update_bits(cs35l35->regmap, + CS35L35_CLK_CTL3, + CS35L35_SP_SCLKS_MASK, sp_sclks << + CS35L35_SP_SCLKS_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set fsclk %d\n", ret); + return ret; + } + } + + return ret; +} + +static const unsigned int cs35l35_src_rates[] = { + 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_constraints = { + .count = ARRAY_SIZE(cs35l35_src_rates), + .list = cs35l35_src_rates, +}; + +static int cs35l35_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + if (!substream->runtime) + return 0; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l35_constraints); + + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, + 0 << CS35L35_PDM_MODE_SHIFT); + + return 0; +} + +static const unsigned int cs35l35_pdm_rates[] = { + 44100, 48000, 88200, 96000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l35_pdm_constraints = { + .count = ARRAY_SIZE(cs35l35_pdm_rates), + .list = cs35l35_pdm_rates, +}; + +static int cs35l35_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + if (!substream->runtime) + return 0; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l35_pdm_constraints); + + regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, + CS35L35_PDM_MODE_MASK, + 1 << CS35L35_PDM_MODE_SHIFT); + + return 0; +} + +static int cs35l35_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + + /* Need the SCLK Frequency regardless of sysclk source for I2S */ + cs35l35->sclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops cs35l35_ops = { + .startup = cs35l35_pcm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_hw_params, + .set_sysclk = cs35l35_dai_set_sysclk, +}; + +static const struct snd_soc_dai_ops cs35l35_pdm_ops = { + .startup = cs35l35_pdm_startup, + .set_fmt = cs35l35_set_dai_fmt, + .hw_params = cs35l35_hw_params, +}; + +static struct snd_soc_dai_driver cs35l35_dai[] = { + { + .name = "cs35l35-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_ops, + .symmetric_rates = 1, + }, + { + .name = "cs35l35-pdm", + .id = 1, + .playback = { + .stream_name = "PDM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L35_FORMATS, + }, + .ops = &cs35l35_pdm_ops, + }, +}; + +static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + int clksrc; + int ret = 0; + + switch (clk_id) { + case 0: + clksrc = CS35L35_CLK_SOURCE_MCLK; + break; + case 1: + clksrc = CS35L35_CLK_SOURCE_SCLK; + break; + case 2: + clksrc = CS35L35_CLK_SOURCE_PDM; + break; + default: + dev_err(codec->dev, "Invalid CLK Source\n"); + return -EINVAL; + }; + + switch (freq) { + case 5644800: + case 6144000: + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 22579200: + case 24000000: + case 24576000: + case 26000000: + cs35l35->sysclk = freq; + break; + default: + dev_err(codec->dev, "Invalid CLK Frequency Input : %d\n", freq); + return -EINVAL; + } + + ret = regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_CLK_SOURCE_MASK, + clksrc << CS35L35_CLK_SOURCE_SHIFT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set sysclk %d\n", ret); + return ret; + } + + return ret; +} + +static int cs35l35_codec_probe(struct snd_soc_codec *codec) +{ + struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); + struct classh_cfg *classh = &cs35l35->pdata.classh_algo; + struct monitor_cfg *monitor_config = &cs35l35->pdata.mon_cfg; + int ret; + + cs35l35->codec = codec; + + /* Set Platform Data */ + if (cs35l35->pdata.bst_vctl) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_CVTR_V_CTL, + CS35L35_BST_CTL_MASK, + cs35l35->pdata.bst_vctl); + + if (cs35l35->pdata.bst_ipk) + regmap_update_bits(cs35l35->regmap, CS35L35_BST_PEAK_I, + CS35L35_BST_IPK_MASK, + cs35l35->pdata.bst_ipk << + CS35L35_BST_IPK_SHIFT); + + if (cs35l35->pdata.gain_zc) + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_GAIN_ZC_MASK, + cs35l35->pdata.gain_zc << + CS35L35_AMP_GAIN_ZC_SHIFT); + + if (cs35l35->pdata.aud_channel) + regmap_update_bits(cs35l35->regmap, + CS35L35_AUDIN_RXLOC_CTL, + CS35L35_AUD_IN_LR_MASK, + cs35l35->pdata.aud_channel << + CS35L35_AUD_IN_LR_SHIFT); + + if (cs35l35->pdata.stereo) { + regmap_update_bits(cs35l35->regmap, + CS35L35_ADVIN_RXLOC_CTL, + CS35L35_ADV_IN_LR_MASK, + cs35l35->pdata.adv_channel << + CS35L35_ADV_IN_LR_SHIFT); + if (cs35l35->pdata.shared_bst) + regmap_update_bits(cs35l35->regmap, CS35L35_CLASS_H_CTL, + CS35L35_CH_STEREO_MASK, + 1 << CS35L35_CH_STEREO_SHIFT); + ret = snd_soc_add_codec_controls(codec, cs35l35_adv_controls, + ARRAY_SIZE(cs35l35_adv_controls)); + if (ret) + return ret; + } + + if (cs35l35->pdata.sp_drv_str) + regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, + CS35L35_SP_DRV_MASK, + cs35l35->pdata.sp_drv_str << + CS35L35_SP_DRV_SHIFT); + + if (classh->classh_algo_enable) { + if (classh->classh_bst_override) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, + CS35L35_CH_BST_OVR_MASK, + classh->classh_bst_override << + CS35L35_CH_BST_OVR_SHIFT); + if (classh->classh_bst_max_limit) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, + CS35L35_CH_BST_LIM_MASK, + classh->classh_bst_max_limit << + CS35L35_CH_BST_LIM_SHIFT); + if (classh->classh_mem_depth) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_CTL, + CS35L35_CH_MEM_DEPTH_MASK, + classh->classh_mem_depth << + CS35L35_CH_MEM_DEPTH_SHIFT); + if (classh->classh_headroom) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_HEADRM_CTL, + CS35L35_CH_HDRM_CTL_MASK, + classh->classh_headroom << + CS35L35_CH_HDRM_CTL_SHIFT); + if (classh->classh_release_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_RELEASE_RATE, + CS35L35_CH_REL_RATE_MASK, + classh->classh_release_rate << + CS35L35_CH_REL_RATE_SHIFT); + if (classh->classh_wk_fet_disable) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_DIS_MASK, + classh->classh_wk_fet_disable << + CS35L35_CH_WKFET_DIS_SHIFT); + if (classh->classh_wk_fet_delay) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_DEL_MASK, + classh->classh_wk_fet_delay << + CS35L35_CH_WKFET_DEL_SHIFT); + if (classh->classh_wk_fet_thld) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_FET_DRIVE_CTL, + CS35L35_CH_WKFET_THLD_MASK, + classh->classh_wk_fet_thld << + CS35L35_CH_WKFET_THLD_SHIFT); + if (classh->classh_vpch_auto) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, + CS35L35_CH_VP_AUTO_MASK, + classh->classh_vpch_auto << + CS35L35_CH_VP_AUTO_SHIFT); + if (classh->classh_vpch_rate) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, + CS35L35_CH_VP_RATE_MASK, + classh->classh_vpch_rate << + CS35L35_CH_VP_RATE_SHIFT); + if (classh->classh_vpch_man) + regmap_update_bits(cs35l35->regmap, + CS35L35_CLASS_H_VP_CTL, + CS35L35_CH_VP_MAN_MASK, + classh->classh_vpch_man << + CS35L35_CH_VP_MAN_SHIFT); + } + + if (monitor_config->is_present) { + if (monitor_config->vmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, + CS35L35_VMON_DEPTH_MASK, + monitor_config->vmon_dpth << + CS35L35_VMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vmon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vmon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->imon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SPKMON_DEPTH_CTL, + CS35L35_IMON_DEPTH_MASK, + monitor_config->imon_dpth << + CS35L35_IMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->imon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_IMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->imon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_VPMON_DEPTH_MASK, + monitor_config->vpmon_dpth << + CS35L35_VPMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vpmon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vpmon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vbstmon_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_VBSTMON_DEPTH_MASK, + monitor_config->vpmon_dpth << + CS35L35_VBSTMON_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vbstmon_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VBSTMON_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vbstmon_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->vpbrstat_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_VPBRSTAT_DEPTH_MASK, + monitor_config->vpbrstat_dpth << + CS35L35_VPBRSTAT_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->vpbrstat_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_VPBR_STATUS_TXLOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->vpbrstat_frm << + CS35L35_MON_FRM_SHIFT); + } + if (monitor_config->zerofill_specs) { + regmap_update_bits(cs35l35->regmap, + CS35L35_SUPMON_DEPTH_CTL, + CS35L35_ZEROFILL_DEPTH_MASK, + monitor_config->zerofill_dpth << + CS35L35_ZEROFILL_DEPTH_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, + CS35L35_MON_TXLOC_MASK, + monitor_config->zerofill_loc << + CS35L35_MON_TXLOC_SHIFT); + regmap_update_bits(cs35l35->regmap, + CS35L35_ZERO_FILL_LOC_CTL, + CS35L35_MON_FRM_MASK, + monitor_config->zerofill_frm << + CS35L35_MON_FRM_SHIFT); + } + } + + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l35 = { + .probe = cs35l35_codec_probe, + .set_sysclk = cs35l35_codec_set_sysclk, + .component_driver = { + .dapm_widgets = cs35l35_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l35_dapm_widgets), + + .dapm_routes = cs35l35_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l35_audio_map), + + .controls = cs35l35_aud_controls, + .num_controls = ARRAY_SIZE(cs35l35_aud_controls), + }, + +}; + +static struct regmap_config cs35l35_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L35_MAX_REGISTER, + .reg_defaults = cs35l35_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l35_reg), + .volatile_reg = cs35l35_volatile_register, + .readable_reg = cs35l35_readable_register, + .precious_reg = cs35l35_precious_register, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l35_irq(int irq, void *data) +{ + struct cs35l35_private *cs35l35 = data; + struct snd_soc_codec *codec = cs35l35->codec; + unsigned int sticky1, sticky2, sticky3, sticky4; + unsigned int mask1, mask2, mask3, mask4, current1; + + /* ack the irq by reading all status registers */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_4, &sticky4); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_3, &sticky3); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_2, &sticky2); + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &sticky1); + + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_4, &mask4); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_3, &mask3); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_2, &mask2); + regmap_read(cs35l35->regmap, CS35L35_INT_MASK_1, &mask1); + + /* Check to see if unmasked bits are active */ + if (!(sticky1 & ~mask1) && !(sticky2 & ~mask2) && !(sticky3 & ~mask3) + && !(sticky4 & ~mask4)) + return IRQ_NONE; + + if (sticky2 & CS35L35_PDN_DONE) + complete(&cs35l35->pdn_done); + + /* read the current values */ + regmap_read(cs35l35->regmap, CS35L35_INT_STATUS_1, &current1); + + /* handle the interrupts */ + if (sticky1 & CS35L35_CAL_ERR) { + dev_crit(codec->dev, "Calibration Error\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_CAL_ERR)) { + pr_debug("%s : Cal error release\n", __func__); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_CAL_ERR_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_CAL_ERR_RLS, + CS35L35_CAL_ERR_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_CAL_ERR_RLS, 0); + } + } + + if (sticky1 & CS35L35_AMP_SHORT) { + dev_crit(codec->dev, "AMP Short Error\n"); + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_AMP_SHORT)) { + dev_dbg(codec->dev, "Amp short error release\n"); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_SHORT_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_SHORT_RLS, + CS35L35_SHORT_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_SHORT_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTW) { + dev_warn(codec->dev, "Over temperature warning\n"); + + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTW)) { + dev_dbg(codec->dev, "Over temperature warn release\n"); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTW_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTW_RLS, + CS35L35_OTW_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTW_RLS, 0); + } + } + + if (sticky1 & CS35L35_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + /* error is no longer asserted; safe to reset */ + if (!(current1 & CS35L35_OTE)) { + dev_dbg(codec->dev, "Over temperature error release\n"); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTE_RLS, 0); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTE_RLS, + CS35L35_OTE_RLS); + regmap_update_bits(cs35l35->regmap, + CS35L35_PROT_RELEASE_CTL, + CS35L35_OTE_RLS, 0); + } + } + + if (sticky3 & CS35L35_BST_HIGH) { + dev_crit(codec->dev, "VBST error: powering off!\n"); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky3 & CS35L35_LBST_SHORT) { + dev_crit(codec->dev, "LBST error: powering off!\n"); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_AMP, CS35L35_PDN_AMP); + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL1, + CS35L35_PDN_ALL, CS35L35_PDN_ALL); + } + + if (sticky2 & CS35L35_VPBR_ERR) + dev_dbg(codec->dev, "Error: Reactive Brownout\n"); + + if (sticky4 & CS35L35_VMON_OVFL) + dev_dbg(codec->dev, "Error: VMON overflow\n"); + + if (sticky4 & CS35L35_IMON_OVFL) + dev_dbg(codec->dev, "Error: IMON overflow\n"); + + return IRQ_HANDLED; +} + + +static int cs35l35_handle_of_data(struct i2c_client *i2c_client, + struct cs35l35_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct device_node *classh, *signal_format; + struct classh_cfg *classh_config = &pdata->classh_algo; + struct monitor_cfg *monitor_config = &pdata->mon_cfg; + unsigned int val32 = 0; + u8 monitor_array[3]; + int ret = 0; + + if (!np) + return 0; + + pdata->bst_pdn_fet_on = of_property_read_bool(np, + "cirrus,boost-pdn-fet-on"); + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val32); + if (ret >= 0) { + if (val32 < 2600 || val32 > 9000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val32); + return -EINVAL; + } + pdata->bst_vctl = ((val32 - 2600) / 100) + 1; + } + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val32); + if (ret >= 0) { + if (val32 < 1680 || val32 > 4480) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val32); + return -EINVAL; + } + + pdata->bst_ipk = (val32 - 1680) / 110; + } + + if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) + pdata->sp_drv_str = val32; + + pdata->stereo = of_property_read_bool(np, "cirrus,stereo-config"); + + if (pdata->stereo) { + ret = of_property_read_u32(np, "cirrus,audio-channel", &val32); + if (ret >= 0) + pdata->aud_channel = val32; + + ret = of_property_read_u32(np, "cirrus,advisory-channel", + &val32); + if (ret >= 0) + pdata->adv_channel = val32; + + pdata->shared_bst = of_property_read_bool(np, + "cirrus,shared-boost"); + } + + pdata->gain_zc = of_property_read_bool(np, "cirrus,amp-gain-zc"); + + classh = of_get_child_by_name(np, "cirrus,classh-internal-algo"); + classh_config->classh_algo_enable = classh ? true : false; + + if (classh_config->classh_algo_enable) { + classh_config->classh_bst_override = + of_property_read_bool(np, "cirrus,classh-bst-overide"); + + ret = of_property_read_u32(classh, + "cirrus,classh-bst-max-limit", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_bst_max_limit = val32; + } + + ret = of_property_read_u32(classh, + "cirrus,classh-bst-max-limit", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_bst_max_limit = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-mem-depth", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_mem_depth = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-release-rate", + &val32); + if (ret >= 0) + classh_config->classh_release_rate = val32; + + ret = of_property_read_u32(classh, "cirrus,classh-headroom", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_headroom = val32; + } + + ret = of_property_read_u32(classh, + "cirrus,classh-wk-fet-disable", + &val32); + if (ret >= 0) + classh_config->classh_wk_fet_disable = val32; + + ret = of_property_read_u32(classh, "cirrus,classh-wk-fet-delay", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_wk_fet_delay = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-wk-fet-thld", + &val32); + if (ret >= 0) + classh_config->classh_wk_fet_thld = val32; + + ret = of_property_read_u32(classh, "cirrus,classh-vpch-auto", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_vpch_auto = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-vpch-rate", + &val32); + if (ret >= 0) { + val32 |= CS35L35_VALID_PDATA; + classh_config->classh_vpch_rate = val32; + } + + ret = of_property_read_u32(classh, "cirrus,classh-vpch-man", + &val32); + if (ret >= 0) + classh_config->classh_vpch_man = val32; + } + of_node_put(classh); + + /* frame depth location */ + signal_format = of_get_child_by_name(np, "cirrus,monitor-signal-format"); + monitor_config->is_present = signal_format ? true : false; + if (monitor_config->is_present) { + ret = of_property_read_u8_array(signal_format, "cirrus,imon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->imon_specs = true; + monitor_config->imon_dpth = monitor_array[0]; + monitor_config->imon_loc = monitor_array[1]; + monitor_config->imon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vmon_specs = true; + monitor_config->vmon_dpth = monitor_array[0]; + monitor_config->vmon_loc = monitor_array[1]; + monitor_config->vmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpmon_specs = true; + monitor_config->vpmon_dpth = monitor_array[0]; + monitor_config->vpmon_loc = monitor_array[1]; + monitor_config->vpmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vbstmon", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vbstmon_specs = true; + monitor_config->vbstmon_dpth = monitor_array[0]; + monitor_config->vbstmon_loc = monitor_array[1]; + monitor_config->vbstmon_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,vpbrstat", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->vpbrstat_specs = true; + monitor_config->vpbrstat_dpth = monitor_array[0]; + monitor_config->vpbrstat_loc = monitor_array[1]; + monitor_config->vpbrstat_frm = monitor_array[2]; + } + ret = of_property_read_u8_array(signal_format, "cirrus,zerofill", + monitor_array, ARRAY_SIZE(monitor_array)); + if (!ret) { + monitor_config->zerofill_specs = true; + monitor_config->zerofill_dpth = monitor_array[0]; + monitor_config->zerofill_loc = monitor_array[1]; + monitor_config->zerofill_frm = monitor_array[2]; + } + } + of_node_put(signal_format); + + return 0; +} + +/* Errata Rev A0 */ +static const struct reg_sequence cs35l35_errata_patch[] = { + + { 0x7F, 0x99 }, + { 0x00, 0x99 }, + { 0x52, 0x22 }, + { 0x04, 0x14 }, + { 0x6D, 0x44 }, + { 0x24, 0x10 }, + { 0x58, 0xC4 }, + { 0x00, 0x98 }, + { 0x18, 0x08 }, + { 0x00, 0x00 }, + { 0x7F, 0x00 }, +}; + +static int cs35l35_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l35_private *cs35l35; + struct cs35l35_platform_data *pdata = + dev_get_platdata(&i2c_client->dev); + int i; + int ret; + unsigned int devid = 0; + unsigned int reg; + + cs35l35 = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_private), + GFP_KERNEL); + if (!cs35l35) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l35); + cs35l35->regmap = devm_regmap_init_i2c(i2c_client, &cs35l35_regmap); + if (IS_ERR(cs35l35->regmap)) { + ret = PTR_ERR(cs35l35->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) + cs35l35->supplies[i].supply = cs35l35_supplies[i]; + cs35l35->num_supplies = ARRAY_SIZE(cs35l35_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l35->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs35l35_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + if (i2c_client->dev.of_node) { + ret = cs35l35_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + cs35l35->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l35->num_supplies, + cs35l35->supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + return ret; + } + + /* returning NULL can be valid if in stereo mode */ + cs35l35->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l35->reset_gpio)) { + ret = PTR_ERR(cs35l35->reset_gpio); + if (ret == -EBUSY) { + dev_info(&i2c_client->dev, + "Reset line busy, assuming shared reset\n"); + cs35l35->reset_gpio = NULL; + } else { + dev_err(&i2c_client->dev, + "Failed to get reset GPIO: %d\n", ret); + goto err; + } + } + + gpiod_set_value_cansleep(cs35l35->reset_gpio, 1); + + init_completion(&cs35l35->pdn_done); + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l35_irq, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l35", cs35l35); + if (ret != 0) { + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + /* initialize codec */ + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_AB, &reg); + + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l35->regmap, CS35L35_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L35_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L35 Device ID (%X). Expected ID %X\n", + devid, CS35L35_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l35->regmap, CS35L35_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed: %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l35->regmap, cs35l35_errata_patch, + ARRAY_SIZE(cs35l35_errata_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to apply errata patch: %d\n", + ret); + goto err; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L35 (%x), Revision: %02X\n", devid, + ret & 0xFF); + + /* Set the INT Masks for critical errors */ + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_1, + CS35L35_INT1_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_2, + CS35L35_INT2_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_3, + CS35L35_INT3_CRIT_MASK); + regmap_write(cs35l35->regmap, CS35L35_INT_MASK_4, + CS35L35_INT4_CRIT_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PWR2_PDN_MASK, + CS35L35_PWR2_PDN_MASK); + + if (cs35l35->pdata.bst_pdn_fet_on) + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETON_SHIFT); + else + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL2, + CS35L35_PDN_BST_MASK, + 1 << CS35L35_PDN_BST_FETOFF_SHIFT); + + regmap_update_bits(cs35l35->regmap, CS35L35_PWRCTL3, + CS35L35_PWR3_PDN_MASK, + CS35L35_PWR3_PDN_MASK); + + regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, + CS35L35_AMP_MUTE_MASK, 1 << CS35L35_AMP_MUTE_SHIFT); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l35, cs35l35_dai, + ARRAY_SIZE(cs35l35_dai)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Failed to register codec: %d\n", ret); + goto err; + } + +err: + regulator_bulk_disable(cs35l35->num_supplies, + cs35l35->supplies); + /* should we release reset-gpio? */ + return ret; +} + +static int cs35l35_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id cs35l35_of_match[] = { + {.compatible = "cirrus,cs35l35"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l35_of_match); + +static const struct i2c_device_id cs35l35_id[] = { + {"cs35l35", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l35_id); + +static struct i2c_driver cs35l35_i2c_driver = { + .driver = { + .name = "cs35l35", + .of_match_table = cs35l35_of_match, + }, + .id_table = cs35l35_id, + .probe = cs35l35_i2c_probe, + .remove = cs35l35_i2c_remove, +}; + +module_i2c_driver(cs35l35_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L35 driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h new file mode 100644 index 0000000..e27a7fe --- /dev/null +++ b/sound/soc/codecs/cs35l35.h @@ -0,0 +1,284 @@ +/* + * cs35l35.h -- CS35L35 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Brian Austin <brian.austin(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 __CS35L35_H__ +#define __CS35L35_H__ + +#define CS35L35_FIRSTREG 0x01 +#define CS35L35_LASTREG 0x7E +#define CS35L35_CHIP_ID 0x00035A35 +#define CS35L35_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L35_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L35_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L35_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L35_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L35_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L35_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L35_PWRCTL3 0x08 /* Power Ctl 3 */ +#define CS35L35_CLK_CTL1 0x0A /* Clocking Ctl 1 */ +#define CS35L35_CLK_CTL2 0x0B /* Clocking Ctl 2 */ +#define CS35L35_CLK_CTL3 0x0C /* Clocking Ctl 3 */ +#define CS35L35_SP_FMT_CTL1 0x0D /* Serial Port Format CTL1 */ +#define CS35L35_SP_FMT_CTL2 0x0E /* Serial Port Format CTL2 */ +#define CS35L35_SP_FMT_CTL3 0x0F /* Serial Port Format CTL3 */ +#define CS35L35_MAG_COMP_CTL 0x13 /* Magnitude Comp CTL */ +#define CS35L35_AMP_INP_DRV_CTL 0x14 /* Amp Input Drive Ctl */ +#define CS35L35_AMP_DIG_VOL_CTL 0x15 /* Amplifier Dig Volume Ctl */ +#define CS35L35_AMP_DIG_VOL 0x16 /* Amplifier Dig Volume */ +#define CS35L35_ADV_DIG_VOL 0x17 /* Advisory Digital Volume */ +#define CS35L35_PROTECT_CTL 0x18 /* Amp Gain - Prot Ctl Param */ +#define CS35L35_AMP_GAIN_AUD_CTL 0x19 /* Amp Serial Port Gain Ctl */ +#define CS35L35_AMP_GAIN_PDM_CTL 0x1A /* Amplifier Gain PDM Ctl */ +#define CS35L35_AMP_GAIN_ADV_CTL 0x1B /* Amplifier Gain Ctl */ +#define CS35L35_GPI_CTL 0x1C /* GPI Ctl */ +#define CS35L35_BST_CVTR_V_CTL 0x1D /* Boost Conv Voltage Ctl */ +#define CS35L35_BST_PEAK_I 0x1E /* Boost Conv Peak Current */ +#define CS35L35_BST_RAMP_CTL 0x20 /* Boost Conv Soft Ramp Ctl */ +#define CS35L35_BST_CONV_COEF_1 0x21 /* Boost Conv Coefficients 1 */ +#define CS35L35_BST_CONV_COEF_2 0x22 /* Boost Conv Coefficients 2 */ +#define CS35L35_BST_CONV_SLOPE_COMP 0x23 /* Boost Conv Slope Comp */ +#define CS35L35_BST_CONV_SW_FREQ 0x24 /* Boost Conv L BST SW Freq */ +#define CS35L35_CLASS_H_CTL 0x30 /* CLS H Control */ +#define CS35L35_CLASS_H_HEADRM_CTL 0x31 /* CLS H Headroom Ctl */ +#define CS35L35_CLASS_H_RELEASE_RATE 0x32 /* CLS H Release Rate */ +#define CS35L35_CLASS_H_FET_DRIVE_CTL 0x33 /* CLS H Weak FET Drive Ctl */ +#define CS35L35_CLASS_H_VP_CTL 0x34 /* CLS H VP Ctl */ +#define CS35L35_CLASS_H_STATUS 0x38 /* CLS H Status */ +#define CS35L35_VPBR_CTL 0x3A /* VPBR Ctl */ +#define CS35L35_VPBR_VOL_CTL 0x3B /* VPBR Volume Ctl */ +#define CS35L35_VPBR_TIMING_CTL 0x3C /* VPBR Timing Ctl */ +#define CS35L35_VPBR_MODE_VOL_CTL 0x3D /* VPBR Mode/Attack Vol Ctl */ +#define CS35L35_VPBR_ATTEN_STATUS 0x4B /* VPBR Attenuation Status */ +#define CS35L35_SPKR_MON_CTL 0x4E /* Speaker Monitoring Ctl */ +#define CS35L35_IMON_SCALE_CTL 0x51 /* IMON Scale Ctl */ +#define CS35L35_AUDIN_RXLOC_CTL 0x52 /* Audio Input RX Loc Ctl */ +#define CS35L35_ADVIN_RXLOC_CTL 0x53 /* Advisory Input RX Loc Ctl */ +#define CS35L35_VMON_TXLOC_CTL 0x54 /* VMON TX Loc Ctl */ +#define CS35L35_IMON_TXLOC_CTL 0x55 /* IMON TX Loc Ctl */ +#define CS35L35_VPMON_TXLOC_CTL 0x56 /* VPMON TX Loc Ctl */ +#define CS35L35_VBSTMON_TXLOC_CTL 0x57 /* VBSTMON TX Loc Ctl */ +#define CS35L35_VPBR_STATUS_TXLOC_CTL 0x58 /* VPBR Status TX Loc Ctl */ +#define CS35L35_ZERO_FILL_LOC_CTL 0x59 /* Zero Fill Loc Ctl */ +#define CS35L35_AUDIN_DEPTH_CTL 0x5A /* Audio Input Depth Ctl */ +#define CS35L35_SPKMON_DEPTH_CTL 0x5B /* SPK Mon Output Depth Ctl */ +#define CS35L35_SUPMON_DEPTH_CTL 0x5C /* Supply Mon Out Depth Ctl */ +#define CS35L35_ZEROFILL_DEPTH_CTL 0x5D /* Zero Fill Mon Output Ctl */ +#define CS35L35_MULT_DEV_SYNCH1 0x62 /* Multidevice Synch */ +#define CS35L35_MULT_DEV_SYNCH2 0x63 /* Multidevice Synch 2 */ +#define CS35L35_PROT_RELEASE_CTL 0x64 /* Protection Release Ctl */ +#define CS35L35_DIAG_MODE_REG_LOCK 0x68 /* Diagnostic Mode Reg Lock */ +#define CS35L35_DIAG_MODE_CTL_1 0x69 /* Diagnostic Mode Ctl 1 */ +#define CS35L35_DIAG_MODE_CTL_2 0x6A /* Diagnostic Mode Ctl 2 */ +#define CS35L35_INT_MASK_1 0x70 /* Interrupt Mask 1 */ +#define CS35L35_INT_MASK_2 0x71 /* Interrupt Mask 2 */ +#define CS35L35_INT_MASK_3 0x72 /* Interrupt Mask 3 */ +#define CS35L35_INT_MASK_4 0x73 /* Interrupt Mask 4 */ +#define CS35L35_INT_STATUS_1 0x74 /* Interrupt Status 1 */ +#define CS35L35_INT_STATUS_2 0x75 /* Interrupt Status 2 */ +#define CS35L35_INT_STATUS_3 0x76 /* Interrupt Status 3 */ +#define CS35L35_INT_STATUS_4 0x77 /* Interrupt Status 4 */ +#define CS35L35_PLL_STATUS 0x78 /* PLL Status */ +#define CS35L35_OTP_TRIM_STATUS 0x7E /* OTP Trim Status */ + +#define CS35L35_MAX_REGISTER 0x7F + +/* CS35L35_PWRCTL1 */ +#define CS35L35_SFT_RST 0x80 +#define CS35L35_DISCHG_FLT 0x02 +#define CS35L35_PDN_ALL 0x01 + +/* CS35L35_PWRCTL2 */ +#define CS35L35_PDN_VMON 0x80 +#define CS35L35_PDN_IMON 0x40 +#define CS35L35_PDN_CLASSH 0x20 +#define CS35L35_PDN_VPBR 0x10 +#define CS35L35_PDN_BST 0x04 +#define CS35L35_PDN_AMP 0x01 + +/* CS35L35_PWRCTL3 */ +#define CS35L35_PDN_VBSTMON_OUT 0x10 +#define CS35L35_PDN_VMON_OUT 0x08 + +#define CS35L35_AUDIN_DEPTH_MASK 0x03 +#define CS35L35_AUDIN_DEPTH_SHIFT 0 +#define CS35L35_ADVIN_DEPTH_MASK 0x0C +#define CS35L35_ADVIN_DEPTH_SHIFT 2 +#define CS35L35_SDIN_DEPTH_8 0x01 +#define CS35L35_SDIN_DEPTH_16 0x02 +#define CS35L35_SDIN_DEPTH_24 0x03 + +#define CS35L35_SDOUT_DEPTH_8 0x01 +#define CS35L35_SDOUT_DEPTH_12 0x02 +#define CS35L35_SDOUT_DEPTH_16 0x03 + +#define CS35L35_AUD_IN_LR_MASK 0x80 +#define CS35L35_AUD_IN_LR_SHIFT 7 +#define CS35L35_ADV_IN_LR_MASK 0x80 +#define CS35L35_ADV_IN_LR_SHIFT 7 +#define CS35L35_AUD_IN_LOC_MASK 0x0F +#define CS35L35_AUD_IN_LOC_SHIFT 0 +#define CS35L35_ADV_IN_LOC_MASK 0x0F +#define CS35L35_ADV_IN_LOC_SHIFT 0 + +#define CS35L35_IMON_DEPTH_MASK 0x03 +#define CS35L35_IMON_DEPTH_SHIFT 0 +#define CS35L35_VMON_DEPTH_MASK 0x0C +#define CS35L35_VMON_DEPTH_SHIFT 2 +#define CS35L35_VBSTMON_DEPTH_MASK 0x03 +#define CS35L35_VBSTMON_DEPTH_SHIFT 0 +#define CS35L35_VPMON_DEPTH_MASK 0x0C +#define CS35L35_VPMON_DEPTH_SHIFT 2 +#define CS35L35_VPBRSTAT_DEPTH_MASK 0x30 +#define CS35L35_VPBRSTAT_DEPTH_SHIFT 4 +#define CS35L35_ZEROFILL_DEPTH_MASK 0x03 +#define CS35L35_ZEROFILL_DEPTH_SHIFT 0x00 + +#define CS35L35_MON_TXLOC_MASK 0x3F +#define CS35L35_MON_TXLOC_SHIFT 0 +#define CS35L35_MON_FRM_MASK 0x80 +#define CS35L35_MON_FRM_SHIFT 7 + +#define CS35L35_MS_MASK 0x80 +#define CS35L35_MS_SHIFT 7 +#define CS35L35_SPMODE_MASK 0x40 +#define CS35L35_SP_DRV_MASK 0x10 +#define CS35L35_SP_DRV_SHIFT 4 +#define CS35L35_CLK_CTL2_MASK 0xFF +#define CS35L35_PDM_MODE_MASK 0x40 +#define CS35L35_PDM_MODE_SHIFT 6 +#define CS35L35_CLK_SOURCE_MASK 0x03 +#define CS35L35_CLK_SOURCE_SHIFT 0 +#define CS35L35_CLK_SOURCE_MCLK 0 +#define CS35L35_CLK_SOURCE_SCLK 1 +#define CS35L35_CLK_SOURCE_PDM 2 + +#define CS35L35_SP_SCLKS_MASK 0x0F +#define CS35L35_SP_SCLKS_SHIFT 0x00 +#define CS35L35_SP_SCLKS_16FS 0x03 +#define CS35L35_SP_SCLKS_32FS 0x07 +#define CS35L35_SP_SCLKS_48FS 0x0B +#define CS35L35_SP_SCLKS_64FS 0x0F +#define CS35L35_SP_RATE_MASK 0xC0 + +#define CS35L35_PDN_BST_MASK 0x06 +#define CS35L35_PDN_BST_FETON_SHIFT 1 +#define CS35L35_PDN_BST_FETOFF_SHIFT 2 +#define CS35L35_PWR2_PDN_MASK 0xE0 +#define CS35L35_PWR3_PDN_MASK 0x1E +#define CS35L35_PDN_ALL_MASK 0x01 +#define CS35L35_DISCHG_FILT_MASK 0x02 +#define CS35L35_DISCHG_FILT_SHIFT 1 +#define CS35L35_MCLK_DIS_MASK 0x04 +#define CS35L35_MCLK_DIS_SHIFT 2 + +#define CS35L35_BST_CTL_MASK 0x7F +#define CS35L35_BST_CTL_SHIFT 0 +#define CS35L35_BST_IPK_MASK 0x1F +#define CS35L35_BST_IPK_SHIFT 0 +#define CS35L35_AMP_MUTE_MASK 0x20 +#define CS35L35_AMP_MUTE_SHIFT 5 +#define CS35L35_AMP_GAIN_ZC_MASK 0x10 +#define CS35L35_AMP_GAIN_ZC_SHIFT 4 + +/* Class H Algorithm Control */ +#define CS35L35_CH_STEREO_MASK 0x40 +#define CS35L35_CH_STEREO_SHIFT 6 +#define CS35L35_CH_BST_OVR_MASK 0x04 +#define CS35L35_CH_BST_OVR_SHIFT 2 +#define CS35L35_CH_BST_LIM_MASK 0x08 +#define CS35L35_CH_BST_LIM_SHIFT 3 +#define CS35L35_CH_MEM_DEPTH_MASK 0x01 +#define CS35L35_CH_MEM_DEPTH_SHIFT 0 +#define CS35L35_CH_HDRM_CTL_MASK 0x3F +#define CS35L35_CH_HDRM_CTL_SHIFT 0 +#define CS35L35_CH_REL_RATE_MASK 0xFF +#define CS35L35_CH_REL_RATE_SHIFT 0 +#define CS35L35_CH_WKFET_DIS_MASK 0x80 +#define CS35L35_CH_WKFET_DIS_SHIFT 7 +#define CS35L35_CH_WKFET_DEL_MASK 0x70 +#define CS35L35_CH_WKFET_DEL_SHIFT 4 +#define CS35L35_CH_WKFET_THLD_MASK 0x0F +#define CS35L35_CH_WKFET_THLD_SHIFT 0 +#define CS35L35_CH_VP_AUTO_MASK 0x80 +#define CS35L35_CH_VP_AUTO_SHIFT 7 +#define CS35L35_CH_VP_RATE_MASK 0x60 +#define CS35L35_CH_VP_RATE_SHIFT 5 +#define CS35L35_CH_VP_MAN_MASK 0x1F +#define CS35L35_CH_VP_MAN_SHIFT 0 + +/* CS35L35_PROT_RELEASE_CTL */ +#define CS35L35_CAL_ERR_RLS 0x80 +#define CS35L35_SHORT_RLS 0x04 +#define CS35L35_OTW_RLS 0x02 +#define CS35L35_OTE_RLS 0x01 + +/* INT Mask Registers */ +#define CS35L35_INT1_CRIT_MASK 0x38 +#define CS35L35_INT2_CRIT_MASK 0xEF +#define CS35L35_INT3_CRIT_MASK 0xEE +#define CS35L35_INT4_CRIT_MASK 0xFF + +/* PDN DONE Masks */ +#define CS35L35_M_PDN_DONE_SHIFT 4 +#define CS35L35_M_PDN_DONE_MASK 0x10 + +/* CS35L35_INT_1 */ +#define CS35L35_CAL_ERR 0x80 +#define CS35L35_OTP_ERR 0x40 +#define CS35L35_LRCLK_ERR 0x20 +#define CS35L35_SPCLK_ERR 0x10 +#define CS35L35_MCLK_ERR 0x08 +#define CS35L35_AMP_SHORT 0x04 +#define CS35L35_OTW 0x02 +#define CS35L35_OTE 0x01 + +/* CS35L35_INT_2 */ +#define CS35L35_PDN_DONE 0x10 +#define CS35L35_VPBR_ERR 0x02 +#define CS35L35_VPBR_CLR 0x01 + +/* CS35L35_INT_3 */ +#define CS35L35_BST_HIGH 0x10 +#define CS35L35_BST_HIGH_FLAG 0x08 +#define CS35L35_BST_IPK_FLAG 0x04 +#define CS35L35_LBST_SHORT 0x01 + +/* CS35L35_INT_4 */ +#define CS35L35_VMON_OVFL 0x08 +#define CS35L35_IMON_OVFL 0x04 + +#define CS35L35_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct cs35l35_private { + struct snd_soc_codec *codec; + struct cs35l35_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int sysclk; + int sclk; + bool pdm_mode; + bool i2s_mode; + bool slave_mode; + /* GPIO for /RST */ + struct gpio_desc *reset_gpio; + struct completion pdn_done; +}; + +static const char * const cs35l35_supplies[] = { + "VA", + "VP", +}; + +#endif -- 1.9.1

2 1

[alsa-devel] [BUG] Meter causes freeze when closing audio device
by Alistair Buxton 06 Mar '17

06 Mar '17

To reproduce, put the following in asound.conf or .asoundrc: pcm.mymeter { type meter slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" period_time 0 period_size 1024 buffer_size 4096 rate 48000 } bindings { 0 0 1 1 } } ctl.dmixer { type hw card 0 } If pulseaudio is running, suspend it: pasuspender sleep 1d Now run aplay (you may need to change sample format): aplay -v -fS16_LE -r48000 -c2 -d1 -Dmymeter /dev/zero Expected result: aplay should play one second of silence and then exit. Actual result: About 50% of the time, aplay plays one second of silence and then freezes forever in poll(). Playing directly to dmixer does not freeze. Backtrace: (gdb) bt #0 0xb6ce5b80 in poll () at ../sysdeps/unix/syscall-template.S:81 #1 0xb6e5c428 in poll (__timeout=-1, __nfds=1, __fds=0xbe96c408) at /usr/include/arm-linux-gnueabihf/bits/poll2.h:46 #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xc69570, timeout=timeout@entry=-1) at pcm.c:2418 #3 0xb6e961d8 in snd_pcm_dmix_drain (pcm=0xc69570) at pcm_dmix.c:649 #4 0x0000de90 in ?? () Backtrace stopped: previous frame identical to this frame (corrupt stack?) Also reproducible with other audio players such as sox and ogg123. Reproducible on Ubuntu 17.04 beta with libasound2 1.1.3 and virtualbox audio device. Reproducible on Ubuntu 16.04 with libasound2 1.1.0 and Intel HD audio. Reproducible on Raspbian Jessie with libasound 1.0.28 and snd_soc_hifiberry_dac. -- Alistair Buxton a.j.buxton(a)gmail.com

1 0

[alsa-devel] [RFC v2 0/7] Add STM32 DFSDM support
by Arnaud Pouliquen 05 Mar '17

05 Mar '17

Hello, This RFC is following first patch-set sent for DFSDM driver (https://www.spinics.net/lists/arm-kernel/msg557133.html). AS MFD driver has been rejected, a new way to integrate DFSDM has to be defined. Aim of this RFC is to provide a base to discuss this redesign. 1) DFSDM hardware overview: ----------------------- The Digital Filter for Sigma Delta is a module dedicated to interface external sigma delta modulators to STM32 micro-controllers. It is targeted for: - Audio mode signal with Pulse density modulation (PDM) microphone - motor and Sensing mode with sigma delta ADC modulator Main Features: - Up to 8 multiplexed input digital serial channel (SIT) . SPI (PDM compatible) or Manchester interface . Slave or master ( one main SPI CLK OUT for all interface) - Alternative inputs for 8 internal digital parallel channels (PIT) . 16 bits resolution . internal sources: ADC or memory - Digital signal processing: . 4 instances . Sincx filter (x : order 1 to 5) oversampling ratio up to 1024 . integrator: oversampling ratio up to 256 - Channel multiplexer . allow to connect SIT or PIT to filter . channel n can be connected SIT n or SIT n+1 - up to 24-bit output resolution . signed . sampling rate and resolution depend on filter parameters. - data offset correction - detector: . analog watchdog . short circuit detection . clock absence detection . extremes For details on DFSDM IP, please refer to STM32F413 data-sheet chapter 15: http://www.st.com/content/ccc/resource/technical/document/reference_manual/… 2) SW design proposal: --------------------- Patch-set associated to this RFC proposes an implementation of the DFSDM features shared between ASoC and IIO frameworks. Patch-set is only a Skeleton of the drivers, so a base to discuss and validate a design. It contains minimum code to allow probing (with DT) and to expose the ASoC and IIO ABI. Hope that is sufficent in a first step to allow to focus on APIs. In this patch-set there are two new APIs used: - IIO in-kern API: based on hw_customer API proposed by Lars - ASOC <-> IIO API inspired by API defined for hdmi-codec for ASoC/DRM interconnect. API is dedicated to DFSDM only. Notice also that this design is based on following assumption: - Audio stream ABI interface is ASoC, no access to data through IIO ABI for PDM. - ASoC DMA should be used for audio transfer as designed for real time stream. - Need some runtime parameters exchange between ASoC and IIO due to the correlation between the sample frequency, the DFSDM decimation factor and the associated scaling. - "ASoC: dmaengine_pcm: add copy support" patch: I added a patch in ASoC that allows to implement a copy function to process data after DMA transfer. Requested, as DFSDM samples captured contain channel ID on 8-LSB bits and need also a potential rescale to present DAT on 24-bits. - "IIO: ADC: add sigma delta modulator support" patch: Simple dummy driver created to support external Sigma delta modulator. It is binded to DFSDM driver through hw_customer API. Regards Arnaud Arnaud Pouliquen (6): iio: Add hardware consumer support IIO: Add bindings for simple sigma delta adc IIO: ADC: add sigma delta modulator support ASoC: stm32: add DFSDM DAI support IIO: add bindings for stm32 DFSDM filter IIO: ADC: add stm32 DFSDM support olivier moysan (1): ASoC: dmaengine_pcm: add copy support .../devicetree/bindings/iio/adc/simple_sd_adc.txt | 13 + .../bindings/iio/adc/st,stm32-dfsdm-adc.txt | 125 ++++++ drivers/iio/Kconfig | 6 + drivers/iio/Makefile | 1 + drivers/iio/adc/Kconfig | 24 + drivers/iio/adc/Makefile | 2 + drivers/iio/adc/simple_sd_adc.c | 112 +++++ drivers/iio/adc/stm32-dfsdm-adc.c | 483 +++++++++++++++++++++ drivers/iio/adc/stm32-dfsdm-core.c | 273 ++++++++++++ drivers/iio/adc/stm32-dfsdm.h | 141 ++++++ drivers/iio/hw_consumer.c | 156 +++++++ include/linux/iio/hw_consumer.h | 12 + include/sound/dmaengine_pcm.h | 3 + include/sound/stm32-adfsdm.h | 80 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/soc-generic-dmaengine-pcm.c | 37 +- sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 2 + sound/soc/stm/stm32_adfsdm.c | 365 ++++++++++++++++ 20 files changed, 1845 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/iio/adc/simple_sd_adc.txt create mode 100644 Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt create mode 100644 drivers/iio/adc/simple_sd_adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-core.c create mode 100644 drivers/iio/adc/stm32-dfsdm.h create mode 100644 drivers/iio/hw_consumer.c create mode 100644 include/linux/iio/hw_consumer.h create mode 100644 include/sound/stm32-adfsdm.h create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_adfsdm.c -- 1.9.1

4 32

[alsa-devel] [GIT PULL] Gemini lake audio firmware
by Vinod Koul 03 Mar '17

03 Mar '17

Hi, Please pull to receive the Geminilake audio firmware used in Intel Geminilake devices with Audio DSP. The following changes since commit 4b9559fb0565f582a21a4070a81a3b6a7d77610b: linux-firmware: update firmware file for atusb (2016-12-07 01:01:04 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git glk for you to fetch changes up to dc76591484e43cc031a28eeecae68196a98a1550: linux-firmware: intel: Add Geminilake audio firmware (2017-03-03 15:40:35 +0530) ---------------------------------------------------------------- Subhransu S. Prusty (1): linux-firmware: intel: Add Geminilake audio firmware WHENCE | 6 ++++++ intel/dsp_fw_glk.bin | 1 + intel/dsp_fw_glk_v1814.bin | Bin 0 -> 505608 bytes 3 files changed, 7 insertions(+) create mode 120000 intel/dsp_fw_glk.bin create mode 100644 intel/dsp_fw_glk_v1814.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH 00/10] ALSA SoC MAX98927 driver - revision
by Ryan Lee 03 Mar '17

03 Mar '17

The initial driver had many textual and structural problems. I have added next patches to modify existing problems. There was concern about the design of keeping separate regmaps for each instance of the amplifier explicitly instead of instantiating two instances of a single, simpler driver. This was major change of this release. Coding style problem like indentation also have been fixed. Ryan Lee (10): ALSA SoC MAX98927 driver - Initial release Updated max98927_reg table with physical defaults. Replaced max98927.h for better legibility Removed the secondary device initialization from the primary device initialization. Removed manual register configuration from devicetree Modified indentation Replaced pr_err by dev_err. Modified debug message Added mask variable to apply it in one round after the switch Modified initialization code of VI sensing Added DAI Sel Mux and minor updates Added ACPI support. Changed snd_kcontrol_chip to snd_soc_kcontrol_codec Added vendor prefix. Added range information .../devicetree/bindings/sound/max98927.txt | 39 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98927.c | 1039 ++++++++++++++++++++ sound/soc/codecs/max98927.h | 267 +++++ 5 files changed, 1352 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/max98927.txt mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/max98927.c create mode 100755 sound/soc/codecs/max98927.h -- 2.7.4

1 8

[alsa-devel] [PATCH] ASoC: Intel: Update bxt_da7219_max98357a to add a new platform clock widget
by Harsha Priya 03 Mar '17

03 Mar '17

This patch adds a platform clock widget to turn off the clock only when both headset capture and headset playback are not in use. This removes turning off the clock in hw_free so that the clock is on when either capture or playback of headset is in progress. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 58 +++++++++++++++++++-------- 1 file changed, 41 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 2cda06c..ed58809 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -55,6 +55,42 @@ enum { BXT_DPCM_AUDIO_HDMI3_PB, }; +static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI, + strlen(BXT_DIALOG_CODEC_DAI))) + return rtd->codec_dai; + } + + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = bxt_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to stop PLL\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + } + + return ret; +} + static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -69,6 +105,8 @@ enum { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route broxton_map[] = { @@ -109,6 +147,9 @@ enum { /* DMIC */ {"dmic01_hifi", NULL, "DMIC01 Rx"}, {"DMIC01 Rx", NULL, "DMIC AIF"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, }; static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -265,25 +306,8 @@ static int broxton_da7219_hw_params(struct snd_pcm_substream *substream, return ret; } -static int broxton_da7219_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - static const struct snd_soc_ops broxton_da7219_ops = { .hw_params = broxton_da7219_hw_params, - .hw_free = broxton_da7219_hw_free, }; static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, -- 1.9.1

2 1

[alsa-devel] [PATCH v2 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 03 Mar '17

03 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..ad0e99e 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -64,6 +64,8 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) + snd_config_delete((snd_config_t*)plug->rate_converter); assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1110,13 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; + if (rate_converter) { + if ((err = snd_config_copy((snd_config_t**)&plug->rate_converter, + (snd_config_t *)rate_converter)) < 0) + free(plug); + return err; + } + plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1119,6 +1127,8 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, err = snd_pcm_new(&pcm, SND_PCM_TYPE_PLUG, name, slave->stream, slave->mode); if (err < 0) { + if (plug->rate_converter) + snd_config_delete((snd_config_t*)plug->rate_converter); free(plug); return err; } -- 2.7.4

3 2

[alsa-devel] [PATCH] conf/cards: add VC4-HDMI card
by Boris Brezillon 02 Mar '17

02 Mar '17

Add a conf file for the VC4-HDMI sound card. Signed-off-by: Boris Brezillon <boris.brezillon(a)free-electrons.com> --- Hi, This patch is adding a card config file for the audio sound card whose driver has been submitted here [1] (not accepted yet). Since I am a total newbie to the alsa world, I'd like to get some feedback on this patch. Also, the card only supports 2 to 8 channels, and I wonder if we should add a plug element to support mono streams, and where this element should be added (after or before the iec958 element). Regards, Boris [1]https://www.spinics.net/lists/arm-kernel/msg565182.html src/conf/cards/Makefile.am | 1 + src/conf/cards/aliases.conf | 1 + src/conf/cards/vc4-hdmi.conf | 64 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 66 insertions(+) create mode 100644 src/conf/cards/vc4-hdmi.conf diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index e8b530e80a8f..00999f0186d6 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -51,6 +51,7 @@ cfg_files = aliases.conf \ TRID4DWAVENX.conf \ USB-Audio.conf \ YMF744.conf \ + vc4-hdmi.conf \ VIA686A.conf \ VIA8233.conf \ VIA8233A.conf \ diff --git a/src/conf/cards/aliases.conf b/src/conf/cards/aliases.conf index 60f9d26974fd..18a920f41e46 100644 --- a/src/conf/cards/aliases.conf +++ b/src/conf/cards/aliases.conf @@ -56,6 +56,7 @@ AV200 cards.CMI8788 CMI8786 cards.CMI8788 CMI8787 cards.CMI8788 pistachio cards.pistachio-card +VC4-HDMI cards.vc4-hdmi <confdir:pcm/default.conf> <confdir:pcm/dmix.conf> diff --git a/src/conf/cards/vc4-hdmi.conf b/src/conf/cards/vc4-hdmi.conf new file mode 100644 index 000000000000..027804a145d1 --- /dev/null +++ b/src/conf/cards/vc4-hdmi.conf @@ -0,0 +1,64 @@ +# +# Configuration for the VC4-HDMI sound card using software IEC958 +# subframe conversion +# + +<confdir:pcm/front.conf> + +vc4-hdmi.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD +} + +# default with dmix +vc4-hdmi.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + @func concat + strings [ "dmix:" $CARD ] + } + } +} + +<confdir:pcm/iec958.conf> + +vc4-hdmi.pcm.iec958.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type iec958 + slave { + format IEC958_SUBFRAME_LE + pcm { + type plug + slave.pcm { + type hw + card $CARD + } + } + } + status [ $AES0 $AES1 $AES2 $AES3 ] +} -- 2.7.4

2 1

Re: [alsa-devel] playback discontinuities on buffer boundaries (SOLVED)
by Paul Stone 02 Mar '17

02 Mar '17

Please disregard this question. I changed the chunk size from 0x80 to 0x100 and resolved the issue. Playback h/w wants to be fed a larger chunk size, or it tends to get starved. Best Regards, Paul Stone On Mar 01, 2017, at 11:51 AM, Paul Stone <metabagel(a)icloud.com> wrote: Correction: the discontinuity pattern seems to repeat itself every 3 buffers (24 ms). There are 128 stereo frames per buffer. -Paul Stone On Mar 01, 2017, at 10:57 AM, Paul Stone <metabagel(a)icloud.com> wrote: Hi all, Summary: playback discontinuities that occur when PCM is configured for 16 kHz go away when it's configured at 8 kHz, even though the output still runs at 16 kHz. I have written an ALSA driver for an embedded system. If I feed a synthetic sine wave to my output, it looks fine. However, when I use the following test, I see (with an oscilloscope) discontinuity in the sine wave on a buffer boundary (every 8 ms). Additionally, the pattern of the discontinuity repeats every 4 buffers (32 ms). Here is the test I am running: # gst-launch audiotestsrc ! 'audio/x-raw-int, channels=2, rate=16000, depth=16' ! alsasink device=plughw:3,0 The audio components of the system are a bit old: gst-launch-0.10 version 0.10.36 GStreamer 0.10.36 alsa-lib-1.0.27.2-r1.cortexa9hf_vfp_neon I have not added any configuration for this device in /etc/asound.conf. Audio buffers are interleaved 16-bit stereo, output to FPGA which is configured with sampling frequency of 16 kHz. Buffer size is 128 frames. * hardware definition */ static struct snd_pcm_hardware snd_imx_audio_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER ), .formats = SNDRV_PCM_FMTBIT_S16_LE, // Signed 16 bits .rates = SNDRV_PCM_RATE_16000, .rate_min = 16000, .rate_max = 16000, .channels_min = 2, // stereo .channels_max = 2, .buffer_bytes_max = MAX_BUFFER, // 10 buffers * BUFFER_SIZE .period_bytes_min = BUFFER_SIZE, // 128 frames per buffer * 2 stereo samples per frame * 2 bytes per sample (16-bit) .period_bytes_max = MAX_BUFFER, // 10 buffers * BUFFER_SIZE .periods_min = 1, .periods_max = BUFFERS_PER_PAIR, // 10 }; If I modify the test script to 8 kHz as follows: # gst-launch audiotestsrc ! 'audio/x-raw-int, channels=2, rate=8000, depth=16' ! alsasink device=plughw:3,0 and modify the PCM h/w definition to 8 kHz as follows: * hardware definition */ static struct snd_pcm_hardware snd_imx_audio_playback_hw = { . .rates = SNDRV_PCM_RATE_8000, .rate_min = 8000, .rate_max = 8000, . }; ... then the discontinuities go away. I get a pretty good sine wave output. Additionally, the output runs at 16 kHz, so it appears that even though ALSA has been told that the PCM runs at 8 kHz, it is keeping up with the true output rate of 16 kHz. I don't believe the problem is in the FPGA, because the output looks fine if I synthesize a sine wave inside the driver. I am trying to investigate buffer usage, but any suggestions on how to approach this would be most welcome. I'm guessing that maybe ALSA is overflowing on every buffer boundary (dropped audio data), but I don't see any warning messages to that effect, and if that's the case I don't know why that would be happening or how to address it. Best Regards, Paul Stone -Paul Stone

1 0

[alsa-devel] Likely offline for a while
by Takashi Iwai 01 Mar '17

01 Mar '17

Hi, JFYI: I'm going to travel from tomorrow for 2.5 weeks. Since I'm not sure how good the net connection in Sri Lanka will be, I'll be less responsive for a while. After all, a vacation is the time to enjoy on a shiny beach and not to discuss about messy bugs, no? :) That is, I'll handle only super urgent issues, and postpone the rest. thanks, Takashi

1 0

Re: [alsa-devel] playback discontinuities on buffer boundaries
by Paul Stone 01 Mar '17

01 Mar '17

Correction: the discontinuity pattern seems to repeat itself every 3 buffers (24 ms). There are 128 stereo frames per buffer. -Paul Stone On Mar 01, 2017, at 10:57 AM, Paul Stone <metabagel(a)icloud.com> wrote: Hi all, Summary: playback discontinuities that occur when PCM is configured for 16 kHz go away when it's configured at 8 kHz, even though the output still runs at 16 kHz. I have written an ALSA driver for an embedded system. If I feed a synthetic sine wave to my output, it looks fine. However, when I use the following test, I see (with an oscilloscope) discontinuity in the sine wave on a buffer boundary (every 8 ms). Additionally, the pattern of the discontinuity repeats every 4 buffers (32 ms). Here is the test I am running: # gst-launch audiotestsrc ! 'audio/x-raw-int, channels=2, rate=16000, depth=16' ! alsasink device=plughw:3,0 The audio components of the system are a bit old: gst-launch-0.10 version 0.10.36 GStreamer 0.10.36 alsa-lib-1.0.27.2-r1.cortexa9hf_vfp_neon I have not added any configuration for this device in /etc/asound.conf. Audio buffers are interleaved 16-bit stereo, output to FPGA which is configured with sampling frequency of 16 kHz. Buffer size is 128 frames. * hardware definition */ static struct snd_pcm_hardware snd_imx_audio_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER ), .formats = SNDRV_PCM_FMTBIT_S16_LE, // Signed 16 bits .rates = SNDRV_PCM_RATE_16000, .rate_min = 16000, .rate_max = 16000, .channels_min = 2, // stereo .channels_max = 2, .buffer_bytes_max = MAX_BUFFER, // 10 buffers * BUFFER_SIZE .period_bytes_min = BUFFER_SIZE, // 128 frames per buffer * 2 stereo samples per frame * 2 bytes per sample (16-bit) .period_bytes_max = MAX_BUFFER, // 10 buffers * BUFFER_SIZE .periods_min = 1, .periods_max = BUFFERS_PER_PAIR, // 10 }; If I modify the test script to 8 kHz as follows: # gst-launch audiotestsrc ! 'audio/x-raw-int, channels=2, rate=8000, depth=16' ! alsasink device=plughw:3,0 and modify the PCM h/w definition to 8 kHz as follows: * hardware definition */ static struct snd_pcm_hardware snd_imx_audio_playback_hw = { . .rates = SNDRV_PCM_RATE_8000, .rate_min = 8000, .rate_max = 8000, . }; ... then the discontinuities go away. I get a pretty good sine wave output. Additionally, the output runs at 16 kHz, so it appears that even though ALSA has been told that the PCM runs at 8 kHz, it is keeping up with the true output rate of 16 kHz. I don't believe the problem is in the FPGA, because the output looks fine if I synthesize a sine wave inside the driver. I am trying to investigate buffer usage, but any suggestions on how to approach this would be most welcome. I'm guessing that maybe ALSA is overflowing on every buffer boundary (dropped audio data), but I don't see any warning messages to that effect, and if that's the case I don't know why that would be happening or how to address it. Best Regards, Paul Stone -Paul Stone

1 0

[alsa-devel] playback discontinuities on buffer boundaries
by Paul Stone 01 Mar '17

01 Mar '17

Hi all, Summary: playback discontinuities that occur when PCM is configured for 16 kHz go away when it's configured at 8 kHz, even though the output still runs at 16 kHz. I have written an ALSA driver for an embedded system. If I feed a synthetic sine wave to my output, it looks fine. However, when I use the following test, I see (with an oscilloscope) discontinuity in the sine wave on a buffer boundary (every 8 ms). Additionally, the pattern of the discontinuity repeats every 4 buffers (32 ms). Here is the test I am running: # gst-launch audiotestsrc ! 'audio/x-raw-int, channels=2, rate=16000, depth=16' ! alsasink device=plughw:3,0 The audio components of the system are a bit old: gst-launch-0.10 version 0.10.36 GStreamer 0.10.36 alsa-lib-1.0.27.2-r1.cortexa9hf_vfp_neon I have not added any configuration for this device in /etc/asound.conf. Audio buffers are interleaved 16-bit stereo, output to FPGA which is configured with sampling frequency of 16 kHz. Buffer size is 128 frames. * hardware definition */ static struct snd_pcm_hardware snd_imx_audio_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER ), .formats = SNDRV_PCM_FMTBIT_S16_LE, // Signed 16 bits .rates = SNDRV_PCM_RATE_16000, .rate_min = 16000, .rate_max = 16000, .channels_min = 2, // stereo .channels_max = 2, .buffer_bytes_max = MAX_BUFFER, // 10 buffers * BUFFER_SIZE .period_bytes_min = BUFFER_SIZE, // 128 frames per buffer * 2 stereo samples per frame * 2 bytes per sample (16-bit) .period_bytes_max = MAX_BUFFER, // 10 buffers * BUFFER_SIZE .periods_min = 1, .periods_max = BUFFERS_PER_PAIR, // 10 }; If I modify the test script to 8 kHz as follows: # gst-launch audiotestsrc ! 'audio/x-raw-int, channels=2, rate=8000, depth=16' ! alsasink device=plughw:3,0 and modify the PCM h/w definition to 8 kHz as follows: * hardware definition */ static struct snd_pcm_hardware snd_imx_audio_playback_hw = { . .rates = SNDRV_PCM_RATE_8000, .rate_min = 8000, .rate_max = 8000, . }; ... then the discontinuities go away. I get a pretty good sine wave output. Additionally, the output runs at 16 kHz, so it appears that even though ALSA has been told that the PCM runs at 8 kHz, it is keeping up with the true output rate of 16 kHz. I don't believe the problem is in the FPGA, because the output looks fine if I synthesize a sine wave inside the driver. I am trying to investigate buffer usage, but any suggestions on how to approach this would be most welcome. I'm guessing that maybe ALSA is overflowing on every buffer boundary (dropped audio data), but I don't see any warning messages to that effect, and if that's the case I don't know why that would be happening or how to address it. Best Regards, Paul Stone -Paul Stone

1 0

[alsa-devel] usb headset with mic Jabra GN2400 (0b0e:2400) regression bug on linux 4.4.0-21-generic
by Fabrizio Lombardozzi 01 Mar '17

01 Mar '17

Hi, I've found that the headset gn2400 is not working properly any more on latest kernels, while it was working fine up to 3.8.0-19 (Ubuntu 13.04). The headset is steadily playing squeaky sound (probably a kind of rate mismatch) with any application that try to set a playback sample rate below 32Khz, e.g.: mplayer -srate 8000 song.mp3 Instead it can records properly at any rate: arecord -f S16_LE -r 8000 -D hw:1,0 -d 5 t.wav (but then you can't hear it) After "a few" modules recompilations I found an acceptable solution: --- ../../../linux-4.4_originale/linux-4.4/sound/usb/quirks.c 2017-02-24 17:22:57.021006325 +0100 +++ quirks.c 2017-03-01 18:36:51.877837419 +0100 @@ -1113,6 +1113,18 @@ set_format_emu_quirk(subs, fmt); break; } + + if (subs->stream->chip->usb_id == USB_ID(0x0b0e, 0x2400)) /* Jabra BIZ-2400 */ + { + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + { + int flr,rate=48000; + printk(KERN_ALERT "snd_usb_set_format_quirk() Jabra BIZ-2400 PLAYBACK forced rate:%d\n",rate); + for (flr=0;flr<fmt->nr_rates;flr++) fmt->rate_table[fmt->nr_rates] = rate; + fmt->rate_min=rate; + fmt->rate_max=rate; + } + } } bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) I hope it can be useful for others. Here are some other info root@fab-desktop:~/linux-4.4/sound/usb# lsusb -vvv -d 0b0e:2400 Bus 005 Device 059: ID 0b0e:2400 GN Netcom Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x0b0e GN Netcom idProduct 0x2400 bcdDevice 1.41 iManufacturer 0 iProduct 2 Jabra BIZ 2400 Mono USB iSerial 3 1C48F9103BE5x014100 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 256 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 100mA 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 103 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 10 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 2 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 4 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 7 bNrInPins 1 baSource( 0) 5 iSelector 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 7 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 8 bSourceID 4 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 9 bNrInPins 1 baSource( 0) 8 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 10 bNrInPins 2 baSourceID( 0) 1 baSourceID( 1) 9 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 bmControls 0x00 iMixer 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 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 3 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 48000 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x81 Sampling Frequency MaxPacketsOnly bLockDelayUnits 2 Decoded PCM samples wLockDelay 0 Decoded PCM samples Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0060 1x 96 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 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 6 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 16000 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 0 Decoded PCM samples Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 339 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0001 Self Powered -- Saluti Fabrizio Ing.Fabrizio Lombardozzi Coord. Servizi Ippofono Capodarco Via Ostiense, 131L asc.B 00154 ROMA Tel. +39 0657 060 230 Cell. +39 335 585 3047 Email: fabrizio.lombardozzi(a)sociale.it Ai sensi e per effetti della legge sulla tutela della riservatezza personale (D.lgs n.196/2003), questa @mail e' destinata unicamente alle persone sopra indicate e le informazioni in essa contenute sono da considerarsi strettamente riservate. E' proibito leggere, copiare, usare o diffondere il contenuto della presente @mail senza autorizzazione. Se avete ricevuto questo messaggio per errore, siete pregati di rispedire la stessa al mittente. Grazie -- "Ai sensi e per gli effetti della legge sulla tutela dei dati personali (D.lgs 196/2003), le informazioni contenute nella presente @mail sono di natura riservata e destinate ad un uso aziendale-lavorativo con esclusione di utilizzi ad uso personale; come tali, pertanto, sono riservate esclusivamente ai destinatari sopra indicati. E' proibito leggere, copiare, usare o diffondere il contenuto della presente @mail senza autorizzazione. Se avete ricevuto questa @mail per errore, siete pregati di rispedire la stessa al mittente. Grazie"

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[alsa-devel] [PATCH 1/9] ARM: dts: sun8i: add device tree for Lichee Pi Zero with Dock
by Icenowy Zheng 01 Mar '17

01 Mar '17

Lichee Pi Zero features a dock, which adds some functions, and should be soldered with the core board of Lichee Pi Zero. Add support for the dock as a new dts, as soldering is needed to make the dock usable and there's functions enabled on the dock that is unavailable at the 2.54mm pins of the core board. Currently no new feature on core board is available in the mainline kernel, so the dock dt file only includes the core board device tree. Signed-off-by: Icenowy Zheng <icenowy(a)aosc.xyz> --- arch/arm/boot/dts/Makefile | 3 +- arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts | 49 ++++++++++++++++++++++ 2 files changed, 51 insertions(+), 1 deletion(-) create mode 100644 arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts diff --git a/arch/arm/boot/dts/Makefile b/arch/arm/boot/dts/Makefile index 011808490fed..1aa365306479 100644 --- a/arch/arm/boot/dts/Makefile +++ b/arch/arm/boot/dts/Makefile @@ -876,7 +876,8 @@ dtb-$(CONFIG_MACH_SUN8I) += \ sun8i-h3-orangepi-plus.dtb \ sun8i-h3-orangepi-plus2e.dtb \ sun8i-r16-parrot.dtb \ - sun8i-v3s-licheepi-zero.dtb + sun8i-v3s-licheepi-zero.dtb \ + sun8i-v3s-licheepi-zero-dock.dtb dtb-$(CONFIG_MACH_SUN9I) += \ sun9i-a80-optimus.dtb \ sun9i-a80-cubieboard4.dtb diff --git a/arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts b/arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts new file mode 100644 index 000000000000..68c114d6b939 --- /dev/null +++ b/arch/arm/boot/dts/sun8i-v3s-licheepi-zero-dock.dts @@ -0,0 +1,49 @@ +/* + * Copyright (C) 2016 Icenowy Zheng <icenowy(a)aosc.xyz> + * + * This file is dual-licensed: you can use it either under the terms + * of the GPL or the X11 license, at your option. Note that this dual + * licensing only applies to this file, and not this project as a + * whole. + * + * a) This file 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. + * + * This file is distributed in the hope that 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. + * + * Or, alternatively, + * + * b) Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ + +#include "sun8i-v3s-licheepi-zero.dts" + +/ { + model = "Lichee Pi Zero with Dock"; + compatible = "licheepi,licheepi-zero-dock", "licheepi,licheepi-zero", + "allwinner,sun8i-v3s"; +}; -- 2.11.1

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[alsa-devel] [PATCH RESEND 0/4] ASoC: es8328: various improvements
by Romain Perier 01 Mar '17

01 Mar '17

This set of patches fixes clock handling in slave mode, enables sampling frequency of 192khz in slave mode and adds support for this frame rate in the driver rk3288_hdmi_analog.c. Note: Due to an email configuration issue, some of my patches were not received on infradead.org or vger.kernel.org. It is now fixed, so I resend this set of patches for this reason. Romain Perier (4): ASoC: es8328: Simplify rates definition ASoC: es8328: Let device auto detect ratios in slave mode ASoC: es8328: Enabling support for 192k ASoC: rockchip: Enable 192khz in hw_params operation sound/soc/codecs/es8328.c | 51 +++++++++++++++++++-------------- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 ++ 2 files changed, 32 insertions(+), 22 deletions(-) -- 2.9.3

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[alsa-devel] [sound:topic/cht-whiskey-cove 1/3] include/acpi/acpi_bus.h:598:9: error: 'AE_SUPPORT' undeclared
by kbuild test robot 01 Mar '17

01 Mar '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/cht-whiskey-cove head: fe0334695c856dbdbeea861a5eb76f5ddc0dbaa9 commit: fbba8603f93e60c68116197326cdbbceee0b875b [1/3] mfd: Add Cherrytrail WhiskeyCove PMIC driver config: x86_64-randconfig-r0-03011404 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout fbba8603f93e60c68116197326cdbbceee0b875b # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): acpi_status acpi_evaluate_ej0(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:59:1: error: unknown type name 'acpi_status' acpi_status acpi_evaluate_lck(acpi_handle handle, int lock); ^~~~~~~~~~~ include/acpi/acpi_bus.h:59:31: error: unknown type name 'acpi_handle' acpi_status acpi_evaluate_lck(acpi_handle handle, int lock); ^~~~~~~~~~~ include/acpi/acpi_bus.h:60:21: error: unknown type name 'acpi_handle' bool acpi_ata_match(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:61:21: error: unknown type name 'acpi_handle' bool acpi_bay_match(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:62:22: error: unknown type name 'acpi_handle' bool acpi_dock_match(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:64:21: error: unknown type name 'acpi_handle' bool acpi_check_dsm(acpi_handle handle, const u8 *uuid, u64 rev, u64 funcs); ^~~~~~~~~~~ include/acpi/acpi_bus.h:65:38: error: unknown type name 'acpi_handle' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, ^~~~~~~~~~~ include/acpi/acpi_bus.h:69:25: error: unknown type name 'acpi_handle' acpi_evaluate_dsm_typed(acpi_handle handle, const u8 *uuid, u64 rev, u64 func, ^~~~~~~~~~~ include/acpi/acpi_bus.h:70:30: error: unknown type name 'acpi_object_type' union acpi_object *argv4, acpi_object_type type) ^~~~~~~~~~~~~~~~ In file included from drivers/mfd/intel_soc_pmic_chtwc.c:18:0: include/acpi/acpi_bus.h:286:40: error: 'ACPI_D_STATE_COUNT' undeclared here (not in a function) struct acpi_device_power_state states[ACPI_D_STATE_COUNT]; /* Power states (D0-D3Cold) */ ^~~~~~~~~~~~~~~~~~ include/acpi/acpi_bus.h:326:2: error: unknown type name 'acpi_handle' acpi_handle gpe_device; ^~~~~~~~~~~ include/acpi/acpi_bus.h:356:2: error: unknown type name 'acpi_handle' acpi_handle handle; /* no handle for fixed hardware */ ^~~~~~~~~~~ include/acpi/acpi_bus.h:387:2: error: unknown type name 'acpi_handle' acpi_handle handle; ^~~~~~~~~~~ include/acpi/acpi_bus.h:478:49: error: expected ')' before 'void' void acpi_bus_private_data_handler(acpi_handle, void *); ^~~~ include/acpi/acpi_bus.h:479:44: error: expected ')' before 'void' int acpi_bus_get_private_data(acpi_handle, void **); ^~~~ include/acpi/acpi_bus.h:480:47: error: expected ')' before 'void' int acpi_bus_attach_private_data(acpi_handle, void *); ^~~~ include/acpi/acpi_bus.h:481:1: warning: parameter names (without types) in function declaration void acpi_bus_detach_private_data(acpi_handle); ^~~~ include/acpi/acpi_bus.h:490:25: error: unknown type name 'acpi_handle' int acpi_bus_get_device(acpi_handle handle, struct acpi_device **device); ^~~~~~~~~~~ include/acpi/acpi_bus.h:491:46: error: unknown type name 'acpi_handle' struct acpi_device *acpi_bus_get_acpi_device(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:493:1: error: unknown type name 'acpi_status' acpi_status acpi_bus_get_status_handle(acpi_handle handle, ^~~~~~~~~~~ include/acpi/acpi_bus.h:493:40: error: unknown type name 'acpi_handle' acpi_status acpi_bus_get_status_handle(acpi_handle handle, ^~~~~~~~~~~ include/acpi/acpi_bus.h:497:24: error: unknown type name 'acpi_handle' int acpi_bus_set_power(acpi_handle handle, int state); ^~~~~~~~~~~ include/acpi/acpi_bus.h:503:27: error: unknown type name 'acpi_handle' int acpi_bus_update_power(acpi_handle handle, int *state_p); ^~~~~~~~~~~ include/acpi/acpi_bus.h:505:32: error: unknown type name 'acpi_handle' bool acpi_bus_power_manageable(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:510:40: error: unknown type name 'acpi_handle' static inline bool acpi_bus_can_wakeup(acpi_handle handle) { return false; } ^~~~~~~~~~~ include/acpi/acpi_bus.h:520:19: error: unknown type name 'acpi_handle' int acpi_bus_scan(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:522:1: error: unknown type name 'acpi_status' acpi_status acpi_bus_get_ejd(acpi_handle handle, acpi_handle * ejd); ^~~~~~~~~~~ include/acpi/acpi_bus.h:522:30: error: unknown type name 'acpi_handle' acpi_status acpi_bus_get_ejd(acpi_handle handle, acpi_handle * ejd); ^~~~~~~~~~~ include/acpi/acpi_bus.h:522:50: error: unknown type name 'acpi_handle' acpi_status acpi_bus_get_ejd(acpi_handle handle, acpi_handle * ejd); ^~~~~~~~~~~ include/acpi/acpi_bus.h:581:1: warning: parameter names (without types) in function declaration int acpi_is_root_bridge(acpi_handle); ^~~ include/acpi/acpi_bus.h:582:42: error: unknown type name 'acpi_handle' struct acpi_pci_root *acpi_pci_find_root(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:594:15: error: unknown type name 'acpi_status' static inline acpi_status acpi_add_pm_notifier(struct acpi_device *adev, ^~~~~~~~~~~ include/acpi/acpi_bus.h: In function 'acpi_add_pm_notifier': >> include/acpi/acpi_bus.h:598:9: error: 'AE_SUPPORT' undeclared (first use in this function) return AE_SUPPORT; ^~~~~~~~~~ include/acpi/acpi_bus.h:598:9: note: each undeclared identifier is reported only once for each function it appears in include/acpi/acpi_bus.h: At top level: include/acpi/acpi_bus.h:600:15: error: unknown type name 'acpi_status' static inline acpi_status acpi_remove_pm_notifier(struct acpi_device *adev) ^~~~~~~~~~~ include/acpi/acpi_bus.h: In function 'acpi_remove_pm_notifier': include/acpi/acpi_bus.h:602:9: error: 'AE_SUPPORT' undeclared (first use in this function) return AE_SUPPORT; ^~~~~~~~~~ include/acpi/acpi_bus.h: In function 'acpi_pm_device_sleep_state': >> include/acpi/acpi_bus.h:607:8: error: 'ACPI_STATE_D0' undeclared (first use in this function) *p = ACPI_STATE_D0; ^~~~~~~~~~~~~ include/acpi/acpi_bus.h:609:37: error: 'ACPI_STATE_D3_COLD' undeclared (first use in this function) return (m >= ACPI_STATE_D0 && m <= ACPI_STATE_D3_COLD) ? ^~~~~~~~~~~~~~~~~~ include/acpi/acpi_bus.h: In function 'acpi_target_system_state': include/acpi/acpi_bus.h:630:59: error: 'ACPI_STATE_S0' undeclared (first use in this function) static inline u32 acpi_target_system_state(void) { return ACPI_STATE_S0; } ^~~~~~~~~~~~~ include/acpi/acpi_bus.h: In function 'acpi_device_can_poweroff': include/acpi/acpi_bus.h:645:28: error: 'ACPI_STATE_D3_COLD' undeclared (first use in this function) return adev->power.states[ACPI_STATE_D3_COLD].flags.valid || ^~~~~~~~~~~~~~~~~~ include/acpi/acpi_bus.h:646:5: error: 'acpi_gbl_FADT' undeclared (first use in this function) ((acpi_gbl_FADT.header.revision < 6) && ^~~~~~~~~~~~~ include/acpi/acpi_bus.h:647:22: error: 'ACPI_STATE_D3_HOT' undeclared (first use in this function) adev->power.states[ACPI_STATE_D3_HOT].flags.explicit_set); ^~~~~~~~~~~~~~~~~ In file included from drivers/mfd/intel_soc_pmic_chtwc.c:19:0: include/linux/acpi.h: In function 'acpi_device_handle': include/linux/acpi.h:51:29: warning: pointer/integer type mismatch in conditional expression return adev ? adev->handle : NULL; ^ drivers/mfd/intel_soc_pmic_chtwc.c: In function 'cht_wc_probe': drivers/mfd/intel_soc_pmic_chtwc.c:104:11: error: implicit declaration of function 'acpi_evaluate_integer' [-Werror=implicit-function-declaration] status = acpi_evaluate_integer(handle, "_HRV", NULL, &hrv); ^~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/AE_SUPPORT +598 include/acpi/acpi_bus.h d760a1baf2 Lorenzo Pieralisi 2016-11-21 576 void acpi_dma_configure(struct device *dev, enum dev_dma_attr attr); d760a1baf2 Lorenzo Pieralisi 2016-11-21 577 void acpi_dma_deconfigure(struct device *dev); b84f196d96 Suthikulpanit, Suravee 2015-10-28 578 d9fef0c4d2 Rafael J. Wysocki 2013-11-28 579 struct acpi_device *acpi_find_child_device(struct acpi_device *parent, d9fef0c4d2 Rafael J. Wysocki 2013-11-28 580 u64 address, bool check_children); 275582031f Alexander Chiang 2009-06-10 581 int acpi_is_root_bridge(acpi_handle); 76d56de57a Alex Chiang 2009-07-23 @582 struct acpi_pci_root *acpi_pci_find_root(acpi_handle handle); 4e10d12a3d David Shaohua Li 2005-03-18 583 b67ea76172 Rafael J. Wysocki 2010-02-17 584 int acpi_enable_wakeup_device_power(struct acpi_device *dev, int state); b67ea76172 Rafael J. Wysocki 2010-02-17 585 int acpi_disable_wakeup_device_power(struct acpi_device *dev); b67ea76172 Rafael J. Wysocki 2010-02-17 586 aa33860158 Rafael J. Wysocki 2011-02-11 587 #ifdef CONFIG_PM c072530f39 Rafael J. Wysocki 2014-07-23 588 acpi_status acpi_add_pm_notifier(struct acpi_device *adev, struct device *dev, c072530f39 Rafael J. Wysocki 2014-07-23 589 void (*work_func)(struct work_struct *work)); c072530f39 Rafael J. Wysocki 2014-07-23 590 acpi_status acpi_remove_pm_notifier(struct acpi_device *adev); ee85f54371 Huang Ying 2012-06-23 591 int acpi_pm_device_sleep_state(struct device *, int *, int); 5de21bb998 Rafael J. Wysocki 2014-11-27 592 int acpi_pm_device_run_wake(struct device *, bool); 761afb869f Rafael J. Wysocki 2010-10-14 593 #else ec2cd81ccf Rafael J. Wysocki 2012-11-02 594 static inline acpi_status acpi_add_pm_notifier(struct acpi_device *adev, c072530f39 Rafael J. Wysocki 2014-07-23 595 struct device *dev, c072530f39 Rafael J. Wysocki 2014-07-23 596 void (*work_func)(struct work_struct *work)) ec2cd81ccf Rafael J. Wysocki 2012-11-02 597 { ec2cd81ccf Rafael J. Wysocki 2012-11-02 @598 return AE_SUPPORT; ec2cd81ccf Rafael J. Wysocki 2012-11-02 599 } c072530f39 Rafael J. Wysocki 2014-07-23 @600 static inline acpi_status acpi_remove_pm_notifier(struct acpi_device *adev) ec2cd81ccf Rafael J. Wysocki 2012-11-02 601 { ec2cd81ccf Rafael J. Wysocki 2012-11-02 602 return AE_SUPPORT; ec2cd81ccf Rafael J. Wysocki 2012-11-02 603 } b25c77efa7 Rafael J. Wysocki 2013-06-16 604 static inline int acpi_pm_device_sleep_state(struct device *d, int *p, int m) 296699de6b Rafael J. Wysocki 2007-07-29 605 { 296699de6b Rafael J. Wysocki 2007-07-29 606 if (p) 296699de6b Rafael J. Wysocki 2007-07-29 @607 *p = ACPI_STATE_D0; b25c77efa7 Rafael J. Wysocki 2013-06-16 608 8ad928d52e Rafael J. Wysocki 2013-07-30 609 return (m >= ACPI_STATE_D0 && m <= ACPI_STATE_D3_COLD) ? 8ad928d52e Rafael J. Wysocki 2013-07-30 610 m : ACPI_STATE_D0; :::::: The code at line 598 was first introduced by commit :::::: ec2cd81ccfc055155ef4ca673f207168f516d287 ACPI / PM: Move routines for adding/removing device wakeup notifiers :::::: TO: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> :::::: CC: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] sounds: deadlocked processed in snd_seq_pool_done
by Dmitry Vyukov 28 Feb '17

28 Feb '17

Hello, The following program creates deadlocked processed in snd_seq_pool_done: https://gist.githubusercontent.com/dvyukov/1c9b1e4641f46c16a42cdb4458bbe8fd… # cat /proc/2964/status Name: a.out State: D (disk sleep) Tgid: 2964 Ngid: 0 Pid: 2964 PPid: 2956 TracerPid: 0 Uid: 0 0 0 0 Gid: 0 0 0 0 FDSize: 0 Groups: 0 NStgid: 2964 NSpid: 2964 NSpgid: 2964 NSsid: 2956 Threads: 1 SigQ: 1/3304 SigPnd: 0000000000000000 ShdPnd: 0000000000000100 SigBlk: 0000000000000000 SigIgn: 0000000000000000 SigCgt: 0000000180000000 CapInh: 0000000000000000 CapPrm: 0000003fffffffff CapEff: 0000003fffffffff CapBnd: 0000003fffffffff CapAmb: 0000000000000000 NoNewPrivs: 0 Seccomp: 0 Cpus_allowed: f Cpus_allowed_list: 0-3 Mems_allowed: 00000000,00000001 Mems_allowed_list: 0 voluntary_ctxt_switches: 103997 nonvoluntary_ctxt_switches: 0 # cat /proc/2964/task/*/stack [<ffffffff8356c96b>] snd_seq_pool_done+0x31b/0x620 sound/core/seq/seq_memory.c:435 [<ffffffff83571961>] snd_seq_fifo_delete+0x161/0x1f0 sound/core/seq/seq_fifo.c:83 [<ffffffff83562b4f>] snd_seq_release+0x7f/0xe0 sound/core/seq/seq_clientmgr.c:368 [<ffffffff81a8aae2>] __fput+0x332/0x7f0 fs/file_table.c:208 [<ffffffff81a8b025>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff814c782a>] task_work_run+0x18a/0x260 kernel/task_work.c:116 [<ffffffff81451a86>] exit_task_work include/linux/task_work.h:21 [inline] [<ffffffff81451a86>] do_exit+0x1956/0x2900 kernel/exit.c:873 [<ffffffff81457579>] do_group_exit+0x149/0x420 kernel/exit.c:977 [<ffffffff8145786d>] SYSC_exit_group kernel/exit.c:988 [inline] [<ffffffff8145786d>] SyS_exit_group+0x1d/0x20 kernel/exit.c:986 [<ffffffff8457c781>] entry_SYSCALL_64_fastpath+0x1f/0xc2 [<ffffffffffffffff>] 0xffffffffffffffff

2 1

[alsa-devel] [PATCH 1/5] ALSA: hda/realtek - Add support for Acer Aspire E5-475 headset mic
by Daniel Drake 28 Feb '17

28 Feb '17

From: Chris Chiu <chiu(a)endlessm.com> The Acer laptop Aspire E5-475 with ALC255 can't detect the headset microphone until we modify a pin definition. Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2da176..2b1f74f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4833,6 +4833,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, ALC255_FIXUP_HEADSET_MODE, @@ -5287,6 +5288,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC255_FIXUP_ACER_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5852,6 +5862,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x21, 0x03211020} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE, + {0x12, 0x90a601c0}, + {0x14, 0x90171120}, + {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, -- 2.9.3

2 5

[alsa-devel] [sound:topic/cht-whiskey-cove 1/3] include/acpi/acpi_bus.h:32:21: sparse: expected ; at end of declaration
by kbuild test robot 28 Feb '17

28 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/cht-whiskey-cove head: fe0334695c856dbdbeea861a5eb76f5ddc0dbaa9 commit: fbba8603f93e60c68116197326cdbbceee0b875b [1/3] mfd: Add Cherrytrail WhiskeyCove PMIC driver reproduce: # apt-get install sparse git checkout fbba8603f93e60c68116197326cdbbceee0b875b make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) include/linux/compiler.h:253:8: sparse: attribute 'no_sanitize_address': unknown attribute >> include/acpi/acpi_bus.h:32:21: sparse: expected ; at end of declaration include/acpi/acpi_bus.h:32:21: sparse: Expected } at end of struct-union-enum-specifier include/acpi/acpi_bus.h:32:21: sparse: got handles include/acpi/acpi_bus.h:33:1: sparse: Expected ; at the end of type declaration include/acpi/acpi_bus.h:33:1: sparse: got } include/acpi/acpi_bus.h:37:1: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:37:1: sparse: got acpi_extract_package include/acpi/acpi_bus.h:40:1: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:40:1: sparse: got acpi_evaluate_integer include/acpi/acpi_bus.h:44:1: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:44:1: sparse: got acpi_evaluate_reference include/acpi/acpi_bus.h:49:1: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:49:1: sparse: got acpi_evaluate_ost include/acpi/acpi_bus.h:53:1: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:53:1: sparse: got acpi_get_physical_device_location include/acpi/acpi_bus.h:55:34: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:55:34: sparse: got handle include/acpi/acpi_bus.h:56:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:56:13: sparse: got acpi_execute_simple_method include/acpi/acpi_bus.h:58:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:58:13: sparse: got acpi_evaluate_ej0 include/acpi/acpi_bus.h:59:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:59:13: sparse: got acpi_evaluate_lck include/acpi/acpi_bus.h:60:33: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:60:33: sparse: got handle include/acpi/acpi_bus.h:61:33: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:61:33: sparse: got handle include/acpi/acpi_bus.h:62:34: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:62:34: sparse: got handle include/acpi/acpi_bus.h:64:33: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:64:33: sparse: got handle include/acpi/acpi_bus.h:65:50: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:65:50: sparse: got handle include/acpi/acpi_bus.h:69:37: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:69:37: sparse: got handle include/acpi/acpi_bus.h:75:17: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:75:17: sparse: got && >> include/acpi/acpi_bus.h:75:9: sparse: Trying to use reserved word 'if' as identifier include/acpi/acpi_bus.h:78:9: sparse: Expected ; at the end of type declaration include/acpi/acpi_bus.h:78:9: sparse: got } include/acpi/acpi_bus.h:81:1: sparse: Expected ; at the end of type declaration include/acpi/acpi_bus.h:81:1: sparse: got } include/acpi/acpi_bus.h:326:21: sparse: expected ; at end of declaration include/acpi/acpi_bus.h:326:21: sparse: Expected } at end of struct-union-enum-specifier include/acpi/acpi_bus.h:326:21: sparse: got gpe_device include/acpi/acpi_bus.h:334:1: sparse: Expected ; at the end of type declaration include/acpi/acpi_bus.h:334:1: sparse: got } include/acpi/acpi_bus.h:356:21: sparse: expected ; at end of declaration include/acpi/acpi_bus.h:356:21: sparse: Expected } at end of struct-union-enum-specifier include/acpi/acpi_bus.h:356:21: sparse: got handle include/acpi/acpi_bus.h:382:1: sparse: Expected ; at the end of type declaration include/acpi/acpi_bus.h:382:1: sparse: got } include/acpi/acpi_bus.h:387:21: sparse: expected ; at end of declaration include/acpi/acpi_bus.h:387:21: sparse: Expected } at end of struct-union-enum-specifier include/acpi/acpi_bus.h:387:21: sparse: got handle include/acpi/acpi_bus.h:393:1: sparse: Expected ; at the end of type declaration include/acpi/acpi_bus.h:393:1: sparse: got } include/acpi/acpi_bus.h:478:47: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:478:47: sparse: got , include/acpi/acpi_bus.h:479:42: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:479:42: sparse: got , include/acpi/acpi_bus.h:480:45: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:480:45: sparse: got , include/acpi/acpi_bus.h:490:37: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:490:37: sparse: got handle include/acpi/acpi_bus.h:491:58: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:491:58: sparse: got handle include/acpi/acpi_bus.h:493:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:493:13: sparse: got acpi_bus_get_status_handle include/acpi/acpi_bus.h:497:36: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:497:36: sparse: got handle include/acpi/acpi_bus.h:503:39: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:503:39: sparse: got handle include/acpi/acpi_bus.h:505:44: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:505:44: sparse: got handle include/acpi/acpi_bus.h:508:38: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:508:38: sparse: got handle include/acpi/acpi_bus.h:520:31: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:520:31: sparse: got handle include/acpi/acpi_bus.h:522:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:522:13: sparse: got acpi_bus_get_ejd include/acpi/acpi_bus.h:582:54: sparse: Expected ) in function declarator include/acpi/acpi_bus.h:582:54: sparse: got handle include/acpi/acpi_bus.h:588:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:588:13: sparse: got acpi_add_pm_notifier include/acpi/acpi_bus.h:590:13: sparse: Expected ; at end of declaration include/acpi/acpi_bus.h:590:13: sparse: got acpi_remove_pm_notifier include/acpi/acpi_bus.h:74:33: sparse: undefined identifier 'handle' include/acpi/acpi_bus.h:77:17: sparse: symbol 'obj' has multiple initializers (originally initialized at include/acpi/acpi_bus.h:74) include/acpi/acpi_bus.h:364:34: sparse: symbol 'flags' redeclared with different type (originally declared at include/acpi/acpi_bus.h:330) - different base types include/acpi/acpi_bus.h:286:47: sparse: undefined identifier 'ACPI_D_STATE_COUNT' >> include/acpi/acpi_bus.h:286:47: sparse: bad constant expression type include/acpi/acpi_bus.h:409:17: sparse: no member 'fwnode' in struct acpi_device >> include/acpi/acpi_bus.h:409:17: sparse: unknown member >> include/acpi/acpi_bus.h:409:17: sparse: cast from unknown type >> include/acpi/acpi_bus.h:408:44: sparse: incompatible types in conditional expression (different types) include/linux/acpi.h:51:27: sparse: no member 'handle' in struct acpi_device drivers/mfd/intel_soc_pmic_chtwc.c:104:18: sparse: undefined identifier 'acpi_evaluate_integer' In file included from drivers/mfd/intel_soc_pmic_chtwc.c:18:0: include/acpi/acpi_bus.h:32:2: error: unknown type name 'acpi_handle' acpi_handle handles[ACPI_MAX_HANDLES]; ^~~~~~~~~~~ include/acpi/acpi_bus.h:36:1: error: unknown type name 'acpi_status' acpi_status ^~~~~~~~~~~ include/acpi/acpi_bus.h:38:15: warning: 'struct acpi_buffer' declared inside parameter list will not be visible outside of this definition or declaration struct acpi_buffer *format, struct acpi_buffer *buffer); ^~~~~~~~~~~ include/acpi/acpi_bus.h:37:28: warning: 'union acpi_object' declared inside parameter list will not be visible outside of this definition or declaration acpi_extract_package(union acpi_object *package, ^~~~~~~~~~~ include/acpi/acpi_bus.h:39:1: error: unknown type name 'acpi_status' acpi_status ^~~~~~~~~~~ include/acpi/acpi_bus.h:40:23: error: unknown type name 'acpi_handle' acpi_evaluate_integer(acpi_handle handle, ^~~~~~~~~~~ include/acpi/acpi_bus.h:41:9: error: unknown type name 'acpi_string' acpi_string pathname, ^~~~~~~~~~~ include/acpi/acpi_bus.h:43:1: error: unknown type name 'acpi_status' acpi_status ^~~~~~~~~~~ include/acpi/acpi_bus.h:44:25: error: unknown type name 'acpi_handle' acpi_evaluate_reference(acpi_handle handle, ^~~~~~~~~~~ include/acpi/acpi_bus.h:45:4: error: unknown type name 'acpi_string' acpi_string pathname, ^~~~~~~~~~~ include/acpi/acpi_bus.h:48:1: error: unknown type name 'acpi_status' acpi_status ^~~~~~~~~~~ include/acpi/acpi_bus.h:49:19: error: unknown type name 'acpi_handle' acpi_evaluate_ost(acpi_handle handle, u32 source_event, u32 status_code, ^~~~~~~~~~~ include/acpi/acpi_bus.h:52:1: error: unknown type name 'acpi_status' acpi_status ^~~~~~~~~~~ include/acpi/acpi_bus.h:53:35: error: unknown type name 'acpi_handle' acpi_get_physical_device_location(acpi_handle handle, struct acpi_pld_info **pld); ^~~~~~~~~~~ include/acpi/acpi_bus.h:55:22: error: unknown type name 'acpi_handle' bool acpi_has_method(acpi_handle handle, char *name); ^~~~~~~~~~~ include/acpi/acpi_bus.h:56:1: error: unknown type name 'acpi_status' acpi_status acpi_execute_simple_method(acpi_handle handle, char *method, ^~~~~~~~~~~ include/acpi/acpi_bus.h:56:40: error: unknown type name 'acpi_handle' acpi_status acpi_execute_simple_method(acpi_handle handle, char *method, ^~~~~~~~~~~ include/acpi/acpi_bus.h:58:1: error: unknown type name 'acpi_status' acpi_status acpi_evaluate_ej0(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:58:31: error: unknown type name 'acpi_handle' acpi_status acpi_evaluate_ej0(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:59:1: error: unknown type name 'acpi_status' acpi_status acpi_evaluate_lck(acpi_handle handle, int lock); ^~~~~~~~~~~ include/acpi/acpi_bus.h:59:31: error: unknown type name 'acpi_handle' acpi_status acpi_evaluate_lck(acpi_handle handle, int lock); ^~~~~~~~~~~ include/acpi/acpi_bus.h:60:21: error: unknown type name 'acpi_handle' bool acpi_ata_match(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:61:21: error: unknown type name 'acpi_handle' bool acpi_bay_match(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:62:22: error: unknown type name 'acpi_handle' bool acpi_dock_match(acpi_handle handle); ^~~~~~~~~~~ include/acpi/acpi_bus.h:64:21: error: unknown type name 'acpi_handle' bool acpi_check_dsm(acpi_handle handle, const u8 *uuid, u64 rev, u64 funcs); ^~~~~~~~~~~ include/acpi/acpi_bus.h:65:38: error: unknown type name 'acpi_handle' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, ^~~~~~~~~~~ include/acpi/acpi_bus.h:69:25: error: unknown type name 'acpi_handle' acpi_evaluate_dsm_typed(acpi_handle handle, const u8 *uuid, u64 rev, u64 func, ^~~~~~~~~~~ include/acpi/acpi_bus.h:70:30: error: unknown type name 'acpi_object_type' union acpi_object *argv4, acpi_object_type type) ^~~~~~~~~~~~~~~~ In file included from drivers/mfd/intel_soc_pmic_chtwc.c:18:0: include/acpi/acpi_bus.h:286:40: error: 'ACPI_D_STATE_COUNT' undeclared here (not in a function) struct acpi_device_power_state states[ACPI_D_STATE_COUNT]; /* Power states (D0-D3Cold) */ ^~~~~~~~~~~~~~~~~~ include/acpi/acpi_bus.h:326:2: error: unknown type name 'acpi_handle' acpi_handle gpe_device; ^~~~~~~~~~~ include/acpi/acpi_bus.h:356:2: error: unknown type name 'acpi_handle' acpi_handle handle; /* no handle for fixed hardware */ ^~~~~~~~~~~ include/acpi/acpi_bus.h:387:2: error: unknown type name 'acpi_handle' acpi_handle handle; ^~~~~~~~~~~ vim +32 include/acpi/acpi_bus.h 8a0662d9 Rafael J. Wysocki 2014-11-04 26 #include <linux/property.h> ^1da177e Linus Torvalds 2005-04-16 27 ^1da177e Linus Torvalds 2005-04-16 28 /* TBD: Make dynamic */ ^1da177e Linus Torvalds 2005-04-16 29 #define ACPI_MAX_HANDLES 10 ^1da177e Linus Torvalds 2005-04-16 30 struct acpi_handle_list { ^1da177e Linus Torvalds 2005-04-16 31 u32 count; ^1da177e Linus Torvalds 2005-04-16 @32 acpi_handle handles[ACPI_MAX_HANDLES]; ^1da177e Linus Torvalds 2005-04-16 33 }; ^1da177e Linus Torvalds 2005-04-16 34 ^1da177e Linus Torvalds 2005-04-16 35 /* acpi_utils.h */ ^1da177e Linus Torvalds 2005-04-16 36 acpi_status 4be44fcd Len Brown 2005-08-05 37 acpi_extract_package(union acpi_object *package, 4be44fcd Len Brown 2005-08-05 38 struct acpi_buffer *format, struct acpi_buffer *buffer); ^1da177e Linus Torvalds 2005-04-16 39 acpi_status 4be44fcd Len Brown 2005-08-05 40 acpi_evaluate_integer(acpi_handle handle, ^1da177e Linus Torvalds 2005-04-16 41 acpi_string pathname, 27663c58 Matthew Wilcox 2008-10-10 42 struct acpi_object_list *arguments, unsigned long long *data); ^1da177e Linus Torvalds 2005-04-16 43 acpi_status 4be44fcd Len Brown 2005-08-05 44 acpi_evaluate_reference(acpi_handle handle, ^1da177e Linus Torvalds 2005-04-16 45 acpi_string pathname, ^1da177e Linus Torvalds 2005-04-16 46 struct acpi_object_list *arguments, ^1da177e Linus Torvalds 2005-04-16 47 struct acpi_handle_list *list); 275c58d7 Toshi Kani 2012-05-23 48 acpi_status 05730c19 Jiang Liu 2014-02-19 49 acpi_evaluate_ost(acpi_handle handle, u32 source_event, u32 status_code, 05730c19 Jiang Liu 2014-02-19 50 struct acpi_buffer *status_buf); ^1da177e Linus Torvalds 2005-04-16 51 38ac0f1b Matthew Garrett 2012-05-11 52 acpi_status 8ede06ab Feng Tang 2012-08-21 53 acpi_get_physical_device_location(acpi_handle handle, struct acpi_pld_info **pld); 952c63e9 Jiang Liu 2013-06-29 54 952c63e9 Jiang Liu 2013-06-29 55 bool acpi_has_method(acpi_handle handle, char *name); 0db98202 Jiang Liu 2013-06-29 56 acpi_status acpi_execute_simple_method(acpi_handle handle, char *method, 0db98202 Jiang Liu 2013-06-29 57 u64 arg); 7d2421f8 Jiang Liu 2013-06-29 58 acpi_status acpi_evaluate_ej0(acpi_handle handle); 7d2421f8 Jiang Liu 2013-06-29 59 acpi_status acpi_evaluate_lck(acpi_handle handle, int lock); ebf4df8d Jiang Liu 2013-06-29 60 bool acpi_ata_match(acpi_handle handle); ebf4df8d Jiang Liu 2013-06-29 61 bool acpi_bay_match(acpi_handle handle); ebf4df8d Jiang Liu 2013-06-29 @62 bool acpi_dock_match(acpi_handle handle); 952c63e9 Jiang Liu 2013-06-29 63 c7e16e52 Jerry Hoemann 2016-04-11 64 bool acpi_check_dsm(acpi_handle handle, const u8 *uuid, u64 rev, u64 funcs); a65ac520 Jiang Liu 2013-12-19 65 union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, c7e16e52 Jerry Hoemann 2016-04-11 66 u64 rev, u64 func, union acpi_object *argv4); a65ac520 Jiang Liu 2013-12-19 67 a65ac520 Jiang Liu 2013-12-19 68 static inline union acpi_object * c7e16e52 Jerry Hoemann 2016-04-11 69 acpi_evaluate_dsm_typed(acpi_handle handle, const u8 *uuid, u64 rev, u64 func, a65ac520 Jiang Liu 2013-12-19 70 union acpi_object *argv4, acpi_object_type type) a65ac520 Jiang Liu 2013-12-19 71 { a65ac520 Jiang Liu 2013-12-19 72 union acpi_object *obj; a65ac520 Jiang Liu 2013-12-19 73 a65ac520 Jiang Liu 2013-12-19 74 obj = acpi_evaluate_dsm(handle, uuid, rev, func, argv4); a65ac520 Jiang Liu 2013-12-19 @75 if (obj && obj->type != type) { a65ac520 Jiang Liu 2013-12-19 76 ACPI_FREE(obj); a65ac520 Jiang Liu 2013-12-19 77 obj = NULL; a65ac520 Jiang Liu 2013-12-19 78 } :::::: The code at line 32 was first introduced by commit :::::: 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Linux-2.6.12-rc2 :::::: TO: Linus Torvalds <torvalds(a)ppc970.osdl.org> :::::: CC: Linus Torvalds <torvalds(a)ppc970.osdl.org> --- 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] ALSA: hda - Add subwoofer support for Dell Inspiron 17 7000 Gaming
by Takashi Iwai 28 Feb '17

28 Feb '17

Dell Inspiron 17 7000 Gaming laptop needs a similar quirk like Inspiron 7599 to support its subwoofer speaker. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=194191 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73a00460b5c1..fb03c122e0e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5606,6 +5606,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), 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), -- 2.11.1

1 0

[alsa-devel] [PATCH] ALSA: ctxfi: Fallback DMA mask to 32bit
by Takashi Iwai 28 Feb '17

28 Feb '17

Currently ctxfi driver tries to set only the 64bit DMA mask on 64bit architectures, and bails out if it fails. This causes a problem on some platforms since the 64bit DMA isn't always guaranteed. We should fall back to the default 32bit DMA when 64bit DMA fails. Fixes: 6d74b86d3c0f ("ALSA: ctxfi - Allow 64bit DMA") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ctxfi/cthw20k1.c | 19 ++++++------------- sound/pci/ctxfi/cthw20k2.c | 19 ++++++------------- 2 files changed, 12 insertions(+), 26 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 9667cbfb0ca2..ab4cdab5cfa5 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -27,12 +27,6 @@ #include "cthw20k1.h" #include "ct20k1reg.h" -#if BITS_PER_LONG == 32 -#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */ -#else -#define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */ -#endif - struct hw20k1 { struct hw hw; spinlock_t reg_20k1_lock; @@ -1904,19 +1898,18 @@ static int hw_card_start(struct hw *hw) { int err; struct pci_dev *pci = hw->pci; + const unsigned int dma_bits = BITS_PER_LONG; err = pci_enable_device(pci); if (err < 0) return err; /* Set DMA transfer mask */ - if (dma_set_mask(&pci->dev, CT_XFI_DMA_MASK) < 0 || - dma_set_coherent_mask(&pci->dev, CT_XFI_DMA_MASK) < 0) { - dev_err(hw->card->dev, - "architecture does not support PCI busmaster DMA with mask 0x%llx\n", - CT_XFI_DMA_MASK); - err = -ENXIO; - goto error1; + if (dma_set_mask(&pci->dev, DMA_BIT_MASK(dma_bits))) { + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(dma_bits)); + } else { + dma_set_mask(&pci->dev, DMA_BIT_MASK(32)); + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(32)); } if (!hw->io_base) { diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 6414ecf93efa..18ee7768b7c4 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -26,12 +26,6 @@ #include "cthw20k2.h" #include "ct20k2reg.h" -#if BITS_PER_LONG == 32 -#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */ -#else -#define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */ -#endif - struct hw20k2 { struct hw hw; /* for i2c */ @@ -2029,19 +2023,18 @@ static int hw_card_start(struct hw *hw) int err = 0; struct pci_dev *pci = hw->pci; unsigned int gctl; + const unsigned int dma_bits = BITS_PER_LONG; err = pci_enable_device(pci); if (err < 0) return err; /* Set DMA transfer mask */ - if (dma_set_mask(&pci->dev, CT_XFI_DMA_MASK) < 0 || - dma_set_coherent_mask(&pci->dev, CT_XFI_DMA_MASK) < 0) { - dev_err(hw->card->dev, - "architecture does not support PCI busmaster DMA with mask 0x%llx\n", - CT_XFI_DMA_MASK); - err = -ENXIO; - goto error1; + if (!dma_set_mask(&pci->dev, DMA_BIT_MASK(dma_bits))) { + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(dma_bits)); + } else { + dma_set_mask(&pci->dev, DMA_BIT_MASK(32)); + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(32)); } if (!hw->io_base) { -- 2.11.1

1 0

[alsa-devel] sound: spinlock lockup in snd_timer_user_tinterrupt
by Dmitry Vyukov 28 Feb '17

28 Feb '17

Hello, The following program locks up system: https://gist.githubusercontent.com/dvyukov/1b5cdca7f0fc6254afd4816901160e4c… BUG: spinlock lockup suspected on CPU#1, syz-executor0/6018 lock: kvm_count_lock+0x0/0x60, .magic: dead4ead, .owner: syz-executor0/6001, .owner_cpu: 2 CPU: 1 PID: 6018 Comm: syz-executor0 Not tainted 4.10.0-rc8+ #228 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:15 [inline] dump_stack+0x2ee/0x3ef lib/dump_stack.c:51 spin_dump+0x14d/0x280 kernel/locking/spinlock_debug.c:67 __spin_lock_debug kernel/locking/spinlock_debug.c:117 [inline] do_raw_spin_lock+0x2c1/0x5b0 kernel/locking/spinlock_debug.c:137 __raw_spin_lock include/linux/spinlock_api_smp.h:145 [inline] _raw_spin_lock+0x3b/0x50 kernel/locking/spinlock.c:151 hardware_enable_all arch/x86/kvm/../../../virt/kvm/kvm_main.c:3324 [inline] kvm_create_vm arch/x86/kvm/../../../virt/kvm/kvm_main.c:632 [inline] kvm_dev_ioctl_create_vm arch/x86/kvm/../../../virt/kvm/kvm_main.c:3173 [inline] kvm_dev_ioctl+0x984/0x11b0 arch/x86/kvm/../../../virt/kvm/kvm_main.c:3217 vfs_ioctl fs/ioctl.c:43 [inline] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:683 SYSC_ioctl fs/ioctl.c:698 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:689 entry_SYSCALL_64_fastpath+0x1f/0xc2 RIP: 0033:0x4458b9 RSP: 002b:00007fb0cb7b7b58 EFLAGS: 00000286 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000017 RCX: 00000000004458b9 RDX: 0000000000000000 RSI: 000000000000ae01 RDI: 0000000000000017 RBP: 00000000006dfa10 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000286 R12: 0000000000708150 R13: 0000000020000000 R14: 0000000000fff000 R15: 0000000000000003 NMI backtrace for cpu 1 CPU: 1 PID: 6018 Comm: syz-executor0 Not tainted 4.10.0-rc8+ #228 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:15 [inline] dump_stack+0x2ee/0x3ef lib/dump_stack.c:51 nmi_cpu_backtrace+0xfd/0x120 lib/nmi_backtrace.c:99 nmi_trigger_cpumask_backtrace+0x11d/0x170 lib/nmi_backtrace.c:60 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:37 trigger_all_cpu_backtrace include/linux/nmi.h:58 [inline] __spin_lock_debug kernel/locking/spinlock_debug.c:119 [inline] do_raw_spin_lock+0x2cf/0x5b0 kernel/locking/spinlock_debug.c:137 __raw_spin_lock include/linux/spinlock_api_smp.h:145 [inline] _raw_spin_lock+0x3b/0x50 kernel/locking/spinlock.c:151 hardware_enable_all arch/x86/kvm/../../../virt/kvm/kvm_main.c:3324 [inline] kvm_create_vm arch/x86/kvm/../../../virt/kvm/kvm_main.c:632 [inline] kvm_dev_ioctl_create_vm arch/x86/kvm/../../../virt/kvm/kvm_main.c:3173 [inline] kvm_dev_ioctl+0x984/0x11b0 arch/x86/kvm/../../../virt/kvm/kvm_main.c:3217 vfs_ioctl fs/ioctl.c:43 [inline] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:683 SYSC_ioctl fs/ioctl.c:698 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:689 entry_SYSCALL_64_fastpath+0x1f/0xc2 RIP: 0033:0x4458b9 RSP: 002b:00007fb0cb7b7b58 EFLAGS: 00000286 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000017 RCX: 00000000004458b9 RDX: 0000000000000000 RSI: 000000000000ae01 RDI: 0000000000000017 RBP: 00000000006dfa10 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000286 R12: 0000000000708150 R13: 0000000020000000 R14: 0000000000fff000 R15: 0000000000000003 Sending NMI from CPU 1 to CPUs 0,2-3: NMI backtrace for cpu 0 skipped: idling at pc 0xffffffff84288cb6 NMI backtrace for cpu 2 CPU: 2 PID: 6001 Comm: syz-executor0 Not tainted 4.10.0-rc8+ #228 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff880049720640 task.stack: ffff880024a38000 RIP: 0010:rep_nop arch/x86/include/asm/processor.h:616 [inline] RIP: 0010:cpu_relax arch/x86/include/asm/processor.h:621 [inline] RIP: 0010:csd_lock_wait kernel/smp.c:99 [inline] RIP: 0010:smp_call_function_many+0x75c/0x930 kernel/smp.c:455 RSP: 0018:ffff880024a3f6a0 EFLAGS: 00000246 RAX: 0000000000010000 RBX: ffffe8ffffd052d0 RCX: ffffc90000ab5000 RDX: 0000000000010000 RSI: ffffffff8167f79a RDI: ffffffff84ba1ad8 RBP: ffff880024a3f8c0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000006 R11: 0000000000000000 R12: ffffe8ffffd052e8 R13: dffffc0000000000 R14: 0000000000000003 R15: ffffed0004947eeb FS: 00007fb0cb7fa700(0000) GS:ffff880089e00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000006dfa10 CR3: 0000000051171000 CR4: 00000000000006e0 DR0: 0000000020000000 DR1: 0000000020000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000600 Call Trace: smp_call_function kernel/smp.c:479 [inline] on_each_cpu+0x109/0x330 kernel/smp.c:589 hardware_enable_all arch/x86/kvm/../../../virt/kvm/kvm_main.c:3329 [inline] kvm_create_vm arch/x86/kvm/../../../virt/kvm/kvm_main.c:632 [inline] kvm_dev_ioctl_create_vm arch/x86/kvm/../../../virt/kvm/kvm_main.c:3173 [inline] kvm_dev_ioctl+0xbe2/0x11b0 arch/x86/kvm/../../../virt/kvm/kvm_main.c:3217 vfs_ioctl fs/ioctl.c:43 [inline] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:683 SYSC_ioctl fs/ioctl.c:698 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:689 entry_SYSCALL_64_fastpath+0x1f/0xc2 RIP: 0033:0x4458b9 RSP: 002b:00007fb0cb7f9b58 EFLAGS: 00000286 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000017 RCX: 00000000004458b9 RDX: 0000000000000000 RSI: 000000000000ae01 RDI: 0000000000000017 RBP: 00000000006dfa10 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000286 R12: 0000000000708000 R13: 0000000020000000 R14: 0000000000fff000 R15: 0000000000000003 Code: 3c 03 0f 8e d0 01 00 00 44 89 b5 98 fe ff ff 41 83 e6 01 41 c6 07 f8 74 51 4d 89 e6 41 83 e6 07 41 83 c6 03 e8 66 af 0a 00 f3 90 <4c> 89 e0 41 c6 07 04 48 c1 e8 03 42 0f b6 04 28 41 38 c6 7c 08 NMI backtrace for cpu 3 CPU: 3 PID: 5984 Comm: syz-executor3 Not tainted 4.10.0-rc8+ #228 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88004a566500 task.stack: ffff88002f9f0000 RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:767 [inline] RIP: 0010:lock_release+0x86c/0xe90 kernel/locking/lockdep.c:3775 RSP: 0018:ffff880089f073e8 EFLAGS: 00000082 RAX: dffffc0000000000 RBX: ffff88004a566500 RCX: 1ffff100113e0ea2 RDX: 1ffffffff09d8a95 RSI: 0000000000000001 RDI: 0000000000000082 RBP: ffff880089f07638 R08: ffffe8ffffd32500 R09: 0000000000000001 R10: 1ffff100113e0e6e R11: ffff88009fff70dd R12: ffff880086136d60 R13: 0000000000000001 R14: ffff88004a566d20 R15: 0000000000000002 FS: 00007f366abbe700(0000) GS:ffff880089f00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00005642b47a21e8 CR3: 000000003d684000 CR4: 00000000000006e0 Call Trace: <IRQ> __raw_spin_unlock include/linux/spinlock_api_smp.h:152 [inline] _raw_spin_unlock+0x1a/0x30 kernel/locking/spinlock.c:183 spin_unlock include/linux/spinlock.h:347 [inline] snd_timer_user_tinterrupt+0x54d/0x950 sound/core/timer.c:1322 snd_timer_interrupt.part.12+0xa8f/0x1480 sound/core/timer.c:799 snd_timer_interrupt+0x88/0xb0 sound/core/timer.c:719 snd_hrtimer_callback+0x1f0/0x3c0 sound/core/hrtimer.c:64 __run_hrtimer kernel/time/hrtimer.c:1238 [inline] __hrtimer_run_queues+0x385/0xf80 kernel/time/hrtimer.c:1302 hrtimer_interrupt+0x1c2/0x5e0 kernel/time/hrtimer.c:1336 local_apic_timer_interrupt+0x6f/0xe0 arch/x86/kernel/apic/apic.c:936 smp_apic_timer_interrupt+0x71/0xa0 arch/x86/kernel/apic/apic.c:960 apic_timer_interrupt+0x93/0xa0 arch/x86/entry/entry_64.S:487 RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:767 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:162 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0x144/0x200 kernel/locking/spinlock.c:191 RSP: 0018:ffff88002f9f7568 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff10 RAX: 1ffffffff09d8a95 RBX: 1ffff10005f3eead RCX: 0000000000000006 RDX: 1ffff10005f3eed1 RSI: ffff88004a566cf8 RDI: 0000000000000282 RBP: ffff88002f9f7728 R08: ffffe8ffffd32500 R09: 0000000000000000 R10: 0000000000000006 R11: 0000000000000000 R12: dffffc0000000000 R13: 0000000000000282 R14: ffff88004c077bf0 R15: 0000000000000000 </IRQ> spin_unlock_irqrestore include/linux/spinlock.h:362 [inline] snd_timer_start1+0x1e3/0x950 sound/core/timer.c:480 snd_timer_start+0x5d/0xa0 sound/core/timer.c:589 snd_timer_user_start.isra.7+0xde/0x130 sound/core/timer.c:1823 __snd_timer_user_ioctl sound/core/timer.c:1910 [inline] snd_timer_user_ioctl+0x79d/0x34a0 sound/core/timer.c:1931 vfs_ioctl fs/ioctl.c:43 [inline] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:683 SYSC_ioctl fs/ioctl.c:698 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:689 entry_SYSCALL_64_fastpath+0x1f/0xc2 RIP: 0033:0x4458b9 RSP: 002b:00007f366abbdb58 EFLAGS: 00000286 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000017 RCX: 00000000004458b9 RDX: 0000000000000000 RSI: 00000000000054a0 RDI: 0000000000000017 RBP: 00000000006e0680 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000286 R12: 00000000007080a8 R13: 000000002002f000 R14: 0000000000001000 R15: 0000000000000003 Code: 04 02 f8 48 89 fa 48 c1 ea 03 80 3c 02 00 0f 85 99 03 00 00 48 83 3d fb 94 92 03 00 0f 84 c4 02 00 00 48 8b bd 00 fe ff ff 57 9d <0f> 1f 44 00 00 48 b8 00 00 00 00 00 fc ff df 48 03 85 10 fe ff Or: [ 393.912269] NMI backtrace for cpu 2 [ 393.912271] CPU: 2 PID: 2937 Comm: a.out Not tainted 4.10.0+ #229 [ 393.912274] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 [ 393.912276] task: ffff88006353c400 task.stack: ffff880066da0000 [ 393.912277] RIP: 0010:__lock_acquire+0xc9/0x3430 kernel/locking/lockdep.c:3204 [ 393.912279] RSP: 0018:ffff88006d106f68 EFLAGS: 00000086 [ 393.912282] RAX: ffffed000da20e03 RBX: 1ffff1000da20ea0 RCX: 0000000000000000 [ 393.912285] RDX: dffffc0000000000 RSI: 0000000000000000 RDI: ffff8800681416c8 [ 393.912287] RBP: ffff88006d1074a0 R08: 0000000000000001 R09: 0000000000000001 [ 393.912289] R10: ffff8800681416c8 R11: 0000000000000000 R12: ffff88006d107720 [ 393.912291] R13: ffff88006353c400 R14: 0000000000000000 R15: 0000000000000001 [ 393.912294] FS: 000000000082f880(0000) GS:ffff88006d100000(0000) knlGS:0000000000000000 [ 393.912295] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 393.912298] CR2: 000000002001cfb0 CR3: 000000006ae13000 CR4: 00000000001406e0 [ 393.912299] Call Trace: [ 393.912300] <IRQ> [ 393.912336] lock_acquire+0x2a1/0x630 kernel/locking/lockdep.c:3753 [ 393.912359] __raw_spin_lock include/linux/spinlock_api_smp.h:142 [inline] [ 393.912359] _raw_spin_lock+0x33/0x50 kernel/locking/spinlock.c:151 [ 393.912362] spin_lock include/linux/spinlock.h:299 [inline] [ 393.912362] snd_timer_interrupt.part.12+0xaa0/0x1480 sound/core/timer.c:800 [ 393.912401] snd_timer_interrupt+0x88/0xb0 sound/core/timer.c:719 [ 393.912403] snd_hrtimer_callback+0x1f0/0x3c0 sound/core/hrtimer.c:64 [ 393.912404] __run_hrtimer kernel/time/hrtimer.c:1210 [inline] [ 393.912404] __hrtimer_run_queues+0x349/0xe50 kernel/time/hrtimer.c:1274 [ 393.912422] hrtimer_interrupt+0x1c2/0x5e0 kernel/time/hrtimer.c:1308 [ 393.912424] local_apic_timer_interrupt+0x6f/0xe0 arch/x86/kernel/apic/apic.c:937 [ 393.912425] smp_apic_timer_interrupt+0x71/0xa0 arch/x86/kernel/apic/apic.c:961 [ 393.912427] apic_timer_interrupt+0x93/0xa0 arch/x86/entry/entry_64.S:487 [ 393.912429] RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:778 [inline] [ 393.912429] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] [ 393.912429] RIP: 0010:_raw_spin_unlock_irqrestore+0x144/0x200 kernel/locking/spinlock.c:191 [ 393.912431] RSP: 0018:ffff880066da7568 EFLAGS: 00000286 ORIG_RAX: ffffffffffffff10 [ 393.912436] RAX: 1ffffffff09d8eb9 RBX: 1ffff1000cdb4ead RCX: 0000000000000006 [ 393.912438] RDX: 1ffff1000cdb4ed1 RSI: ffff88006353cbe0 RDI: 0000000000000286 [ 393.912440] RBP: ffff880066da7728 R08: ffff88007fff709c R09: 0000000000000000 [ 393.912443] R10: 0000000000000006 R11: 0000000000000000 R12: dffffc0000000000 [ 393.912445] R13: 0000000000000286 R14: ffff8800681416b0 R15: 0000000000000000 [ 393.912446] </IRQ> [ 393.912467] spin_unlock_irqrestore include/linux/spinlock.h:354 [inline] [ 393.912467] snd_timer_start1+0x1e3/0x950 sound/core/timer.c:480 [ 393.912479] snd_timer_start+0x5d/0xa0 sound/core/timer.c:589 [ 393.912481] snd_timer_user_start.isra.7+0xde/0x130 sound/core/timer.c:1823 [ 393.912482] __snd_timer_user_ioctl sound/core/timer.c:1910 [inline] [ 393.912482] snd_timer_user_ioctl+0x79d/0x34a0 sound/core/timer.c:1931 [ 393.912509] vfs_ioctl fs/ioctl.c:43 [inline] [ 393.912509] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:683 [ 393.912524] SYSC_ioctl fs/ioctl.c:698 [inline] [ 393.912524] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:689 [ 393.912525] entry_SYSCALL_64_fastpath+0x1f/0xc2 [ 393.912527] RIP: 0033:0x434d89 [ 393.912528] RSP: 002b:00007ffd0ad83a48 EFLAGS: 00000217 ORIG_RAX: 0000000000000010 [ 393.912532] RAX: ffffffffffffffda RBX: 0000000000401ae0 RCX: 0000000000434d89 [ 393.912534] RDX: 0000000000000000 RSI: 00000000000054a0 RDI: 0000000000000003 [ 393.912536] RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 [ 393.912538] R10: 0000000000000000 R11: 0000000000000217 R12: 00000000004002b0 [ 393.912540] R13: 0000000000401ae0 R14: 0000000000401b70 R15: 0000000000000000 [ 393.912542] Code: f2 c7 40 28 f2 f2 f2 f2 c7 40 2c 04 f2 f2 f2 c7 40 30 f2 f2 f2 f2 c7 40 34 00 f2 f2 f2 c7 40 38 f2 f2 f2 f2 c7 40 3c 00 f2 f2 f2 <c7> 40 40 f2 f2 f2 f2 c7 40 44 00 f2 f2 f2 c7 40 48 f2 f2 f2 f2 On commit e5d56efc97f8240d0b5d66c03949382b6d7e5570

2 2

[alsa-devel] [PATCH -next] ASoC: hdmi-codec: fix non static symbol warnings
by Wei Yongjun 28 Feb '17

28 Feb '17

From: Wei Yongjun <weiyongjun1(a)huawei.com> Fixes the following sparse warnings: sound/soc/codecs/hdmi-codec.c:34:1: warning: symbol 'hdmi_device_list' was not declared. Should it be static? sound/soc/codecs/hdmi-codec.c:80:33: warning: symbol 'hdmi_codec_stereo_chmaps' was not declared. Should it be static? sound/soc/codecs/hdmi-codec.c:87:33: warning: symbol 'hdmi_codec_8ch_chmaps' was not declared. Should it be static? sound/soc/codecs/hdmi-codec.c:354:6: warning: symbol 'hdmi_codec_eld_chmap' was not declared. Should it be static? Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/hdmi-codec.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index dc6715a..4982c91 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -31,7 +31,7 @@ struct hdmi_device { int cnt; }; #define pos_to_hdmi_device(pos) container_of((pos), struct hdmi_device, list) -LIST_HEAD(hdmi_device_list); +static LIST_HEAD(hdmi_device_list); #define DAI_NAME_SIZE 16 @@ -77,14 +77,14 @@ struct hdmi_codec_cea_spk_alloc { }; /* Channel maps stereo HDMI */ -const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = { +static const struct snd_pcm_chmap_elem hdmi_codec_stereo_chmaps[] = { { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { } }; /* Channel maps for multi-channel playbacks, up to 8 n_ch */ -const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { +static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { { .channels = 2, /* CA_ID 0x00 */ .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, /* CA_ID 0x01 */ @@ -351,7 +351,7 @@ static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) return spk_mask; } -void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp) +static void hdmi_codec_eld_chmap(struct hdmi_codec_priv *hcp) { u8 spk_alloc; unsigned long spk_mask;

3 2

[alsa-devel] dmesg complains with CONFIG_SND_VERBOSE_PROCFS
by Linus Walleij 27 Feb '17

27 Feb '17

If CONFIG_SND_VERBOSE_PROCFS is enabled I got a bunch of nasty messages from duplicate entries when registering the DAIs like this: [ 3.144805] entry->name == "info" [ 3.148132] entry->name == "info" [ 3.151519] entry->name == "hw_params" [ 3.155273] entry->name == "sw_params" [ 3.159027] entry->name == "status" [ 3.164001] entry->name == "prealloc" [ 3.167663] entry->name == "prealloc_max" [ 3.173034] entry->name == "prealloc" [ 3.176696] ------------[ cut here ]------------ [ 3.181335] WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 [ 3.189422] proc_dir_entry 'sub0/prealloc' already registered (...) [ 3.414642] entry->name == "prealloc_max" [ 3.418670] ------------[ cut here ]------------ [ 3.423309] WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 [ 3.431396] proc_dir_entry 'sub0/prealloc_max' already registered (...) [ 3.658020] snd-soc-mop500 soc:sound: ab8500-codec-dai.0 <-> 80124000.msp mapping ok [ 3.665893] entry->name == "info" [ 3.669219] entry->name == "info" [ 3.672546] entry->name == "hw_params" [ 3.676300] entry->name == "sw_params" [ 3.680053] entry->name == "status" [ 3.684906] entry->name == "prealloc" [ 3.688568] entry->name == "prealloc_max" [ 3.693939] entry->name == "prealloc" [ 3.697601] ------------[ cut here ]------------ [ 3.702239] WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 [ 3.710296] proc_dir_entry 'sub0/prealloc' already registered (...) [ 3.936645] entry->name == "prealloc_max" [ 3.940673] ------------[ cut here ]------------ [ 3.945281] WARNING: CPU: 0 PID: 95 at ../fs/proc/generic.c:346 proc_register+0xf0/0x110 [ 3.953369] proc_dir_entry 'sub0/prealloc_max' already registered [ 4.180023] snd-soc-mop500 soc:sound: ab8500-codec-dai.1 <-> 80125000.msp mapping ok What I can't seem to figure out is whether the DAI in my card (sound/soc/ux500/mop500.c) is misconfigured some way, or if CONFIG_SND_VERBOSE_PROCFS is not handling this the right way? If I take out the config option everything seems to work smoothly but I guess it should be possible to have this on? Any hints? Yours, Linus Walleij

2 2

[alsa-devel] [PATCH v1 1/3] ASoC: zx-96p22: add documentation for zte's aud96p22 controller
by Baoyou Xie 27 Feb '17

27 Feb '17

This patch adds dt-binding documentation for zte's aud96p22 controller. Signed-off-by: Baoyou Xie <baoyou.xie(a)linaro.org> --- .../devicetree/bindings/sound/zte,zx-96p22.txt | 24 ++++++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-96p22.txt diff --git a/Documentation/devicetree/bindings/sound/zte,zx-96p22.txt b/Documentation/devicetree/bindings/sound/zte,zx-96p22.txt new file mode 100644 index 0000000..4184566 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zte,zx-96p22.txt @@ -0,0 +1,24 @@ +ZTE zx96p22 controller + +Required properties: + - compatible : Must be "zte,zx-aud96p22" + - #sound-dai-cells: Should be 0 + - reg : Offset of I2C register for zx96p22 + +Example: + + audio_i2c0: audio_i2c0@1486000 { + compatible = "zte,zx296718-i2c"; + reg = <0x01486000 0x1000>; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + #address-cells = <1>; + #size-cells = <0>; + clocks = <&audiocrm AUDIO_I2C0_WCLK>; + clock-frequency = <1600000>; + status = "ok"; + inner_codec: aud96p22@22 { + compatible = "zte,zx-aud96p22"; + #sound-dai-cells = <0>; + reg = <0x22>; + }; + }; -- 2.7.4

4 8

[alsa-devel] [PATCH] ALSA: hda: Add Geminilake HDMI codec ID
by Vinod Koul 27 Feb '17

27 Feb '17

Geminilake HDMI codec 0x280d is similar to previous platforms, so add it with similar ops as previous. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3fc201c3b95a..6254c3ba4473 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3658,6 +3658,7 @@ static int patch_via_hdmi(struct hda_codec *codec) HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hda - Add Geminilake PCI ID
by Vinod Koul 27 Feb '17

27 Feb '17

Geminilake is another Intel part, so need to add PCI ID for it. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 16108f0eb688..c8256a89375a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2255,6 +2255,9 @@ static void azx_shutdown(struct pci_dev *pci) /* Broxton-T */ { PCI_DEVICE(0x8086, 0x1a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, + /* Gemini-Lake */ + { PCI_DEVICE(0x8086, 0x3198), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, -- 1.9.1

2 3

[alsa-devel] [sound:topic/brcfmac-fixes 1/4] drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c:164:2: error: implicit declaration of function 'brcm_err'
by kbuild test robot 27 Feb '17

27 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/brcfmac-fixes head: c61fbd73e341032a7843e6da9dc63414152f7381 commit: e66cb6d7f8abcc90969fa5cf0a87f12954bdee9e [1/4] brcmfmac: Do not print the firmware version as an error config: openrisc-allmodconfig (attached as .config) compiler: or32-linux-gcc (GCC) 4.5.1-or32-1.0rc1 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 e66cb6d7f8abcc90969fa5cf0a87f12954bdee9e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c: In function 'brcmf_c_preinit_dcmds': >> drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c:164:2: error: implicit declaration of function 'brcm_err' vim +/brcm_err +164 drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c 158 goto done; 159 } 160 ptr = (char *)buf; 161 strsep(&ptr, "\n"); 162 163 /* Print fw version info */ > 164 brcmf_info("Firmware version = %s\n", buf); 165 166 /* locate firmware version number for ethtool */ 167 ptr = strrchr(buf, ' ') + 1; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [sound:topic/brcfmac-fixes 1/4] drivers/net/wireless/broadcom/brcm80211/brcmfmac/debug.h:69:20: error: implicit declaration of function 'brcm_err'
by kbuild test robot 27 Feb '17

27 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/brcfmac-fixes head: c61fbd73e341032a7843e6da9dc63414152f7381 commit: e66cb6d7f8abcc90969fa5cf0a87f12954bdee9e [1/4] brcmfmac: Do not print the firmware version as an error config: i386-randconfig-x007-201709 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout e66cb6d7f8abcc90969fa5cf0a87f12954bdee9e # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): In file included from drivers/net/wireless/broadcom/brcm80211/brcmfmac/bus.h:20:0, from drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c:24: drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c: In function 'brcmf_c_preinit_dcmds': >> drivers/net/wireless/broadcom/brcm80211/brcmfmac/debug.h:69:20: error: implicit declaration of function 'brcm_err' [-Werror=implicit-function-declaration] #define brcmf_info brcm_err ^ >> drivers/net/wireless/broadcom/brcm80211/brcmfmac/common.c:164:2: note: in expansion of macro 'brcmf_info' brcmf_info("Firmware version = %s\n", buf); ^~~~~~~~~~ cc1: some warnings being treated as errors vim +/brcm_err +69 drivers/net/wireless/broadcom/brcm80211/brcmfmac/debug.h 63 #else 64 __printf(2, 3) 65 void __brcmf_err(const char *func, const char *fmt, ...); 66 #define brcmf_err(fmt, ...) \ 67 __brcmf_err(__func__, fmt, ##__VA_ARGS__) 68 /* For tracing purposes treat info messages as errors */ > 69 #define brcmf_info brcm_err 70 #endif 71 72 #if defined(DEBUG) || defined(CONFIG_BRCM_TRACING) --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/1] ASoC: Add AU1761 audio codec as selectable option
by Bryce Ferguson 27 Feb '17

27 Feb '17

This commit adds the ADI AU1761 audio codec as a selectable option in the kernel config. Currently the driver can only be selected for ADI blackfin devices or if SND_SOC_ALL_CODECS is enabled. Signed-off-by: Bryce Ferguson <bryce.ferguson(a)rockwellcollins.com> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e49e9da..43539c3b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -303,12 +303,14 @@ config SND_SOC_ADAU1761 select SND_SOC_ADAU17X1 config SND_SOC_ADAU1761_I2C - tristate + tristate "Analog Devices AU1761 CODEC - I2C" + depends on I2C select SND_SOC_ADAU1761 select REGMAP_I2C config SND_SOC_ADAU1761_SPI - tristate + tristate "Analog Devices AU1761 CODEC - SPI" + depends on SPI select SND_SOC_ADAU1761 select REGMAP_SPI -- 2.7.4

1 1

[alsa-devel] [sound:topic/pwm-cherrytrail 12/12] pwm-lpss.c:undefined reference to `__udivdi3'
by kbuild test robot 27 Feb '17

27 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/pwm-cherrytrail head: 6644f014dbc7a6cb6c2e410e60aab183d08a8df7 commit: 6644f014dbc7a6cb6c2e410e60aab183d08a8df7 [12/12] pwm: lpss: Add get_state callback config: i386-randconfig-n0-201709 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 6644f014dbc7a6cb6c2e410e60aab183d08a8df7 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): drivers/built-in.o: In function `pwm_lpss_get_state': >> pwm-lpss.c:(.text+0x20360): undefined reference to `__udivdi3' pwm-lpss.c:(.text+0x203b8): undefined reference to `__udivdi3' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix micmute hotkey problem for a lenovo AIO machine
by Hui Wang 27 Feb '17

27 Feb '17

On this machine, the micmute button is connected to Line2 of the codec and the micmute led is connected to GPIO2 of the codec. After applying this quirk, both hotkey and led work well. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 73a0046..e2da176 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5724,6 +5724,7 @@ enum { SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), + SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 1.9.1

2 1

[alsa-devel] [PATCH] alsa-lib: Add htimestamp operation in plugin file
by sutar.mounesh＠gmail.com 27 Feb '17

27 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> PCM operation htimestamp is not implemented in plugin file. Calling snd_pcm_htimestamp() on a plugin file crashes. This scenario is considered now. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Joshua Frkuska <joshua_frkuska(a)mentor.com> --- --- a/src/pcm/pcm_file.c 2013-07-08 14:31:36.000000000 +0200 +++ b/src/pcm/pcm_file.c 2015-05-04 16:26:10.413615403 +0200 @@ -698,6 +698,7 @@ .readi = snd_pcm_file_readi, .readn = snd_pcm_file_readn, .avail_update = snd_pcm_generic_avail_update, + .htimestamp = snd_pcm_generic_htimestamp, .mmap_commit = snd_pcm_file_mmap_commit, .poll_descriptors_count = snd_pcm_generic_poll_descriptors_count, .poll_descriptors = snd_pcm_generic_poll_descriptors,

4 8

[alsa-devel] [PATCH 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 27 Feb '17

27 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..6d0c03d 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -64,6 +64,8 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) + snd_config_delete((snd_config_t*)plug->rate_converter); assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1110,15 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; + if (rate_converter) { + if ((err = snd_config_copy((snd_config_t**)&plug->rate_converter, + (snd_config_t *)rate_converter)) < 0) + free(plug); + return err; + } + else + snd_config_delete((snd_config_t*)plug->rate_converter); + plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1119,6 +1129,8 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, err = snd_pcm_new(&pcm, SND_PCM_TYPE_PLUG, name, slave->stream, slave->mode); if (err < 0) { + if (plug->rate_converter) + snd_config_delete((snd_config_t*)plug->rate_converter); free(plug); return err; } -- 2.7.4

3 2

[alsa-devel] [PATCH 6/6 V2] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 27 Feb '17

27 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..ad0e99e 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -64,6 +64,8 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) + snd_config_delete((snd_config_t*)plug->rate_converter); assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1110,13 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; + if (rate_converter) { + if ((err = snd_config_copy((snd_config_t**)&plug->rate_converter, + (snd_config_t *)rate_converter)) < 0) + free(plug); + return err; + } + plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1119,6 +1127,8 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, err = snd_pcm_new(&pcm, SND_PCM_TYPE_PLUG, name, slave->stream, slave->mode); if (err < 0) { + if (plug->rate_converter) + snd_config_delete((snd_config_t*)plug->rate_converter); free(plug); return err; } -- 2.7.4

1 0

[alsa-devel] [PATCH v2 0/7][resend] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 27 Feb '17

27 Feb '17

Hi Mark, Rob These are v2 of resend of OF-graph base simple-card patch-set. I added 3 OF-graph helper functions for it. 1) - 3) : new OF-graph helper functions 4) - 7) : OF-graph base simple-card Kuninori Morimoto (7): 1) of_graph: add of_graph_get_remote_endpoint() 2) of_graph: add of_graph_get_port_parent() 3) of_graph: add of_graph_get_endpoint_count() 4) ASoC: add snd_soc_get_dai_id() 5) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 6) ASoC: add simple-graph-card document 7) ASoC: add simple-graph-card support .../bindings/sound/simple-graph-card.txt | 137 ++++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 22 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 7 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-card-utils.c | 36 +++ sound/soc/generic/simple-graph-card.c | 303 +++++++++++++++++++++ sound/soc/soc-core.c | 23 ++ 10 files changed, 591 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-card.txt create mode 100644 sound/soc/generic/simple-graph-card.c -- 1.9.1

2 9

[alsa-devel] [PATCH] ASoC: bcm2835: Add 8 channel (multitrack) capability
by Matt Flax 26 Feb '17

26 Feb '17

This patch adds 8 channel capability by allowing DSP modes. It also increases the channels_max (in the bcm2835_i2s_dai structure) to 8. The driver guards against multitrack mode when the codec isn't master. The driver guards against multitrack capability when in I2S mode. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..022ca13 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -276,11 +276,16 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* otherwise calculate a fitting block ratio */ bclk_ratio = 2 * data_length; - /* Clock should only be set up here if CPU is clock master */ + /* Clock should only be set up here if CPU is clock master. + If codec isn't master guard against multichannel mode. */ switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: case SND_SOC_DAIFMT_CBS_CFM: clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + case SND_SOC_DAIFMT_CBM_CFS: + /* Codec must be master in multichannel mode */ + if (params_channels(params) > 2) + return -EINVAL; break; default: break; @@ -294,8 +299,14 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, format |= BCM2835_I2S_CHWID((data_length-8)&0xf); + /* Default data delay to 1 bit. + In I2S mode, we must have 2 channels */ switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + if (params_channels(params) != 2) + return -EINVAL; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: data_delay = 1; break; default: @@ -312,6 +323,7 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 2: + case 8: format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); @@ -577,7 +589,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .probe = bcm2835_i2s_dai_probe, .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE @@ -585,7 +597,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE -- 2.7.4

9 36

[alsa-devel] [PATCH] ASoC: rockchip: i2s: Remove unnecessary 'out of memory' message
by Mihai-Valentin DUMITRU 26 Feb '17

26 Feb '17

This was reported by checkpatch.pl Signed-off-by: Mihai-Valentin DUMITRU <mihai.dumitru2201(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 974915cb4c4f..d02a1045fc0b 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -551,10 +551,8 @@ static int rockchip_i2s_probe(struct platform_device *pdev) int val; i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); - if (!i2s) { - dev_err(&pdev->dev, "Can't allocate rk_i2s_dev\n"); + if (!i2s) return -ENOMEM; - } i2s->dev = &pdev->dev; -- 2.11.1

2 1

[alsa-devel] [sound:topic/dollar-cove-ti 3/4] drivers/mfd/intel_soc_pmic_bxtwc.c:24:31: fatal error: asm/intel_pmc_ipc.h: No such file or directory
by kbuild test robot 26 Feb '17

26 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dollar-cove-ti head: 8dcecd0695c309e4f53fab087490fbc2217da5bf commit: 2546db2d230a263672858ec1b1ed51d0487a535a [3/4] mfd: intel-soc-pmic: Allow module build config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 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 2546db2d230a263672858ec1b1ed51d0487a535a # save the attached .config to linux build tree make.cross ARCH=sh All errors (new ones prefixed by >>): >> drivers/mfd/intel_soc_pmic_bxtwc.c:24:31: fatal error: asm/intel_pmc_ipc.h: No such file or directory #include <asm/intel_pmc_ipc.h> ^ compilation terminated. vim +24 drivers/mfd/intel_soc_pmic_bxtwc.c 39d047c0 Qipeng Zha 2015-09-15 8 * version 2, as published by the Free Software Foundation. 39d047c0 Qipeng Zha 2015-09-15 9 * 39d047c0 Qipeng Zha 2015-09-15 10 * This program is distributed in the hope it will be useful, but WITHOUT 39d047c0 Qipeng Zha 2015-09-15 11 * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or 39d047c0 Qipeng Zha 2015-09-15 12 * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for 39d047c0 Qipeng Zha 2015-09-15 13 * more details. 39d047c0 Qipeng Zha 2015-09-15 14 */ 39d047c0 Qipeng Zha 2015-09-15 15 39d047c0 Qipeng Zha 2015-09-15 16 #include <linux/module.h> 39d047c0 Qipeng Zha 2015-09-15 17 #include <linux/acpi.h> 39d047c0 Qipeng Zha 2015-09-15 18 #include <linux/err.h> 39d047c0 Qipeng Zha 2015-09-15 19 #include <linux/delay.h> 39d047c0 Qipeng Zha 2015-09-15 20 #include <linux/interrupt.h> 39d047c0 Qipeng Zha 2015-09-15 21 #include <linux/kernel.h> 39d047c0 Qipeng Zha 2015-09-15 22 #include <linux/mfd/core.h> 39d047c0 Qipeng Zha 2015-09-15 23 #include <linux/mfd/intel_bxtwc.h> 39d047c0 Qipeng Zha 2015-09-15 @24 #include <asm/intel_pmc_ipc.h> 39d047c0 Qipeng Zha 2015-09-15 25 39d047c0 Qipeng Zha 2015-09-15 26 /* PMIC device registers */ 39d047c0 Qipeng Zha 2015-09-15 27 #define REG_ADDR_MASK 0xFF00 39d047c0 Qipeng Zha 2015-09-15 28 #define REG_ADDR_SHIFT 8 39d047c0 Qipeng Zha 2015-09-15 29 #define REG_OFFSET_MASK 0xFF 39d047c0 Qipeng Zha 2015-09-15 30 39d047c0 Qipeng Zha 2015-09-15 31 /* Interrupt Status Registers */ 39d047c0 Qipeng Zha 2015-09-15 32 #define BXTWC_IRQLVL1 0x4E02 :::::: The code at line 24 was first introduced by commit :::::: 39d047c0b1c812e9f0014e7100e372e61f2de3de mfd: add Intel Broxton Whiskey Cove PMIC driver :::::: TO: Qipeng Zha <qipeng.zha(a)intel.com> :::::: CC: Lee Jones <lee.jones(a)linaro.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: vx: remove 'out of memory' message
by Mihai Burduselu 26 Feb '17

26 Feb '17

Reported by checkpatch.pl Signed-off-by: Mihai Burduselu <michelcatalin(a)gmail.com> --- sound/drivers/vx/vx_core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 289f041706cd..f684fffd1397 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -795,10 +795,8 @@ struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw, return NULL; chip = kzalloc(sizeof(*chip) + extra_size, GFP_KERNEL); - if (! chip) { - snd_printk(KERN_ERR "vx_core: no memory\n"); + if (! chip) return NULL; - } mutex_init(&chip->lock); chip->irq = -1; chip->hw = hw; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: pxa: The open brace is placed with the previous line
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/raumfeld.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 552b763..47c9173 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -228,14 +228,12 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, .codec_name = "spi0.0", \ } -static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = -{ +static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { DAI_LINK_CS4270, DAI_LINK_AK4104, }; -static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = -{ +static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { DAI_LINK_CS4270, }; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa-ssp: Line up *s in block comments
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 72a5e87..f3285a6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -592,13 +592,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ if (ssp->type != PXA3xx_SSP) return -EINVAL; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa: Remove unneeded else after return statement
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/pxa2xx-ac97.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 2e2fb18..f49bf02 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -140,9 +140,8 @@ static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, - &pxa2xx_ac97_pcm_mic_mono_in); + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); return 0; } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: sh: Remove redundant out of memory message
by Cristian Done 25 Feb '17

25 Feb '17

Signed-off-by: Cristian Done <done.cristian(a)gmail.com> --- sound/soc/sh/fsi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ead520182e26..6793dc359114 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1964,7 +1964,6 @@ static int fsi_probe(struct platform_device *pdev) master = devm_kzalloc(&pdev->dev, sizeof(*master), GFP_KERNEL); if (!master) { - dev_err(&pdev->dev, "Could not allocate master\n"); return -ENOMEM; } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: pxa: Added */ to the next line
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/hx4700.c | 3 ++- sound/soc/pxa/pxa-ssp.c | 3 ++- sound/soc/pxa/pxa2xx-pcm.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 8548304..17c2e67 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -70,7 +70,8 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, return ret; /* inform codec driver about clock freq * - * (PXA I2S always uses divider 256) */ + * (PXA I2S always uses divider 256) + */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params), SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3cad990..3bdace6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -254,7 +254,8 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* The SSP clock must be disabled when changing SSP clock mode - * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + * on PXA2xx. On PXA3xx it must be enabled when doing so. + */ if (ssp->type != PXA3xx_SSP) clk_disable_unprepare(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index b51d7a0..3926cd8 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -31,7 +31,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ + * codec <--> BT codec or GSM modem -- lg + */ if (!dma) return 0; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa: Remove unneeded return statement in void function
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/mmp-pcm.c | 1 - sound/soc/pxa/mmp-sspa.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 96df9b2..5b5f1a4 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -166,7 +166,6 @@ static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) buf->area = NULL; } - return; } static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index ca8b23f..9cc3501 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -119,7 +119,6 @@ static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, clk_disable(priv->sspa->clk); clk_disable(priv->sysclk); - return; } /* -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa2xx-i2s: Remove unneeded init of static variable
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/pxa2xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60ba..744edc6 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -80,7 +80,7 @@ struct pxa_i2s_port { }; static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; -static int clk_ena = 0; +static int clk_ena; static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa: Remove spaces before tabs
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/pxa2xx-i2s.c | 6 +++--- sound/soc/pxa/pxa2xx-pcm.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 27aa0e4..3fb60ba 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -46,10 +46,10 @@ #define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */ #define SACR0_EFWR (1 << 4) /* Enable EFWR Function */ #define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */ -#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ +#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ #define SACR0_ENB (1 << 0) /* Enable I2S Link */ #define SACR1_ENLBF (1 << 5) /* Enable Loopback */ -#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ +#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ #define SACR1_DREC (1 << 3) /* Disable Recording Function */ #define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */ @@ -60,7 +60,7 @@ #define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */ #define SASR0_BSY (1 << 2) /* I2S Busy */ #define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */ -#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ +#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ #define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */ #define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */ diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 410d48b..b51d7a0 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -85,7 +85,7 @@ static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) } static struct snd_soc_platform_driver pxa2xx_soc_platform = { - .ops = &pxa2xx_pcm_ops, + .ops = &pxa2xx_pcm_ops, .pcm_new = pxa2xx_soc_pcm_new, .pcm_free = pxa2xx_pcm_free_dma_buffers, }; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa: Add space around ':' and '('
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/tosa.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 0389cf7..27aa0e4 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -119,7 +119,7 @@ static int pxa_i2s_wait(void) int i; /* flush the Rx FIFO */ - for(i = 0; i < 16; i++) + for (i = 0; i < 16; i++) SADR; return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index e022b2a..1812748 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -133,7 +133,7 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol, static int tosa_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0); + gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); return 0; } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: pxa: Remove space before semicolon
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/pxa/spitz.c | 4 ++-- sound/soc/pxa/z2.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 07d77cd..d38a2b5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -230,8 +230,8 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget *w, {"Headset Jack", NULL, "ROUT1"}, /* ext speaker connected to LOUT2, ROUT2 */ - {"Ext Spk", NULL , "ROUT2"}, - {"Ext Spk", NULL , "LOUT2"}, + {"Ext Spk", NULL, "ROUT2"}, + {"Ext Spk", NULL, "LOUT2"}, /* mic is connected to input 1 - with bias */ {"LINPUT1", NULL, "Mic Bias"}, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 990b1aa..6d88d9a 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -119,8 +119,8 @@ static int z2_hw_params(struct snd_pcm_substream *substream, {"Headphone Jack", NULL, "ROUT1"}, /* ext speaker connected to LOUT2, ROUT2 */ - {"Ext Spk", NULL , "ROUT2"}, - {"Ext Spk", NULL , "LOUT2"}, + {"Ext Spk", NULL, "ROUT2"}, + {"Ext Spk", NULL, "LOUT2"}, /* mic is connected to R input 2 - with bias */ {"RINPUT2", NULL, "Mic Bias"}, -- 1.9.1

1 0

[alsa-devel] [PATCH] SoC: nuc900: Remove space before semicolon
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/nuc900/nuc900-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index b6615af..1cacfb4 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -76,7 +76,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, goto out; } - val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ; + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); val &= ~SLOT1_VALID; AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: ux500: Added blank line after declarations
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/ux500/ux500_msp_dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 36e9bb2..ec5152a 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -133,6 +133,7 @@ static int setup_pcm_framing(struct snd_soc_dai *dai, unsigned int rate, struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); u32 frame_length = MSP_FRAME_LEN_1; + prot_desc->frame_width = 0; switch (drvdata->slots) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: ux500: Remove unuseful break after return
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/ux500/ux500_msp_i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 959d7b4..bd5266a 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -604,7 +604,6 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) break; default: return -EINVAL; - break; } return 0; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: tegra: Remove unnecessary 'Block comments should align the * on each line' message
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut Grosu <codrut.cristian.grosu(a)gmail.com> --- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 0509902..1f648f9 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,5 +1,5 @@ /* -* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver + * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver * * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> * Copyright (C) 2012 - NVIDIA, Inc. diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 93a3568..cf706d0 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -1,5 +1,5 @@ /* -* tegra_rt5640.c - Tegra machine ASoC driver for boards using RT5640 codec. + * tegra_rt5640.c - Tegra machine ASoC driver for boards using RT5640 codec. * * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. * diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index ebf58d0..0ed2eca 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -1,5 +1,5 @@ /* -* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. * * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. * 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: Add a blank line after declarations
by Anda-Alexandra Dorneanu 25 Feb '17

25 Feb '17

From: andadrn <andadrn(a)gmail.com> This was reported by checkpatch.pl Signed-off-by: Anda-Alexandra Dorneanu <andadrn(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index dafd22e..0af5d46 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -24,6 +24,7 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; + return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai); } @@ -91,6 +92,7 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { const struct mbus_dram_window *cs = dram->cs + i; + if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 1.9.1

2 1

[alsa-devel] [PATCH] davinci: davinci-evm: Put spaces before '='
by sergiu.weisz 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: sergiu.weisz <sergiu121(a)gmail.com> --- sound/soc/davinci/davinci-evm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7a369e0..c5d80e8 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -195,7 +195,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", .stream_name = "AIC3X", - .cpu_dai_name= "davinci-mcasp.0", + .cpu_dai_name = "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "davinci-mcasp.0", .codec_name = "tlv320aic3x-codec.0-001a", @@ -207,7 +207,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "McASP", .stream_name = "spdif", - .cpu_dai_name= "davinci-mcasp.1", + .cpu_dai_name = "davinci-mcasp.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif_dit", .platform_name = "davinci-mcasp.1", @@ -232,7 +232,7 @@ static struct snd_soc_dai_link da830_evm_dai = { static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", - .cpu_dai_name= "davinci-mcasp.0", + .cpu_dai_name = "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-mcasp.0", -- 2.9.3

1 0

[alsa-devel] [PATCH] ASoC: sun4i-codec: Remove spaces before tabs.
by Codrut Grosu 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Codrut GROSU <codrut.cristian.grosu(a)gmail.com> --- sound/soc/sunxi/sun4i-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index c3aab10..7ba6849 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1139,8 +1139,8 @@ static int sun4i_codec_dai_probe(struct snd_soc_dai *dai) .stream_name = "Capture", .channels_min = 1, .channels_max = 2, - .rates = SUN4I_CODEC_RATES, - .formats = SUN4I_CODEC_FORMATS, + .rates = SUN4I_CODEC_RATES, + .formats = SUN4I_CODEC_FORMATS, .sig_bits = 24, }, }; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: mxs-saif: Removing whitespace before a pointer and its name
by Camelia Moise 25 Feb '17

25 Feb '17

Signed-off-by: Camelia Moise <camelia.moise21(a)gmail.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b42f301..729249a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -78,7 +78,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * from its master_id. * Note that the master could be itself. */ -static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif *saif) { return mxs_saif[saif->master_id]; } -- 2.7.4

1 0

[alsa-devel] [PATCH] soc: xtensa: Replace "unsigned" with "unsigned int"
by Corina Palade 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl. Signed-off-by: Corina Palade <corina.palade103(a)gmail.com> --- sound/soc/xtensa/xtfpga-i2s.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 8382ffa..01a2622 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -76,22 +76,22 @@ struct xtfpga_i2s { * handler by means of synchronize_rcu call. */ struct snd_pcm_substream __rcu *tx_substream; - unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + unsigned int (*tx_fn)(struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, - unsigned tx_ptr); - unsigned tx_ptr; /* next frame index in the sample buffer */ + unsigned int tx_ptr); + unsigned int tx_ptr; /* next frame index in the sample buffer */ /* current fifo level estimate. * Doesn't have to be perfectly accurate, but must be not less than * the actual FIFO level in order to avoid stall on push attempt. */ - unsigned tx_fifo_level; + unsigned int tx_fifo_level; /* FIFO level at which level interrupt occurs */ - unsigned tx_fifo_low; + unsigned int tx_fifo_low; /* maximal FIFO level */ - unsigned tx_fifo_high; + unsigned int tx_fifo_high; }; static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) @@ -130,9 +130,9 @@ static const struct regmap_config xtfpga_i2s_regmap_config = { * the LSB of each word is used. */ #define xtfpga_pcm_tx_fn(channels, sample_bits) \ -static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ +static unsigned int xtfpga_pcm_tx_##channels##x##sample_bits( \ struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ - unsigned tx_ptr) \ + unsigned int tx_ptr) \ { \ const u##sample_bits (*p)[channels] = \ (void *)runtime->dma_area; \ @@ -165,8 +165,8 @@ static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) tx_substream = rcu_dereference(i2s->tx_substream); tx_active = tx_substream && snd_pcm_running(tx_substream); if (tx_active) { - unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); - unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + unsigned int tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned int new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, tx_ptr); cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); @@ -178,8 +178,8 @@ static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) { - unsigned int_status; - unsigned i; + unsigned int int_status; + unsigned int i; regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); @@ -227,7 +227,7 @@ static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) { struct xtfpga_i2s *i2s = dev_id; struct snd_pcm_substream *tx_substream; - unsigned config, int_status, int_mask; + unsigned int config, int_status, int_mask; regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); @@ -284,11 +284,11 @@ static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned srate = params_rate(params); - unsigned channels = params_channels(params); - unsigned period_size = params_period_size(params); - unsigned sample_size = snd_pcm_format_width(params_format(params)); - unsigned freq, ratio, level; + unsigned int srate = params_rate(params); + unsigned int channels = params_channels(params); + unsigned int period_size = params_period_size(params); + unsigned int sample_size = snd_pcm_format_width(params_format(params)); + unsigned int freq, ratio, level; int err; regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, @@ -393,7 +393,7 @@ static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, int ret; struct snd_pcm_runtime *runtime = substream->runtime; struct xtfpga_i2s *i2s = runtime->private_data; - unsigned channels = params_channels(hw_params); + unsigned int channels = params_channels(hw_params); switch (channels) { case 1: -- 2.7.4

1 0

[alsa-devel] [PATCH] soc: atmel: Add line after declaration.
by Banu Calin 25 Feb '17

25 Feb '17

This was reported by checkpatch.pl Signed-off-by: Banu Calin <banucaliniulian(a)gmail.com> --- sound/atmel/ac97c.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 6dad042..4a0a48e 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -95,12 +95,14 @@ struct atmel_ac97c { static void atmel_ac97c_dma_playback_period_done(void *arg) { struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->playback_substream); } static void atmel_ac97c_dma_capture_period_done(void *arg) { struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->capture_substream); } @@ -297,6 +299,7 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (cpu_is_at32ap7000()) { if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_free(chip->dma.tx_chan); @@ -307,6 +310,7 @@ static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (cpu_is_at32ap7000()) { if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_free(chip->dma.rx_chan); @@ -659,6 +663,7 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) if (sr & AC97C_SR_CAEVT) { struct snd_pcm_runtime *runtime; int offset, next_period, block_size; + dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", @@ -804,6 +809,7 @@ static int atmel_ac97c_pcm_new(struct atmel_ac97c *chip) static int atmel_ac97c_mixer_new(struct atmel_ac97c *chip) { struct snd_ac97_template template; + memset(&template, 0, sizeof(template)); template.private_data = chip; return snd_ac97_mixer(chip->ac97_bus, &template, &chip->ac97); -- 2.7.4

1 0

[alsa-devel] [PATCH] sound: soc: generic: Removed unnecessary blank line
by Andrei Hirgau 25 Feb '17

25 Feb '17

Reported by checkpatch.pl Signed-off-by: Andrei Hirgau <hirgauandrei(a)gmail.com> --- sound/soc/generic/simple-card.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 85b4f18..53bcf95 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -448,7 +448,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) priv->snd_card.num_links = num; if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv); if (ret < 0) { if (ret != -EPROBE_DEFER) -- 2.7.4

1 0

[alsa-devel] [PATCH] soc: blackfin: remove 'out of memory' message
by Mihai Burduselu 25 Feb '17

25 Feb '17

reported by checkpatch.pl Signed-off-by: Mihai Burduselu <michelcatalin(a)gmail.com> --- sound/soc/blackfin/bf5xx-sport.c | 8 ++------ sound/soc/blackfin/bf6xx-sport.c | 4 +--- 2 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 9dfa1241ea66..c445d6cc875e 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -876,10 +876,8 @@ struct sport_device *sport_init(struct platform_device *pdev, } sport = kzalloc(sizeof(*sport), GFP_KERNEL); - if (!sport) { - dev_err(dev, "failed to allocate for sport device\n"); + if (!sport) goto __init_err0; - } sport->num = param.num; sport->dma_rx_chan = param.dma_rx_chan; @@ -921,10 +919,8 @@ struct sport_device *sport_init(struct platform_device *pdev, sport->dummy_count = param.dummy_count; sport->private_data = kzalloc(priv_size, GFP_KERNEL); - if (!sport->private_data) { - dev_err(dev, "could not alloc priv data %zu bytes\n", priv_size); + if (!sport->private_data) goto __init_err4; - } if (L1_DATA_A_LENGTH) sport->dummy_buf = l1_data_sram_zalloc(param.dummy_count * 2); diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index dfb744381c42..b56ff75d4c13 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -388,10 +388,8 @@ struct sport_device *sport_create(struct platform_device *pdev) int ret; sport = kzalloc(sizeof(*sport), GFP_KERNEL); - if (!sport) { - dev_err(dev, "Unable to allocate memory for sport device\n"); + if (!sport) return NULL; - } sport->pdev = pdev; ret = sport_get_resource(sport); -- 2.11.0

1 0

[alsa-devel] [sound:topic/dollar-cove-ti 1/3] undefined reference to `intel_soc_pmic_config_dc_ti'
by kbuild test robot 24 Feb '17

24 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/dollar-cove-ti head: 86509a5113f54303e3f4d20161f025492f8f8a2a commit: f93a261ed16864f3e96140952e6822040feaea68 [1/3] mfd: intel_soc_pmic: Add support for Dollar Cove TI PMIC config: i386-randconfig-h0-02250144 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout f93a261ed16864f3e96140952e6822040feaea68 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): drivers/built-in.o: In function `lkdtm_rodata_do_nothing': >> (.rodata+0x2d844): undefined reference to `intel_soc_pmic_config_dc_ti' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/9] tools subsystem refcounter conversions
by Elena Reshetova 24 Feb '17

24 Feb '17

Now when new refcount_t type and API are finally merged (see include/linux/refcount.h), the following patches convert various refcounters in the tools susystem from atomic_t to refcount_t. By doing this we prevent intentional or accidental underflows or overflows that can led to use-after-free vulnerabilities. The below patches are fully independent and can be cherry-picked separately. Since we convert all kernel subsystems in the same fashion, resulting in about 300 patches, we have to group them for sending at least in some fashion to be manageable. Please excuse the long cc list. Elena Reshetova (9): tools: convert cgroup_sel.refcnt from atomic_t to refcount_t tools: convert cpu_map.refcnt from atomic_t to refcount_t tools: convert comm_str.refcnt from atomic_t to refcount_t tools: convert dso.refcnt from atomic_t to refcount_t tools: convert map.refcnt from atomic_t to refcount_t tools: convert map_groups.refcnt from atomic_t to refcount_t tools: convert perf_map.refcnt from atomic_t to refcount_t tools: convert thread.refcnt from atomic_t to refcount_t tools: convert thread_map.refcnt from atomic_t to refcount_t tools/perf/util/cgroup.c | 6 +++--- tools/perf/util/cgroup.h | 4 ++-- tools/perf/util/comm.c | 13 +++++-------- tools/perf/util/cpumap.c | 16 ++++++++-------- tools/perf/util/cpumap.h | 4 ++-- tools/perf/util/dso.c | 6 +++--- tools/perf/util/dso.h | 4 ++-- tools/perf/util/evlist.c | 18 +++++++++--------- tools/perf/util/evlist.h | 4 ++-- tools/perf/util/map.c | 10 +++++----- tools/perf/util/map.h | 10 +++++----- tools/perf/util/thread.c | 6 +++--- tools/perf/util/thread.h | 4 ++-- tools/perf/util/thread_map.c | 20 ++++++++++---------- tools/perf/util/thread_map.h | 4 ++-- 15 files changed, 63 insertions(+), 66 deletions(-) -- 2.7.4

4 35

[alsa-devel] [sound:topic/asus-e100h-4.10 5/26] arch/x86/include/asm/iosf_mbi.h:137:55: warning: 'struct notifier_block' declared inside parameter list will not be visible outside of this definition or declaration
by kbuild test robot 24 Feb '17

24 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.10 head: c86c0067e6111324a39f523c250c8b1089b1d822 commit: d97e0b9238852206e9dce1de994ff8d2791f6429 [5/26] x86/platform/intel/iosf_mbi: Add a PMIC bus access notifier config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout d97e0b9238852206e9dce1de994ff8d2791f6429 # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): In file included from arch/x86/platform/intel-quark/imr.c:30:0: >> arch/x86/include/asm/iosf_mbi.h:137:55: warning: 'struct notifier_block' declared inside parameter list will not be visible outside of this definition or declaration int iosf_mbi_register_pmic_bus_access_notifier(struct notifier_block *nb); ^~~~~~~~~~~~~~ arch/x86/include/asm/iosf_mbi.h:144:57: warning: 'struct notifier_block' declared inside parameter list will not be visible outside of this definition or declaration int iosf_mbi_unregister_pmic_bus_access_notifier(struct notifier_block *nb); ^~~~~~~~~~~~~~ vim +137 arch/x86/include/asm/iosf_mbi.h 121 122 /** 123 * iosf_mbi_register_pmic_bus_access_notifier - Register PMIC bus notifier 124 * 125 * This function can be used by drivers which may need to acquire P-Unit 126 * managed resources from interrupt context, where iosf_mbi_punit_acquire() 127 * can not be used. 128 * 129 * This function allows a driver to register a notifier to get notified (in a 130 * process context) before other drivers start accessing the PMIC bus. 131 * 132 * This allows the driver to acquire any resources, which it may need during 133 * the window the other driver is accessing the PMIC, before hand. 134 * 135 * @nb: notifier_block to register 136 */ > 137 int iosf_mbi_register_pmic_bus_access_notifier(struct notifier_block *nb); 138 139 /** 140 * iosf_mbi_register_pmic_bus_access_notifier - Unregister PMIC bus notifier 141 * 142 * @nb: notifier_block to unregister 143 */ 144 int iosf_mbi_unregister_pmic_bus_access_notifier(struct notifier_block *nb); 145 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] [sound:topic/asus-e100h-4.10 18/26] drivers/clk/x86/clk-pmc-atom.c:156:29: error: storage size of 'plt_clk_ops' isn't known
by kbuild test robot 23 Feb '17

23 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.10 head: c86c0067e6111324a39f523c250c8b1089b1d822 commit: 19a80107e5cf1d2d02b678bb086f305a05e4f692 [18/26] clk: x86: Add Atom PMC platform clocks config: x86_64-randconfig-b0-02240418 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 19a80107e5cf1d2d02b678bb086f305a05e4f692 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): ^~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:141:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^~~~~~~~~~ include/linux/kernel.h:850:48: note: (near initialization for 'clk') const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:141:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_is_enabled': include/linux/kernel.h:850:48: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:148:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^~~~~~~~~~ include/linux/kernel.h:850:48: note: (near initialization for 'clk') const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:148:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c: At top level: drivers/clk/x86/clk-pmc-atom.c:156:21: error: variable 'plt_clk_ops' has initializer but incomplete type static const struct clk_ops plt_clk_ops = { ^~~~~~~ drivers/clk/x86/clk-pmc-atom.c:157:2: error: unknown field 'enable' specified in initializer .enable = plt_clk_enable, ^ drivers/clk/x86/clk-pmc-atom.c:157:12: warning: excess elements in struct initializer .enable = plt_clk_enable, ^~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:157:12: note: (near initialization for 'plt_clk_ops') drivers/clk/x86/clk-pmc-atom.c:158:2: error: unknown field 'disable' specified in initializer .disable = plt_clk_disable, ^ drivers/clk/x86/clk-pmc-atom.c:158:13: warning: excess elements in struct initializer .disable = plt_clk_disable, ^~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:158:13: note: (near initialization for 'plt_clk_ops') drivers/clk/x86/clk-pmc-atom.c:159:2: error: unknown field 'is_enabled' specified in initializer .is_enabled = plt_clk_is_enabled, ^ drivers/clk/x86/clk-pmc-atom.c:159:16: warning: excess elements in struct initializer .is_enabled = plt_clk_is_enabled, ^~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:159:16: note: (near initialization for 'plt_clk_ops') drivers/clk/x86/clk-pmc-atom.c:160:2: error: unknown field 'get_parent' specified in initializer .get_parent = plt_clk_get_parent, ^ drivers/clk/x86/clk-pmc-atom.c:160:16: warning: excess elements in struct initializer .get_parent = plt_clk_get_parent, ^~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:160:16: note: (near initialization for 'plt_clk_ops') drivers/clk/x86/clk-pmc-atom.c:161:2: error: unknown field 'set_parent' specified in initializer .set_parent = plt_clk_set_parent, ^ drivers/clk/x86/clk-pmc-atom.c:161:16: warning: excess elements in struct initializer .set_parent = plt_clk_set_parent, ^~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:161:16: note: (near initialization for 'plt_clk_ops') drivers/clk/x86/clk-pmc-atom.c:162:2: error: unknown field 'determine_rate' specified in initializer .determine_rate = __clk_mux_determine_rate, ^ drivers/clk/x86/clk-pmc-atom.c:162:20: error: '__clk_mux_determine_rate' undeclared here (not in a function) .determine_rate = __clk_mux_determine_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:162:20: warning: excess elements in struct initializer drivers/clk/x86/clk-pmc-atom.c:162:20: note: (near initialization for 'plt_clk_ops') drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_register': drivers/clk/x86/clk-pmc-atom.c:171:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ drivers/clk/x86/clk-pmc-atom.c:188:8: error: implicit declaration of function 'devm_clk_hw_register' [-Werror=implicit-function-declaration] ret = devm_clk_hw_register(&pdev->dev, &pclk->hw); ^~~~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:171:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_register_fixed_rate': drivers/clk/x86/clk-pmc-atom.c:221:14: error: implicit declaration of function 'clk_hw_register_fixed_rate' [-Werror=implicit-function-declaration] pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, ^~~~~~~~~~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c:221:12: warning: assignment makes pointer from integer without a cast [-Wint-conversion] pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, ^ drivers/clk/x86/clk-pmc-atom.c:228:3: error: implicit declaration of function 'clk_hw_unregister_fixed_rate' [-Werror=implicit-function-declaration] clk_hw_unregister_fixed_rate(pclk->clk); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ drivers/clk/x86/clk-pmc-atom.c: At top level: >> drivers/clk/x86/clk-pmc-atom.c:156:29: error: storage size of 'plt_clk_ops' isn't known static const struct clk_ops plt_clk_ops = { ^~~~~~~~~~~ cc1: some warnings being treated as errors vim +156 drivers/clk/x86/clk-pmc-atom.c 150 151 value = readl(clk->reg); 152 153 return plt_reg_to_enabled(value); 154 } 155 > 156 static const struct clk_ops plt_clk_ops = { 157 .enable = plt_clk_enable, 158 .disable = plt_clk_disable, 159 .is_enabled = plt_clk_is_enabled, 160 .get_parent = plt_clk_get_parent, 161 .set_parent = plt_clk_set_parent, 162 .determine_rate = __clk_mux_determine_rate, 163 }; 164 165 static struct clk_plt *plt_clk_register(struct platform_device *pdev, int id, 166 void __iomem *base, 167 const char **parent_names, 168 int num_parents) 169 { 170 struct clk_plt *pclk; 171 struct clk_init_data init; 172 int ret; 173 174 pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); 175 if (!pclk) 176 return ERR_PTR(-ENOMEM); 177 178 init.name = kasprintf(GFP_KERNEL, "%s_%d", PLT_CLK_NAME_BASE, id); 179 init.ops = &plt_clk_ops; 180 init.flags = 0; 181 init.parent_names = parent_names; 182 init.num_parents = num_parents; 183 184 pclk->hw.init = &init; 185 pclk->reg = base + PMC_CLK_CTL_OFFSET + id * PMC_CLK_CTL_SIZE; 186 spin_lock_init(&pclk->lock); 187 188 ret = devm_clk_hw_register(&pdev->dev, &pclk->hw); 189 if (ret) { 190 pclk = ERR_PTR(ret); 191 goto err_free_init; 192 } 193 194 pclk->lookup = clkdev_hw_create(&pclk->hw, init.name, NULL); 195 if (!pclk->lookup) { 196 pclk = ERR_PTR(-ENOMEM); 197 goto err_free_init; 198 } 199 200 err_free_init: 201 kfree(init.name); 202 return pclk; 203 } 204 205 static void plt_clk_unregister(struct clk_plt *pclk) 206 { 207 clkdev_drop(pclk->lookup); 208 } 209 210 static struct clk_plt_fixed *plt_clk_register_fixed_rate(struct platform_device *pdev, 211 const char *name, 212 const char *parent_name, 213 unsigned long fixed_rate) 214 { 215 struct clk_plt_fixed *pclk; 216 217 pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); 218 if (!pclk) 219 return ERR_PTR(-ENOMEM); 220 > 221 pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, 222 0, fixed_rate); 223 if (IS_ERR(pclk->clk)) 224 return ERR_CAST(pclk->clk); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [sound:topic/asus-e100h-4.10 18/26] drivers/clk/x86/clk-pmc-atom.c:44:16: error: field 'hw' has incomplete type
by kbuild test robot 23 Feb '17

23 Feb '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asus-e100h-4.10 head: c86c0067e6111324a39f523c250c8b1089b1d822 commit: 19a80107e5cf1d2d02b678bb086f305a05e4f692 [18/26] clk: x86: Add Atom PMC platform clocks config: i386-randconfig-i1-02231015 (attached as .config) compiler: gcc-4.8 (Debian 4.8.4-1) 4.8.4 reproduce: git checkout 19a80107e5cf1d2d02b678bb086f305a05e4f692 # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): >> drivers/clk/x86/clk-pmc-atom.c:44:16: error: field 'hw' has incomplete type struct clk_hw hw; ^ In file included from include/asm-generic/bug.h:13:0, from arch/x86/include/asm/bug.h:35, from include/linux/bug.h:4, from include/linux/io.h:23, from include/linux/clk-provider.h:14, from drivers/clk/x86/clk-pmc-atom.c:17: drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_set_parent': include/linux/kernel.h:850:48: warning: initialization from incompatible pointer type [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:113:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ include/linux/kernel.h:850:48: warning: (near initialization for 'clk') [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:113:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_get_parent': include/linux/kernel.h:850:48: warning: initialization from incompatible pointer type [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:122:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ include/linux/kernel.h:850:48: warning: (near initialization for 'clk') [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:122:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_enable': include/linux/kernel.h:850:48: warning: initialization from incompatible pointer type [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:132:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ include/linux/kernel.h:850:48: warning: (near initialization for 'clk') [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:132:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_disable': include/linux/kernel.h:850:48: warning: initialization from incompatible pointer type [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:141:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ include/linux/kernel.h:850:48: warning: (near initialization for 'clk') [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:141:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_is_enabled': include/linux/kernel.h:850:48: warning: initialization from incompatible pointer type [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:148:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ include/linux/kernel.h:850:48: warning: (near initialization for 'clk') [enabled by default] const typeof( ((type *)0)->member ) *__mptr = (ptr); \ ^ drivers/clk/x86/clk-pmc-atom.c:51:25: note: in expansion of macro 'container_of' #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) ^ drivers/clk/x86/clk-pmc-atom.c:148:24: note: in expansion of macro 'to_clk_plt' struct clk_plt *clk = to_clk_plt(hw); ^ drivers/clk/x86/clk-pmc-atom.c: At top level: >> drivers/clk/x86/clk-pmc-atom.c:156:21: error: variable 'plt_clk_ops' has initializer but incomplete type static const struct clk_ops plt_clk_ops = { ^ >> drivers/clk/x86/clk-pmc-atom.c:157:2: error: unknown field 'enable' specified in initializer .enable = plt_clk_enable, ^ >> drivers/clk/x86/clk-pmc-atom.c:157:2: warning: excess elements in struct initializer [enabled by default] drivers/clk/x86/clk-pmc-atom.c:157:2: warning: (near initialization for 'plt_clk_ops') [enabled by default] >> drivers/clk/x86/clk-pmc-atom.c:158:2: error: unknown field 'disable' specified in initializer .disable = plt_clk_disable, ^ drivers/clk/x86/clk-pmc-atom.c:158:2: warning: excess elements in struct initializer [enabled by default] drivers/clk/x86/clk-pmc-atom.c:158:2: warning: (near initialization for 'plt_clk_ops') [enabled by default] >> drivers/clk/x86/clk-pmc-atom.c:159:2: error: unknown field 'is_enabled' specified in initializer .is_enabled = plt_clk_is_enabled, ^ drivers/clk/x86/clk-pmc-atom.c:159:2: warning: excess elements in struct initializer [enabled by default] drivers/clk/x86/clk-pmc-atom.c:159:2: warning: (near initialization for 'plt_clk_ops') [enabled by default] >> drivers/clk/x86/clk-pmc-atom.c:160:2: error: unknown field 'get_parent' specified in initializer .get_parent = plt_clk_get_parent, ^ drivers/clk/x86/clk-pmc-atom.c:160:2: warning: excess elements in struct initializer [enabled by default] drivers/clk/x86/clk-pmc-atom.c:160:2: warning: (near initialization for 'plt_clk_ops') [enabled by default] >> drivers/clk/x86/clk-pmc-atom.c:161:2: error: unknown field 'set_parent' specified in initializer .set_parent = plt_clk_set_parent, ^ drivers/clk/x86/clk-pmc-atom.c:161:2: warning: excess elements in struct initializer [enabled by default] drivers/clk/x86/clk-pmc-atom.c:161:2: warning: (near initialization for 'plt_clk_ops') [enabled by default] >> drivers/clk/x86/clk-pmc-atom.c:162:2: error: unknown field 'determine_rate' specified in initializer .determine_rate = __clk_mux_determine_rate, ^ >> drivers/clk/x86/clk-pmc-atom.c:162:20: error: '__clk_mux_determine_rate' undeclared here (not in a function) .determine_rate = __clk_mux_determine_rate, ^ drivers/clk/x86/clk-pmc-atom.c:162:2: warning: excess elements in struct initializer [enabled by default] .determine_rate = __clk_mux_determine_rate, ^ drivers/clk/x86/clk-pmc-atom.c:162:2: warning: (near initialization for 'plt_clk_ops') [enabled by default] drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_register': >> drivers/clk/x86/clk-pmc-atom.c:171:23: error: storage size of 'init' isn't known struct clk_init_data init; ^ >> drivers/clk/x86/clk-pmc-atom.c:188:2: error: implicit declaration of function 'devm_clk_hw_register' [-Werror=implicit-function-declaration] ret = devm_clk_hw_register(&pdev->dev, &pclk->hw); ^ drivers/clk/x86/clk-pmc-atom.c:171:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^ drivers/clk/x86/clk-pmc-atom.c: In function 'plt_clk_register_fixed_rate': >> drivers/clk/x86/clk-pmc-atom.c:221:2: error: implicit declaration of function 'clk_hw_register_fixed_rate' [-Werror=implicit-function-declaration] pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, ^ >> drivers/clk/x86/clk-pmc-atom.c:221:12: warning: assignment makes pointer from integer without a cast [enabled by default] pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, ^ >> drivers/clk/x86/clk-pmc-atom.c:228:3: error: implicit declaration of function 'clk_hw_unregister_fixed_rate' [-Werror=implicit-function-declaration] clk_hw_unregister_fixed_rate(pclk->clk); ^ cc1: some warnings being treated as errors vim +/hw +44 drivers/clk/x86/clk-pmc-atom.c 38 struct clk_plt_fixed { 39 struct clk_hw *clk; 40 struct clk_lookup *lookup; 41 }; 42 43 struct clk_plt { > 44 struct clk_hw hw; 45 void __iomem *reg; 46 struct clk_lookup *lookup; 47 /* protect access to PMC registers */ 48 spinlock_t lock; 49 }; 50 > 51 #define to_clk_plt(_hw) container_of(_hw, struct clk_plt, hw) 52 53 struct clk_plt_data { 54 struct clk_plt_fixed **parents; 55 u8 nparents; 56 struct clk_plt *clks[PMC_CLK_NUM]; 57 }; 58 59 /* Return an index in parent table */ 60 static inline int plt_reg_to_parent(int reg) 61 { 62 switch (reg & PMC_MASK_CLK_FREQ) { 63 default: 64 case PMC_CLK_FREQ_XTAL: 65 return 0; 66 case PMC_CLK_FREQ_PLL: 67 return 1; 68 } 69 } 70 71 /* Return clk index of parent */ 72 static inline int plt_parent_to_reg(int index) 73 { 74 switch (index) { 75 default: 76 case 0: 77 return PMC_CLK_FREQ_XTAL; 78 case 1: 79 return PMC_CLK_FREQ_PLL; 80 } 81 } 82 83 /* Abstract status in simpler enabled/disabled value */ 84 static inline int plt_reg_to_enabled(int reg) 85 { 86 switch (reg & PMC_MASK_CLK_CTL) { 87 case PMC_CLK_CTL_GATED_ON_D3: 88 case PMC_CLK_CTL_FORCE_ON: 89 return 1; /* enabled */ 90 case PMC_CLK_CTL_FORCE_OFF: 91 case PMC_CLK_CTL_RESERVED: 92 default: 93 return 0; /* disabled */ 94 } 95 } 96 97 static void plt_clk_reg_update(struct clk_plt *clk, u32 mask, u32 val) 98 { 99 u32 tmp; 100 unsigned long flags; 101 102 spin_lock_irqsave(&clk->lock, flags); 103 104 tmp = readl(clk->reg); 105 tmp = (tmp & ~mask) | (val & mask); 106 writel(tmp, clk->reg); 107 108 spin_unlock_irqrestore(&clk->lock, flags); 109 } 110 111 static int plt_clk_set_parent(struct clk_hw *hw, u8 index) 112 { 113 struct clk_plt *clk = to_clk_plt(hw); 114 115 plt_clk_reg_update(clk, PMC_MASK_CLK_FREQ, plt_parent_to_reg(index)); 116 117 return 0; 118 } 119 120 static u8 plt_clk_get_parent(struct clk_hw *hw) 121 { 122 struct clk_plt *clk = to_clk_plt(hw); 123 u32 value; 124 125 value = readl(clk->reg); 126 127 return plt_reg_to_parent(value); 128 } 129 130 static int plt_clk_enable(struct clk_hw *hw) 131 { 132 struct clk_plt *clk = to_clk_plt(hw); 133 134 plt_clk_reg_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_ON); 135 136 return 0; 137 } 138 139 static void plt_clk_disable(struct clk_hw *hw) 140 { 141 struct clk_plt *clk = to_clk_plt(hw); 142 143 plt_clk_reg_update(clk, PMC_MASK_CLK_CTL, PMC_CLK_CTL_FORCE_OFF); 144 } 145 146 static int plt_clk_is_enabled(struct clk_hw *hw) 147 { > 148 struct clk_plt *clk = to_clk_plt(hw); 149 u32 value; 150 151 value = readl(clk->reg); 152 153 return plt_reg_to_enabled(value); 154 } 155 > 156 static const struct clk_ops plt_clk_ops = { > 157 .enable = plt_clk_enable, > 158 .disable = plt_clk_disable, > 159 .is_enabled = plt_clk_is_enabled, > 160 .get_parent = plt_clk_get_parent, > 161 .set_parent = plt_clk_set_parent, > 162 .determine_rate = __clk_mux_determine_rate, 163 }; 164 165 static struct clk_plt *plt_clk_register(struct platform_device *pdev, int id, 166 void __iomem *base, 167 const char **parent_names, 168 int num_parents) 169 { 170 struct clk_plt *pclk; > 171 struct clk_init_data init; 172 int ret; 173 174 pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); 175 if (!pclk) 176 return ERR_PTR(-ENOMEM); 177 178 init.name = kasprintf(GFP_KERNEL, "%s_%d", PLT_CLK_NAME_BASE, id); 179 init.ops = &plt_clk_ops; 180 init.flags = 0; 181 init.parent_names = parent_names; 182 init.num_parents = num_parents; 183 184 pclk->hw.init = &init; 185 pclk->reg = base + PMC_CLK_CTL_OFFSET + id * PMC_CLK_CTL_SIZE; 186 spin_lock_init(&pclk->lock); 187 > 188 ret = devm_clk_hw_register(&pdev->dev, &pclk->hw); 189 if (ret) { 190 pclk = ERR_PTR(ret); 191 goto err_free_init; 192 } 193 194 pclk->lookup = clkdev_hw_create(&pclk->hw, init.name, NULL); 195 if (!pclk->lookup) { 196 pclk = ERR_PTR(-ENOMEM); 197 goto err_free_init; 198 } 199 200 err_free_init: 201 kfree(init.name); 202 return pclk; 203 } 204 205 static void plt_clk_unregister(struct clk_plt *pclk) 206 { 207 clkdev_drop(pclk->lookup); 208 } 209 210 static struct clk_plt_fixed *plt_clk_register_fixed_rate(struct platform_device *pdev, 211 const char *name, 212 const char *parent_name, 213 unsigned long fixed_rate) 214 { 215 struct clk_plt_fixed *pclk; 216 217 pclk = devm_kzalloc(&pdev->dev, sizeof(*pclk), GFP_KERNEL); 218 if (!pclk) 219 return ERR_PTR(-ENOMEM); 220 > 221 pclk->clk = clk_hw_register_fixed_rate(&pdev->dev, name, parent_name, 222 0, fixed_rate); 223 if (IS_ERR(pclk->clk)) 224 return ERR_CAST(pclk->clk); 225 226 pclk->lookup = clkdev_hw_create(pclk->clk, name, NULL); 227 if (!pclk->lookup) { > 228 clk_hw_unregister_fixed_rate(pclk->clk); 229 return ERR_PTR(-ENOMEM); 230 } 231 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] [PATCH 00/35] treewide trivial patches converting pr_warning to pr_warn
by Rob Herring 23 Feb '17

23 Feb '17

On Fri, Feb 17, 2017 at 1:11 AM, Joe Perches <joe(a)perches.com> wrote: > There are ~4300 uses of pr_warn and ~250 uses of the older > pr_warning in the kernel source tree. > > Make the use of pr_warn consistent across all kernel files. > > This excludes all files in tools/ as there is a separate > define pr_warning for that directory tree and pr_warn is > not used in tools/. > > Done with 'sed s/\bpr_warning\b/pr_warn/' and some emacsing. > > Miscellanea: > > o Coalesce formats and realign arguments > > Some files not compiled - no cross-compilers > > Joe Perches (35): > alpha: Convert remaining uses of pr_warning to pr_warn > ARM: ep93xx: Convert remaining uses of pr_warning to pr_warn > arm64: Convert remaining uses of pr_warning to pr_warn > arch/blackfin: Convert remaining uses of pr_warning to pr_warn > ia64: Convert remaining use of pr_warning to pr_warn > powerpc: Convert remaining uses of pr_warning to pr_warn > sh: Convert remaining uses of pr_warning to pr_warn > sparc: Convert remaining use of pr_warning to pr_warn > x86: Convert remaining uses of pr_warning to pr_warn > drivers/acpi: Convert remaining uses of pr_warning to pr_warn > block/drbd: Convert remaining uses of pr_warning to pr_warn > gdrom: Convert remaining uses of pr_warning to pr_warn > drivers/char: Convert remaining use of pr_warning to pr_warn > clocksource: Convert remaining use of pr_warning to pr_warn > drivers/crypto: Convert remaining uses of pr_warning to pr_warn > fmc: Convert remaining use of pr_warning to pr_warn > drivers/gpu: Convert remaining uses of pr_warning to pr_warn > drivers/ide: Convert remaining uses of pr_warning to pr_warn > drivers/input: Convert remaining uses of pr_warning to pr_warn > drivers/isdn: Convert remaining uses of pr_warning to pr_warn > drivers/macintosh: Convert remaining uses of pr_warning to pr_warn > drivers/media: Convert remaining use of pr_warning to pr_warn > drivers/mfd: Convert remaining uses of pr_warning to pr_warn > drivers/mtd: Convert remaining uses of pr_warning to pr_warn > drivers/of: Convert remaining uses of pr_warning to pr_warn > drivers/oprofile: Convert remaining uses of pr_warning to pr_warn > drivers/platform: Convert remaining uses of pr_warning to pr_warn > drivers/rapidio: Convert remaining use of pr_warning to pr_warn > drivers/scsi: Convert remaining use of pr_warning to pr_warn > drivers/sh: Convert remaining use of pr_warning to pr_warn > drivers/tty: Convert remaining uses of pr_warning to pr_warn > drivers/video: Convert remaining uses of pr_warning to pr_warn > kernel/trace: Convert remaining uses of pr_warning to pr_warn > lib: Convert remaining uses of pr_warning to pr_warn > sound/soc: Convert remaining uses of pr_warning to pr_warn Where's the removal of pr_warning so we don't have more sneak in? Rob

3 3

[alsa-devel] [Bug report] dmix on armv8
by Rémi Pincent 23 Feb '17

23 Feb '17

Here is my alsa config : http://www.alsa-project.org/db/?f=d8c96ba81442526650f277e0b7d43dfb6fd668a4 On raspberry I cannot use dmix plugin, when I run a speaker-test with a very simple config using dmix (refer given config in alsa result), its stays stuck. When I use default alsa config (direct use of hw:0,0 without dmix), mixing is done as said here : NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs. Here are some users having same problem http://raspberrypi.stackexchange.com/questions/57787/using-the-alsa-dmix-pl… . I'm not sure it is an alsa problem. But alsa developers could help solving this issue. regards lah

1 0

[alsa-devel] [PATCH] sound: convert snd_seq_subscribers.ref_count from atomic_t to refcount_t
by Elena Reshetova 23 Feb '17

23 Feb '17

refcount_t type and corresponding API should be used instead of atomic_t when the variable is used as a reference counter. This allows to avoid accidental refcounter overflows that might lead to use-after-free situations. Signed-off-by: Elena Reshetova <elena.reshetova(a)intel.com> Signed-off-by: Hans Liljestrand <ishkamiel(a)gmail.com> Signed-off-by: Kees Cook <keescook(a)chromium.org> Signed-off-by: David Windsor <dwindsor(a)gmail.com> --- sound/core/seq/seq_clientmgr.c | 2 +- sound/core/seq/seq_ports.c | 7 +++---- sound/core/seq/seq_ports.h | 3 ++- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 4c93520..3440f76 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -666,7 +666,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, down_read(&grp->list_mutex); list_for_each_entry(subs, &grp->list_head, src_list) { /* both ports ready? */ - if (atomic_read(&subs->ref_count) != 2) + if (refcount_read(&subs->ref_count) != 2) continue; event->dest = subs->info.dest; if (subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index fe686ee..1ca896b 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -241,7 +241,7 @@ static void clear_subscriber_list(struct snd_seq_client *client, * we decrease the counter, and when both ports are deleted * remove the subscriber info */ - if (atomic_dec_and_test(&subs->ref_count)) + if (refcount_dec_and_test(&subs->ref_count)) kfree(subs); continue; } @@ -520,7 +520,6 @@ static int check_and_subscribe_port(struct snd_seq_client *client, else list_add_tail(&subs->dest_list, &grp->list_head); grp->exclusive = exclusive; - atomic_inc(&subs->ref_count); write_unlock_irq(&grp->list_lock); err = 0; @@ -570,7 +569,6 @@ int snd_seq_port_connect(struct snd_seq_client *connector, return -ENOMEM; subs->info = *info; - atomic_set(&subs->ref_count, 0); INIT_LIST_HEAD(&subs->src_list); INIT_LIST_HEAD(&subs->dest_list); @@ -587,6 +585,7 @@ int snd_seq_port_connect(struct snd_seq_client *connector, if (err < 0) goto error_dest; + refcount_set(&subs->ref_count, 2); return 0; error_dest: @@ -613,7 +612,7 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, /* look for the connection */ list_for_each_entry(subs, &src->list_head, src_list) { if (match_subs_info(info, &subs->info)) { - atomic_dec(&subs->ref_count); /* mark as not ready */ + refcount_dec(&subs->ref_count); /* mark as not ready */ err = 0; break; } diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h index 26bd71f..d09e396 100644 --- a/sound/core/seq/seq_ports.h +++ b/sound/core/seq/seq_ports.h @@ -21,6 +21,7 @@ #ifndef __SND_SEQ_PORTS_H #define __SND_SEQ_PORTS_H +#include <linux/refcount.h> #include <sound/seq_kernel.h> #include "seq_lock.h" @@ -44,7 +45,7 @@ struct snd_seq_subscribers { struct snd_seq_port_subscribe info; /* additional info */ struct list_head src_list; /* link of sources */ struct list_head dest_list; /* link of destinations */ - atomic_t ref_count; + refcount_t ref_count; }; struct snd_seq_port_subs_info { -- 2.7.4

3 2

[alsa-devel] [PATCH v2 4/8] sound: soc: pxa: Suppress SND_MMP_SOC for arm64
by Andreas Färber 23 Feb '17

23 Feb '17

SND_MMP_SOC selects SND_ARM, which is limited to ARM, as well as MMP_SRAM, which is a non-DT driver in arch/arm/mach-arm/. Avoid Kconfig warnings by not making this driver available on ARM64. While at it, tidy the title (Soc -> SoC). Signed-off-by: Andreas Färber <afaerber(a)suse.de> --- v2: new sound/soc/pxa/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 823b5a2..f8e1c06 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,8 +8,8 @@ config SND_PXA2XX_SOC to select the audio interfaces to support below. config SND_MMP_SOC - bool "Soc Audio for Marvell MMP chips" - depends on ARCH_MMP + bool "SoC Audio for Marvell MMP chips" + depends on ARCH_MMP && ARM select MMP_SRAM select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM -- 2.10.2

3 4

[alsa-devel] [PATCH] aserver: fix building on ppc64le
by leitao＠debian.org 23 Feb '17

23 Feb '17

From: Breno Leitao <leitao(a)debian.org> Currently aserver fails to build when using parameter --without-versioned, due to an workaround for ppc64 (06221f86d207cb33ddd4867ca5301eeb247c4400). This workaround is not required anymore on the ppc64 ABI v2, and, in fact is breaking the compilation. Reverting this commit Signed-off-by: Breno Leitao <leitao(a)debian.org> --- include/alsa-symbols.h | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/include/alsa-symbols.h b/include/alsa-symbols.h index 51cb982..eeacfdb 100644 --- a/include/alsa-symbols.h +++ b/include/alsa-symbols.h @@ -29,19 +29,10 @@ #define INTERNAL_CONCAT2_2(Pre, Post) Pre##Post #define INTERNAL(Name) INTERNAL_CONCAT2_2(__, Name) -#ifdef __powerpc64__ -# define symbol_version(real, name, version) \ - __asm__ (".symver " ASM_NAME(#real) "," ASM_NAME(#name) "@" #version); \ - __asm__ (".symver ." ASM_NAME(#real) ",." ASM_NAME(#name) "@" #version) -# define default_symbol_version(real, name, version) \ - __asm__ (".symver " ASM_NAME(#real) "," ASM_NAME(#name) "@@" #version); \ - __asm__ (".symver ." ASM_NAME(#real) ",." ASM_NAME(#name) "@@" #version) -#else # define symbol_version(real, name, version) \ __asm__ (".symver " ASM_NAME(#real) "," ASM_NAME(#name) "@" #version) # define default_symbol_version(real, name, version) \ __asm__ (".symver " ASM_NAME(#real) "," ASM_NAME(#name) "@@" #version) -#endif #ifdef USE_VERSIONED_SYMBOLS #define use_symbol_version(real, name, version) \ @@ -50,13 +41,6 @@ default_symbol_version(real, name, version) #else #define use_symbol_version(real, name, version) /* nothing */ -#ifdef __powerpc64__ -#define use_default_symbol_version(real, name, version) \ - __asm__ (".weak " ASM_NAME(#name)); \ - __asm__ (".weak ." ASM_NAME(#name)); \ - __asm__ (".set " ASM_NAME(#name) "," ASM_NAME(#real)); \ - __asm__ (".set ." ASM_NAME(#name) ",." ASM_NAME(#real)) -#else #if defined(__alpha__) || defined(__mips__) #define use_default_symbol_version(real, name, version) \ __asm__ (".weak " ASM_NAME(#name)); \ @@ -67,6 +51,5 @@ __asm__ (".set " ASM_NAME(#name) "," ASM_NAME(#real)) #endif #endif -#endif #endif /* __ALSA_SYMBOLS_H */ -- 1.9.3

2 1

[alsa-devel] [bug report] ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk
by Dan Carpenter 22 Feb '17

22 Feb '17

Hello Detlef Urban, This is a semi-automatic email about new static checker warnings. The patch d2bb390a2081: "ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk" from Feb 20, 2017, leads to the following Smatch complaint: sound/usb/mixer_us16x08.c:716 snd_us16x08_meter_get() warn: variable dereferenced before check 'elem' (see line 711) sound/usb/mixer_us16x08.c 710 struct usb_mixer_elem_info *elem = kcontrol->private_data; 711 struct snd_usb_audio *chip = elem->head.mixer->chip; ^^^^^^^^^^^^^^^^^^^^^^ Dereference. 712 struct snd_us16x08_meter_store *store = elem->private_data; 713 u8 meter_urb[64]; 714 char tmp[max(sizeof(mix_init_msg1), sizeof(mix_init_msg2))]; 715 716 if (elem) { ^^^^ Check. 717 store = (struct snd_us16x08_meter_store *) elem->private_data; 718 chip = elem->head.mixer->chip; regards, dan carpenter

2 1

[alsa-devel] [bug report] ASoC: hdac_hdmi: Add support for multiple ports to a PCM
by Dan Carpenter 22 Feb '17

22 Feb '17

Hello Jeeja KP, The patch e0e5d3e5a53b: "ASoC: hdac_hdmi: Add support for multiple ports to a PCM" from Feb 7, 2017, leads to the following static checker warning: sound/soc/codecs/hdac_hdmi.c:2011 hdac_hdmi_dev_remove() error: potential NULL dereference 'port'. sound/soc/codecs/hdac_hdmi.c 1995 static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) 1996 { 1997 struct hdac_hdmi_priv *hdmi = edev->private_data; 1998 struct hdac_hdmi_pin *pin, *pin_next; 1999 struct hdac_hdmi_cvt *cvt, *cvt_next; 2000 struct hdac_hdmi_pcm *pcm, *pcm_next; 2001 struct hdac_hdmi_port *port; 2002 int i; 2003 2004 snd_soc_unregister_codec(&edev->hdac.dev); 2005 2006 list_for_each_entry_safe(pcm, pcm_next, &hdmi->pcm_list, head) { 2007 pcm->cvt = NULL; 2008 if (list_empty(&pcm->port_list)) 2009 continue; 2010 2011 list_for_each_entry(port, &pcm->port_list, head) 2012 port = NULL; It's not clear to me what this loop is supposed to do. To me it looks like it will Oops every single time without fail. 2013 2014 list_del(&pcm->head); 2015 kfree(pcm); 2016 } 2017 2018 list_for_each_entry_safe(cvt, cvt_next, &hdmi->cvt_list, head) { 2019 list_del(&cvt->head); 2020 kfree(cvt->name); 2021 kfree(cvt); 2022 } 2023 2024 list_for_each_entry_safe(pin, pin_next, &hdmi->pin_list, head) { 2025 for (i = 0; i < pin->num_ports; i++) 2026 pin->ports[i].pin = NULL; 2027 kfree(pin->ports); 2028 list_del(&pin->head); 2029 kfree(pin); 2030 } 2031 2032 return 0; 2033 } regards, dan carpenter

1 0

[alsa-devel] [PATCH 0/5] ALSA: usb-audio: fixes for quirks of Tascam US-16x08
by Takashi Sakamoto 22 Feb '17

22 Feb '17

Hi, A commit d2bb390a2081 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk") was merged just now, while static code checker reports some apparent bugs. This patchset is to solve the bugs. Takashi Sakamoto (5): ALSA: usb-audio: localize one-referer variable ALSA: usb-audio: purge needless variable length array ALSA: usb-audio: localize function without external linkage ALSA: usb-audio: deallocate memory objects in error path ALSA: usb-audio: fix msleep timeout due to minimum resolution of task scheduling sound/usb/mixer_us16x08.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) -- 2.9.3

2 21

[alsa-devel] Custom USB sound card configuration
by Stephen Wilson 22 Feb '17

22 Feb '17

Hello, I'm developing an application using the pjSip library/framework using the ALSA library as the interface to the PCM audio devices. The audio device is a USB PIC presenting a single sound card with 4 devices (PIC code is still in development so ignore the examples names) - see output of aplay -l below... When I launch my pjSip application (and the pjSIP test application) only the first device on my sound card is listed - see below.... My thoughts are that I need a custom card configuration file for my sound card so all four devices are present as specific devices but I cannot work out how to do this from the ALSA Project documentation. I assume I need a new entry in the /usr/share/alsa/cards/aliases.conf file and an equivalent [example].conf file. What needs to go into both of these files? Any help would be greatly appreciated... Thanks Steve [root@##########]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 3: USB Audio [USB Audio #3] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@################]# pjsystest-i686-pc-linux-gnu 09:15:12.615 os_core_unix.c !pjlib 2.5.5 for POSIX initialized 09:15:12.621 sip_endpoint.c .Creating endpoint instance... 09:15:12.623 pjlib .select() I/O Queue created (0xa20aa00) 09:15:12.623 sip_endpoint.c .Module "mod-msg-print" registered 09:15:12.623 sip_transport. .Transport manager created. 09:15:12.624 pjsua_core.c .PJSUA state changed: NULL --> CREATED 09:15:12.624 sip_endpoint.c .Module "mod-pjsua-log" registered 09:15:12.624 sip_endpoint.c .Module "mod-tsx-layer" registered 09:15:12.624 sip_endpoint.c .Module "mod-stateful-util" registered 09:15:12.624 sip_endpoint.c .Module "mod-ua" registered 09:15:12.625 sip_endpoint.c .Module "mod-100rel" registered 09:15:12.625 sip_endpoint.c .Module "mod-pjsua" registered 09:15:12.625 sip_endpoint.c .Module "mod-invite" registered 09:15:12.897 alsa_dev.c ..ALSA driver found 18 devices 09:15:12.897 alsa_dev.c ..ALSA initialized 09:15:12.898 pjlib ..select() I/O Queue created (0xa23b4fc) WARNING: no real random source present! 09:15:12.910 sip_endpoint.c .Module "mod-evsub" registered 09:15:12.910 sip_endpoint.c .Module "mod-presence" registered 09:15:12.910 sip_endpoint.c .Module "mod-mwi" registered 09:15:12.910 sip_endpoint.c .Module "mod-refer" registered 09:15:12.910 sip_endpoint.c .Module "mod-pjsua-pres" registered 09:15:12.910 sip_endpoint.c .Module "mod-pjsua-im" registered 09:15:12.910 sip_endpoint.c .Module "mod-pjsua-options" registered 09:15:12.910 pjsua_core.c .1 SIP worker threads created 09:15:12.910 pjsua_core.c .pjsua version 2.5.5 for Linux-2.6.34.130/i686/glibc-2.12 initialized 09:15:12.910 pjsua_core.c .PJSUA state changed: CREATED --> INIT 09:15:12.910 pjsua_core.c PJSUA state changed: INIT --> STARTING 09:15:12.910 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 09:15:12.911 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING M E N U : --------- 0: Tests 00: Run test wizard 01: Device Test 02: Play Tone 03: Play WAV File1 04: Play WAV File2 05: Record Audio 06: Latency Test 07: AEC/AES Test 09: Exit 1: Options 10: View Devices 11: View Settings Enter the menu number: 10 09:15:13.463 systest.c Running Audio Device List Audio Device List Found 18 devices 0: ALSA [default:CARD=Intel] (0/1) 1: ALSA [sysdefault:CARD=Intel] (0/1) 2: ALSA [front:CARD=Intel,DEV=0] (1/1) 3: ALSA [surround40:CARD=Intel,DEV=0] (1/1) 4: ALSA [surround51:CARD=Intel,DEV=0] (1/1) 5: ALSA [surround71:CARD=Intel,DEV=0] (1/1) 6: ALSA [iec958:CARD=Intel,DEV=0] (0/1) 7: ALSA [hdmi:CARD=Intel,DEV=0] (0/1) 8: ALSA [default:CARD=Example] (1/1) 9: ALSA [sysdefault:CARD=Example] (1/1) 10: ALSA [front:CARD=Example,DEV=0] (1/1) 11: ALSA [surround21:CARD=Example,DEV=0] (1/1) 12: ALSA [surround40:CARD=Example,DEV=0] (1/1) 13: ALSA [surround41:CARD=Example,DEV=0] (1/1) 14: ALSA [surround50:CARD=Example,DEV=0] (1/1) 15: ALSA [surround51:CARD=Example,DEV=0] (1/1) 16: ALSA [surround71:CARD=Example,DEV=0] (1/1) 17: ALSA [iec958:CARD=Example,DEV=0] (1/1)

1 0

[alsa-devel] [PATCH v2] ALSA: usb-audio: purge needless variable length array
by Takashi Sakamoto 22 Feb '17

22 Feb '17

Variable length array is used in 'snd_us16x08_meter_get()', while there is no need. It's better to purge it because variable length array has overhead for stack handling. This commit replaces the array with static length. Sparse generated below warning. sound/usb/mixer_us16x08.c:714:18: warning: Variable length array is used. Fixes: d2bb390a2081 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/usb/mixer_us16x08.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 7ac45ec..73a0b9a 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -711,7 +711,7 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = elem->head.mixer->chip; struct snd_us16x08_meter_store *store = elem->private_data; u8 meter_urb[64]; - char tmp[max(sizeof(mix_init_msg1), sizeof(mix_init_msg2))]; + char tmp[sizeof(mix_init_msg2)] = {0}; if (elem) { store = (struct snd_us16x08_meter_store *) elem->private_data; @@ -721,8 +721,8 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, switch (kcontrol->private_value) { case 0: - memcpy(tmp, mix_init_msg1, sizeof(mix_init_msg1)); - snd_us16x08_send_urb(chip, tmp, 4); + snd_us16x08_send_urb(chip, (char *)mix_init_msg1, + sizeof(mix_init_msg1)); snd_us16x08_recv_urb(chip, meter_urb, sizeof(meter_urb)); kcontrol->private_value++; @@ -740,7 +740,7 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, case 3: memcpy(tmp, mix_init_msg2, sizeof(mix_init_msg2)); tmp[2] = snd_get_meter_comp_index(store); - snd_us16x08_send_urb(chip, tmp, 10); + snd_us16x08_send_urb(chip, tmp, sizeof(mix_init_msg2)); snd_us16x08_recv_urb(chip, meter_urb, sizeof(meter_urb)); kcontrol->private_value = 0; -- 2.9.3

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: zx-tdm: add bindings doc for zte's tdm controller
by Baoyou Xie 22 Feb '17

22 Feb '17

This patch adds dt-binding documentation for zte's tdm controller. Signed-off-by: Baoyou Xie <baoyou.xie(a)linaro.org> --- .../devicetree/bindings/sound/zte,tdm.txt | 30 ++++++++++++++++++++++ 1 file changed, 30 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt diff --git a/Documentation/devicetree/bindings/sound/zte,tdm.txt b/Documentation/devicetree/bindings/sound/zte,tdm.txt new file mode 100644 index 0000000..8ffc39e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zte,tdm.txt @@ -0,0 +1,30 @@ +ZTE TDM DAI driver + +Required properties: + +- compatible : should be one of the following. + * zte,zx296718-tdm +- reg : physical base address of the controller and length of memory mapped + region. +- clocks : Pairs of phandle and specifier referencing the controller's clocks. +- clock-names: "wclk" for the wclk. + "pclk" for the pclk. +-#clock-cells: should be 1. +- zte,tdm-dma-sysctrl : Reference to the sysctrl controller controlling + the dma. includes: + phandle of sysctrl. + register offset in sysctrl for control dma. + mask of the register that be written to sysctrl. + +Example: + + tdm: tdm@1487000 { + compatible = "zte,zx296718-tdm"; + reg = <0x01487000 0x1000>; + clocks = <&audiocrm AUDIO_TDM_WCLK>, <&audiocrm AUDIO_TDM_PCLK>; + clock-names = "wclk", "pclk"; + #clock-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = <&tdm_global_pin>; + zte,tdm-dma-sysctrl = <&sysctrl 0x10c 4>; + }; -- 2.7.4

3 6

[alsa-devel] Applied "ASoC: rsnd: drop useles self-assignments" to the asoc tree
by Mark Brown 22 Feb '17

22 Feb '17

The patch ASoC: rsnd: drop useles self-assignments has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 34157f7bec8276b4296cf2ec172fc13385ac8af7 Mon Sep 17 00:00:00 2001 From: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Date: Mon, 20 Feb 2017 21:58:34 +0100 Subject: [PATCH] ASoC: rsnd: drop useles self-assignments Coverity reported (CID 1397992) this self-assignment. I think the code stays readable even with the assignments removed. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 948c5ec87980..72966bdd3daa 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -674,12 +674,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_IF: - rdai->bit_clk_inv = rdai->bit_clk_inv; rdai->frm_clk_inv = !rdai->frm_clk_inv; break; case SND_SOC_DAIFMT_IB_NF: rdai->bit_clk_inv = !rdai->bit_clk_inv; - rdai->frm_clk_inv = rdai->frm_clk_inv; break; case SND_SOC_DAIFMT_IB_IF: rdai->bit_clk_inv = !rdai->bit_clk_inv; -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: mediatek: add I2C dependency for CS42XX8
by Arnd Bergmann 22 Feb '17

22 Feb '17

We should not select drivers that depend on I2C when that is disabled, as it results in a build error: warning: (SND_SOC_MT2701_CS42448) selects SND_SOC_CS42XX8_I2C which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && I2C) sound/soc/codecs/cs42xx8-i2c.c:60:1: warning: data definition has no type or storage class module_i2c_driver(cs42xx8_i2c_driver); sound/soc/codecs/cs42xx8-i2c.c:60:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] Fixes: 1f458d53f76c ("ASoC: mediatek: Add mt2701-cs42448 driver and config option.") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 05cf809cf9e1..d7013bde6f45 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -13,7 +13,7 @@ config SND_SOC_MT2701 config SND_SOC_MT2701_CS42448 tristate "ASoc Audio driver for MT2701 with CS42448 codec" - depends on SND_SOC_MT2701 + depends on SND_SOC_MT2701 && I2C select SND_SOC_CS42XX8_I2C select SND_SOC_BT_SCO help -- 2.9.0

2 1

[alsa-devel] Applied "ASoC: rsnd: check return value of init function" to the asoc tree
by Mark Brown 22 Feb '17

22 Feb '17

The patch ASoC: rsnd: check return value of init function has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 56d2c61d611a50e58dba521be1325dc90f9cc933 Mon Sep 17 00:00:00 2001 From: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Date: Mon, 20 Feb 2017 22:05:07 +0100 Subject: [PATCH] ASoC: rsnd: check return value of init function Currently, this function cannot fail for the ADG case. Still, let's apply defensive programming techniques to make sure we fail gracefully whenever rsnd_mod_init() gets extended with another failure case. Reported by Coverity (CID 1397893). Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/adg.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 85a33ac0a5c4..54146f66538c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -564,6 +564,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; + int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -571,8 +572,10 @@ int rsnd_adg_probe(struct rsnd_priv *priv) return -ENOMEM; } - rsnd_mod_init(priv, &adg->mod, &adg_ops, + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, NULL, 0, 0); + if (ret) + return ret; rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); -- 2.11.0

1 0

[alsa-devel] [PATCH] ALSA: x86: hdmi: select CONFIG_SND_PCM
by Arnd Bergmann 22 Feb '17

22 Feb '17

The newly added driver uses the PCM helpers and fails to link if they are disabled: sound/built-in.o: In function `hdmi_lpe_audio_runtime_suspend': intel_hdmi_audio.c:(.text+0x15906): undefined reference to `snd_pcm_suspend' sound/built-in.o: In function `had_pcm_hw_params': intel_hdmi_audio.c:(.text+0x15ac7): undefined reference to `snd_pcm_lib_malloc_pages' sound/built-in.o: In function `had_pcm_open': intel_hdmi_audio.c:(.text+0x15d49): undefined reference to `snd_pcm_hw_constraint_integer' This uses a Kconfig 'select' statement to make sure they are enabled. Fixes: 5dab11d89777 ("ALSA: x86: hdmi: Add audio support for BYT and CHT") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/x86/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 30d066e4885e..84c8f8fc597c 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -9,6 +9,7 @@ if SND_X86 config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" depends on DRM_I915 + select SND_PCM help Choose this option to support HDMI LPE Audio mode -- 2.9.0

2 1

[alsa-devel] [PATCH] pcm_rate: Do not discard slave reported delay in status result.
by Alan Young 21 Feb '17

21 Feb '17

Similar to recent dshare patch.

3 31

[alsa-devel] [PATCH] ALSA: usb-audio: constify snd_kcontrol_new structures
by Bhumika Goyal 21 Feb '17

21 Feb '17

Declare snd_kcontrol_new structures as const as they are only passed as an argument to the function add_new_ctl. This agrument is of type const, so snd_kcontrol_new structures having this property can be made const too. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/usb/mixer_scarlett.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..c33e237 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -477,7 +477,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, return 0; } -static struct snd_kcontrol_new usb_scarlett_ctl_switch = { +static const struct snd_kcontrol_new usb_scarlett_ctl_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", .info = scarlett_ctl_switch_info, @@ -487,7 +487,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, static const DECLARE_TLV_DB_SCALE(db_scale_scarlett_gain, -12800, 100, 0); -static struct snd_kcontrol_new usb_scarlett_ctl = { +static const struct snd_kcontrol_new usb_scarlett_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, @@ -499,7 +499,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, .tlv = { .p = db_scale_scarlett_gain } }; -static struct snd_kcontrol_new usb_scarlett_ctl_master = { +static const struct snd_kcontrol_new usb_scarlett_ctl_master = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, @@ -511,7 +511,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, .tlv = { .p = db_scale_scarlett_gain } }; -static struct snd_kcontrol_new usb_scarlett_ctl_enum = { +static const struct snd_kcontrol_new usb_scarlett_ctl_enum = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", .info = scarlett_ctl_enum_info, @@ -519,7 +519,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, .put = scarlett_ctl_enum_put, }; -static struct snd_kcontrol_new usb_scarlett_ctl_dynamic_enum = { +static const struct snd_kcontrol_new usb_scarlett_ctl_dynamic_enum = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", .info = scarlett_ctl_enum_dynamic_info, @@ -527,7 +527,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, .put = scarlett_ctl_enum_put, }; -static struct snd_kcontrol_new usb_scarlett_ctl_sync = { +static const struct snd_kcontrol_new usb_scarlett_ctl_sync = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .name = "", -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: add driver support for Conexant CX2092X DSP
by simon.ho.cnxt＠gmail.com 21 Feb '17

21 Feb '17

From: Simon Ho <simon.ho(a)conexant.com> This patch adds support for Conexant CX20921/CX20924 I2S AUDIO DSP Simon Ho (2): ASoC: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add driver for CX2092X I2S Audio DSP .../devicetree/bindings/sound/cx2092x.txt | 30 ++ sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/cx2092x-i2c.c | 54 +++ sound/soc/codecs/cx2092x-spi.c | 57 +++ sound/soc/codecs/cx2092x.c | 419 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 26 ++ 7 files changed, 609 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x-i2c.c create mode 100644 sound/soc/codecs/cx2092x-spi.c create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

3 7

[alsa-devel] powerpc64 workaround
by Breno Leitao 21 Feb '17

21 Feb '17

Hello Takashi, Fourteen years ago you commited a patch in alsa-libs entitled "added workaround for ppc64." and I am curious about it. What is this workaround for? Is this still needed? Currently it seems to be causing some issues on ppc64le (ABI v2) when building this package, and we are reverting it downstream in order to avoid FTBFS on alsa-libs: https://github.com/alpinelinux/aports/pull/911 https://github.com/buildroot/buildroot/blob/master/package/alsa-lib/0005-re… Anyway, do you think it is still important? Thank you, Breno

2 2

[alsa-devel] [PATCH] ASoC: Intel: Atom: ix spelling mistake: "allocationf" -> "allocation"
by Colin King 21 Feb '17

21 Feb '17

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/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 21cac1c..e741191 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -690,7 +690,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - dev_err(rtd->dev, "dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocation fail\n"); return retval; } } -- 2.10.2

2 1

[alsa-devel] [PATCH v3 0/2] ASoC: Intel: Skylake: Driver updates on DSP firmware
by jeeja.kp＠intel.com 21 Feb '17

21 Feb '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides update on DSP firmware download by optimizing o updating the dsp register poll method to use accurate timeout. o optimizing ROM init retries. o saving the firmware/library context at boot and use this ctx for downloading the firmware to DSP memory. o cleanup to store the library manifest data o release the firmware in cleanup routine. Changes in v2: - fixed the title to reflect the change as this is optimization. - removed already merged 4 patchs Changes in v3: - rebased on branch topic/intel latest Jeeja KP (2): ASoC: Intel: bxtn: Store the FW/Library context at boot ASoC: Intel: bxtn: optimize ROM init retries sound/soc/intel/skylake/bxt-sst.c | 82 +++++++++++++++++++++++++-------------- 1 file changed, 52 insertions(+), 30 deletions(-) -- 2.5.0

2 3

[alsa-devel] [alsa_devel] [PATCH v3 0/2] ASoC: add driver support for Conexant CX2092X DSP
by simon.ho.cnxt＠gmail.com 21 Feb '17

21 Feb '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Mark, Takashi I corrected the driver based on your input as below. - GPIO specifier names have to be pluralised. - Using gpiod_ instead of gpio APIs. - Using cansleep () gpio version. - Using cansleep () gpio version. - Moving the driver information to patch description. Thanks, Simoon Simon Ho (2): doc: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add driver support fro Conexant CX2092X DSP .../devicetree/bindings/sound/cnxt,cx2092x.txt | 29 ++ sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/cx2092x-i2c.c | 54 +++ sound/soc/codecs/cx2092x-spi.c | 57 +++ sound/soc/codecs/cx2092x.c | 411 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 26 ++ 7 files changed, 600 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cnxt,cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x-i2c.c create mode 100644 sound/soc/codecs/cx2092x-spi.c create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: add driver support for Conexant CX2092X DSP
by simon.ho.cnxt＠gmail.com 21 Feb '17

21 Feb '17

From: Simon Ho <simon.ho(a)conexant.com> This patch adds support for Conexant CX20921/CX20924 I2S AUDIO DSP The CX2092X devices are designed for virtual assisant application need to be always open, listening for users to summon it. There is no any power saving mode support on this device. The processed voice data will be sent to automatic speech recognition (ASR) application for further processing. Simon Ho (2): doc: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add driver support fro Conexant CX2092X DSP .../devicetree/bindings/sound/cnxt,cx2092x.txt | 29 ++ sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/cx2092x-i2c.c | 54 +++ sound/soc/codecs/cx2092x-spi.c | 57 +++ sound/soc/codecs/cx2092x.c | 411 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 26 ++ 7 files changed, 600 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cnxt,cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x-i2c.c create mode 100644 sound/soc/codecs/cx2092x-spi.c create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

4 5

[alsa-devel] [PATCH] ASoC: nau8825: automatic BCLK and LRC divde in master mode
by John Hsu 21 Feb '17

21 Feb '17

configurable LRC and BCLK divide. The driver will make configurations of LRC and BCLK automatically according to BCLK and FS information in master mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 4576f98..97fbeba 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1231,7 +1231,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, osr; + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; nau8825_sema_acquire(nau8825, 3 * HZ); @@ -1261,6 +1261,24 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, osr_adc_sel[osr].clk_src << NAU8825_CLK_ADC_SRC_SFT); } + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, &ctrl_val); + if (ctrl_val & NAU8825_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / params_rate(params); + if (bclk_fs <= 32) + bclk_div = 2; + else if (bclk_fs <= 64) + bclk_div = 1; + else if (bclk_fs <= 128) + bclk_div = 0; + else + return -EINVAL; + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, + ((bclk_div + 1) << NAU8825_I2S_LRC_DIV_SFT) | bclk_div); + } + switch (params_width(params)) { case 16: val_len |= NAU8825_I2S_DL_16; -- 2.6.4 . =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

2 5

[alsa-devel] [GIT PULL] ASoC updates for v4.11
by Mark Brown 20 Feb '17

20 Feb '17

The following changes since commit 566cf877a1fcb6d6dc0126b076aad062054c2637: Linux 4.10-rc6 (2017-01-29 14:25:17 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.11 for you to fetch changes up to 141dee78c40ac2c43aa4ff306688d625e1c731de: Merge remote-tracking branches 'asoc/topic/wm8753' and 'asoc/topic/zte' into asoc-next (2017-02-19 16:37:07 +0000) ---------------------------------------------------------------- ASoC: Updates for v4.11 Another release that's mainly focused on drivers rather than core changes, highlights include: - A huge batch of updates to the Intel drivers, mainly around DisplayPort and HDMI with some additional board support too. - Channel mapping support for HDMI. - Support for AllWinner A31 and A33, Everest Semiconductor ES8328, Nuvoton NAU8540. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da7218: Set DAI output pin high impedance when not in use Alexander Shiyan (1): ASoC: wm8753: Add control to allow swapping HiFi DAC channels Alexandre Belloni (1): ASoC: fsl_sai: support more than 2 channels Alexandrov Stansilav (2): ASoC: rt5640: Add "10EC3276" ACPI ID ASoC: Intel: Atom: Add HP Pavilion x2 10-p000 machine entry Andy Shevchenko (4): ASoC: Intel: remove redundant select SND_SOC_INTEL_SST ASoC: Intel: rename SND_SST_MFLD_PLATFORM to SND_SST_ATOM_HIFI2_PLATFORM ASoC: Intel: select DW_DMAC_CORE since it's mandatory ASoC: Intel: remove ignored dependencies Arnaud Pouliquen (3): DRM: add help to get ELD speaker allocation ASoC: core: add optional pcm_new callback for DAI driver ASoC: hdmi-codec: add channel mapping control Arnd Bergmann (1): ASoC: fix ES8328_I2C/SPI dependencies Axel Lin (1): ASoC: rt5665: Make SND_SOC_RT5665 entry sort in Kconfig and Makefile Baoyou Xie (2): ASoC: zx-i2s: Add the info of pclk to the binding document for zx2967 family ASoC: zx-i2s: introduce pclk for zx2967 family Bard Liao (2): ASoC: rt5640: move DAC2 Power to rt5640_dapm_widgets ASoC: rt5645: set high voltage for capless power Bjorn Andersson (2): ASoC: qcom: lpass-cpu: Remove unnecessary clock checks ASoC: qcom: Drop __func__ usage from log prints Charles Keepax (6): ASoC: wm_adsp: Add mechanism to preload firmware on a core ASoC: Fixup some small kernel-doc typos ASoC: arizona: Propagate errors from arizona_spk_init ASoC: wm_adsp: Correct some missing locking ASoC: wm_adsp: Set booted/running flags at the end of bring up ASoC: wm_adsp: Fixup wm_adsp2_boot_work error paths Chen-Yu Tsai (1): ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route Corentin Labbe (3): ASoC: rt5514-spi: Remove unneeded linux/miscdevice.h include ASoC: rt5677: Remove unneeded linux/miscdevice.h include ASoC: wm0010: Remove unneeded linux/miscdevice.h include Daniel Baluta (1): ASoC: Drop unnecessary debugfs ifdef Enric Balletbo i Serra (1): ASoC: mt8173-max98090: remove the call to snd_soc_jack_add_pins. Fabian Frederick (2): ASoC: samsung: include gpio consumer.h ASoC: samsung: add GPIOLIB dependency Garlic Tseng (2): ASoC: mediatek: add power-domain get/put ctrl for mt2701 ASoC: mediatek: add power-domains for mt2701-afe-pcm.txt Harunobu Kurokawa (1): ASoC: ak4642: Replace mdelay function to msleep Icenowy Zheng (1): ASoC: sunxi: allow the analog codec driver to be built on ARM64 Jeeja KP (39): ASoC: Intel: Skylake: Update link_index and format in pipe params ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Removed unused skl_get_format() ASoC: Intel: Skylake: Clean up manifest info ASoC: Intel: Common: Update dsp register poll implementation ASoC: Intel: bxtn: Use DSP poll API to poll FW status ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare ASoC: hdac_hdmi: Register widget event handlers ASoC: Intel: Skylake: Use set_tdm_slot to set the dma channel ASoC: hdac_hdmi: Move channel info from pin to PCM structure ASoC: Intel: bxt: add channel map support in rt298 machine ASoC: Intel: Skylake: Add route change to rt286 machine ASoC: Intel: bxt: Add route change to da7219_max98357a machine ASoC: Intel: bxt: add channel map support in bxt_da7219_max98357a machine ASoC: hdac_hdmi: Begin to add support for DP Multi-stream audio ASoC: Intel: bxt: Add route change to rt298 machine ASoC: Intel: Skylake: Add route change to nau88l25_max98357a machine ASoC: Intel: Skylake: Add route change to nau88l25_ssm4567 machine ASoC: hdac_hdmi: Add support to handle MST capable pin ASoC: hdac_hdmi: Add MST verb support ASoC: hdac_hdmi: Handle MST pin jack detection at boot/resume ASoc: hdac_hdmi: Configure pin verbs for MST ASoC: hdac_hdmi: Add support for multiple ports to a PCM ASoC: hdac_hdmi: Use ASoC jack instead of snd_jack ASoC: Intel: Skylake: Create ASoC jack for hdmi in rt286 machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in skl_nau88l25_max98357a machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in nau88l25_ssm4567 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_rt298 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_da7219_max98357 machine ASoC: hdac_hdmi: Add machine pin widget for each port ASoC: Intel: Skylake: Add jack port initialize in rt286 machine ASoC: Intel: Skylake: Add jack port initialize in nau88l25_max98357a machine ASoC: Intel: Skylake: Add jack port initialize in nau88l25_ssm4567 machine ASoC: Intel: bxt: Add jack port initialize in da7219_max98357a machine ASoC: Intel: bxt: Add jack port initialize in bxt_rt298 machine John Hsu (2): ASoC: nau8540: new codec driver ASoC: nau8825: automatic BCLK and LRC divde in master mode Jose Abreu (2): ASoC: dwc: Add record capability in PIO mode ASoC: dwc: Enable 24 bit sample size in PIO mode Jörg Krause (3): ASoC: wm8731: Adjust clk definitions so that simple card can work ASoC: mxs-saif: fix setting SAIF1 register ASoC: mxs-saif: fix setting master base rate Krzysztof Kozlowski (1): ASoC: samsung: Remove tests of member address Kuninori Morimoto (13): ASoC: add Component level pcm_new/pcm_free ASoC: remove .delay from snd_soc_platform_driver ASoC: remove .bespoke_trigger from snd_soc_platform_driver ASoC: remove snd_soc_platform_trigger() ASoC: rsnd: fixup for_each_rsnd_mod_array{s} iterator increment ASoC: rsnd: fixup for_each_rsnd_mod_array{s} iterator increment ASoC: simple-card: use devm_get_clk_from_child() ASoC: cq93vc: remove MFD_DAVINCI_VOICECODEC dependency from CQ0093VC ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_audio_routing ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_audio_simple_widgets ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_audio_prefix ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_card_name ASoC: rsnd: fixup reset timing of sync convert_rate Lars-Peter Clausen (2): ASoC: pxa2xx-ac97: Remove unused DAI ID defines ASoC: mpc5200_psc_ac97: Remove unused DAI ID defines Liam Girdwood (1): ASoC: core: Add API to use DMI name in sound card long name Linus Walleij (2): ASoC: dapm: handle probe deferrals ASoC: dapm: fix some pointer error handling Marcus Cooper (4): ASoC: sun4i-spdif: remove legacy dapm components ASoC: sun4i-spdif: Add quirks to the spdif driver ASoC: sunxi: Add bindings for sun8i to SPDIF ASoC: sun4i-spdif: Add support for the H3 SoC Marek Szyprowski (5): ASoC: samsung: i2s: Remove virtual device for secondary DAI ASoC: samsung: i2s: Ensure proper runtime PM state of I2S device ASoC: samsung: i2s: Move saving and restoring regs to runtime pm operations ASoC: samsung: i2s: Let runtime PM operations to control op_clk too ASoC: samsung: i2s: Provide I2S device for registered clocks Mark Brown (26): Merge branch 'topic/hda-ext-decouple' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge tag 'v4.10-rc1' into asoc-samsung Merge tag 'v4.10-rc1' into asoc-intel Merge branch 'topic/hdac-hdmi' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'fix/intel' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge tag 'asoc-fix-v4.10-rc3' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/fsl-mxs-saif' and 'asoc/fix/sunxi' into asoc-linus Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/dmaengine' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3168a' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/adau17x1', 'asoc/topic/adsp', 'asoc/topic/ak4642', 'asoc/topic/amd' and 'asoc/topic/arizona' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/chmap', 'asoc/topic/cq93vc' and 'asoc/topic/da7218' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/debugfs' and 'asoc/topic/doc' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/es8328', 'asoc/topic/fsl', 'asoc/topic/fsl-sai' and 'asoc/topic/graph' into asoc-next Merge remote-tracking branches 'asoc/topic/img' and 'asoc/topic/max98090' into asoc-next Merge remote-tracking branches 'asoc/topic/max9867', 'asoc/topic/mtk', 'asoc/topic/nau8540', 'asoc/topic/nau8825' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5514', 'asoc/topic/rt5640', 'asoc/topic/rt5645', 'asoc/topic/rt5659' and 'asoc/topic/rt5660' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5665', 'asoc/topic/rt5677', 'asoc/topic/samsung', 'asoc/topic/simple' and 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic3x', 'asoc/topic/topology', 'asoc/topic/wm0010', 'asoc/topic/wm8731' and 'asoc/topic/wm8741' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8753' and 'asoc/topic/zte' into asoc-next Matt Ranostay (1): ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches Mengdong Lin (1): ASoC: Intel: Use DMI name for sound card long name in Broadwell machine driver Mylène Josserand (6): ASoC: sun4i-i2s: Increase DMA max burst to 8 ASoC: sun8i-codec-analog: Add amplifier event to fix first delay ASoC: Add sun8i digital audio codec ASoC: codecs: Add sun8i-a33 binding documentation ASoC: sun4i-i2s: Add quirks to handle a31 compatible ASoC: sun4i-i2s: Update binding documentation to include A31 Nicholas Mc Guire (6): ASoC: rt5640: use msleep() for long delays ASoC: rt5659: use msleep() for long delays ASoC: rt5659: declare rt5659_i2c_driver static ASoC: rt5659: drop double const ASoC: rt5660: remove double const ASoC: rt5660: use msleep() for long delay Peter Rosin (2): misc: atmel-ssc: register as sound DAI if #sound-dai-cells is present ASoC: atmel: tse850: rely on the ssc to register as a cpu dai by itself Peter Ujfalusi (1): ASoC: tlv320aic3x: Add delay after power on and register sync Pierre-Louis Bossart (16): ASoC: Intel: boards: remove .pm_ops in all Atom/DPCM machine drivers ASoC: Intel: atom: fix frame polarity ASoC: Intel: cht_bsw_rt5645: add Baytrail MCLK support ASoC: Intel: add support for Realtek 5651 on Cherrytrail ASoC: codecs: rt5670: add quirk for Lenovo Thinkpad 10 ASoC: rt5645: add support for RT5648 ASoC: Intel: Atom: add machine driver for baytrail-rt5645 hardware ASoC: Intel: cht_bsw_rt5645: harden ACPI device detection ASoC: Intel: Baytrail: add quirk for Lenovo Thinkpad 10 ASoc: rt5645: add ACPI ID 10EC3270 ASoC: Intel: cht-bsw-rt5645: add quirks for SSP0/AIF1/AIF2 routing ASoC: Intel: cht-bsw-rt5645: select ASRC source based on routing quirk ASoC: Intel: add support for ALC3270 codec ASoC: Intel: cht-bsw-rt5645: fix DAI formats ASoC: rt5645: fix error handling for gpio detection ASoC: cht-bsw-rt5645: fix unused variable compiler warning Romain Perier (4): ASoC: es8328-i2c: Add compatible for ES8388 ASoC: es8328: Add support for slave mode ASoC: Allow to select ES8328_I2C and ES8328_SPI directly ASoC: rockchip: Add machine driver for RK3288 boards that use analog/HDMI Sandeep Tayal (1): ASoC: hdac_hdmi: use audio component framework to read ELD Sebastien Guiriec (1): ASoC: Intel: Atom: Configure media_loop1 and sprot_loop in stereo Senthilnathan Veppur (1): ASoC: Intel: Skylake: Check device type to get endpoint configuration Sergej Sawazki (1): ASoC: wm8741: Remove unused WM8741_NUM_RATES macro Shailendra Verma (1): ASoC: davinci - Fix possible NULL derefrence. Subhransu S. Prusty (1): ASoC: Intel: Skylake: Report Platform ID info from NHLT Sylwester Nawrocki (2): ASoC: Revert "samsung: Remove unneeded initialization of chan_name" ASoC: Revert "Drop SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag" Vincent Abriou (1): ASoC: hdmi-codec: remove HDMI device unregister Vinod Koul (17): ASoC: Intel: Skylake: remove unused 'ret' ASoC: Intel: sst: remove unused 'ops' ASoC: Intel: sst: remove unused 'msg_high' ASoC: topology: remove unused 'err' ASoC: hdac_hdmi: remove unused 'dai_map' ASoC: max98090: remove superflous check for 'micbias' ASoC: AMD: remove unused ‘dma_buffer’ ASoC: adau17x1: remove unused ‘ret’ ASoC: max9867: remove unused ‘ret’ ASoC: pcm3168a: remove unused ‘format’ ASoC: img: remove unused ‘format’ ASoC: Intel: sst: remove unused ‘ret_val’ ASoC: samsung: smdk_wm8580: remove unused ‘bfs’ ASoC: zx296702-i2s: remove unused ‘format’ ASoC: rt298: Add DMI match for Geminilake reference platform ASoC: Intel: Skylake: Add Geminlake IDs ASoC: hdac_hdmi: Add device id for Geminilake Wei Yongjun (1): ASoC: sun4i-spdif: drop unnessary snd_soc_unregister_component() Yong Zhi (1): ASoC: Intel: boards: Remove ignore_suspend for WoV streams kbuild test robot (1): ASoC: rt5659: fix platform_no_drv_owner.cocci warnings youling257 (1): ASoC: Intel: bytcr_rt5640: quirks for Insyde devices .../devicetree/bindings/misc/atmel-ssc.txt | 2 + .../bindings/sound/axentia,tse850-pcm5142.txt | 11 +- Documentation/devicetree/bindings/sound/es8328.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 2 + .../devicetree/bindings/sound/nau8540.txt | 16 + .../bindings/sound/rockchip,rk3288-hdmi-analog.txt | 36 + .../devicetree/bindings/sound/sun4i-i2s.txt | 5 + .../devicetree/bindings/sound/sun8i-a33-codec.txt | 63 + .../bindings/sound/sunxi,sun4i-spdif.txt | 1 + .../devicetree/bindings/sound/zte,zx-i2s.txt | 14 +- arch/arm/mach-s3c64xx/dev-audio.c | 4 +- drivers/misc/atmel-ssc.c | 50 + include/drm/drm_edid.h | 13 + include/linux/atmel-ssc.h | 1 + include/linux/platform_data/asoc-s3c.h | 6 +- include/sound/dmaengine_pcm.h | 6 + include/sound/simple_card_utils.h | 11 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 52 +- sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/amd/acp-pcm-dma.c | 3 - sound/soc/atmel/tse850-pcm5142.c | 23 +- sound/soc/codecs/Kconfig | 25 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/adau17x1.c | 3 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/da7218.c | 3 +- sound/soc/codecs/es8328-i2c.c | 2 + sound/soc/codecs/es8328.c | 20 +- sound/soc/codecs/hdac_hdmi.c | 1327 ++++++++++++-------- sound/soc/codecs/hdac_hdmi.h | 5 +- sound/soc/codecs/hdmi-codec.c | 403 +++++- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 5 +- sound/soc/codecs/nau8540.c | 835 ++++++++++++ sound/soc/codecs/nau8540.h | 222 ++++ sound/soc/codecs/nau8825.c | 20 +- sound/soc/codecs/pcm3168a.c | 2 - sound/soc/codecs/rt298.c | 7 + sound/soc/codecs/rt5514-spi.c | 1 - sound/soc/codecs/rt5640.c | 13 +- sound/soc/codecs/rt5645.c | 14 +- sound/soc/codecs/rt5659.c | 91 +- sound/soc/codecs/rt5660.c | 6 +- sound/soc/codecs/rt5670.c | 1 + sound/soc/codecs/rt5677-spi.c | 1 - sound/soc/codecs/tlv320aic3x.c | 6 + sound/soc/codecs/wm0010.c | 1 - sound/soc/codecs/wm5102.c | 7 +- sound/soc/codecs/wm5110.c | 10 +- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8741.c | 2 - sound/soc/codecs/wm8753.c | 3 +- sound/soc/codecs/wm8997.c | 6 +- sound/soc/codecs/wm8998.c | 6 +- sound/soc/codecs/wm_adsp.c | 77 +- sound/soc/codecs/wm_adsp.h | 11 + sound/soc/davinci/davinci-evm.c | 13 +- sound/soc/dwc/designware_i2s.c | 9 +- sound/soc/dwc/designware_pcm.c | 99 +- sound/soc/dwc/local.h | 9 +- sound/soc/fsl/efika-audio-fabric.c | 1 - sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/mpc5200_psc_ac97.c | 1 - sound/soc/fsl/mpc5200_psc_ac97.h | 13 - sound/soc/generic/simple-card-utils.c | 8 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/generic/simple-scu-card.c | 4 +- sound/soc/img/img-parallel-out.c | 2 - sound/soc/intel/Kconfig | 51 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/Makefile | 7 +- sound/soc/intel/atom/sst-atom-controls.c | 10 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 5 +- sound/soc/intel/atom/sst/sst_acpi.c | 52 +- sound/soc/intel/atom/sst/sst_ipc.c | 2 - sound/soc/intel/atom/sst/sst_stream.c | 2 - sound/soc/intel/boards/broadwell.c | 2 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 76 +- sound/soc/intel/boards/bxt_rt298.c | 78 +- sound/soc/intel/boards/bytcr_rt5640.c | 15 +- sound/soc/intel/boards/bytcr_rt5651.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 397 +++++- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 33 +- sound/soc/intel/boards/skl_rt286.c | 30 +- sound/soc/intel/common/sst-dsp.c | 52 +- sound/soc/intel/skylake/bxt-sst.c | 64 +- sound/soc/intel/skylake/skl-messages.c | 7 + sound/soc/intel/skylake/skl-nhlt.c | 58 +- sound/soc/intel/skylake/skl-pcm.c | 174 +-- sound/soc/intel/skylake/skl-sst-dsp.h | 4 +- sound/soc/intel/skylake/skl-sst-ipc.h | 5 +- sound/soc/intel/skylake/skl-topology.c | 97 +- sound/soc/intel/skylake/skl-topology.h | 19 + sound/soc/intel/skylake/skl-tplg-interface.h | 12 - sound/soc/intel/skylake/skl.c | 12 + sound/soc/intel/skylake/skl.h | 5 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 + sound/soc/mediatek/mt8173/mt8173-max98090.c | 14 +- sound/soc/mxs/mxs-saif.c | 34 +- sound/soc/omap/mcbsp.h | 3 + sound/soc/omap/omap-mcbsp.c | 48 +- sound/soc/pxa/e740_wm9705.c | 3 - sound/soc/pxa/e750_wm9705.c | 2 - sound/soc/pxa/e800_wm9712.c | 2 - sound/soc/pxa/em-x270.c | 2 - sound/soc/pxa/mioa701_wm9713.c | 1 - sound/soc/pxa/palm27x.c | 2 - sound/soc/pxa/pxa2xx-ac97.c | 2 - sound/soc/pxa/pxa2xx-ac97.h | 17 - sound/soc/pxa/tosa.c | 2 - sound/soc/pxa/zylonite.c | 1 - sound/soc/qcom/lpass-apq8016.c | 15 +- sound/soc/qcom/lpass-cpu.c | 107 +- sound/soc/qcom/lpass-platform.c | 106 +- sound/soc/qcom/storm.c | 22 +- sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3288_hdmi_analog.c | 299 +++++ sound/soc/samsung/Kconfig | 2 + sound/soc/samsung/dmaengine.c | 8 +- sound/soc/samsung/i2s.c | 206 ++- sound/soc/samsung/s3c2412-i2s.c | 2 + sound/soc/samsung/s3c24xx-i2s.c | 2 + sound/soc/samsung/smdk_wm8580.c | 5 +- sound/soc/samsung/tm2_wm5110.c | 1 + sound/soc/sh/rcar/core.c | 2 - sound/soc/sh/rcar/rsnd.h | 4 +- sound/soc/sh/rcar/src.c | 6 +- sound/soc/soc-ac97.c | 2 +- sound/soc/soc-core.c | 225 +++- sound/soc/soc-dapm.c | 62 + sound/soc/soc-generic-dmaengine-pcm.c | 12 +- sound/soc/soc-ops.c | 2 +- sound/soc/soc-pcm.c | 55 +- sound/soc/soc-topology.c | 19 +- sound/soc/sunxi/Kconfig | 13 +- sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-codec.c | 1 + sound/soc/sunxi/sun4i-i2s.c | 61 +- sound/soc/sunxi/sun4i-spdif.c | 60 +- sound/soc/sunxi/sun8i-codec-analog.c | 30 +- sound/soc/sunxi/sun8i-codec.c | 498 ++++++++ sound/soc/zte/zx-i2s.c | 43 +- 147 files changed, 5466 insertions(+), 1521 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nau8540.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt create mode 100644 sound/soc/codecs/nau8540.c create mode 100644 sound/soc/codecs/nau8540.h delete mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h delete mode 100644 sound/soc/pxa/pxa2xx-ac97.h create mode 100644 sound/soc/rockchip/rk3288_hdmi_analog.c create mode 100644 sound/soc/sunxi/sun8i-codec.c

2 1

[alsa-devel] [PATCH] ASoC: rsnd: check return value of init function
by Wolfram Sang 20 Feb '17

20 Feb '17

Currently, this function cannot fail for the ADG case. Still, let's apply defensive programming techniques to make sure we fail gracefully whenever rsnd_mod_init() gets extended with another failure case. Reported by Coverity (CID 1397893). Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- sound/soc/sh/rcar/adg.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 85a33ac0a5c443..54146f66538c18 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -564,6 +564,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; + int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -571,8 +572,10 @@ int rsnd_adg_probe(struct rsnd_priv *priv) return -ENOMEM; } - rsnd_mod_init(priv, &adg->mod, &adg_ops, + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, NULL, 0, 0); + if (ret) + return ret; rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); -- 2.11.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: drop useles self-assignments
by Wolfram Sang 20 Feb '17

20 Feb '17

Coverity reported (CID 1397992) this self-assignment. I think the code stays readable even with the assignments removed. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- sound/soc/sh/rcar/core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 47b370cb2d3b3a..24adb3cc17aa63 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -674,12 +674,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_IF: - rdai->bit_clk_inv = rdai->bit_clk_inv; rdai->frm_clk_inv = !rdai->frm_clk_inv; break; case SND_SOC_DAIFMT_IB_NF: rdai->bit_clk_inv = !rdai->bit_clk_inv; - rdai->frm_clk_inv = rdai->frm_clk_inv; break; case SND_SOC_DAIFMT_IB_IF: rdai->bit_clk_inv = !rdai->bit_clk_inv; -- 2.11.0

1 0

[alsa-devel] New platform CherryTrail/ES8316 audio output is silent
by Daniel Drake 20 Feb '17

20 Feb '17

Hi, I'm trying to get audio working on the Weibu F3C miniPC which has a 3.5mm headphone jack and 3.5mm mic jack. The SoC is Intel Cherry Trail x5-Z8300. The ES8316 codec included is not supported by Linux at the moment. I'd like to fix that and upstream the driver but so far am I stuck with no audio output, only silence. I do have a working Windows setup that I can poke at for more info. I would appreciate any suggestions or ideas. Here is a description of my efforts so far, followed by the work in progress code: I started with the Cherry Trail ES8316 platform driver + ES8316 codec driver which I found in https://github.com/JideTechnology/remixos-kernel/tree/jide_chuwi_vi10_plus_… First I did the required updates to get it building on Linux 4.9 including adding the driver for the codec's MCLK input which is done with some patches currently under review: clk: x86: Add Atom PMC platform clocks arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 platform/x86: Enable Atom PMC platform clocks I confirmed that I am manipulating the right clock by disabling it under Windows and observing that audio output suddenly stops. I created an audio loop within the codec itself, by enabling MIC1 as an input to the Left HP Mux, enabling the LLIN Switch, and enabling Left HP Mux as an input to Left Hp Mixer. Now when I make noise in the microphone, I can hear it on the headphone. This verifies that most of the audio path through the codec is working fine, and also that it is clocked. I used asoc debugfs to trace the path through the DAI, platform and codec DAPM graphs. Everything is On and active as expected. I confirmed that it is basically the same as another Cherry Trail platform I have here (except for the codec which is rt5645 on the other platform), including mixer levels. I also dumped the codec registers over I2C from Windows and set them to the exact same values on Linux. Still not working. At this point I am suspecting that this is likely a problem with the audio interface and its linkage to the codec (SSP/I2S). I modified the platform driver to use ssp1 instead of ssp2, because the device schematics say that I2S_1 is being used. Then I ran into send_ssp_cmd() complaining that ssp1 is not supported, and found a point of curiosity here. This code is basically to set the value of sst_cmd_sba_hw_set_ssp.selection which is documented as: /* 0:SSP0(def), 1:SSP1, 2:SSP2 */ But other platforms like cht_bsw_rt5645 use ssp2, for which send_ssp_cmd() will use SSP_CODEC, which has value 3. So it looks like the above comment is wrong? Either way, I tried selection values 0, 1, 2, and 3, and still no audio output. Here's the code, apologies that it is a bit unclean, I'll continue improving it once I get something working. Any suggestions, or do I need to get a hardware scope to diganose further? --- drivers/base/regmap/regmap-debugfs.c | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8316.c | 1386 ++++++++++++++++++++++++++++ sound/soc/codecs/es8316.h | 124 +++ sound/soc/intel/Kconfig | 12 + sound/soc/intel/atom/sst-atom-controls.c | 5 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cht_es8316.c | 1462 ++++++++++++++++++++++++++++++ 10 files changed, 2999 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h create mode 100644 sound/soc/intel/boards/cht_es8316.c diff --git a/drivers/base/regmap/regmap-debugfs.c b/drivers/base/regmap/regmap-debugfs.c index 36ce351..6d3dc14 100644 --- a/drivers/base/regmap/regmap-debugfs.c +++ b/drivers/base/regmap/regmap-debugfs.c @@ -269,7 +269,7 @@ static ssize_t regmap_map_read_file(struct file *file, char __user *user_buf, count, ppos); } -#undef REGMAP_ALLOW_WRITE_DEBUGFS +#define REGMAP_ALLOW_WRITE_DEBUGFS #ifdef REGMAP_ALLOW_WRITE_DEBUGFS /* * This can be dangerous especially when we have clients such as diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..c74846c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_GTM601 @@ -512,6 +513,9 @@ config SND_SOC_HDMI_CODEC select SND_PCM_IEC958 select HDMI +config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 958cd49..63a9329 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -61,6 +61,7 @@ snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -286,6 +287,7 @@ obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 0000000..6566895 --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,1386 @@ +/* + * es8316.c -- es8316 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.com> + * + * Based on es8316.c + * 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. + */ +#define DEBUG +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/of_gpio.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <linux/proc_fs.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/regmap.h> +#include "es8316.h" + +#if 1 +#define DBG(x...) printk(x) +#else +#define DBG(x...) do { } while (0) +#endif +#define alsa_dbg DBG + +#define dmic_used 1 +#define amic_used 0 + +#define INVALID_GPIO -1 +int es8316_spk_con_gpio = INVALID_GPIO; +int es8316_hp_con_gpio = INVALID_GPIO; +int es8316_hp_det_gpio = INVALID_GPIO; +//static int HP_IRQ=0; +//static int mutex_status=0; +//static int hp_irq_flag = 0; +int es8316_init_reg = 0; + +#define GPIO_LOW 0 +#define GPIO_HIGH 1 +#ifndef es8316_DEF_VOL +#define es8316_DEF_VOL 0x1e +#endif + +struct snd_soc_codec *es8316_codec; +static int es8316_init_regs(struct snd_soc_codec *codec); +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); + +static const struct reg_default es8316_reg_defaults[] = { + {0x00, 0x03}, {0x01 ,0x03}, {0x02, 0x00}, {0x03, 0x20}, + {0x04, 0x11}, {0x05, 0x00}, {0x06, 0x11}, {0x07, 0x00}, + {0x08, 0x00}, {0x09, 0x01}, {0x0a, 0x00}, {0x0b, 0x00}, + {0x0c, 0xf8}, {0x0d, 0x3f}, {0x0e, 0x00}, {0x0f, 0x00}, + {0x10, 0x01}, {0x11, 0xfc}, {0x12, 0x28}, {0x13, 0x00}, + {0x14, 0x00}, {0x15, 0x33}, {0x16, 0x00}, {0x17, 0x00}, + {0x18, 0x88}, {0x19, 0x07}, {0x1a, 0x22}, {0x1b, 0x03}, + {0x1c, 0x0f}, {0x1d, 0x00}, {0x1e, 0x80}, {0x1f, 0x80}, + {0x20, 0x00}, {0x21, 0x00}, {0x22, 0xc0}, {0x23, 0x00}, + {0x24, 0x01}, {0x25, 0x08}, {0x26, 0x10}, {0x27, 0xc0}, + {0x28, 0x00}, {0x29, 0x1c}, {0x2a, 0x00}, {0x2b, 0xb0}, + {0x2c, 0x32}, {0x2d, 0x03}, {0x2e, 0x00}, {0x2f, 0x11}, + {0x30, 0x10}, {0x31, 0x00}, {0x32, 0x00}, {0x33, 0xc0}, + {0x34, 0xc0}, {0x35, 0x1f}, {0x36, 0xf7}, {0x37, 0xfd}, + {0x38, 0xff}, {0x39, 0x1f}, {0x3a, 0xf7}, {0x3b, 0xfd}, + {0x3c, 0xff}, {0x3d, 0x1f}, {0x3e, 0xf7}, {0x3f, 0xfd}, + {0x40, 0xff}, {0x41, 0x1f}, {0x42, 0xf7}, {0x43, 0xfd}, + {0x44, 0xff}, {0x45, 0x1f}, {0x46, 0xf7}, {0x47, 0xfd}, + {0x48, 0xff}, {0x49, 0x1f}, {0x4a, 0xf7}, {0x4b, 0xfd}, + {0x4c, 0xff}, {0x4d, 0x00}, {0x4e, 0x00}, {0x4f, 0xff}, + {0x50, 0x00}, {0x51, 0x00}, {0x52, 0x00}, {0x53, 0x00}, + }; + +static bool es8316_writeable(struct device *dev, unsigned int reg) +{ + if (reg <= 90) + return true; + else + return false; +} +static bool es8316_readable(struct device *dev, unsigned int reg) +{ + if (reg <= 90) + return true; + else + return false; +} +static bool es8316_volatile(struct device *dev, unsigned int reg) +{ + if (reg <= 90) + return true; + else + return false; +} +/* codec private data */ +struct es8316_priv { + struct regmap *regmap; + unsigned int dmic_amic; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct delayed_work pcm_pop_work; +}; +struct snd_soc_codec *tron_codec; +/* +* es8316_reset +* write value 0xff to reg0x00, the chip will be in reset mode +* then, writer 0x00 to reg0x00, unreset the chip +*/ +static int es8316_reset(struct snd_soc_codec *codec) +{ + /* snd_soc_write(codec, ES8316_RESET_REG00, 0x3F); */ + msleep(5); + /* return snd_soc_write(codec, ES8316_RESET_REG00, 0x03); */ + return 0; +} +/* +* es8316_hp_state_query +* Check the register 0x4F, return the register value +* User should update this code according the headphone jack configuratio +*/ + static int es8316_hp_state_query(struct snd_soc_codec *codec) + { + unsigned int retv,tmp; + + retv = snd_soc_read(codec, + 0); + tmp = retv & 0x06; + + return(tmp); + } +/* + * es8316S Controls + */ +//#define DECLARE_TLV_DB_SCALE(name, min, step, mute) +//static const DECLARE_TLV_DB_SCALE(hpout_vol_tlv, -4800, 1200, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(mic_bst_tlv, 0, 1200, 0); +//static const DECLARE_TLV_DB_SCALE(linin_pga_tlv, 0, 300, 0); +/* {0, +3, +6, +9, +12, +15, +18, +21, +24,+27,+30,+33} dB */ +static unsigned int linin_pga_tlv[] = { + TLV_DB_RANGE_HEAD(12), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(600, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(900, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(1200, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1500, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1800, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(2100, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(2400, 0, 0), +}; +static unsigned int hpout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 3, TLV_DB_SCALE_ITEM(-4800, 1200, 0), + +}; +static const char *alc_func_txt[] = {"Off", "On"}; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(ES8316_ADC_ALC1_REG29, 6, 2, alc_func_txt); + +static const char *ng_type_txt[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(ES8316_ADC_ALC6_REG2E, 6, 2, ng_type_txt); + +static const char *adcpol_txt[] = {"Normal", "Invert"}; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(ES8316_ADC_MUTE_REG26, 1, 2, adcpol_txt); +static const char *dacpol_txt[] = {"Normal", "R Invert", "L Invert", + "L + R Invert"}; +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8316_DAC_SET1_REG30, 0, 4, dacpol_txt); + +static const struct snd_kcontrol_new es8316_snd_controls[] = { + /* HP OUT VOLUME */ + SOC_DOUBLE_TLV("HP Playback Volume", ES8316_CPHP_ICAL_VOL_REG18, + 4, 0, 0, 1, hpout_vol_tlv), + /* HPMIXER VOLUME Control */ + SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL_REG16, + 0, 4, 7, 0, hpmixer_gain_tlv), + + /* DAC Digital controls */ + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL_REG33, + ES8316_DAC_VOLR_REG34, 0, 0xC0, 1, dac_vol_tlv), + + SOC_SINGLE("Enable DAC Soft Ramp", ES8316_DAC_SET1_REG30, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1_REG30, 2, 4, 0), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE("DAC Notch Filter", ES8316_DAC_SET2_REG31, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Mode", ES8316_DAC_SET2_REG31, 7, 1, 0), + SOC_SINGLE("DAC Volume Control-LeR", ES8316_DAC_SET2_REG31, 2, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3_REG32, 0, 7, 0), + + /* +20dB D2SE PGA Control */ + SOC_SINGLE_TLV("MIC Boost", ES8316_ADC_D2SEPGA_REG24, + 0, 1, 0, mic_bst_tlv), + /* 0-+24dB Lineinput PGA Control */ + SOC_SINGLE_TLV("Input PGA", ES8316_ADC_PGAGAIN_REG23, + 4, 8, 0, linin_pga_tlv), + + /* ADC Digital Control */ + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME_REG27, + 0, 0xC0, 1, adc_vol_tlv), + SOC_SINGLE("ADC Soft Ramp", ES8316_ADC_MUTE_REG26, 4, 1, 0), + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("ADC Double FS Mode", ES8316_ADC_DMIC_REG25, 4, 1, 0), + /* ADC ALC Control */ + SOC_SINGLE("ALC Capture Target Volume", ES8316_ADC_ALC3_REG2B, 4, 10, 0), + SOC_SINGLE("ALC Capture Max PGA", ES8316_ADC_ALC1_REG29, 0, 28, 0), + SOC_SINGLE("ALC Capture Min PGA", ES8316_ADC_ALC2_REG2A, 0, 28, 0), + SOC_ENUM("ALC Capture Function", alc_func), + SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3_REG2B, 0, 10, 0), + SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4_REG2C, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4_REG2C, 0, 10, 0), + SOC_SINGLE("ALC Capture NG Threshold", ES8316_ADC_ALC6_REG2E, 0, 31, 0), + SOC_ENUM("ALC Capture NG Type", ng_type), + SOC_SINGLE("ALC Capture NG Switch", ES8316_ADC_ALC6_REG2E, 5, 1, 0), +}; + +/* Analog Input MUX */ +static const char * const es8316_analog_in_txt[] = { + "lin1-rin1", + "lin2-rin2", + "lin1-rin1 with 20db Boost", + "lin2-rin2 with 20db Boost" + }; +static const unsigned int es8316_analog_in_values[] = { + 0,/*1,*/ + 1, + 2, + 3 + }; +static const struct soc_enum es8316_analog_input_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL_REG22, 4, 3, + ARRAY_SIZE(es8316_analog_in_txt), + es8316_analog_in_txt, + es8316_analog_in_values); +static const struct snd_kcontrol_new es8316_analog_in_mux_controls = + SOC_DAPM_ENUM("Route", es8316_analog_input_enum); + +/* Dmic MUX */ +static const char * const es8316_dmic_txt[] = { + "dmic disable", + "dmic data at high level", + "dmic data at low level", + }; +static const unsigned int es8316_dmic_values[] = { + 0,/*1,*/ + 1, + 2 + }; +static const struct soc_enum es8316_dmic_src_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC_REG25, 0, 3, + ARRAY_SIZE(es8316_dmic_txt), + es8316_dmic_txt, + es8316_dmic_values); +static const struct snd_kcontrol_new es8316_dmic_src_controls = + SOC_DAPM_ENUM("Route", es8316_dmic_src_enum); + +/* hp mixer mux */ +static const char *es8316_hpmux_texts[] = { + "lin1-rin1", + "lin2-rin2", + "lin-rin with Boost", + "lin-rin with Boost and PGA" + }; + +static const unsigned int es8316_hpmux_values[] = { + 0, 1, 2, 3}; + +static const struct soc_enum es8316_left_hpmux_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_HPMIX_SEL_REG13, 4, 7, + ARRAY_SIZE(es8316_hpmux_texts), + es8316_hpmux_texts, + es8316_hpmux_values); +static const struct snd_kcontrol_new es8316_left_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum); + +static const struct soc_enum es8316_right_hpmux_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_HPMIX_SEL_REG13, 0, 7, + ARRAY_SIZE(es8316_hpmux_texts), + es8316_hpmux_texts, + es8316_hpmux_values); +static const struct snd_kcontrol_new es8316_right_hpmux_controls = + SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum); + +/* headphone Output Mixer */ +static const struct snd_kcontrol_new es8316_out_left_mix[] = { + SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH_REG14, + 6, 1, 0), + SOC_DAPM_SINGLE("Left DAC Switch", 0x30, + 5, 1, 0), +}; +static const struct snd_kcontrol_new es8316_out_right_mix[] = { + SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH_REG14, + 2, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", 0x30, + 5, 1, 0), +}; + +/* DAC data source mux */ +static const char *es8316_dacsrc_texts[] = { + "LDATA TO LDAC, RDATA TO RDAC", + "LDATA TO LDAC, LDATA TO RDAC", + "RDATA TO LDAC, RDATA TO RDAC", + "RDATA TO LDAC, LDATA TO RDAC", + }; + +static const unsigned int es8316_dacsrc_values[] = { + 0, 1, 2, 3}; + +static const struct soc_enum es8316_dacsrc_mux_enum = + SOC_VALUE_ENUM_SINGLE(ES8316_DAC_SET1_REG30, 6, 4, + ARRAY_SIZE(es8316_dacsrc_texts), + es8316_dacsrc_texts, + es8316_dacsrc_values); +static const struct snd_kcontrol_new es8316_dacsrc_mux_controls = + SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum); + + +static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), +// SND_SOC_DAPM_MICBIAS("micbias", ES8316_SYS_PDN_REG0D, +// 5, 1), + + SND_SOC_DAPM_MICBIAS("micbias", SND_SOC_NOPM, + 0, 0), + /* Input MUX */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + SND_SOC_DAPM_PGA("Line input PGA", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* ADCs */ + //SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL_REG22, 6, 0), + SND_SOC_DAPM_ADC("Mono ADC", NULL, SND_SOC_NOPM, 0, 0), + + /* Dmic MUX */ + SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0, + &es8316_dmic_src_controls), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "Capture", 1, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "Playback", 0, + SND_SOC_NOPM, 0, 0), + + /* DACs DATA SRC MUX */ + SND_SOC_DAPM_MUX("DAC SRC Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + /* DACs */ + SND_SOC_DAPM_DAC("Right DAC", NULL, SND_SOC_NOPM, 0, 0), + + //SND_SOC_DAPM_DAC("Left DAC", NULL, ES8316_DAC_PDN_REG2F, 4, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, SND_SOC_NOPM, 0, 0), + + /* Headphone Output Side */ + /* hpmux for hp mixer */ + SND_SOC_DAPM_MUX("Left Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + /* Output mixer */ +/* SND_SOC_DAPM_MIXER("Left Hp mixer", ES8316_HPMIX_PDN_REG15, + 4, 1, &es8316_out_left_mix[0], ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", ES8316_HPMIX_PDN_REG15, + 0, 1, &es8316_out_right_mix[0], ARRAY_SIZE(es8316_out_right_mix)), */ + SND_SOC_DAPM_MIXER("Left Hp mixer", SND_SOC_NOPM, + 4, 1, &es8316_out_left_mix[0], ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", SND_SOC_NOPM, + 0, 1, &es8316_out_right_mix[0], ARRAY_SIZE(es8316_out_right_mix)), + + //SND_SOC_DAPM_MIXER("Left Hp mixer", 0x32, + // 5, 1, &es8316_out_left_mix[0], ARRAY_SIZE(es8316_out_left_mix)), + //SND_SOC_DAPM_MIXER("Right Hp mixer", 0x32, + // 4, 1, &es8316_out_right_mix[0], ARRAY_SIZE(es8316_out_right_mix)), + + /* Ouput charge pump */ +/* + SND_SOC_DAPM_PGA("HPCP L", 0x17, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPCP R", 0x17, + 2, 0, NULL, 0), +*/ + /* Ouput Driver */ + SND_SOC_DAPM_PGA("HPVOL L", SND_SOC_NOPM, + 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("HPVOL R", SND_SOC_NOPM, + 7, 1, NULL, 0), + /* Ouput Driver */ + SND_SOC_DAPM_PGA("SPKVOL L", SND_SOC_NOPM, + 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("SPKVOL R", SND_SOC_NOPM, + 7, 1, NULL, 0), + + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKOL"), + SND_SOC_DAPM_OUTPUT("SPKOR"), +}; + +static const struct snd_soc_dapm_route es8316_dapm_routes[] = { + /* + * record route map + */ + {"MIC1", NULL, "micbias"}, + {"MIC2", NULL, "micbias"}, + {"DMIC", NULL, "micbias"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "Line input PGA"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + {"Digital Mic Mux", "dmic data at high level", "DMIC"}, + {"Digital Mic Mux", "dmic data at low level", "DMIC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + /* + * playback route map + */ + {"DAC SRC Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + {"DAC SRC Mux", "LDATA TO LDAC, LDATA TO RDAC", "I2S IN"}, + {"DAC SRC Mux", "RDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + {"DAC SRC Mux", "RDATA TO LDAC, LDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC SRC Mux"}, + {"Right DAC", NULL, "DAC SRC Mux"}, + + + {"Left Hp mux", "lin1-rin1", "MIC1"}, + {"Left Hp mux", "lin2-rin2", "MIC2"}, + {"Left Hp mux", "lin-rin with Boost", "Differential Mux"}, + {"Left Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Right Hp mux", "lin1-rin1", "MIC1"}, + {"Right Hp mux", "lin2-rin2", "MIC2"}, + {"Right Hp mux", "lin-rin with Boost", "Differential Mux"}, + {"Right Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Hp mixer", "LLIN Switch", "Left Hp mux"}, + {"Left Hp mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Hp mixer", "RLIN Switch", "Right Hp mux"}, + {"Right Hp mixer", "Right DAC Switch", "Right DAC"}, + + {"HPVOL L", NULL, "Left Hp mixer"}, + {"HPVOL R", NULL, "Right Hp mixer"}, +/* + {"HPVOL L", NULL, "HPCP L"}, + {"HPVOL R", NULL, "HPCP R"}, +*/ + {"HPOL", NULL, "HPVOL L"}, + {"HPOR", NULL, "HPVOL R"}, + + {"SPKOL", NULL, "SPKVOL L"}, + {"SPKOR", NULL, "SPKVOL R"}, + + {"SPKVOL L", NULL, "Left Hp mixer"}, + {"SPKVOL R", NULL, "Right Hp mixer"}, +}; + +struct _coeff_div { + u32 mclk; //mclk frequency + u32 rate; //sample rate + u8 div; //adcclk and dacclk divider + u8 lrck_h; //adclrck divider and daclrck divider + u8 lrck_l; + u8 sr; //sclk divider + u8 osr; //adc osr +}; + + +/* codec hifi mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 8k */ + {12288000, 8000 , 6 , 0x06, 0x00, 21, 32}, + {11289600, 8000 , 6 , 0x05, 0x83, 20, 29}, + {18432000, 8000 , 9 , 0x09, 0x00, 27, 32}, + {16934400, 8000 , 8 , 0x08, 0x44, 25, 33}, + {12000000, 8000 , 7 , 0x05, 0xdc, 21, 25}, + {19200000, 8000 , 12, 0x09, 0x60, 27, 25}, + + /* 11.025k */ + {11289600, 11025, 4 , 0x04, 0x00, 16, 32}, + {16934400, 11025, 6 , 0x06, 0x00, 21, 32}, + {12000000, 11025, 4 , 0x04, 0x40, 17, 34}, + + /* 16k */ + {12288000, 16000, 3 , 0x03, 0x00, 12, 32}, + {18432000, 16000, 5 , 0x04, 0x80, 18, 25}, + {12000000, 16000, 3 , 0x02, 0xee, 12, 31}, + {19200000, 16000, 6 , 0x04, 0xb0, 18, 25}, + + /* 22.05k */ + {11289600, 22050, 2 , 0x02, 0x00, 8 , 32}, + {16934400, 22050, 3 , 0x03, 0x00, 12, 32}, + {12000000, 22050, 2 , 0x02, 0x20, 8 , 34}, + + /* 32k */ + {12288000, 32000, 1 , 0x01, 0x80, 6 , 48}, + {18432000, 32000, 2 , 0x02, 0x40, 9 , 32}, + {12000000, 32000, 1 , 0x01, 0x77, 6 , 31}, + {19200000, 32000, 3 , 0x02, 0x58, 10, 25}, + + /* 44.1k */ + {11289600, 44100, 1 , 0x01, 0x00, 4 , 32}, + {16934400, 44100, 1 , 0x01, 0x80, 6 , 32}, + {12000000, 44100, 1 , 0x01, 0x10, 4 , 34}, + + /* 48k */ + {12288000, 48000, 1 , 0x01, 0x00, 4 , 32}, + {18432000, 48000, 1 , 0x01, 0x80, 6 , 32}, + {12000000, 48000, 1 , 0x00, 0xfa, 4 , 31}, + {19200000, 48000, 2 , 0x01, 0x90, 6, 25}, + + /* 88.2k */ + {11289600, 88200, 1 , 0x00, 0x80, 2 , 32}, + {16934400, 88200, 1 , 0x00, 0xc0, 3 , 48}, + {12000000, 88200, 1 , 0x00, 0x88, 2 , 34}, + + /* 96k */ + {12288000, 96000, 1 , 0x00, 0x80, 2 , 32}, + {18432000, 96000, 1 , 0x00, 0xc0, 3 , 48}, + {12000000, 96000, 1 , 0x00, 0x7d, 1 , 31}, + {19200000, 96000, 1 , 0x00, 0xc8, 3 , 25}, +}; +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +/* The set of rates we can generate from the above for each SYSCLK */ + +static unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + +/* + * Note that this should be called from init rather than from hw_params. + */ +static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + + DBG("Enter::%s----%d, freq:%d\n",__FUNCTION__,__LINE__, freq); + + switch (freq) { + case 11289600: + case 18432000: + case 22579200: + case 36864000: + es8316->sysclk_constraints = &constraints_112896; + es8316->sysclk = freq; + return 0; + + case 12288000: + case 19200000: + case 16934400: + case 24576000: + case 33868800: + es8316->sysclk_constraints = &constraints_12288; + es8316->sysclk = freq; + return 0; + + case 12000000: + case 24000000: + es8316->sysclk_constraints = &constraints_12; + es8316->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface = 0; + u8 adciface = 0; + u8 daciface = 0; + + printk("%s----%d, fmt[%02x]\n",__FUNCTION__,__LINE__,fmt); + + iface = snd_soc_read(codec, ES8316_IFACE); + adciface = snd_soc_read(codec, ES8316_ADC_IFACE); + daciface = snd_soc_read(codec, ES8316_DAC_IFACE); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: // MASTER MODE + alsa_dbg("es8316 in master mode"); + iface |= 0x80; + break; + case SND_SOC_DAIFMT_CBS_CFS: // SLAVE MODE + alsa_dbg("es8316 in slave mode"); + iface &= 0x7F; + break; + default: + return -EINVAL; + } + + /* interface format */ + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adciface &= 0xFC; + daciface &= 0xFC; + break; + case SND_SOC_DAIFMT_RIGHT_J: + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + adciface &= 0xFC; + daciface &= 0xFC; + adciface |= 0x01; + daciface |= 0x01; + break; + case SND_SOC_DAIFMT_DSP_A: + adciface &= 0xDC; + daciface &= 0xDC; + adciface |= 0x03; + daciface |= 0x03; + break; + case SND_SOC_DAIFMT_DSP_B: + adciface &= 0xDC; + daciface &= 0xDC; + adciface |= 0x23; + daciface |= 0x23; + break; + default: + return -EINVAL; + } + + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface &= 0xDF; + adciface &= 0xDF; + daciface &= 0xDF; + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x20; + adciface |= 0x20; + daciface |= 0x20; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x20; + adciface &= 0xDF; + daciface &= 0xDF; + break; + case SND_SOC_DAIFMT_NB_IF: + iface &= 0xDF; + adciface |= 0x20; + daciface |= 0x20; + break; + default: + return -EINVAL; + } + snd_soc_write(codec, ES8316_IFACE, iface); + snd_soc_write(codec, ES8316_ADC_IFACE, adciface); + snd_soc_write(codec, ES8316_DAC_IFACE, daciface); + return 0; +} + +static void pcm_pop_work_events(struct work_struct *work) +{ + int ret; + //printk("es8316--------pcm_pop_work_events\n"); + snd_soc_write(tron_codec,ES8316_CPHP_OUTEN_REG17,0x66); + ret=snd_soc_read(tron_codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) { + gpio_set_value(368, 1); + //printk("2222 gpio_set_value 1\n"); + } + es8316_init_reg=1; +} + +static int es8316_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int ret; + //printk("es8316_pcm_startup Enter::%s----%d es8316->sysclk=%d\n",__FUNCTION__,__LINE__,es8316->sysclk); + if(playback) + { + snd_soc_write(codec, 0x50, 0xAA); + if(es8316_init_reg==0){ + //printk("==================>>>>>>>>es8316_pcm_startup es8316_init_reg=0\n"); + schedule_delayed_work(&es8316->pcm_pop_work,msecs_to_jiffies(100)); + } + else{ + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) + { + gpio_set_value(368, 1); + // printk("es8316_pcm_startup gpio_set_value 1\n"); + //msleep(250); + } + } + } + + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!es8316->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + //return -EINVAL; + } else { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, es8316->sysclk_constraints); +} + + return 0; +} + +static void es8316_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + struct snd_soc_codec *codec = dai->codec; + if(playback) + { + snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x30);//close �� + gpio_set_value(368, 0); + snd_soc_write(codec, 0x50, 0xA0); + //printk("es8316_pcm_shutdown----gpio_set_value 0\n"); + //printk("%s playback\n",__FUNCTION__); + } +} + + +static int es8316_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 es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + int retv; + int coeff; + u16 osrate = snd_soc_read(codec, ES8316_CLKMGR_ADCOSR_REG03) & 0xc0; + u16 mclkdiv = snd_soc_read(codec, ES8316_CLKMGR_CLKSW_REG01) & 0x7f; + u16 srate = snd_soc_read(codec, ES8316_SDP_MS_BCKDIV_REG09) & 0xE0; + u16 adciface = snd_soc_read(codec, ES8316_SDP_ADCFMT_REG0A) & 0xE3; + u16 daciface = snd_soc_read(codec, ES8316_SDP_DACFMT_REG0B) & 0xE3; + u16 adcdiv = snd_soc_read(codec, ES8316_CLKMGR_ADCDIV1_REG04); + u16 adclrckdiv_l = snd_soc_read(codec, ES8316_CLKMGR_ADCDIV2_REG05) & 0x00; + u16 dacdiv = snd_soc_read(codec, ES8316_CLKMGR_DACDIV1_REG06); + u16 daclrckdiv_l = snd_soc_read(codec, ES8316_CLKMGR_DACDIV2_REG07) & 0x00; + u16 adclrckdiv_h = adcdiv & 0xf0; + u16 daclrckdiv_h = dacdiv & 0xf0; + int ret; + adcdiv &= 0x0f; + dacdiv &= 0x0f; + + u16 tmp = snd_soc_read(codec, ES8316_CLKMGR_DACDIV2_REG07); + + coeff = get_coeff(es8316->sysclk, params_rate(params)); + if (coeff < 0) { + coeff = get_coeff(es8316->sysclk / 2, params_rate(params)); + mclkdiv |= 0x80; + } + if (coeff < 0) { + dev_err(codec->dev, + "Unable to configure sample rate %dHz with %dHz MCLK\n", + params_rate(params), es8316->sysclk); + return coeff; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + adciface |= 0x000C; + daciface |= 0x000C; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + adciface |= 0x0004; + daciface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + break; + case SNDRV_PCM_FORMAT_S32_LE: + adciface |= 0x0010; + daciface |= 0x0010; + break; + } + + /* set iface & srate*/ + snd_soc_update_bits(codec,ES8316_SDP_DACFMT_REG0B, 0xe3, daciface); + snd_soc_update_bits(codec, ES8316_SDP_ADCFMT_REG0A, 0xe3, adciface); + if (coeff >= 0) { + osrate = coeff_div[coeff].osr; + osrate &= 0x3f; + + srate |= coeff_div[coeff].sr; + srate &= 0x1f; + + adcdiv |= (coeff_div[coeff].div << 4); + adclrckdiv_h |= coeff_div[coeff].lrck_h; + adcdiv &= 0xf0; + adclrckdiv_h &= 0x0f; + adcdiv |= adclrckdiv_h; + adclrckdiv_l = coeff_div[coeff].lrck_l; + + dacdiv |= (coeff_div[coeff].div << 4); + daclrckdiv_h |= coeff_div[coeff].lrck_h; + dacdiv &= 0xf0; + daclrckdiv_h &= 0x0f; + dacdiv |= daclrckdiv_h; + daclrckdiv_l = coeff_div[coeff].lrck_l; + /* + * must do codec power on initialization at here becauses MCLK and I2S CLK always startup at here + */ + retv = snd_soc_read(codec, 0x14) ; + //pr_err("%s:%d, retv=%x \n", __func__, __LINE__, retv); + if(es8316_init_reg>0){ + //printk("===%s es8316_init_reg>0\n", __func__); + snd_soc_write(codec, 0X17, 0x66); + snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x10); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) { + if(es8316_init_reg>0){ + gpio_set_value(368, 1); + // printk("gpio_set_value 1\n"); + //msleep(250); + } + } + + }else{ + //printk("===%s es8316_init_reg=0\n", __func__); + snd_soc_write(codec, 0x01, 0x7F); + snd_soc_write(codec, 0x2F, 0x00); + snd_soc_write(codec, 0X18, 0x00); + snd_soc_write(codec, 0X1b, 0x30); + snd_soc_write(codec, 0X1a, 0x10); + snd_soc_write(codec, 0X19, 0x03); + //msleep(10); + snd_soc_write(codec, 0X14, 0x88); + snd_soc_write(codec, 0X15, 0x00); + snd_soc_write(codec, 0X16, 0xbb); + snd_soc_write(codec, 0x00, 0xC0); //CHARGE PUMP DIVIDER + msleep(50); + snd_soc_write(codec, 0X17, 0x66); + snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x00); + snd_soc_write(codec, ES8316_GPIO_SEL_REG4D, 0x00); //set gpio2 to GM SHORT + snd_soc_write(codec, ES8316_GPIO_DEBUNCE_INT_REG4E, 0xf3); //maximum debance time, enable interrupt, low active + es8316_set_bias_level(codec, SND_SOC_BIAS_ON); + //es8316_init_reg = 1; + } + //pr_err("%s:%d, retv=%x \n", __func__, __LINE__, retv); + } + + //snd_soc_write(codec,ES8316_DAC_SET2_REG31, 0x20); + retv = snd_soc_read(codec, ES8316_GPIO_FLAG); + //pr_err("%s:%d, retv=%x \n", __func__, __LINE__, retv); + return 0; +} + +static int es8316_mute(struct snd_soc_dai *dai, int mute) +{ + //printk("===enter %s\n", __func__); + #if 0 + struct snd_soc_codec *codec = dai->codec; + int ret; + dev_dbg(codec->dev, "%s %d\n", __func__, mute); + if (mute){ + //snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x20);//close �� + gpio_set_value(368, 0); + }else{ + //snd_soc_write(codec, ES8316_DAC_SET1_REG30, 0x00); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x04)==0x04) { //remove + gpio_set_value(368, 1); + } + } + #endif + return 0; +} + +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dev_dbg(codec->dev, "%s on\n", __func__); + break; + case SND_SOC_BIAS_PREPARE: + dev_dbg(codec->dev, "%s prepare\n", __func__); + if(es8316_init_reg>0) + { + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x7F); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x88); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x00); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0xBB); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x10); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x03); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x00); + snd_soc_write(codec, ES8316_RESET_REG00, 0xC0); + //snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x66); + } + break; + case SND_SOC_BIAS_STANDBY: + dev_dbg(codec->dev, "%s standby\n", __func__); + if(es8316_init_reg>0) // if codec intialized, + { + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x7F); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x88); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x00); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0xBB); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x10); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x03); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x00); + snd_soc_write(codec, ES8316_RESET_REG00, 0xC0); + //snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x66); + } + break; + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "%s off\n", __func__); + /*if(es8316_init_reg>0){ + // snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x33); + snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x07); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_write(codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x11); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x3F); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + snd_soc_write(codec, ES8316_RESET_REG00, 0x7F); + }*/ + break; + } + + return 0; +} + +#define es8316_RATES SNDRV_PCM_RATE_8000_96000 + +#define es8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops es8316_ops = { + .startup = es8316_pcm_startup, + .hw_params = es8316_pcm_hw_params, + .set_fmt = es8316_set_dai_fmt, + .set_sysclk = es8316_set_dai_sysclk, + .digital_mute = es8316_mute, + .shutdown = es8316_pcm_shutdown, +}; + +static struct snd_soc_dai_driver es8316_dai = { + .name = "ES8316 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = es8316_RATES, + .formats = es8316_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = es8316_RATES, + .formats = es8316_FORMATS, + }, + .ops = &es8316_ops, + .symmetric_rates = 1, +}; + + +static int es8316_init_regs(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s\n", __func__); + snd_soc_write(codec,0x01, 0x7f); //setup adc volume + snd_soc_write(codec,0x02, 0x00); //disable DAC ds and notch mode + snd_soc_write(codec, 0x03,0x19); //CHARGE PUMP DIVIDER + snd_soc_write(codec, 0x04,0x21); + snd_soc_write(codec, 0x05,0x90); + snd_soc_write(codec, 0x06,0x11); + snd_soc_write(codec, 0x07,0x90); + snd_soc_write(codec, 0x08,0x00); + snd_soc_write(codec, 0x09,0x01); + + snd_soc_write(codec, 0x0A, 0x0C);//ADC FORMAT , i2s-16bit, + snd_soc_write(codec, 0x0B, 0x0C);//DAC FORMAT, i2s-16bit + snd_soc_write(codec, 0x0D,0x00); //CLK CASE + snd_soc_write(codec, 0x22,0x20); //CHARGE PUMP DIVIDER + snd_soc_write(codec, 0x2F,0x00); + snd_soc_write(codec, 0x10,0x11); + snd_soc_write(codec, 0x12,0x28); + snd_soc_write(codec, 0x1C,0x0F); + snd_soc_write(codec, 0x1D,0x0C); + snd_soc_write(codec, 0x00,0xC0); //CHARGE PUMP DIVIDER + msleep(50); + + snd_soc_write(codec, 0x23,0x10); //CLK ON + snd_soc_write(codec, 0x24,0x01); //VMID SEQUENCE + snd_soc_write(codec, 0x26,0x00); //ADC,DAC VREF SET, micbias pup + snd_soc_write(codec, 0x27,0x00); //CSM POWER UP + snd_soc_write(codec, 0x30, 0x10); //INPUT SELECT INPUT1:0X20,INPUT2:0X30, ain and adc power down + snd_soc_write(codec, 0x33, 0x00); //POWER UP DACPOWER + snd_soc_write(codec, 0x34, 0x00); //L,R MIXER SET + snd_soc_write(codec, 0x14,0x88); //MIXER + snd_soc_write(codec, 0x15,0x00); //MIXER + snd_soc_write(codec, 0x16,0xAA); //MIXER + snd_soc_write(codec, 0X18,0x00); //CHARGE PUMP + snd_soc_write(codec, 0X31, 0x00); //CHARGE PUMP + snd_soc_write(codec, 0X29, 0xd2); //CHARGE PUMP + snd_soc_write(codec, 0X2A, 0x08); //vmidlow = 10'b + snd_soc_write(codec, 0X2B,0xA0); //HPOUT SET + + snd_soc_write(codec, 0X2C, 0x05); + snd_soc_write(codec, 0X2D, 0x06); + snd_soc_write(codec, 0X2E, 0x61); + /*alc set*/ + snd_soc_write(codec, 0X4D, 0x00); + snd_soc_write(codec, 0X4E, 0xF3); + snd_soc_write(codec, 0X50, 0xA0); //adc ds mode, HPF enable + snd_soc_write(codec, 0X51, 0x00); //ALC ON, + snd_soc_write(codec, 0X52, 0x00); + //msleep(50); + + snd_soc_write(codec, 0X18, 0x00); + snd_soc_write(codec, 0X1b, 0x30); + snd_soc_write(codec, 0X1a, 0x10); + snd_soc_write(codec, 0X19, 0x03); + snd_soc_write(codec, 0X14, 0x88); + snd_soc_write(codec, 0X15, 0x00); + snd_soc_write(codec, 0X16, 0xbb); + snd_soc_write(codec, 0X17, 0x66); + return 0; +} + +//static int es8316_suspend(struct snd_soc_codec *codec, pm_message_t state) +static int es8316_suspend(struct device *dev) +{ + //printk("%s suspend\n", __func__); + gpio_set_value(368, 0); + // snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x33); + if(es8316_init_reg>0){ + /*if Codec has been initialized before, then enter into suspend directly */ + //printk("===%s es8316_init_reg > 0\n", __func__); + snd_soc_write(tron_codec, ES8316_CPHP_ICAL_VOL_REG18, 0x11); + mdelay(5); + snd_soc_write(tron_codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(tron_codec, 0x2f, 0x11); + snd_soc_write(tron_codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(tron_codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(tron_codec, ES8316_CPHP_PDN1_REG19, 0x06); + snd_soc_write(tron_codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(tron_codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(tron_codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_update_bits(tron_codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0, 0xc0); + snd_soc_write(tron_codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + //snd_soc_write(tron_codec, 0x00,0x7f); //CHARGE PUMP DIVIDER + } else { + #if 0 + /*if Codec has not been initialized before, enter into normal mode firstly, then enter into suspend at last */ + //snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0, 0x00); + //printk("===%s es8316_init_reg = 0\n", __func__); + snd_soc_write(tron_codec, 0x2F, 0x00); + snd_soc_write(tron_codec, 0X18, 0x11); + snd_soc_write(tron_codec, 0X1b, 0x30); + snd_soc_write(tron_codec, 0X1a, 0x10); + snd_soc_write(tron_codec, 0X19, 0x03); + snd_soc_write(tron_codec, 0X14, 0x88); + snd_soc_write(tron_codec, 0X15, 0x00); + snd_soc_write(tron_codec, 0X16, 0xbb); + snd_soc_write(tron_codec, 0x00,0xC0); //CHARGE PUMP DIVIDER + //msleep(50); + //snd_soc_write(tron_codec, 0X17, 0x66); + msleep(200); + snd_soc_write(tron_codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(tron_codec, 0x2f, 0x11); + snd_soc_write(tron_codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(tron_codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(tron_codec, ES8316_CPHP_PDN1_REG19, 0x06); + snd_soc_write(tron_codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(tron_codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(tron_codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_update_bits(tron_codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0, 0xc0); + snd_soc_write(tron_codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + //es8316_init_reg = 1; + #endif + } + //snd_soc_write(tron_codec, 0X4E, 0xF0); + return 0; +} + +//static int es8316_resume(struct snd_soc_codec *codec) +static int es8316_resume(struct device *dev) +{ + //printk("===%s resume\n", __func__); + if(es8316_init_reg>0){ + //printk("es8316 resume\n"); + snd_soc_write(tron_codec, ES8316_CLKMGR_CLKSW_REG01, 0x7f); + snd_soc_write(tron_codec, ES8316_CPHP_ICAL_VOL_REG18, 0x11); + snd_soc_write(tron_codec, ES8316_CPHP_LDOCTL_REG1B, 0x30); + snd_soc_write(tron_codec, ES8316_CPHP_PDN2_REG1A, 0x10); + snd_soc_write(tron_codec, ES8316_CPHP_PDN1_REG19, 0x02); + snd_soc_write(tron_codec, ES8316_HPMIX_SWITCH_REG14, 0x88); + snd_soc_write(tron_codec, ES8316_HPMIX_PDN_REG15, 0x00); + snd_soc_write(tron_codec, ES8316_HPMIX_VOL_REG16, 0xbb); + snd_soc_update_bits(tron_codec, ES8316_ADC_PDN_LINSEL_REG22, 0xc0, 0x00); + snd_soc_write(tron_codec, 0x2f, 0x00); + snd_soc_write(tron_codec, ES8316_CPHP_OUTEN_REG17, 0x66); + } + //snd_soc_write(tron_codec, 0X4E, 0xF3); + return 0; +} + +static int es8316_probe(struct snd_soc_codec *codec) +{ + int ret = 0; +// int ReadBuffer,ReadBuffer1,ReadBuffer2; + struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec); + pr_debug("---%s--start--\n",__FUNCTION__); + tron_codec = codec; + es8316_init_reg = 0; + //gpio_set_value(139, 0); + snd_soc_write(codec, 0x00, 0xc2); + //msleep(100); + //retv = snd_soc_read(codec, ES8316_CLKMGR_ADCDIV2_REG05) ; + snd_soc_write(codec, 0x01, 0x7f); //setup adc volume + snd_soc_write(codec, 0x02, 0x00); //disable DAC ds and notch mode + snd_soc_write(codec, 0x03,0x19); //CHARGE PUMP DIVIDER + snd_soc_write(codec, 0x04,0x21); + snd_soc_write(codec, 0x05,0x90); + snd_soc_write(codec, 0x06,0x11); + snd_soc_write(codec, 0x07,0x90); + snd_soc_write(codec, 0x08,0x00); + snd_soc_write(codec, 0x09,0x01); + + snd_soc_write(codec, 0x0A, 0x0C);//ADC FORMAT , i2s-16bit, + snd_soc_write(codec, 0x0B, 0x0C);//DAC FORMAT, i2s-16bit + snd_soc_write(codec, 0x0D,0x00); //CLK CASE + + snd_soc_write(codec, 0x10, 0x11); + snd_soc_write(codec, 0x12, 0x28); + snd_soc_write(codec, 0x1C, 0x0F); + snd_soc_write(codec, 0x1D, 0x0C); + msleep(50); + + snd_soc_write(codec, 0x23, 0x50); //CLK ON + snd_soc_write(codec, 0x24, 0x00); //VMID SEQUENCE + if(es8316->dmic_amic == dmic_used){ + snd_soc_write(codec, 0x25, 0x0A); //dmic channel + }else { + snd_soc_write(codec, 0x25, 0x08); //amic channel + } + snd_soc_write(codec, 0x26, 0x00); //ADC,DAC VREF SET, micbias pup + snd_soc_write(codec, 0x27, 0x00); //CSM POWER UP + snd_soc_write(codec, 0x30, 0x10); //INPUT SELECT INPUT1:0X20,INPUT2:0X30, ain and adc power down + snd_soc_write(codec, 0x33, 0x00); //POWER UP DACPOWER + snd_soc_write(codec, 0x34, 0x00); //L,R MIXER SET + snd_soc_write(codec, 0X31, 0x00); //CHARGE PUMP + if(es8316->dmic_amic == dmic_used){ + snd_soc_write(codec, 0X29, 0xDC); //CHARGE PUMP + snd_soc_write(codec, 0X2A, 0x1C); //vmidlow = 10'b + }else { + snd_soc_write(codec, 0X29, 0xD0); //CHARGE PUMP + snd_soc_write(codec, 0X2A, 0x08); //vmidlow = 10'b + } + snd_soc_write(codec, 0X2B, 0xA0); //HPOUT SET + + snd_soc_write(codec, 0X2C, 0x05); + snd_soc_write(codec, 0X2D, 0x06); + snd_soc_write(codec, 0X2E, 0x21); + /*alc set*/ + snd_soc_write(codec, 0X4D, 0x00); + snd_soc_write(codec, 0X4E, 0xF0); + snd_soc_write(codec, 0X50, 0xA0); //adc ds mode, HPF enable + snd_soc_write(codec, 0X51, 0x00); //ALC ON, + snd_soc_write(codec, 0X52, 0x00); +#if defined(HS_IRQ) + det_initalize(); +#elif defined(HS_TIMER) + hsdet_init(); +#endif + INIT_DELAYED_WORK(&es8316->pcm_pop_work, pcm_pop_work_events); + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + //.suspend = es8316_suspend, + //.resume = es8316_resume, + .set_bias_level = es8316_set_bias_level, + //.idle_bias_off = true, + + .component_driver = { + .controls = es8316_snd_controls, + .num_controls = ARRAY_SIZE(es8316_snd_controls), + .dapm_widgets = es8316_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets), + .dapm_routes = es8316_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes), + }, +}; + +static const struct regmap_config es8316_regmap = { + .reg_bits = 8, + .val_bits = 8, + .volatile_reg = es8316_volatile, + .readable_reg = es8316_readable, + .max_register = ARRAY_SIZE(es8316_reg_defaults) + 1, + .reg_defaults = es8316_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(es8316_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static void es8316_i2c_shutdown(struct i2c_client *i2c) +{ + struct snd_soc_codec *codec; + + if (!tron_codec) + goto err; + codec = tron_codec; + if(es8316_init_reg>0){ + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL_REG18, 0x00); + snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL_REG1B, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2_REG1A, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1_REG19, 0x06); + snd_soc_write(codec, ES8316_HPMIX_SWITCH_REG14, 0x00); + snd_soc_write(codec, ES8316_HPMIX_PDN_REG15, 0x33); + snd_soc_write(codec, ES8316_HPMIX_VOL_REG16, 0x00); + snd_soc_write(codec, ES8316_ADC_PDN_LINSEL_REG22, 0xC0); + snd_soc_write(codec, ES8316_DAC_PDN_REG2F, 0x11); + snd_soc_write(codec, ES8316_SYS_PDN_REG0D, 0x3F); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW_REG01, 0x03); + snd_soc_write(codec, ES8316_RESET_REG00, 0x7F); + } +err: + return; +} + +static int es8316_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct es8316_priv *es8316; + struct regmap *regmap; + int ret = -1; + + es8316 = kzalloc(sizeof(*es8316), GFP_KERNEL); + if (es8316 == NULL){ + return -ENOMEM; + } + es8316->dmic_amic = dmic_used; //if internal mic is amic + + regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + i2c_set_clientdata(i2c_client, es8316); + ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316, + &es8316_dai, 1); + if (ret < 0) { + kfree(es8316); + return ret; + } + + gpio_request(368,"SPK_OUT_SHUTDOWN"); + gpio_direction_output(368, 0); + return ret; +} + +static int es8316_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const unsigned short normal_i2c[] = {0x11, I2C_CLIENT_END}; +static const struct i2c_device_id es8316_i2c_id[] = { + {"es8316", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8316_acpi_match[] = { + { "ESSX8316", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); +#endif + +static struct dev_pm_ops es8316_pm_ops = { + .suspend = es8316_suspend, + .resume = es8316_resume, +}; + + +static struct i2c_driver es8316_i2c_driver = { + .driver = { + .name = "es8316", + .pm = &es8316_pm_ops, + .acpi_match_table = ACPI_PTR(es8316_acpi_match), + }, + .shutdown = es8316_i2c_shutdown, + .probe = es8316_i2c_probe, + .remove = es8316_i2c_remove, + .id_table = es8316_i2c_id, +}; +module_i2c_driver(es8316_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8316 driver"); +MODULE_AUTHOR("Will <will(a)everset-semi.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 0000000..cb5d035 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,124 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang <yangxiaohua(a)everest-semi.com> + * + * Based on ES8323.h + * + * 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 _ES8316_H +#define _ES8316_H + +/* ES8316 register space */ +/* +* RESET Control +*/ +#define ES8316_RESET_REG00 0x00 +/* +* Clock Managerment +*/ +#define ES8316_CLKMGR_CLKSW_REG01 0x01 +#define ES8316_CLKMGR_CLKSEL_REG02 0x02 +#define ES8316_CLKMGR_ADCOSR_REG03 0x03 +#define ES8316_CLKMGR_ADCDIV1_REG04 0x04 +#define ES8316_CLKMGR_ADCDIV2_REG05 0x05 +#define ES8316_CLKMGR_DACDIV1_REG06 0x06 +#define ES8316_CLKMGR_DACDIV2_REG07 0x07 +#define ES8316_CLKMGR_CPDIV_REG08 0x08 +/* +* SDP Control +*/ +#define ES8316_SDP_MS_BCKDIV_REG09 0x09 +#define ES8316_SDP_ADCFMT_REG0A 0x0a +#define ES8316_SDP_DACFMT_REG0B 0x0b +/* +* System Control +*/ +#define ES8316_SYS_VMIDSEL_REG0C 0x0c +#define ES8316_SYS_PDN_REG0D 0x0d +#define ES8316_SYS_LP1_REG0E 0x0e +#define ES8316_SYS_LP2_REG0F 0x0f +#define ES8316_SYS_VMIDLOW_REG10 0x10 +#define ES8316_SYS_VSEL_REG11 0x11 +#define ES8316_SYS_REF_REG12 0x12 +/* +* HP Mixer +*/ +#define ES8316_HPMIX_SEL_REG13 0x13 +#define ES8316_HPMIX_SWITCH_REG14 0x14 +#define ES8316_HPMIX_PDN_REG15 0x15 +#define ES8316_HPMIX_VOL_REG16 0x16 +/* +* Charge Pump Headphone driver +*/ +#define ES8316_CPHP_OUTEN_REG17 0x17 +#define ES8316_CPHP_ICAL_VOL_REG18 0x18 +#define ES8316_CPHP_PDN1_REG19 0x19 +#define ES8316_CPHP_PDN2_REG1A 0x1a +#define ES8316_CPHP_LDOCTL_REG1B 0x1b +/* +* Calibration +*/ +#define ES8316_CAL_TYPE_REG1C 0x1c +#define ES8316_CAL_SET_REG1D 0x1d +#define ES8316_CAL_HPLIV_REG1E 0x1e +#define ES8316_CAL_HPRIV_REG1F 0x1f +#define ES8316_CAL_HPLMV_REG20 0x20 +#define ES8316_CAL_HPRMV_REG21 0x21 +/* +* ADC Control +*/ +#define ES8316_ADC_PDN_LINSEL_REG22 0x22 +#define ES8316_ADC_PGAGAIN_REG23 0x23 +#define ES8316_ADC_D2SEPGA_REG24 0x24 +#define ES8316_ADC_DMIC_REG25 0x25 +#define ES8316_ADC_MUTE_REG26 0x26 +#define ES8316_ADC_VOLUME_REG27 0x27 +#define ES8316_ADC_ALC1_REG29 0x29 +#define ES8316_ADC_ALC2_REG2A 0x2a +#define ES8316_ADC_ALC3_REG2B 0x2b +#define ES8316_ADC_ALC4_REG2C 0x2c +#define ES8316_ADC_ALC5_REG2D 0x2d +#define ES8316_ADC_ALC6_REG2E 0x2e +/* +* DAC Control +*/ +#define ES8316_DAC_PDN_REG2F 0x2f +#define ES8316_DAC_SET1_REG30 0x30 +#define ES8316_DAC_SET2_REG31 0x31 +#define ES8316_DAC_SET3_REG32 0x32 +#define ES8316_DAC_VOLL_REG33 0x33 +#define ES8316_DAC_VOLR_REG34 0x34 +/* +* GPIO +*/ +#define ES8316_GPIO_SEL_REG4D 0x4D +#define ES8316_GPIO_DEBUNCE_INT_REG4E 0x4E +#define ES8316_GPIO_FLAG 0x4F +/* +* TEST MODE +*/ +#define ES8316_TESTMODE_REG50 0x50 +#define ES8316_TEST1_REG51 0x51 +#define ES8316_TEST2_REG52 0x52 +#define ES8316_TEST3_REG53 0x53 +/* +* es8396 System clock derived from MCLK or BCLK +*/ +#define ES8316_CLKID_MCLK 0 +#define ES8316_CLKID_BCLK 1 +#define ES8316_CLKID_PLLO 2 + + +#define ES8316_IFACE ES8316_SDP_MS_BCKDIV_REG09 +#define ES8316_ADC_IFACE ES8316_SDP_ADCFMT_REG0A +#define ES8316_DAC_IFACE ES8316_SDP_DACFMT_REG0B + +#define ES8316_REGNUM 84 + +#endif diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index fd5d1e0..b3023db 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -210,6 +210,18 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH platforms with MAX98090 audio codec it also can support TI jack chip as aux device. If unsure select "N". +config SND_SOC_INTEL_CHT_ES8316_MACH + tristate "ASoC Audio driver for Intel Cherrytrail with ES8316 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ES8316 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail + platforms with ES8316 audio codec. + If unsure select "N". + config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0838478..25ea25c 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -21,6 +21,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt +#define DEBUG #include <linux/slab.h> #include <sound/soc.h> @@ -894,7 +895,7 @@ int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt) * this can be overwritten by set_dai_xxx APIs */ static const struct sst_ssp_config sst_ssp_configs = { - .ssp_id = SSP_CODEC, + .ssp_id = SSP_BT, .bits_per_slot = 24, .slots = 4, .ssp_mode = SSP_MODE_MASTER, @@ -943,6 +944,8 @@ int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) ssp_id = SSP_MODEM; else if (strcmp(id, "ssp2-port") == 0) ssp_id = SSP_CODEC; + else if (strcmp(id, "ssp1-port") == 0) + ssp_id = SSP_BT; // FIXME what is the right ID? else { dev_dbg(dai->dev, "port %s is not supported\n", id); return -1; diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 0a88537..75997e2 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -458,6 +458,8 @@ static struct sst_acpi_mach sst_acpi_chv[] = { &chv_platform_data }, {"193C9890", "cht-bsw-max98090", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, &chv_platform_data }, + {"ESSX8316", "cht-es8316", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, + &chv_platform_data }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", cht_quirk, &chv_platform_data }, diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 5639f10..2124aaa 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -10,6 +10,7 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-sst-cht-es8316-objs := cht_es8316.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_ES8316_MACH) += snd-soc-sst-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o diff --git a/sound/soc/intel/boards/cht_es8316.c b/sound/soc/intel/boards/cht_es8316.c new file mode 100644 index 0000000..a6e105d --- /dev/null +++ b/sound/soc/intel/boards/cht_es8316.c @@ -0,0 +1,1462 @@ +/* + * cht_cr_es8316.c - ASoc DPCM Machine driver + * for Intel CherryTrail MID platform + * + * Copyright (C) 2014 Intel Corp + * Author: Praveen Diwakar <praveen.diwakar(a)intel.com> + * Bhakte, GurudattaX <gurudattax.bhakte(a)intel.com> + * This file is modified from cht_bl_dpcm_rt5672.c for cherrytrail CR + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that 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. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define DEBUG +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/input.h> +#include <linux/gpio.h> +#include <linux/acpi.h> +#include <linux/slab.h> +#include <linux/input.h> +#include <linux/mutex.h> +#include <linux/clk.h> +#include <asm/intel-mid.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/es8316.h" +#include "../atom/sst-atom-controls.h" + +struct clk *mclk; + +#define CHT_PLAT_CLK_3_HZ 19200000 + +#define CHT_INTR_DEBOUNCE 0 //0 /* updated by everest-david 15-3-4*/ +#define CHT_HS_DEBOUNCE_DELAY 500//500 /* added by everest-david */ +#define CHT_HS_INSERT_DET_DELAY 400//600 +#define CHT_HS_REMOVE_DET_DELAY 40 //100 /* updated by everest-david 15-3-4*/ +#define CHT_BUTTON_PRESS_DELAY 50 +#define CHT_BUTTON_RELEASE_DELAY 50 +#define CHT_HS_DET_POLL_INTRVL 50 +#define CHT_BUTTON_EN_DELAY 0 + +#define CHT_HS_DET_RETRY_COUNT 8 +#define CHT_HS_DEBOUNCE_RETRY_COUNT 3 /* added by everest-david */ +#define CHT_HS_BUTTON_PRESS_CHK_COUNT 3 /* added by everest-david */ +#define CHT_HS_BUTTON_RELEASE_CHK_COUNT 3 /* added by everest-david */ +#define CHT_HS_REMOVE_RETRY_COUNT 1 /* added by everest-david 15-3-4*/ + +#define VLV2_PLAT_CLK_AUDIO 3 +#define PLAT_CLK_FORCE_ON 1 +#define PLAT_CLK_FORCE_OFF 2 + +#define MAINMIC_DMIC_USED 1 +//#define MAINMIC_AMIC_USED 1 + + +struct cht_mc_private { + struct snd_soc_codec *codec; + struct snd_soc_jack jack; + struct delayed_work hs_insert_work; + struct delayed_work hs_remove_work; + struct delayed_work hs_button_press_work; + struct delayed_work hs_button_release_work; + struct delayed_work hs_debounce_work; /* added by everest-david */ + struct delayed_work hs_poll_work; + struct mutex jack_mlock; + struct mutex poll_mlock; + /* To enable button press interrupts after a delay after HS detection. + * This is to avoid spurious button press events during slow + * HS insertion + */ + struct delayed_work hs_button_en_work; + int intr_debounce; + int hs_debounce_delay; /* added by everest-david */ + int hs_debounce_retry; /* added by everest-david */ + int hs_insert_det_delay; + int hs_remove_det_delay; + int hs_remove_retry; /* added by everest-david 15-3-4*/ + int button_press_delay; + int button_press_chk_count; /* added by everest-david */ + int button_release_delay; + int button_release_chk_count; /* added by everest-david */ + int button_en_delay; + int hs_det_poll_intrvl; + int hs_det_retry; + + int same_reg_cnt; + int hs_reg_v; + int poll_cnt; + int poll_status; + int poll_same_cnt; + bool process_button_events; +}; + +static int cht_hs_detection(void *); + +static struct snd_soc_jack_gpio hs_gpio = { + .name = "cht-codec-int", + .report = SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0, + .debounce_time = CHT_INTR_DEBOUNCE, + .jack_status_check = cht_hs_detection, +}; +static inline void cht_force_enable_pin(struct snd_soc_codec *codec, + const char *bias_widget, bool enable) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + pr_debug("%s %s\n", enable ? "enable" : "disable", bias_widget); + if (enable) + snd_soc_dapm_force_enable_pin(dapm, bias_widget); + else + snd_soc_dapm_disable_pin(dapm, bias_widget); +} + +static inline void cht_set_mic_bias(struct snd_soc_codec *codec, bool enable) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + if (enable) + cht_force_enable_pin(codec, "micbias", true); + else + cht_force_enable_pin(codec, "micbias", false); + snd_soc_dapm_sync(dapm); +} + +static inline void cht_set_codec_power(struct snd_soc_codec *codec, + int jack_type) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + switch (jack_type) { + case SND_JACK_HEADSET: + cht_force_enable_pin(codec, "micbias2", true); + cht_force_enable_pin(codec, "JD Power", true); + cht_force_enable_pin(codec, "Mic Det Power", true); + break; + case SND_JACK_HEADPHONE: + cht_force_enable_pin(codec, "JD Power", true); + cht_force_enable_pin(codec, "Mic Det Power", false); + cht_force_enable_pin(codec, "micbias2", false); + break; + case 0: + cht_force_enable_pin(codec, "JD Power", false); + cht_force_enable_pin(codec, "Mic Det Power", false); + cht_force_enable_pin(codec, "micbias2", false); + break; + default: + return; + } + snd_soc_dapm_sync(dapm); +} +static int es8316_headset_detect(struct snd_soc_codec *codec) +{ + + int value; + int status; + + value = snd_soc_read(codec, ES8316_GPIO_FLAG); + //printk("=== es8316_headset_detect: 0x%x\n",value); + switch (value) { + //case 0x24: + case 0x20: + status = SND_JACK_HEADPHONE; + break; + case 0x22: + status = SND_JACK_HEADSET; + break; + default: + status = 0; + break; + } + + return status; +} + +/* Identify the jack type as Headset/Headphone/None */ +static int cht_check_jack_type(struct snd_soc_jack *jack, + struct snd_soc_codec *codec) +{ + int status, jack_type = 0; + struct cht_mc_private *ctx = container_of(jack, + struct cht_mc_private, jack); + + status = es8316_headset_detect(codec); + /* jd status high indicates some accessory has been connected */ + if (status) { + pr_debug("Jack insert intr"); + snd_soc_write(codec,0x30,0x30); + gpio_direction_output(368,0); + + pr_debug("Jack insert intr"); + /* Do not process button events until accessory is + detected as headset*/ + ctx->process_button_events = false; + cht_set_mic_bias(codec, true); + jack_type = es8316_headset_detect(codec); //read flag register to check headphone / headset type / or none + pr_info("jack_type:%d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) { + ctx->process_button_events = true; +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin2rin2 + snd_soc_write(codec, 0x25,0x08); //record path switch to headset mic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin1rin1 +#endif +#endif + + /* If headset is detected, enable button + interrupts after a delay */ + schedule_delayed_work(&ctx->hs_button_en_work, + msecs_to_jiffies(ctx->button_en_delay)); + } + if (jack_type != SND_JACK_HEADSET){ + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + + } + } else { + jack_type = 0; + } + pr_info("Jack type detected:%d\n", jack_type); + return jack_type; +} + +/* Work function invoked by the Jack Infrastructure. + * Other delayed works for jack detection/removal/button + * press are scheduled from this function + */ +static int cht_hs_detection(void *data) +{ + int status, jack_type = 0; + int ret,value; + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(jack, + struct cht_mc_private, jack); + struct snd_soc_codec *codec = ctx->codec; + + pr_info("Enter:%s", __func__); + /* Ack interrupt first */ + mutex_lock(&ctx->jack_mlock); + + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + + /* Initialize jack status with previous status. + The delayed work will confirm the event and + send updated status later */ + + jack_type = jack->status; + value = snd_soc_read(codec, ES8316_GPIO_FLAG); + //printk("=== cht_hs_detection gpio flag: 0x%x\n",value); + if (!jack->status) { + //printk("=== jack->status = 0\n"); + snd_soc_write(codec,0x30,0x30); //mute dac when hp inserted + //snd_soc_write(codec,0x22,0x30); + ctx->hs_det_retry = CHT_HS_DET_RETRY_COUNT; + ctx->hs_debounce_retry = CHT_HS_DEBOUNCE_RETRY_COUNT; + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + ctx->hs_remove_retry = CHT_HS_REMOVE_RETRY_COUNT; //UPDATED BY DAVID,15-3-4 + + ret = schedule_delayed_work(&ctx->hs_insert_work, + msecs_to_jiffies(ctx->hs_insert_det_delay)); + if (!ret){ + pr_info("byt_check_hs_insert_status already queued"); + } else { + pr_info("%s:Check hs insertion after %d msec", + __func__, ctx->hs_insert_det_delay); + } + } else { + /* First check for accessory removal; If not removed, + check for button events*/ + status = es8316_headset_detect(codec); + /* jd status low indicates accessory has been disconnected. + However, confirm the removal in the delayed work */ + if (!status) { + printk("=== >status = 0\n"); + /* Do not process button events while we make sure + accessory is disconnected*/ + ctx->process_button_events = false; + snd_soc_write(codec,0x30,0x20); + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + if (!ret) { + pr_info("byt_check_hs_remove_status already queued"); + }else +{ + pr_info("%s:Check hs removal after %d msec", __func__, + ctx->hs_remove_det_delay); + } + } else { /* Must be button event. + * Confirm the event in delayed work */ + printk("=== >status = 1\n"); + if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) && + ctx->process_button_events) { + ret = schedule_delayed_work(&ctx->hs_button_press_work, + msecs_to_jiffies(ctx->button_press_delay)); + if (!ret) + { + pr_info("byt_check_hs_button_press_status already queued"); + } + else + pr_info("%s:check BP/BR after %d msec", + __func__, + ctx->button_press_delay); + } + else + { + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + if (!ret) { + pr_info("byt_check_hs_remove_status already queued"); + }else { + pr_info("%s:Check hs removal after %d msec", __func__, + ctx->hs_remove_det_delay); + } + } + } + } + + pr_info("Exit:%s", __func__); + mutex_unlock(&ctx->jack_mlock); + + return jack_type; +} +/* Checks jack insertion and identifies the jack type. + * Retries the detection if necessary + */ +static void cht_check_hs_insert_status(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, + struct cht_mc_private, hs_insert_work.work); + int jack_type = 0,ret; + struct snd_soc_codec *codec = ctx->codec; + mutex_lock(&ctx->jack_mlock); + pr_debug("Enter:%s", __func__); + + //jack_type = cht_check_jack_type(jack, codec); + + /* Report jack immediately only if jack is headset. + * If headphone or no jack was detected, dont report it + * until the last HS det try. + * This is to avoid reporting any temporary jack removal or + * accessory change (eg, HP to HS) during the detection tries. + * This provides additional debounce that will help in the case + * of slow insertion. + * This also avoids the pause in audio due to accessory change + * from HP to HS + */ + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if(ctx->hs_reg_v == ret) { + ctx->same_reg_cnt++; + } + else { + ctx->same_reg_cnt = 0; + } + ctx->hs_reg_v = ret; + //if (ctx->hs_det_retry <= 0) /* end of retries; report the status */{ + if (ctx->same_reg_cnt == CHT_HS_DET_RETRY_COUNT) /* end of retries; report the status */{ + jack_type = cht_check_jack_type(jack, codec); + pr_info("%d Jack type sent is %d\n", __LINE__, jack_type); + if(jack_type == 0) { /* if no hp inserted */ + cht_set_mic_bias(codec, true); + ctx->process_button_events = false; + snd_soc_jack_report(jack, jack_type, gpio->report); //report Jack status to Frame-work +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + } else { /* if headset or headphone inserted*/ + snd_soc_jack_report(jack, jack_type, gpio->report); //report Jack status to Frame-work + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); + gpio_direction_output(368, 0); + + } + ctx->same_reg_cnt = 0; + pr_info("jack report:%s,%d\n", __func__, __LINE__); + /* now, schedule one debounce work-quene, to cancel the noise at remove or insert */ + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } else { + /* Schedule another detection try if headphone or + * no jack is detected. + * During slow insertion of headset, first a headphone + * may be detected. + * Hence retry until headset is detected + */ + //ctx->hs_det_retry--; + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + schedule_delayed_work(&ctx->hs_insert_work, + msecs_to_jiffies(ctx->hs_det_poll_intrvl)); + pr_info("%s:re-try hs detection after %d msec", + __func__, ctx->hs_det_poll_intrvl); + } + + mutex_unlock(&ctx->jack_mlock); +} +/* Checks jack removal. */ +static void cht_check_hs_remove_status(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, + struct cht_mc_private, hs_remove_work.work); + struct snd_soc_codec *codec = ctx->codec; + int status = 0, jack_type = 0; + + int ret; + /* Cancel any pending insertion detection. There + could be pending insertion detection in the + case of very slow insertion or insertion and + immediate removal.*/ + + cancel_delayed_work_sync(&ctx->hs_insert_work); + mutex_lock(&ctx->jack_mlock); + pr_info("Enter:%s", __func__); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if(ctx->hs_reg_v == ret) { + ctx->same_reg_cnt++; + } + else { + ctx->same_reg_cnt = 0; + } + ctx->hs_reg_v = ret; + //if(ctx->hs_remove_retry <= 0){ //Updated by david, 15-3-4 + if (ctx->same_reg_cnt == CHT_HS_REMOVE_RETRY_COUNT) /* end of retries; report the status */{ + + ctx->same_reg_cnt = 0; + /* Initialize jack_type with previous status. + If the event was an invalid one, we return the preious state*/ + jack_type = jack->status; + ret = snd_soc_read(codec, 0x50); + if(ret == 0xaa) { + gpio_direction_output(368, 1); //1 + } + + msleep(100); + if (jack->status) { + /* jack is in connected state; look for removal event */ + + status = es8316_headset_detect(codec); + if (!status) { + pr_info("Jack remove event"); + snd_soc_write(codec, 0x30,0x30); + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + //cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + jack_type = 0; + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + + } else if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) && + !ctx->process_button_events) { + /* Jack is still connected. We may come here if + there was a spurious jack removal event. + No state change is done until removal is confirmed + by the check_jd_status above.i.e. jack status + remains Headset or headphone. But as soon as + the interrupt thread(byt_hs_detection) detected a jack + removal, button processing gets disabled. + Hence re-enable button processing in the case of + headset */ + pr_info(" spurious Jack remove event for headset \ + re-enable button events"); + ctx->process_button_events = true; +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin2rin2 + snd_soc_write(codec, 0x25,0x08); //record path switch to headset mic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x20); //switch record channel to lin1rin1 +#endif +#endif + } + } + snd_soc_jack_report(jack, jack_type, gpio->report); + + pr_info("%d Jack type sent is %d\n", __LINE__, jack_type); + if(jack_type == 0) + { + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + //cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + //cancel_delayed_work_sync(&ctx->hs_poll_work); + jack_type = 0; + snd_soc_jack_report(jack, jack_type, gpio->report); + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + snd_soc_write(codec,0x30,0x30); + /*Now, to schedule a debounce work-quenue to cancel the noise at remove moment*/ + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } else + { + snd_soc_jack_report(jack, jack_type, gpio->report); + } + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); + } + else + { + //ctx->hs_remove_retry--; + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + } + pr_info("Exit:%s", __func__); + mutex_unlock(&ctx->jack_mlock); +} +static int es8316_query_btn_press(struct snd_soc_codec *codec) +{ + + int btnstate = -1; + u32 value; + //bool status; + value = snd_soc_read(codec, ES8316_GPIO_FLAG); + + switch (value & 0x6) { + case 0x0: + btnstate = SND_JACK_BTN_0; + break; + case 0x2: + btnstate = 0; + break; + default: + btnstate = 0; + break; + } + return btnstate; +} +/* Check for button press/release */ +static void cht_check_hs_button_press_status(struct work_struct *work) +{ + + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = + container_of(work, struct cht_mc_private, + hs_button_press_work.work); + struct snd_soc_codec *codec = ctx->codec; + int status = 0, jack_type = 0; + int ret; + mutex_lock(&ctx->jack_mlock); + pr_info("Enter:%s\n", __func__); + jack_type = jack->status; + + status = es8316_headset_detect(codec); + if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) + && ctx->process_button_events) { + + status = es8316_headset_detect(codec); + if (status) { /* confirm jack is connected */ + status = es8316_query_btn_press(codec); + if (status & SND_JACK_BTN_0) { + jack_type = SND_JACK_HEADSET | SND_JACK_BTN_0; + pr_info("%d Jack type sent is %d\n", + __LINE__, jack_type); + snd_soc_jack_report(jack, jack_type, gpio->report); + /* Since there is not button_relese interrupt + schedule delayed work to poll for button + release status + */ + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + ret = schedule_delayed_work(&ctx->hs_button_release_work, + msecs_to_jiffies(ctx->button_release_delay)); + } else { + if(ctx->button_press_chk_count >0){ + ctx->button_press_chk_count--; + ret = schedule_delayed_work(&ctx->hs_button_press_work, + msecs_to_jiffies(ctx->button_press_delay)); + } else { + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + } + } + } + else + { + status = es8316_headset_detect(codec); + if(!status){ + ctx->process_button_events = false; + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + + } else { + ctx->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + ret = schedule_delayed_work(&ctx->hs_button_press_work, + msecs_to_jiffies(ctx->button_press_delay)); + } + } + } + if(!jack_type){ + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + jack_type = 0; + cht_set_mic_bias(codec, true); + #ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic + #else + #ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 + #endif + #endif + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } + pr_info("Exit:%s\n", __func__); + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + mutex_unlock(&ctx->jack_mlock); +} + + +/* Check for button release */ +static void cht_check_hs_button_release_status(struct work_struct *work) +{ + + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_release_work.work); + struct snd_soc_codec *codec = ctx->codec; + int status = 0, jack_type = 0; + int ret; + mutex_lock(&ctx->jack_mlock); + pr_info("Enter:%s\n", __func__); + jack_type = jack->status; + + if (((jack->status & SND_JACK_HEADSET) == SND_JACK_HEADSET) + && ctx->process_button_events) { + + status = es8316_headset_detect(codec); + if (status) { /* confirm jack is connected */ + + status = es8316_query_btn_press(codec); + if (!(status & SND_JACK_BTN_0)) { + jack_type = SND_JACK_HEADSET; + pr_info("%d Jack type sent is %d\n", + __LINE__, jack_type); + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + snd_soc_jack_report(jack, jack_type, gpio->report); + } else { + /* Schedule again */ + if(ctx->button_release_chk_count > 0){ + ctx->button_release_chk_count--; + ret = schedule_delayed_work(&ctx->hs_button_release_work, + msecs_to_jiffies(ctx->button_release_delay)); + } else { + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + } + } + + }else { + status = es8316_headset_detect(codec); + if(!status){ + ctx->process_button_events = false; + ret = schedule_delayed_work(&ctx->hs_remove_work, + msecs_to_jiffies(ctx->hs_remove_det_delay)); + } else { + ctx->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + ret = schedule_delayed_work(&ctx->hs_button_release_work, + msecs_to_jiffies(ctx->button_release_delay)); + } + + } + } + if(!jack_type){ + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + jack_type = 0; + cht_set_mic_bias(codec, true); +#ifdef MAINMIC_DMIC_USED + snd_soc_write(codec, 0x22,0x30); //Disable ADC + snd_soc_write(codec, 0x25,0x08); //record path switch to dmic +#else +#ifdef MAINMIC_AMIC_USED + snd_soc_write(codec, 0x22,0x30); //switch record channel to lin2rin2 +#endif +#endif + ret = schedule_delayed_work(&ctx->hs_debounce_work, + msecs_to_jiffies(ctx->hs_debounce_delay)); + } + + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + pr_info("Exit:%s\n", __func__); + mutex_unlock(&ctx->jack_mlock); +} +/* +* added by everest-david +*Used for remove/button debounce +*/ +static void cht_hs_debounce_events(struct work_struct *work) +{ + + int ret; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_release_work.work); + struct snd_soc_codec *codec = ctx->codec; + pr_info("Enter:%s\n", __func__); + + ret = snd_soc_read(codec, ES8316_GPIO_FLAG); + if((ret & 0x06)==0x06){ //remove + ret = snd_soc_read(codec, 0x50); + if(ret == 0xaa) { + gpio_direction_output(368, 1); //1 + } + printk("byt_hs_debounce_remove\n"); + } + else { //inserted + gpio_direction_output(368, 0); + printk("byt_hs_debounce_insert\n"); + } + snd_soc_write(codec,0x30,0x10); +} + +static void cht_hs_poll_events(struct work_struct *work) +{ + + int ret,status; + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_release_work.work); + struct snd_soc_codec *codec = ctx->codec; + //pr_info("Enter:%s\n", __func__); + + mutex_lock(&ctx->poll_mlock); + status = es8316_headset_detect(codec); + if(ctx->poll_status == status){ + ctx->poll_same_cnt++; + } else { + ctx->poll_same_cnt = 0; + } + if(!status){ + //ret = schedule_delayed_work(&ctx->hs_remove_work, + // msecs_to_jiffies(0)); + if(ctx->poll_same_cnt > 2) { + snd_soc_jack_report(jack, status, gpio->report); + ctx->poll_same_cnt = 0; + } else { + if(ctx->poll_cnt < 10){ + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); ctx->poll_cnt++; + } else { + ctx->poll_cnt = 0; + } + } + }else { + if(jack->status != status){ + if(ctx->poll_same_cnt>2) { + snd_soc_jack_report(jack, status, gpio->report); + } + //ret = schedule_delayed_work(&ctx->hs_insert_work, + // msecs_to_jiffies(0)); + } + if(ctx->poll_cnt<10){ + ret = schedule_delayed_work(&ctx->hs_poll_work, + msecs_to_jiffies(500)); + ctx->poll_cnt++; + } else { + ctx->poll_cnt = 0; + } + } + mutex_unlock(&ctx->poll_mlock); + //ret = schedule_delayed_work(&ctx->hs_poll_work, + // msecs_to_jiffies(1000)); + +} + +/* Delayed work for enabling the overcurrent detection circuit + * and interrupt for generating button events */ +static void cht_enable_hs_button_events(struct work_struct *work) +{ +#if 0 + struct snd_soc_jack_gpio *gpio = &hs_gpio; + struct snd_soc_jack *jack = gpio->jack; + struct snd_soc_codec *codec = jack->codec; + struct cht_mc_private *ctx = container_of(work, struct cht_mc_private, + hs_button_en_work.work); + + int status = aic31xx_query_jack_status(codec); + if (status == SND_JACK_HEADSET) + ctx->process_button_events = true; + else + ctx->process_button_events = false; + aic31xx_btn_press_intr_enable(codec, ctx->process_button_events); +#endif +} + +static inline struct snd_soc_codec *cht_get_codec(struct snd_soc_card *card) +{ + bool found = false; + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + printk(KERN_ERR " codec->name = %s\n", codec->component.name); + if (!strstr(codec->component.name, "i2c-ESSX8316:00")) { // "i2c-10ES8316:00")) { + pr_debug("codec was %s", codec->component.name); + continue; + } else { + found = true; + break; + } + } + if (found == false) { + pr_err("%s: cant find codec", __func__); + return NULL; + } + return codec; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_codec *codec; + + codec = cht_get_codec(card); + if (!codec) { + pr_err("Codec not found; Unable to set platform clock\n"); + return -EIO; + } + if (SND_SOC_DAPM_EVENT_ON(event)) { + //vlv2_plat_configure_clock(VLV2_PLAT_CLK_AUDIO, + // PLAT_CLK_FORCE_ON); +clk_prepare_enable(mclk); + pr_info("Platform clk turned ON\n"); + } else { + /* Set codec clock source to internal clock before + * turning off the platform clock. Codec needs clock + * for Jack detection and button press + */ + //vlv2_plat_configure_clock(VLV2_PLAT_CLK_AUDIO, + // PLAT_CLK_FORCE_ON); + clk_disable_unprepare(mclk); + pr_info("Platform clk turned OFF\n"); + } + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_MIC("D-MIC", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU| + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { +#if 0 + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKOUTL"}, + {"Ext Spk", NULL, "SPKOUTR"}, + /* AMIC */ + /* + Don't use RVP resource + for HP detection issue on RVP + */ + {"Headset AMIC", NULL, "MIC Bias"}, + {"MONOINP", NULL, "Headset AMIC"}, + {"MONOINN", NULL, "Headset AMIC"}, + + {"AMIC", NULL, "MIC Bias"}, + {"MIC", NULL, "AMIC"}, +#else + {"Ext Spk", NULL, "SPKOL"}, + {"Ext Spk", NULL, "SPKOR"}, + + {"D-MIC", NULL, "micbias"}, + {"Headset Mic", NULL, "micbias"}, + {"Internal Mic", NULL, "micbias"}, + /* Headset Mic: Headset Mic with bias */ + {"MIC1", NULL, "Headset Mic"}, + {"MIC2", NULL, "Internal Mic"}, + {"DMIC", NULL, "D-MIC"}, + + /* Headset Stereophone(Headphone): HSOL, HSOR */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"D-MIC", NULL, "Platform Clock"}, + {"Internal Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +#endif + {"Playback", NULL, "ssp1 Tx"}, + {"ssp1 Tx", NULL, "codec_out0"}, + {"ssp1 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp1 Rx"}, + {"codec_in1", NULL, "ssp1 Rx"}, + {"ssp1 Rx", NULL, "Capture"}, + {"ssp0 Tx", NULL, "modem_out"}, + {"Playback", NULL, "Platform Clock"}, + {"Capture", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("D-MIC"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + + +static int cht_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; + unsigned int fmt; + int ret; + + pr_debug("Enter:%s", __func__); + + //add for voip call no sound by zm 1211 + if (strncmp(codec_dai->name, "ES8316 HiFi", 11)) + return 0; + + /* I2S Slave Mode*/ + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err("can't set codec DAI configuration %d\n", ret); + return ret; + } + +#if 0 + ret = snd_soc_dai_set_pll(codec_dai, 0, ES8316_PLL_SRC_FRM_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 256); + if (ret < 0) { + pr_err("can't set codec pll***********: %d\n", ret); + return ret; + } + + + if (codec_dai->driver && codec_dai->driver->ops->hw_params) + codec_dai->driver->ops->hw_params(substream, params, codec_dai); + + snd_soc_dai_set_sysclk(codec_dai, ES8316_CLKID_PLLO, CHT_PLAT_CLK_3_HZ, 0); +#endif +#if 0 + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("can't set codec sysclk: %d\n", ret); + return ret; + } +#endif + + snd_soc_dai_set_sysclk(codec_dai, ES8316_CLKID_PLLO, CHT_PLAT_CLK_3_HZ, 0); + + return 0; +} + +static int cht_compr_set_params(struct snd_compr_stream *cstream) +{ + return 0; +} + +static const struct snd_soc_pcm_stream cht_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_compr_ops cht_compr_ops = { + .set_params = cht_compr_set_params, +}; + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + pr_debug("Invoked %s for dailink %s\n", __func__, rtd->dai_link->name); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 4;//2 + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int cht_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + card->dapm.bias_level = level; + pr_debug("card(%s)->bias_level %u\n", card->name, + card->dapm.bias_level); + break; + default: + pr_err("%s: Invalid bias level=%d\n", __func__, level); + ret = -EINVAL; + } + + return ret; +} + +static int cht_audio_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + int ReadBuffer,ReadBuffer1,ReadBuffer2; + struct snd_soc_codec *codec; + struct snd_soc_card *card = runtime->card; + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct gpio_desc *desc; + pr_debug("Enter:%s", __func__); + + codec = cht_get_codec(card); + if (!codec) { + pr_err("Codec not found: %s:failed\n", __func__); + return -EIO; + } + ctx->codec = codec; + desc = devm_gpiod_get_index(codec->dev, NULL, 1, 0); + if (!IS_ERR(desc)) { + hs_gpio.gpio = 307;//desc_to_gpio(desc); + devm_gpiod_put(codec->dev, desc); + pr_info("cht-cr GPIOs - JD/BP-int: %d\n", hs_gpio.gpio); + } else { + hs_gpio.gpio = 307; + pr_err("Failed to get gpio desc for Jack det\n"); + } + pr_err("hs codec gpio %d\n", hs_gpio.gpio); + + /* Set codec bias level */ + cht_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + /* Headset jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &ctx->jack, NULL, 0); + if (ret) { + pr_err("Jack creation failed\n"); +// todo return ret; + } + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + + ret = snd_soc_jack_add_gpios(&ctx->jack, 1, &hs_gpio); + if (ret) { + pr_err("Adding jack GPIO failed with error %d\n", ret); +//todo return ret; + } + + + ret = snd_soc_add_card_controls(card, cht_mc_controls, + ARRAY_SIZE(cht_mc_controls)); + if (ret) { + pr_err("unable to add card controls\n"); + return ret; + } + + ret = snd_soc_dapm_sync(&card->dapm); + if (ret) { + pr_err("unable to sync dapm\n"); + return ret; + } + + ReadBuffer = snd_soc_read(codec, ES8316_GPIO_DEBUNCE_INT_REG4E); + // printk("===probe ReadBuffer: 0x%x\n",ReadBuffer); + ReadBuffer1=ReadBuffer | 0xFE; + ReadBuffer2=ReadBuffer1 & 0x01; + if (ReadBuffer2 & 0x01) + { + // printk("===11111111 ReadBuffer2: 0x%x\n",ReadBuffer2); + snd_soc_write(codec, 0X4E, 0xF2); + } + else + { + //printk("===22222222 ReadBuffer2: 0x%x\n",ReadBuffer2); + snd_soc_write(codec, 0X4E, 0xF3); + } + mdelay(10); + snd_soc_write(codec, 0X4E, 0xF3); + ret = snd_soc_read(codec, 0x4F); + //printk("===read 0x4F reg: 0x%x\n", ret); + + + return ret; +} + +static unsigned int rates_8000_16000[] = { + 8000, + 16000, +}; + +static struct snd_pcm_hw_constraint_list constraints_8000_16000 = { + .count = ARRAY_SIZE(rates_8000_16000), + .list = rates_8000_16000, +}; + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static int cht_8k_16k_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_8000_16000); +} + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + .init = cht_audio_init, + .ignore_suspend = 1, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &cht_aif1_ops, + .ignore_suspend = 1, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .compr_ops = &cht_compr_ops, + }, + /* Back ends */ + { + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "ssp1-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "ES8316 HiFi", + .codec_name = "i2c-ESSX8316:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = cht_codec_fixup, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +#ifdef CONFIG_PM_SLEEP +static int snd_cht_prepare(struct device *dev) +{ + pr_debug("In %s device name\n", __func__); + return snd_soc_suspend(dev); +} + +static void snd_cht_complete(struct device *dev) +{ + pr_debug("In %s\n", __func__); + snd_soc_resume(dev); +} + +static int snd_cht_poweroff(struct device *dev) +{ + pr_debug("In %s\n", __func__); + return snd_soc_poweroff(dev); +} +#else +#define snd_cht_prepare NULL +#define snd_cht_complete NULL +#define snd_cht_poweroff NULL +#endif + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailaud", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .set_bias_level = cht_set_bias_level, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + pr_debug("Entry %s\n", __func__); + + /* Audio Platform clock is on by default. The machine driver requests + * this clock to be turned ON and OFF on playing any stream. But + * until any stream is played the clock remains ON. Hence request the + * clock to be turned OFF initially. + */ + //vlv2_plat_configure_clock(VLV2_PLAT_CLK_AUDIO, PLAT_CLK_FORCE_ON); + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) { + pr_err("allocation failed\n"); + return -ENOMEM; + } + //ret_val = gpio_request(341, "I2S_RESET_GPIO"); + if (ret_val) + { + pr_debug("request gpio fail\n"); + } + + drv->intr_debounce = CHT_INTR_DEBOUNCE; + drv->hs_insert_det_delay = CHT_HS_INSERT_DET_DELAY; + drv->hs_remove_det_delay = CHT_HS_REMOVE_DET_DELAY; + drv->button_press_delay = CHT_BUTTON_PRESS_DELAY; + drv->button_release_delay = CHT_BUTTON_RELEASE_DELAY; + drv->hs_det_poll_intrvl = CHT_HS_DET_POLL_INTRVL; + drv->hs_det_retry = CHT_HS_DET_RETRY_COUNT; + drv->button_press_chk_count = CHT_HS_BUTTON_PRESS_CHK_COUNT; + drv->button_release_chk_count = CHT_HS_BUTTON_RELEASE_CHK_COUNT; + drv->button_en_delay = CHT_BUTTON_EN_DELAY; + drv->hs_debounce_delay = CHT_HS_DEBOUNCE_DELAY; /*added by everest-david*/ + drv->hs_debounce_retry = CHT_HS_DEBOUNCE_RETRY_COUNT; /*added by everest-david*/ + drv->process_button_events = false; + drv->poll_cnt = 0; + drv->poll_status = 0; + drv->poll_same_cnt = 0; + INIT_DELAYED_WORK(&drv->hs_insert_work, cht_check_hs_insert_status); + INIT_DELAYED_WORK(&drv->hs_remove_work, cht_check_hs_remove_status); + INIT_DELAYED_WORK(&drv->hs_button_press_work, + cht_check_hs_button_press_status); + INIT_DELAYED_WORK(&drv->hs_button_release_work, + cht_check_hs_button_release_status); + INIT_DELAYED_WORK(&drv->hs_button_en_work, cht_enable_hs_button_events); + INIT_DELAYED_WORK(&drv->hs_debounce_work, cht_hs_debounce_events); /*added by everest-david*/ + INIT_DELAYED_WORK(&drv->hs_poll_work, cht_hs_poll_events); /*added by everest-david for hampoo, 16-3-3*/ + mutex_init(&drv->jack_mlock); + mutex_init(&drv->poll_mlock); + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + pr_err("snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + pr_info("%s successful\n", __func__); + mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + return ret_val; +} + +static void snd_cht_unregister_jack(struct cht_mc_private *ctx) +{ + /* Set process button events to false so that the button + delayed work will not be scheduled.*/ + ctx->process_button_events = false; + cancel_delayed_work_sync(&ctx->hs_insert_work); + cancel_delayed_work_sync(&ctx->hs_button_en_work); + cancel_delayed_work_sync(&ctx->hs_button_press_work); + cancel_delayed_work_sync(&ctx->hs_button_release_work); + cancel_delayed_work_sync(&ctx->hs_remove_work); + cancel_delayed_work_sync(&ctx->hs_debounce_work); + cancel_delayed_work_sync(&ctx->hs_poll_work); + snd_soc_jack_free_gpios(&ctx->jack, 1, &hs_gpio); +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct cht_mc_private *drv = snd_soc_card_get_drvdata(soc_card); + + pr_debug("In %s\n", __func__); + snd_cht_unregister_jack(drv); + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static void snd_cht_mc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct cht_mc_private *drv = snd_soc_card_get_drvdata(soc_card); + struct snd_soc_codec *codec = drv->codec; + snd_soc_write(codec, ES8316_CPHP_OUTEN_REG17, 0x00); + + pr_debug("In %s\n", __func__); + snd_cht_unregister_jack(drv); +} + +const struct dev_pm_ops snd_cht_cr_mc_pm_ops = { + .prepare = snd_cht_prepare, + .complete = snd_cht_complete, + .poweroff = snd_cht_poweroff, +}; + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .name = "cht-es8316", + .pm = &snd_cht_cr_mc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, + .shutdown = snd_cht_mc_shutdown, +}; + +static int __init snd_cht_driver_init(void) +{ + int ret; + ret = platform_driver_register(&snd_cht_mc_driver); + if (ret) + pr_err("Fail to register Cherrytrail Machine driver cht_es8316\n"); + else + pr_info("Cherrytrail Machine Driver cht_es8316 registerd\n"); + return ret; +} +late_initcall(snd_cht_driver_init); + +static void __exit snd_cht_driver_exit(void) +{ + pr_debug("In %s\n", __func__); + platform_driver_unregister(&snd_cht_mc_driver); +} +module_exit(snd_cht_driver_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) Cherrytrail CR Machine driver"); +MODULE_AUTHOR("Praveen Diwakar <praveen.diwakar(a)intel.com>"); +MODULE_AUTHOR("Bhakte, GurudattaX <gurudattax.bhakte(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht_es8316"); -- 2.9.3

2 3

[alsa-devel] [PATCH 35/35] sound/soc: Convert remaining uses of pr_warning to pr_warn
by Joe Perches 20 Feb '17

20 Feb '17

To enable eventual removal of pr_warning This makes pr_warn use consistent for sound/soc Prior to this patch, there were 5 uses of pr_warning and 10 uses of pr_warn in sound/soc Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/fsl/imx-audmux.c | 6 +++--- sound/soc/samsung/s3c-i2s-v2.c | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..41e877d2b400 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -152,7 +152,7 @@ static void audmux_debugfs_init(void) audmux_debugfs_root = debugfs_create_dir("audmux", NULL); if (!audmux_debugfs_root) { - pr_warning("Failed to create AUDMUX debugfs root\n"); + pr_warn("Failed to create AUDMUX debugfs root\n"); return; } @@ -160,8 +160,8 @@ static void audmux_debugfs_init(void) snprintf(buf, sizeof(buf), "ssi%lu", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) - pr_warning("Failed to create AUDMUX port %lu debugfs file\n", - i); + pr_warn("Failed to create AUDMUX port %lu debugfs file\n", + i); } } diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 644f186fd35c..7423e4257a1b 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -669,13 +669,13 @@ static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) iismod = readl(i2s->regs + S3C2412_IISMOD); if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warning("%s: RXDMA active?\n", __func__); + pr_warn("%s: RXDMA active?\n", __func__); if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warning("%s: TXDMA active?\n", __func__); + pr_warn("%s: TXDMA active?\n", __func__); if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warning("%s: IIS active\n", __func__); + pr_warn("%s: IIS active\n", __func__); } return 0; -- 2.10.0.rc2.1.g053435c

3 5

[alsa-devel] Intel audio fixes on v4.10
by Pierre-Louis Bossart 20 Feb '17

20 Feb '17

As usual since my initial 4.4 HDMI branch, I cherry-picked all the audio-related contributions and fixes already queued for 4.11 on Intel-based platforms and merged them on 4.10. The latest batch contains all the HDMI/DP, Baytrail/Cherrytrail ASoC + clock fixes. If you maintain a distribution or can't get audio to work please look at this branch first: https://github.com/plbossart/sound/tree/topic/v4.10-fixes. Note that I will no longer maintain the legacy HDMI patches now that we have a much better solution queued for upstream and some of the previous hot-plug problems on Asus T100 seem to be solved. Known issues such as support for max98090 with the DPCM driver, rt5645 headset or dmic quality issues, conexant 702x, etc are also not fixed in this branch, they are still work-in-progress on my experimental/codecs branch. If I missed a set of patches please let me know. Enjoy. -Pierre

1 0

[alsa-devel] [PATCH v9 1/1] ALSA: Tascam US-16x08 DSP mixer quirk
by OnkelDead 20 Feb '17

20 Feb '17

constify all URB message templates and string arrays use correkt union elements of snd_ctl_elem_value in get/put route callbacks add sanity check to all integer put callback functions Add mixer quirk for Tascam US-16x08 usb interface. Even that this is an usb compliant device, the input channels and DSP functions (EQ/Compressor) arn't accessible by default. Signed-off-by: Detlef Urban <onkel(a)paraair.de> --- sound/usb/Makefile | 1 + sound/usb/mixer_quirks.c | 5 + sound/usb/mixer_us16x08.c | 1465 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/mixer_us16x08.h | 122 ++++ 4 files changed, 1593 insertions(+) create mode 100644 sound/usb/mixer_us16x08.c create mode 100644 sound/usb/mixer_us16x08.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..42cb33b 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -10,6 +10,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_us16x08.o \ pcm.o \ proc.o \ quirks.o \ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 04991b0..4fa0053 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -43,6 +43,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_us16x08.h" #include "helper.h" extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; @@ -1729,6 +1730,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) return err; switch (mixer->chip->usb_id) { + /* Tascam US-16x08 */ + case USB_ID(0x0644, 0x8047): + err = snd_us16x08_controls_create(mixer); + break; case USB_ID(0x041e, 0x3020): case USB_ID(0x041e, 0x3040): case USB_ID(0x041e, 0x3042): diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c new file mode 100644 index 0000000..301939b --- /dev/null +++ b/sound/usb/mixer_us16x08.c @@ -0,0 +1,1465 @@ +/* + * Tascam US-16x08 ALSA driver + * + * Copyright (c) 2016 by Detlef Urban (onkel(a)paraair.de) + * + * 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. + * + * This program is distributed in the hope that 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. + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/control.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_us16x08.h" + +/* USB control message templates */ +static const char route_msg[] = { + 0x61, + 0x02, + 0x03, /* input from master (0x02) or input from computer bus (0x03) */ + 0x62, + 0x02, + 0x01, /* input index (0x01/0x02 eq. left/right) or bus (0x01-0x08) */ + 0x41, + 0x01, + 0x61, + 0x02, + 0x01, + 0x62, + 0x02, + 0x01, /* output index (0x01-0x08) */ + 0x42, + 0x01, + 0x43, + 0x01, + 0x00, + 0x00 +}; + +static const char mix_init_msg1[] = { + 0x71, 0x01, 0x00, 0x00 +}; + +static const char mix_init_msg2[] = { + 0x62, 0x02, 0x00, 0x61, 0x02, 0x04, 0xb1, 0x01, 0x00, 0x00 +}; + +static const char mix_msg_in[] = { + /* default message head, equal to all mixers */ + 0x61, 0x02, 0x04, 0x62, 0x02, 0x01, + 0x81, /* 0x06: Controller ID */ + 0x02, /* 0x07: */ + 0x00, /* 0x08: Value of common mixer */ + 0x00, + 0x00 +}; + +static const char mix_msg_out[] = { + /* default message head, equal to all mixers */ + 0x61, 0x02, 0x02, 0x62, 0x02, 0x01, + 0x81, /* 0x06: Controller ID */ + 0x02, /* 0x07: */ + 0x00, /* 0x08: Value of common mixer */ + 0x00, + 0x00 +}; + +static const char bypass_msg_out[] = { + 0x45, + 0x02, + 0x01, /* on/off flag */ + 0x00, + 0x00 +}; + +static const char bus_msg_out[] = { + 0x44, + 0x02, + 0x01, /* on/off flag */ + 0x00, + 0x00 +}; + +static const char comp_msg[] = { + /* default message head, equal to all mixers */ + 0x61, 0x02, 0x04, 0x62, 0x02, 0x01, + 0x91, + 0x02, + 0xf0, /* 0x08: Threshold db (8) (e0 ... 00) (+-0dB -- -32dB) x-32 */ + 0x92, + 0x02, + 0x0a, /* 0x0b: Ratio (0a,0b,0d,0f,11,14,19,1e,23,28,32,3c,50,a0,ff) */ + 0x93, + 0x02, + 0x02, /* 0x0e: Attack (0x02 ... 0xc0) (2ms ... 200ms) */ + 0x94, + 0x02, + 0x01, /* 0x11: Release (0x01 ... 0x64) (10ms ... 1000ms) x*10 */ + 0x95, + 0x02, + 0x03, /* 0x14: gain (0 ... 20) (0dB .. 20dB) */ + 0x96, + 0x02, + 0x01, + 0x97, + 0x02, + 0x01, /* 0x1a: main Comp switch (0 ... 1) (off ... on)) */ + 0x00, + 0x00 +}; + +static const char eqs_msq[] = { + /* default message head, equal to all mixers */ + 0x61, 0x02, 0x04, 0x62, 0x02, 0x01, + 0x51, /* 0x06: Controller ID */ + 0x02, + 0x04, /* 0x08: EQ set num (0x01..0x04) (LOW, LOWMID, HIGHMID, HIGH)) */ + 0x52, + 0x02, + 0x0c, /* 0x0b: value dB (0 ... 12) (-12db .. +12db) x-6 */ + 0x53, + 0x02, + 0x0f, /* 0x0e: value freq (32-47) (1.7kHz..18kHz) */ + 0x54, + 0x02, + 0x02, /* 0x11: band width (0-6) (Q16-Q0.25) 2^x/4 (EQ xxMID only) */ + 0x55, + 0x02, + 0x01, /* 0x14: main EQ switch (0 ... 1) (off ... on)) */ + 0x00, + 0x00 +}; + +/* compressor ratio map */ +static const char ratio_map[] = { + 0x0a, 0x0b, 0x0d, 0x0f, 0x11, 0x14, 0x19, 0x1e, + 0x23, 0x28, 0x32, 0x3c, 0x50, 0xa0, 0xff +}; + +/* route enumeration names */ +const const char *route_names[] = { + "Master Left", "Master Right", "Output 1", "Output 2", "Output 3", + "Output 4", "Output 5", "Output 6", "Output 7", "Output 8", +}; + +static int snd_us16x08_recv_urb(struct snd_usb_audio *chip, + unsigned char *buf, int size) +{ + + mutex_lock(&chip->mutex); + snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + SND_US16X08_URB_METER_REQUEST, + SND_US16X08_URB_METER_REQUESTTYPE, 0, 0, buf, size); + mutex_unlock(&chip->mutex); + return 0; +} + +/* wrapper function to send prepared URB buffer to usb device. Return an error + * code if something went wrong + */ +static int snd_us16x08_send_urb(struct snd_usb_audio *chip, char *buf, int size) +{ + int err = 0; + + if (chip) { + err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), + SND_US16X08_URB_REQUEST, SND_US16X08_URB_REQUESTTYPE, + 0, 0, buf, size); + } + + return err; +} + +static int snd_us16x08_route_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_ctl_enum_info(uinfo, 1, 10, route_names); +} + +static int snd_us16x08_route_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + int index = ucontrol->id.index; + + /* route has no bias */ + ucontrol->value.enumerated.item[0] = elem->cache_val[index]; + + return 0; +} + +static int snd_us16x08_route_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + int index = ucontrol->id.index; + char buf[sizeof(route_msg)]; + int val, val_org, err = 0; + + /* prepare the message buffer from template */ + memcpy(buf, route_msg, sizeof(route_msg)); + + /* get the new value (no bias for routes) */ + val = ucontrol->value.enumerated.item[0]; + + /* sanity check */ + if (val < 0 || val > 9) + return -EINVAL; + + if (val < 2) { + /* input comes from a master channel */ + val_org = val; + buf[2] = 0x02; + } else { + /* input comes from a computer channel */ + buf[2] = 0x03; + val_org = val - 2; + } + + /* place new route selection in URB message */ + buf[5] = (unsigned char) (val_org & 0x0f) + 1; + /* place route selector in URB message */ + buf[13] = index + 1; + + err = snd_us16x08_send_urb(chip, buf, sizeof(route_msg)); + + if (err > 0) { + elem->cached |= 1 << index; + elem->cache_val[index] = val; + } else { + usb_audio_dbg(chip, "Failed to set routing, err:%d\n", err); + } + + return err > 0 ? 1 : 0; +} + +static int snd_us16x08_master_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->count = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.max = SND_US16X08_KCMAX(kcontrol); + uinfo->value.integer.min = SND_US16X08_KCMIN(kcontrol); + uinfo->value.integer.step = SND_US16X08_KCSTEP(kcontrol); + return 0; +} + +static int snd_us16x08_master_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + int index = ucontrol->id.index; + + ucontrol->value.integer.value[0] = elem->cache_val[index]; + + return 0; +} + +static int snd_us16x08_master_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + char buf[sizeof(mix_msg_out)]; + int val, err = 0; + int index = ucontrol->id.index; + + /* prepare the message buffer from template */ + memcpy(buf, mix_msg_out, sizeof(mix_msg_out)); + + /* new control value incl. bias*/ + val = ucontrol->value.integer.value[0]; + + /* sanity check */ + if (val < SND_US16X08_KCMIN(kcontrol) + || val > SND_US16X08_KCMAX(kcontrol)) + return -EINVAL; + + buf[8] = val - SND_US16X08_KCBIAS(kcontrol); + buf[6] = elem->head.id; + + /* place channel selector in URB message */ + buf[5] = index + 1; + err = snd_us16x08_send_urb(chip, buf, sizeof(mix_msg_out)); + + if (err > 0) { + elem->cached |= 1 << index; + elem->cache_val[index] = val; + } else { + usb_audio_dbg(chip, "Failed to set master, err:%d\n", err); + } + + return err > 0 ? 1 : 0; +} + +static int snd_us16x08_bus_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + char buf[sizeof(mix_msg_out)]; + int val, err = 0; + + val = ucontrol->value.integer.value[0]; + + /* prepare the message buffer from template */ + switch (elem->head.id) { + case SND_US16X08_ID_BYPASS: + memcpy(buf, bypass_msg_out, sizeof(bypass_msg_out)); + buf[2] = val; + err = snd_us16x08_send_urb(chip, buf, sizeof(bypass_msg_out)); + break; + case SND_US16X08_ID_BUSS_OUT: + memcpy(buf, bus_msg_out, sizeof(bus_msg_out)); + buf[2] = val; + err = snd_us16x08_send_urb(chip, buf, sizeof(bus_msg_out)); + break; + case SND_US16X08_ID_MUTE: + memcpy(buf, mix_msg_out, sizeof(mix_msg_out)); + buf[8] = val; + buf[6] = elem->head.id; + buf[5] = 1; + err = snd_us16x08_send_urb(chip, buf, sizeof(mix_msg_out)); + break; + } + + if (err > 0) { + elem->cached |= 1; + elem->cache_val[0] = val; + } else { + usb_audio_dbg(chip, "Failed to set buss param, err:%d\n", err); + } + + return err > 0 ? 1 : 0; +} + +static int snd_us16x08_bus_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + + switch (elem->head.id) { + case SND_US16X08_ID_BUSS_OUT: + ucontrol->value.integer.value[0] = elem->cache_val[0]; + break; + case SND_US16X08_ID_BYPASS: + ucontrol->value.integer.value[0] = elem->cache_val[0]; + break; + case SND_US16X08_ID_MUTE: + ucontrol->value.integer.value[0] = elem->cache_val[0]; + break; + } + + return 0; +} + +/* gets a current mixer value from common store */ +static int snd_us16x08_channel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + int index = ucontrol->id.index; + + ucontrol->value.integer.value[0] = elem->cache_val[index]; + + return 0; +} + +static int snd_us16x08_channel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + char buf[sizeof(mix_msg_in)]; + int val, err; + int index = ucontrol->id.index; + + /* prepare URB message from template */ + memcpy(buf, mix_msg_in, sizeof(mix_msg_in)); + + val = ucontrol->value.integer.value[0]; + + /* sanity check */ + if (val < SND_US16X08_KCMIN(kcontrol) + || val > SND_US16X08_KCMAX(kcontrol)) + return -EINVAL; + + /* add the bias to the new value */ + buf[8] = val - SND_US16X08_KCBIAS(kcontrol); + buf[6] = elem->head.id; + buf[5] = index + 1; + + err = snd_us16x08_send_urb(chip, buf, sizeof(mix_msg_in)); + + if (err > 0) { + elem->cached |= 1 << index; + elem->cache_val[index] = val; + } else { + usb_audio_dbg(chip, "Failed to set channel, err:%d\n", err); + } + + return err > 0 ? 1 : 0; +} + +static int snd_us16x08_mix_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->count = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.max = SND_US16X08_KCMAX(kcontrol); + uinfo->value.integer.min = SND_US16X08_KCMIN(kcontrol); + uinfo->value.integer.step = SND_US16X08_KCSTEP(kcontrol); + return 0; +} + +static int snd_us16x08_comp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_us16x08_comp_store *store = + ((struct snd_us16x08_comp_store *) elem->private_data); + int index = ucontrol->id.index; + int val_idx = COMP_STORE_IDX(elem->head.id); + + ucontrol->value.integer.value[0] = store->val[val_idx][index]; + + return 0; +} + +static int snd_us16x08_comp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + struct snd_us16x08_comp_store *store = + ((struct snd_us16x08_comp_store *) elem->private_data); + int index = ucontrol->id.index; + char buf[sizeof(comp_msg)]; + int val_idx, val; + int err = 0; + + /* prepare compressor URB message from template */ + memcpy(buf, comp_msg, sizeof(comp_msg)); + + /* new control value incl. bias*/ + val_idx = elem->head.id - SND_US16X08_ID_COMP_BASE; + + val = ucontrol->value.integer.value[0]; + + /* sanity check */ + if (val < SND_US16X08_KCMIN(kcontrol) + || val > SND_US16X08_KCMAX(kcontrol)) + return -EINVAL; + + store->val[val_idx][index] = ucontrol->value.integer.value[0]; + + /* place comp values in message buffer watch bias! */ + buf[8] = store->val[ + COMP_STORE_IDX(SND_US16X08_ID_COMP_THRESHOLD)][index] + - SND_US16X08_COMP_THRESHOLD_BIAS; + buf[11] = ratio_map[store->val[ + COMP_STORE_IDX(SND_US16X08_ID_COMP_RATIO)][index]]; + buf[14] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_ATTACK)][index] + + SND_US16X08_COMP_ATTACK_BIAS; + buf[17] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_RELEASE)][index] + + SND_US16X08_COMP_RELEASE_BIAS; + buf[20] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_GAIN)][index]; + buf[26] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)][index]; + + /* place channel selector in message buffer */ + buf[5] = index + 1; + + err = snd_us16x08_send_urb(chip, buf, sizeof(comp_msg)); + + if (err > 0) { + elem->cached |= 1 << index; + elem->cache_val[index] = val; + } else { + usb_audio_dbg(chip, "Failed to set compressor, err:%d\n", err); + } + + return 1; +} + +static int snd_us16x08_eqswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = 0; + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_us16x08_eq_store *store = + ((struct snd_us16x08_eq_store *) elem->private_data); + int index = ucontrol->id.index; + + /* get low switch from cache is enough, cause all bands are together */ + val = store->val[EQ_STORE_BAND_IDX(elem->head.id)] + [EQ_STORE_PARAM_IDX(elem->head.id)][index]; + ucontrol->value.integer.value[0] = val; + + return 0; +} + +static int snd_us16x08_eqswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + struct snd_us16x08_eq_store *store = + ((struct snd_us16x08_eq_store *) elem->private_data); + int index = ucontrol->id.index; + + char buf[sizeof(eqs_msq)]; + int val, err = 0; + int b_idx; + + /* new control value incl. bias*/ + val = ucontrol->value.integer.value[0] + SND_US16X08_KCBIAS(kcontrol); + + /* prepare URB message from EQ template */ + memcpy(buf, eqs_msq, sizeof(eqs_msq)); + + /* place channel index in URB message */ + buf[5] = index + 1; + for (b_idx = 0; b_idx < SND_US16X08_ID_EQ_BAND_COUNT; b_idx++) { + /* all four EQ bands have to be enabled/disabled in once */ + buf[20] = val; + buf[17] = store->val[b_idx][2][index]; + buf[14] = store->val[b_idx][1][index]; + buf[11] = store->val[b_idx][0][index]; + buf[8] = b_idx + 1; + err = snd_us16x08_send_urb(chip, buf, sizeof(eqs_msq)); + if (err < 0) + break; + store->val[b_idx][3][index] = val; + msleep(15); + } + + if (err > 0) { + elem->cached |= 1 << index; + elem->cache_val[index] = val; + } else { + usb_audio_dbg(chip, "Failed to set eq switch, err:%d\n", err); + } + + return 1; +} + +static int snd_us16x08_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = 0; + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_us16x08_eq_store *store = + ((struct snd_us16x08_eq_store *) elem->private_data); + int index = ucontrol->id.index; + int b_idx = EQ_STORE_BAND_IDX(elem->head.id) - 1; + int p_idx = EQ_STORE_PARAM_IDX(elem->head.id); + + val = store->val[b_idx][p_idx][index]; + + ucontrol->value.integer.value[0] = val; + + return 0; +} + +static int snd_us16x08_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + struct snd_us16x08_eq_store *store = + ((struct snd_us16x08_eq_store *) elem->private_data); + int index = ucontrol->id.index; + char buf[sizeof(eqs_msq)]; + int val, err = 0; + int b_idx = EQ_STORE_BAND_IDX(elem->head.id) - 1; + int p_idx = EQ_STORE_PARAM_IDX(elem->head.id); + + /* copy URB buffer from EQ template */ + memcpy(buf, eqs_msq, sizeof(eqs_msq)); + + val = ucontrol->value.integer.value[0]; + + /* sanity check */ + if (val < SND_US16X08_KCMIN(kcontrol) + || val > SND_US16X08_KCMAX(kcontrol)) + return -EINVAL; + + store->val[b_idx][p_idx][index] = val; + buf[20] = store->val[b_idx][3][index]; + buf[17] = store->val[b_idx][2][index]; + buf[14] = store->val[b_idx][1][index]; + buf[11] = store->val[b_idx][0][index]; + + /* place channel index in URB buffer */ + buf[5] = index + 1; + + /* place EQ band in URB buffer */ + buf[8] = b_idx + 1; + + err = snd_us16x08_send_urb(chip, buf, sizeof(eqs_msq)); + + if (err > 0) { + /* store new value in EQ band cache */ + elem->cached |= 1 << index; + elem->cache_val[index] = val; + } else { + usb_audio_dbg(chip, "Failed to set eq param, err:%d\n", err); + } + + return 1; +} + +static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->count = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.max = 0x7FFF; + uinfo->value.integer.min = 0; + + return 0; +} + +/* calculate compressor index for reduction level request */ +static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) +{ + int ret; + + /* any channel active */ + if (store->comp_active_index) { + /* check for stereo link */ + if (store->comp_active_index & 0x20) { + /* reset comp_index to left channel*/ + if (store->comp_index - + store->comp_active_index > 1) + store->comp_index = + store->comp_active_index; + + ret = store->comp_index++ & 0x1F; + } else { + /* no stereo link */ + ret = store->comp_active_index; + } + } else { + /* skip channels with no compressor active */ + while (!store->comp_store->val[ + COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] + [store->comp_index - 1] + && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + store->comp_index++; + } + ret = store->comp_index++; + if (store->comp_index > SND_US16X08_MAX_CHANNELS) + store->comp_index = 1; + } + return ret; +} + +/* retrieve the meter level values from URB message */ +static void get_meter_levels_from_urb(int s, + struct snd_us16x08_meter_store *store, + u8 *meter_urb) +{ + int val = MUC2(meter_urb, s) + (MUC3(meter_urb, s) << 8); + + if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && + MUA2(meter_urb, s) == 0x04 && MUB0(meter_urb, s) == 0x62) { + if (MUC0(meter_urb, s) == 0x72) + store->meter_level[MUB2(meter_urb, s) - 1] = val; + if (MUC0(meter_urb, s) == 0xb2) + store->comp_level[MUB2(meter_urb, s) - 1] = val; + } + if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && + MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) + store->master_level[MUB2(meter_urb, s) - 1] = val; +} + +/* Function to retrieve current meter values from the device. + * + * The device needs to be polled for meter values with an initial + * requests. It will return with a sequence of different meter value + * packages. The first request (case 0:) initiate this meter response sequence. + * After the third response, an additional request can be placed, + * to retrieve compressor reduction level value for given channel. This round + * trip channel selector will skip all inactive compressors. + * A mixer can interrupt this round-trip by selecting one ore two (stereo-link) + * specific channels. + */ +static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int i, set; + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_usb_audio *chip = elem->head.mixer->chip; + struct snd_us16x08_meter_store *store = elem->private_data; + u8 meter_urb[64]; + char tmp[max(sizeof(mix_init_msg1), sizeof(mix_init_msg2))]; + + if (elem) { + store = (struct snd_us16x08_meter_store *) elem->private_data; + chip = elem->head.mixer->chip; + } else + return 0; + + switch (kcontrol->private_value) { + case 0: + memcpy(tmp, mix_init_msg1, sizeof(mix_init_msg1)); + snd_us16x08_send_urb(chip, tmp, 4); + snd_us16x08_recv_urb(chip, meter_urb, + sizeof(meter_urb)); + kcontrol->private_value++; + break; + case 1: + snd_us16x08_recv_urb(chip, meter_urb, + sizeof(meter_urb)); + kcontrol->private_value++; + break; + case 2: + snd_us16x08_recv_urb(chip, meter_urb, + sizeof(meter_urb)); + kcontrol->private_value++; + break; + case 3: + memcpy(tmp, mix_init_msg2, sizeof(mix_init_msg2)); + tmp[2] = snd_get_meter_comp_index(store); + snd_us16x08_send_urb(chip, tmp, 10); + snd_us16x08_recv_urb(chip, meter_urb, + sizeof(meter_urb)); + kcontrol->private_value = 0; + break; + } + + for (set = 0; set < 6; set++) + get_meter_levels_from_urb(set, store, meter_urb); + + for (i = 0; i < SND_US16X08_MAX_CHANNELS; i++) { + ucontrol->value.integer.value[i] = + store ? store->meter_level[i] : 0; + } + + ucontrol->value.integer.value[i++] = store ? store->master_level[0] : 0; + ucontrol->value.integer.value[i++] = store ? store->master_level[1] : 0; + + for (i = 2; i < SND_US16X08_MAX_CHANNELS + 2; i++) + ucontrol->value.integer.value[i + SND_US16X08_MAX_CHANNELS] = + store ? store->comp_level[i - 2] : 0; + + return 1; +} + +static int snd_us16x08_meter_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kcontrol->private_data; + struct snd_us16x08_meter_store *store = elem->private_data; + int val; + + val = ucontrol->value.integer.value[0]; + + /* sanity check */ + if (val < 0 || val >= SND_US16X08_MAX_CHANNELS) + return -EINVAL; + + store->comp_active_index = val; + store->comp_index = val; + + return 1; +} + +static struct snd_kcontrol_new snd_us16x08_ch_boolean_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_switch_info, + .get = snd_us16x08_channel_get, + .put = snd_us16x08_channel_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1) +}; + +static struct snd_kcontrol_new snd_us16x08_ch_int_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_channel_get, + .put = snd_us16x08_channel_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_FADER_BIAS, 1, 0, 133) +}; + +static struct snd_kcontrol_new snd_us16x08_pan_int_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_channel_get, + .put = snd_us16x08_channel_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_FADER_BIAS, 1, 0, 255) +}; + +static struct snd_kcontrol_new snd_us16x08_master_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 1, + .info = snd_us16x08_master_info, + .get = snd_us16x08_master_get, + .put = snd_us16x08_master_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_FADER_BIAS, 1, 0, 133) +}; + +static struct snd_kcontrol_new snd_us16x08_route_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 8, + .info = snd_us16x08_route_info, + .get = snd_us16x08_route_get, + .put = snd_us16x08_route_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 9) +}; + +static struct snd_kcontrol_new snd_us16x08_bus_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 1, + .info = snd_us16x08_switch_info, + .get = snd_us16x08_bus_get, + .put = snd_us16x08_bus_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1) +}; + +static struct snd_kcontrol_new snd_us16x08_compswitch_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_switch_info, + .get = snd_us16x08_comp_get, + .put = snd_us16x08_comp_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1) +}; + +static struct snd_kcontrol_new snd_us16x08_comp_threshold_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_comp_get, + .put = snd_us16x08_comp_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_COMP_THRESHOLD_BIAS, 1, + 0, 0x20) +}; + +static struct snd_kcontrol_new snd_us16x08_comp_ratio_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_comp_get, + .put = snd_us16x08_comp_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, + sizeof(ratio_map) - 1), /*max*/ +}; + +static struct snd_kcontrol_new snd_us16x08_comp_gain_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_comp_get, + .put = snd_us16x08_comp_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x14) +}; + +static struct snd_kcontrol_new snd_us16x08_comp_attack_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_comp_get, + .put = snd_us16x08_comp_put, + .private_value = + SND_US16X08_KCSET(SND_US16X08_COMP_ATTACK_BIAS, 1, 0, 0xc6), +}; + +static struct snd_kcontrol_new snd_us16x08_comp_release_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_comp_get, + .put = snd_us16x08_comp_put, + .private_value = + SND_US16X08_KCSET(SND_US16X08_COMP_RELEASE_BIAS, 1, 0, 0x63), +}; + +static struct snd_kcontrol_new snd_us16x08_eq_gain_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_eq_get, + .put = snd_us16x08_eq_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 24), +}; + +static struct snd_kcontrol_new snd_us16x08_eq_low_freq_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_eq_get, + .put = snd_us16x08_eq_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x1F), +}; + +static struct snd_kcontrol_new snd_us16x08_eq_mid_freq_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_eq_get, + .put = snd_us16x08_eq_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x3F) +}; + +static struct snd_kcontrol_new snd_us16x08_eq_mid_width_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_eq_get, + .put = snd_us16x08_eq_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x06) +}; + +static struct snd_kcontrol_new snd_us16x08_eq_high_freq_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_mix_info, + .get = snd_us16x08_eq_get, + .put = snd_us16x08_eq_put, + .private_value = + SND_US16X08_KCSET(SND_US16X08_EQ_HIGHFREQ_BIAS, 1, 0, 0x1F) +}; + +static struct snd_kcontrol_new snd_us16x08_eq_switch_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 16, + .info = snd_us16x08_switch_info, + .get = snd_us16x08_eqswitch_get, + .put = snd_us16x08_eqswitch_put, + .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1) +}; + +static struct snd_kcontrol_new snd_us16x08_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .count = 1, + .info = snd_us16x08_meter_info, + .get = snd_us16x08_meter_get, + .put = snd_us16x08_meter_put +}; + +/* control store preparation */ + +/* setup compressor store and assign default value */ +static struct snd_us16x08_comp_store *snd_us16x08_create_comp_store(void) +{ + int i = 0; + struct snd_us16x08_comp_store *tmp = + kmalloc(sizeof(struct snd_us16x08_comp_store), GFP_KERNEL); + + if (tmp == NULL) + return NULL; + + for (i = 0; i < SND_US16X08_MAX_CHANNELS; i++) { + tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_THRESHOLD)][i] + = 0x20; + tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_RATIO)][i] = 0x00; + tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_GAIN)][i] = 0x00; + tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)][i] = 0x00; + tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_ATTACK)][i] = 0x00; + tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_RELEASE)][i] = 0x00; + } + return tmp; +} + +/* setup EQ store and assign default values */ +static struct snd_us16x08_eq_store *snd_us16x08_create_eq_store(void) +{ + int i, b_idx; + struct snd_us16x08_eq_store *tmp = + kmalloc(sizeof(struct snd_us16x08_eq_store), GFP_KERNEL); + + if (tmp == NULL) + return NULL; + + for (i = 0; i < SND_US16X08_MAX_CHANNELS; i++) { + for (b_idx = 0; b_idx < SND_US16X08_ID_EQ_BAND_COUNT; b_idx++) { + tmp->val[b_idx][0][i] = 0x0c; + tmp->val[b_idx][3][i] = 0x00; + switch (b_idx) { + case 0: /* EQ Low */ + tmp->val[b_idx][1][i] = 0x05; + tmp->val[b_idx][2][i] = 0xff; + break; + case 1: /* EQ Mid low */ + tmp->val[b_idx][1][i] = 0x0e; + tmp->val[b_idx][2][i] = 0x02; + break; + case 2: /* EQ Mid High */ + tmp->val[b_idx][1][i] = 0x1b; + tmp->val[b_idx][2][i] = 0x02; + break; + case 3: /* EQ High */ + tmp->val[b_idx][1][i] = 0x2f + - SND_US16X08_EQ_HIGHFREQ_BIAS; + tmp->val[b_idx][2][i] = 0xff; + break; + } + } + } + return tmp; +} + +struct snd_us16x08_meter_store *snd_us16x08_create_meter_store(void) +{ + struct snd_us16x08_meter_store *tmp = + kzalloc(sizeof(struct snd_us16x08_meter_store), GFP_KERNEL); + + if (!tmp) + return NULL; + tmp->comp_index = 1; + tmp->comp_active_index = 0; + return tmp; + +} + +static int add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int val_type, int channels, + const char *name, const void *opt, + void (*freeer)(struct snd_kcontrol *kctl), + struct usb_mixer_elem_info **elem_ret) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + usb_audio_dbg(mixer->chip, "us16x08 add mixer %s\n", name); + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->head.resume = NULL; + elem->control = 0; + elem->idx_off = 0; + elem->head.id = index; + elem->val_type = val_type; + elem->channels = channels; + elem->private_data = (void *) opt; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + + kctl->private_free = freeer; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (elem_ret) + *elem_ret = elem; + + return 0; +} + +static struct snd_us16x08_control_params control_params; + +/* table of EQ controls */ +static struct snd_us16x08_control_params eq_controls[] = { + { /* EQ switch */ + .kcontrol_new = &snd_us16x08_eq_switch_ctl, + .control_id = SND_US16X08_ID_EQENABLE, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "EQ Switch", + .freeer = snd_usb_mixer_elem_free + }, + { /* EQ low gain */ + .kcontrol_new = &snd_us16x08_eq_gain_ctl, + .control_id = SND_US16X08_ID_EQLOWLEVEL, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ Low Volume", + .freeer = snd_usb_mixer_elem_free + }, + { /* EQ low freq */ + .kcontrol_new = &snd_us16x08_eq_low_freq_ctl, + .control_id = SND_US16X08_ID_EQLOWFREQ, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ Low Frequence", + .freeer = NULL + }, + { /* EQ mid low gain */ + .kcontrol_new = &snd_us16x08_eq_gain_ctl, + .control_id = SND_US16X08_ID_EQLOWMIDLEVEL, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ MidLow Volume", + .freeer = snd_usb_mixer_elem_free + }, + { /* EQ mid low freq */ + .kcontrol_new = &snd_us16x08_eq_mid_freq_ctl, + .control_id = SND_US16X08_ID_EQLOWMIDFREQ, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ MidLow Frequence", + .freeer = NULL + }, + { /* EQ mid low Q */ + .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, + .control_id = SND_US16X08_ID_EQLOWMIDWIDTH, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ MidQLow Q", + .freeer = NULL + }, + { /* EQ mid high gain */ + .kcontrol_new = &snd_us16x08_eq_gain_ctl, + .control_id = SND_US16X08_ID_EQHIGHMIDLEVEL, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ MidHigh Volume", + .freeer = snd_usb_mixer_elem_free + }, + { /* EQ mid high freq */ + .kcontrol_new = &snd_us16x08_eq_mid_freq_ctl, + .control_id = SND_US16X08_ID_EQHIGHMIDFREQ, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ MidHigh Frequence", + .freeer = NULL + }, + { /* EQ mid high Q */ + .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, + .control_id = SND_US16X08_ID_EQHIGHMIDWIDTH, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ MidHigh Q", + .freeer = NULL + }, + { /* EQ high gain */ + .kcontrol_new = &snd_us16x08_eq_gain_ctl, + .control_id = SND_US16X08_ID_EQHIGHLEVEL, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ High Volume", + .freeer = snd_usb_mixer_elem_free + }, + { /* EQ low freq */ + .kcontrol_new = &snd_us16x08_eq_high_freq_ctl, + .control_id = SND_US16X08_ID_EQHIGHFREQ, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "EQ High Frequence", + .freeer = NULL + }, +}; + +/* table of compressor controls */ +static struct snd_us16x08_control_params comp_controls[] = { + { /* Comp enable */ + .kcontrol_new = &snd_us16x08_compswitch_ctl, + .control_id = SND_US16X08_ID_COMP_SWITCH, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "Compressor Switch", + .freeer = snd_usb_mixer_elem_free + }, + { /* Comp threshold */ + .kcontrol_new = &snd_us16x08_comp_threshold_ctl, + .control_id = SND_US16X08_ID_COMP_THRESHOLD, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Compressor Threshold Volume", + .freeer = NULL + }, + { /* Comp ratio */ + .kcontrol_new = &snd_us16x08_comp_ratio_ctl, + .control_id = SND_US16X08_ID_COMP_RATIO, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Compressor Ratio", + .freeer = NULL + }, + { /* Comp attack */ + .kcontrol_new = &snd_us16x08_comp_attack_ctl, + .control_id = SND_US16X08_ID_COMP_ATTACK, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Compressor Attack", + .freeer = NULL + }, + { /* Comp release */ + .kcontrol_new = &snd_us16x08_comp_release_ctl, + .control_id = SND_US16X08_ID_COMP_RELEASE, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Compressor Release", + .freeer = NULL + }, + { /* Comp gain */ + .kcontrol_new = &snd_us16x08_comp_gain_ctl, + .control_id = SND_US16X08_ID_COMP_GAIN, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Compressor Volume", + .freeer = NULL + }, +}; + +/* table of channel controls */ +static struct snd_us16x08_control_params channel_controls[] = { + { /* Phase */ + .kcontrol_new = &snd_us16x08_ch_boolean_ctl, + .control_id = SND_US16X08_ID_PHASE, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "Phase Switch", + .freeer = snd_usb_mixer_elem_free, + .default_val = 0 + }, + { /* Fader */ + .kcontrol_new = &snd_us16x08_ch_int_ctl, + .control_id = SND_US16X08_ID_FADER, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Line Volume", + .freeer = NULL, + .default_val = 127 + }, + { /* Mute */ + .kcontrol_new = &snd_us16x08_ch_boolean_ctl, + .control_id = SND_US16X08_ID_MUTE, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "Mute Switch", + .freeer = NULL, + .default_val = 0 + }, + { /* Pan */ + .kcontrol_new = &snd_us16x08_pan_int_ctl, + .control_id = SND_US16X08_ID_PAN, + .type = USB_MIXER_U16, + .num_channels = 16, + .name = "Pan Left-Right Volume", + .freeer = NULL, + .default_val = 127 + }, +}; + +/* table of master controls */ +static struct snd_us16x08_control_params master_controls[] = { + { /* Master */ + .kcontrol_new = &snd_us16x08_master_ctl, + .control_id = SND_US16X08_ID_FADER, + .type = USB_MIXER_U8, + .num_channels = 16, + .name = "Master Volume", + .freeer = NULL, + .default_val = 127 + }, + { /* Bypass */ + .kcontrol_new = &snd_us16x08_bus_ctl, + .control_id = SND_US16X08_ID_BYPASS, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "DSP Bypass Switch", + .freeer = NULL, + .default_val = 0 + }, + { /* Buss out */ + .kcontrol_new = &snd_us16x08_bus_ctl, + .control_id = SND_US16X08_ID_BUSS_OUT, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "Buss Out Switch", + .freeer = NULL, + .default_val = 0 + }, + { /* Master mute */ + .kcontrol_new = &snd_us16x08_bus_ctl, + .control_id = SND_US16X08_ID_MUTE, + .type = USB_MIXER_BOOLEAN, + .num_channels = 16, + .name = "Master Mute Switch", + .freeer = NULL, + .default_val = 0 + }, + +}; + +int snd_us16x08_controls_create(struct usb_mixer_interface *mixer) +{ + int i, j; + int err; + struct usb_mixer_elem_info *elem; + struct snd_us16x08_comp_store *comp_store; + struct snd_us16x08_meter_store *meter_store; + struct snd_us16x08_eq_store *eq_store; + + /* just check for non-MIDI interface */ + if (mixer->hostif->desc.bInterfaceNumber == 3) { + + /* create compressor mixer elements */ + comp_store = snd_us16x08_create_comp_store(); + if (comp_store == NULL) + return -ENOMEM; + + /* create eq store */ + eq_store = snd_us16x08_create_eq_store(); + if (eq_store == NULL) { + kfree(comp_store); + return -ENOMEM; + } + + /* create meters store */ + meter_store = snd_us16x08_create_meter_store(); + if (meter_store == NULL) { + kfree(comp_store); + kfree(eq_store); + return -ENOMEM; + } + + /* add routing control */ + err = add_new_ctl(mixer, &snd_us16x08_route_ctl, + SND_US16X08_ID_ROUTE, USB_MIXER_U8, 8, "Line Out Route", + NULL, NULL, &elem); + if (err < 0) { + usb_audio_dbg(mixer->chip, + "Failed to create route control, err:%d\n", + err); + return err; + } + for (i = 0; i < 8; i++) + elem->cache_val[i] = i < 2 ? i : i + 2; + elem->cached = 0xff; + + /* add master controls */ + for (i = 0; + i < sizeof(master_controls) + / sizeof(control_params); + i++) { + + err = add_new_ctl(mixer, + master_controls[i].kcontrol_new, + master_controls[i].control_id, + master_controls[i].type, + master_controls[i].num_channels, + master_controls[i].name, + comp_store, + master_controls[i].freeer, &elem); + if (err < 0) + return err; + elem->cache_val[0] = master_controls[i].default_val; + elem->cached = 1; + } + + /* add channel controls */ + for (i = 0; + i < sizeof(channel_controls) + / sizeof(control_params); + i++) { + + err = add_new_ctl(mixer, + channel_controls[i].kcontrol_new, + channel_controls[i].control_id, + channel_controls[i].type, + channel_controls[i].num_channels, + channel_controls[i].name, + comp_store, + channel_controls[i].freeer, &elem); + if (err < 0) + return err; + for (j = 0; j < SND_US16X08_MAX_CHANNELS; j++) { + elem->cache_val[j] = + channel_controls[i].default_val; + } + elem->cached = 0xffff; + } + + /* add EQ controls */ + for (i = 0; i < sizeof(eq_controls) / + sizeof(control_params); i++) { + + err = add_new_ctl(mixer, + eq_controls[i].kcontrol_new, + eq_controls[i].control_id, + eq_controls[i].type, + eq_controls[i].num_channels, + eq_controls[i].name, + eq_store, + eq_controls[i].freeer, NULL); + if (err < 0) + return err; + } + + /* add compressor controls */ + for (i = 0; + i < sizeof(comp_controls) + / sizeof(control_params); + i++) { + + err = add_new_ctl(mixer, + comp_controls[i].kcontrol_new, + comp_controls[i].control_id, + comp_controls[i].type, + comp_controls[i].num_channels, + comp_controls[i].name, + comp_store, + comp_controls[i].freeer, NULL); + if (err < 0) + return err; + } + + /* meter function 'get' must access to compressor store + * so place a reference here + */ + meter_store->comp_store = comp_store; + err = add_new_ctl(mixer, &snd_us16x08_meter_ctl, + SND_US16X08_ID_METER, USB_MIXER_U16, 0, "Level Meter", + (void *) meter_store, snd_usb_mixer_elem_free, NULL); + if (err < 0) + return err; + } + + return 0; +} + diff --git a/sound/usb/mixer_us16x08.h b/sound/usb/mixer_us16x08.h new file mode 100644 index 0000000..64f89b5 --- /dev/null +++ b/sound/usb/mixer_us16x08.h @@ -0,0 +1,122 @@ +#ifndef __USB_MIXER_US16X08_H +#define __USB_MIXER_US16X08_H + +#define SND_US16X08_MAX_CHANNELS 16 + +/* define some bias, cause some alsa-mixers wont work with + * negative ranges or if mixer-min != 0 + */ +#define SND_US16X08_NO_BIAS 0 +#define SND_US16X08_FADER_BIAS 127 +#define SND_US16X08_EQ_HIGHFREQ_BIAS 0x20 +#define SND_US16X08_COMP_THRESHOLD_BIAS 0x20 +#define SND_US16X08_COMP_ATTACK_BIAS 2 +#define SND_US16X08_COMP_RELEASE_BIAS 1 + +/* get macro for components of kcontrol private_value */ +#define SND_US16X08_KCBIAS(x) (((x)->private_value >> 24) & 0xff) +#define SND_US16X08_KCSTEP(x) (((x)->private_value >> 16) & 0xff) +#define SND_US16X08_KCMIN(x) (((x)->private_value >> 8) & 0xff) +#define SND_US16X08_KCMAX(x) (((x)->private_value >> 0) & 0xff) +/* set macro for kcontrol private_value */ +#define SND_US16X08_KCSET(bias, step, min, max) \ + (((bias) << 24) | ((step) << 16) | ((min) << 8) | (max)) + +/* the URB request/type to control Tascam mixers */ +#define SND_US16X08_URB_REQUEST 0x1D +#define SND_US16X08_URB_REQUESTTYPE 0x40 + +/* the URB params to retrieve meter ranges */ +#define SND_US16X08_URB_METER_REQUEST 0x1e +#define SND_US16X08_URB_METER_REQUESTTYPE 0xc0 + +#define MUA0(x, y) ((x)[(y) * 10 + 4]) +#define MUA1(x, y) ((x)[(y) * 10 + 5]) +#define MUA2(x, y) ((x)[(y) * 10 + 6]) +#define MUB0(x, y) ((x)[(y) * 10 + 7]) +#define MUB1(x, y) ((x)[(y) * 10 + 8]) +#define MUB2(x, y) ((x)[(y) * 10 + 9]) +#define MUC0(x, y) ((x)[(y) * 10 + 10]) +#define MUC1(x, y) ((x)[(y) * 10 + 11]) +#define MUC2(x, y) ((x)[(y) * 10 + 12]) +#define MUC3(x, y) ((x)[(y) * 10 + 13]) + +/* Common Channel control IDs */ +#define SND_US16X08_ID_BYPASS 0x45 +#define SND_US16X08_ID_BUSS_OUT 0x44 +#define SND_US16X08_ID_PHASE 0x85 +#define SND_US16X08_ID_MUTE 0x83 +#define SND_US16X08_ID_FADER 0x81 +#define SND_US16X08_ID_PAN 0x82 +#define SND_US16X08_ID_METER 0xB1 + +#define SND_US16X08_ID_EQ_BAND_COUNT 4 +#define SND_US16X08_ID_EQ_PARAM_COUNT 4 + +/* EQ level IDs */ +#define SND_US16X08_ID_EQLOWLEVEL 0x01 +#define SND_US16X08_ID_EQLOWMIDLEVEL 0x02 +#define SND_US16X08_ID_EQHIGHMIDLEVEL 0x03 +#define SND_US16X08_ID_EQHIGHLEVEL 0x04 + +/* EQ frequence IDs */ +#define SND_US16X08_ID_EQLOWFREQ 0x11 +#define SND_US16X08_ID_EQLOWMIDFREQ 0x12 +#define SND_US16X08_ID_EQHIGHMIDFREQ 0x13 +#define SND_US16X08_ID_EQHIGHFREQ 0x14 + +/* EQ width IDs */ +#define SND_US16X08_ID_EQLOWMIDWIDTH 0x22 +#define SND_US16X08_ID_EQHIGHMIDWIDTH 0x23 + +#define SND_US16X08_ID_EQENABLE 0x30 + +#define EQ_STORE_BAND_IDX(x) ((x) & 0xf) +#define EQ_STORE_PARAM_IDX(x) (((x) & 0xf0) >> 4) + +#define SND_US16X08_ID_ROUTE 0x00 + +/* Compressor Ids */ +#define SND_US16X08_ID_COMP_BASE 0x32 +#define SND_US16X08_ID_COMP_THRESHOLD SND_US16X08_ID_COMP_BASE +#define SND_US16X08_ID_COMP_RATIO (SND_US16X08_ID_COMP_BASE + 1) +#define SND_US16X08_ID_COMP_ATTACK (SND_US16X08_ID_COMP_BASE + 2) +#define SND_US16X08_ID_COMP_RELEASE (SND_US16X08_ID_COMP_BASE + 3) +#define SND_US16X08_ID_COMP_GAIN (SND_US16X08_ID_COMP_BASE + 4) +#define SND_US16X08_ID_COMP_SWITCH (SND_US16X08_ID_COMP_BASE + 5) +#define SND_US16X08_ID_COMP_COUNT 6 + +#define COMP_STORE_IDX(x) ((x) - SND_US16X08_ID_COMP_BASE) + +struct snd_us16x08_eq_store { + u8 val[SND_US16X08_ID_EQ_BAND_COUNT][SND_US16X08_ID_EQ_PARAM_COUNT] + [SND_US16X08_MAX_CHANNELS]; +}; + +struct snd_us16x08_comp_store { + u8 val[SND_US16X08_ID_COMP_COUNT][SND_US16X08_MAX_CHANNELS]; +}; + +struct snd_us16x08_meter_store { + int meter_level[SND_US16X08_MAX_CHANNELS]; + int master_level[2]; /* level of meter for master output */ + int comp_index; /* round trip channel selector */ + int comp_active_index; /* channel select from user space mixer */ + int comp_level[16]; /* compressor reduction level */ + struct snd_us16x08_comp_store *comp_store; +}; + +struct snd_us16x08_control_params { + struct snd_kcontrol_new *kcontrol_new; + int control_id; + int type; + int num_channels; + const char *name; + void (*freeer)(struct snd_kcontrol *kctl); + int default_val; +}; + +#define snd_us16x08_switch_info snd_ctl_boolean_mono_info + +int snd_us16x08_controls_create(struct usb_mixer_interface *mixer); +#endif /* __USB_MIXER_US16X08_H */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: constify snd_kcontrol_new structures
by Bhumika Goyal 19 Feb '17

19 Feb '17

Declare snd_kcontrol_new structures as const as they are only passed as an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having the same property can be made const too. Done using Coccinelle: @r1 disable optional_qualifier @ identifier i; position p; @@ static struct snd_kcontrol_new i@p = {...}; @ok1@ identifier r1.i; position p; expression e1; @@ snd_ctl_new1(&i@p,e1) @bad@ position p!={r1.p,ok1.p}; identifier r1.i; @@ i@p @depends on !bad disable optional_qualifier@ identifier r1.i; @@ +const struct snd_kcontrol_new i; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/au88x0/au88x0_a3d.c | 2 +- sound/pci/au88x0/au88x0_eq.c | 6 +++--- sound/pci/au88x0/au88x0_pcm.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/bt87x.c | 6 +++--- sound/pci/ca0106/ca0106_mixer.c | 4 ++-- sound/pci/cmipci.c | 6 +++--- sound/pci/cs4281.c | 4 ++-- sound/pci/echoaudio/echoaudio.c | 26 +++++++++++++------------- sound/pci/emu10k1/emu10k1x.c | 6 +++--- sound/pci/emu10k1/emumixer.c | 30 +++++++++++++++--------------- sound/pci/emu10k1/emupcm.c | 2 +- sound/pci/ens1370.c | 4 ++-- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/ice1712/delta.c | 2 +- sound/pci/ice1712/ews.c | 4 ++-- sound/pci/ice1712/ice1712.c | 30 +++++++++++++++--------------- sound/pci/ice1712/ice1724.c | 20 ++++++++++---------- sound/pci/lola/lola_mixer.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/mixart/mixart_mixer.c | 6 +++--- sound/pci/pcxhr/pcxhr_mix22.c | 6 +++--- sound/pci/pcxhr/pcxhr_mixer.c | 22 +++++++++++----------- sound/pci/trident/trident_main.c | 22 +++++++++++----------- sound/pci/via82xx.c | 6 +++--- sound/pci/vx222/vx222_ops.c | 4 ++-- sound/pci/ymfpci/ymfpci_main.c | 14 +++++++------- 28 files changed, 123 insertions(+), 123 deletions(-) diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index ab0f873..7a4558a 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -846,7 +846,7 @@ static void vortex_a3d_translate_filter(a3d_atmos_t filter, int *params) return changed; } -static struct snd_kcontrol_new vortex_a3d_kcontrol = { +static const struct snd_kcontrol_new vortex_a3d_kcontrol = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Playback PCM advanced processing", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index 9585c5c..b566b44 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -757,7 +757,7 @@ static void vortex_Eqlzr_shutdown(vortex_t * vortex) return 1; /* Allways changes */ } -static struct snd_kcontrol_new vortex_eqtoggle_kcontrol = { +static const struct snd_kcontrol_new vortex_eqtoggle_kcontrol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "EQ Enable", .index = 0, @@ -815,7 +815,7 @@ static void vortex_Eqlzr_shutdown(vortex_t * vortex) return changed; } -static struct snd_kcontrol_new vortex_eq_kcontrol = { +static const struct snd_kcontrol_new vortex_eq_kcontrol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = " .", .index = 0, @@ -855,7 +855,7 @@ static void vortex_Eqlzr_shutdown(vortex_t * vortex) return 0; } -static struct snd_kcontrol_new vortex_levels_kcontrol = { +static const struct snd_kcontrol_new vortex_levels_kcontrol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "EQ Peaks", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index df5741a..335979a 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -601,7 +601,7 @@ static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400); -static struct snd_kcontrol_new snd_vortex_pcm_vol = { +static const struct snd_kcontrol_new snd_vortex_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "PCM Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 57bbb87..8356180 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -202,7 +202,7 @@ static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, .pointer = snd_aw2_pcm_pointer_capture, }; -static struct snd_kcontrol_new aw2_control = { +static const struct snd_kcontrol_new aw2_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Capture Route", .index = 0, diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index f2c0709..099efb0 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -598,7 +598,7 @@ static int snd_bt87x_capture_volume_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_bt87x_capture_volume = { +static const struct snd_kcontrol_new snd_bt87x_capture_volume = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Volume", .info = snd_bt87x_capture_volume_info, @@ -634,7 +634,7 @@ static int snd_bt87x_capture_boost_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_bt87x_capture_boost = { +static const struct snd_kcontrol_new snd_bt87x_capture_boost = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Boost", .info = snd_bt87x_capture_boost_info, @@ -676,7 +676,7 @@ static int snd_bt87x_capture_source_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_bt87x_capture_source = { +static const struct snd_kcontrol_new snd_bt87x_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .info = snd_bt87x_capture_source_info, diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 025805c..b4d3415 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -301,7 +301,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in = +static const struct snd_kcontrol_new snd_ca0106_capture_mic_line_in = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Shared Mic/Line in Capture Switch", @@ -310,7 +310,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, .put = snd_ca0106_capture_mic_line_in_put }; -static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out = +static const struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Shared Line in/Side out Capture Switch", diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index aeedc27..227c9d3 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1045,7 +1045,7 @@ static int snd_cmipci_spdif_default_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_cmipci_spdif_default = +static const struct snd_kcontrol_new snd_cmipci_spdif_default = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), @@ -1072,7 +1072,7 @@ static int snd_cmipci_spdif_mask_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_cmipci_spdif_mask = +static const struct snd_kcontrol_new snd_cmipci_spdif_mask = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1119,7 +1119,7 @@ static int snd_cmipci_spdif_stream_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_cmipci_spdif_stream = +static const struct snd_kcontrol_new snd_cmipci_spdif_stream = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 8f0f5f2..470c773 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1055,7 +1055,7 @@ static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(db_scale_dsp, -4650, 150, 0); -static struct snd_kcontrol_new snd_cs4281_fm_vol = +static const struct snd_kcontrol_new snd_cs4281_fm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Synth Playback Volume", @@ -1066,7 +1066,7 @@ static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol, .tlv = { .p = db_scale_dsp }, }; -static struct snd_kcontrol_new snd_cs4281_pcm_vol = +static const struct snd_kcontrol_new snd_cs4281_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Stream Playback Volume", diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 9370717..d15ecf9 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1039,7 +1039,7 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol, #ifdef ECHOCARD_HAS_LINE_OUT_GAIN /* On the Mia this one controls the line-out volume */ -static struct snd_kcontrol_new snd_echo_line_output_gain = { +static const struct snd_kcontrol_new snd_echo_line_output_gain = { .name = "Line Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -1050,7 +1050,7 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol, .tlv = {.p = db_scale_output_gain}, }; #else -static struct snd_kcontrol_new snd_echo_pcm_output_gain = { +static const struct snd_kcontrol_new snd_echo_pcm_output_gain = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, @@ -1120,7 +1120,7 @@ static int snd_echo_input_gain_put(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(db_scale_input_gain, -2500, 50, 0); -static struct snd_kcontrol_new snd_echo_line_input_gain = { +static const struct snd_kcontrol_new snd_echo_line_input_gain = { .name = "Line Capture Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, @@ -1184,7 +1184,7 @@ static int snd_echo_output_nominal_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_echo_output_nominal_level = { +static const struct snd_kcontrol_new snd_echo_output_nominal_level = { .name = "Line Playback Switch (-10dBV)", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_echo_output_nominal_info, @@ -1250,7 +1250,7 @@ static int snd_echo_input_nominal_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_echo_intput_nominal_level = { +static const struct snd_kcontrol_new snd_echo_intput_nominal_level = { .name = "Line Capture Switch (-10dBV)", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_echo_input_nominal_info, @@ -1477,7 +1477,7 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_echo_digital_mode_switch = { +static const struct snd_kcontrol_new snd_echo_digital_mode_switch = { .name = "Digital mode Switch", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = snd_echo_digital_mode_info, @@ -1527,7 +1527,7 @@ static int snd_echo_spdif_mode_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_echo_spdif_mode_switch = { +static const struct snd_kcontrol_new snd_echo_spdif_mode_switch = { .name = "S/PDIF mode Switch", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = snd_echo_spdif_mode_info, @@ -1600,7 +1600,7 @@ static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_echo_clock_source_switch = { +static const struct snd_kcontrol_new snd_echo_clock_source_switch = { .name = "Sample Clock Source", .iface = SNDRV_CTL_ELEM_IFACE_PCM, .info = snd_echo_clock_source_info, @@ -1643,7 +1643,7 @@ static int snd_echo_phantom_power_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_echo_phantom_power_switch = { +static const struct snd_kcontrol_new snd_echo_phantom_power_switch = { .name = "Phantom power Switch", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = snd_echo_phantom_power_info, @@ -1686,7 +1686,7 @@ static int snd_echo_automute_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_echo_automute_switch = { +static const struct snd_kcontrol_new snd_echo_automute_switch = { .name = "Digital Capture Switch (automute)", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .info = snd_echo_automute_info, @@ -1713,7 +1713,7 @@ static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new snd_echo_vumeters_switch = { +static const struct snd_kcontrol_new snd_echo_vumeters_switch = { .name = "VU-meters Switch", .iface = SNDRV_CTL_ELEM_IFACE_CARD, .access = SNDRV_CTL_ELEM_ACCESS_WRITE, @@ -1751,7 +1751,7 @@ static int snd_echo_vumeters_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_echo_vumeters = { +static const struct snd_kcontrol_new snd_echo_vumeters = { .name = "VU-meters", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READ | @@ -1804,7 +1804,7 @@ static int snd_echo_channels_info_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_echo_channels_info = { +static const struct snd_kcontrol_new snd_echo_channels_info = { .name = "Channels info", .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 921037e..f45612e 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1112,7 +1112,7 @@ static int snd_emu10k1x_shared_spdif_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1x_shared_spdif = +static const struct snd_kcontrol_new snd_emu10k1x_shared_spdif = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Analog/Digital Output Jack", @@ -1171,7 +1171,7 @@ static int snd_emu10k1x_spdif_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1x_spdif_mask_control = +static const struct snd_kcontrol_new snd_emu10k1x_spdif_mask_control = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1181,7 +1181,7 @@ static int snd_emu10k1x_spdif_put(struct snd_kcontrol *kcontrol, .get = snd_emu10k1x_spdif_get_mask }; -static struct snd_kcontrol_new snd_emu10k1x_spdif_control = +static const struct snd_kcontrol_new snd_emu10k1x_spdif_control = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 076b117..b2219a7 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -795,7 +795,7 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu1010_internal_clock = +static const struct snd_kcontrol_new snd_emu1010_internal_clock = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -847,7 +847,7 @@ static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu1010_optical_out = { +static const struct snd_kcontrol_new snd_emu1010_optical_out = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Optical Output Mode", @@ -898,7 +898,7 @@ static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu1010_optical_in = { +static const struct snd_kcontrol_new snd_emu1010_optical_in = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Optical Input Mode", @@ -978,7 +978,7 @@ static int snd_audigy_i2c_capture_source_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_audigy_i2c_capture_source = +static const struct snd_kcontrol_new snd_audigy_i2c_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -1177,7 +1177,7 @@ static int snd_emu10k1_spdif_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control = +static const struct snd_kcontrol_new snd_emu10k1_spdif_mask_control = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1187,7 +1187,7 @@ static int snd_emu10k1_spdif_put(struct snd_kcontrol *kcontrol, .get = snd_emu10k1_spdif_get_mask }; -static struct snd_kcontrol_new snd_emu10k1_spdif_control = +static const struct snd_kcontrol_new snd_emu10k1_spdif_control = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), @@ -1293,7 +1293,7 @@ static int snd_emu10k1_send_routing_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_send_routing_control = +static const struct snd_kcontrol_new snd_emu10k1_send_routing_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1364,7 +1364,7 @@ static int snd_emu10k1_send_volume_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_send_volume_control = +static const struct snd_kcontrol_new snd_emu10k1_send_volume_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1429,7 +1429,7 @@ static int snd_emu10k1_attn_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_attn_control = +static const struct snd_kcontrol_new snd_emu10k1_attn_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1501,7 +1501,7 @@ static int snd_emu10k1_efx_send_routing_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_efx_send_routing_control = +static const struct snd_kcontrol_new snd_emu10k1_efx_send_routing_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1568,7 +1568,7 @@ static int snd_emu10k1_efx_send_volume_put(struct snd_kcontrol *kcontrol, } -static struct snd_kcontrol_new snd_emu10k1_efx_send_volume_control = +static const struct snd_kcontrol_new snd_emu10k1_efx_send_volume_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1626,7 +1626,7 @@ static int snd_emu10k1_efx_attn_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_efx_attn_control = +static const struct snd_kcontrol_new snd_emu10k1_efx_attn_control = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1691,7 +1691,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_emu10k1_shared_spdif = +static const struct snd_kcontrol_new snd_emu10k1_shared_spdif = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "SB Live Analog/Digital Output Jack", @@ -1700,7 +1700,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, .put = snd_emu10k1_shared_spdif_put }; -static struct snd_kcontrol_new snd_audigy_shared_spdif = +static const struct snd_kcontrol_new snd_audigy_shared_spdif = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Audigy Analog/Digital Output Jack", @@ -1738,7 +1738,7 @@ static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol, return snd_ac97_update(emu->ac97, AC97_REC_GAIN, val); } -static struct snd_kcontrol_new snd_audigy_capture_boost = +static const struct snd_kcontrol_new snd_audigy_capture_boost = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Mic Extra Boost", diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 37be1e1..ef1cf53 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1542,7 +1542,7 @@ static int snd_emu10k1_pcm_efx_voices_mask_put(struct snd_kcontrol *kcontrol, st return change; } -static struct snd_kcontrol_new snd_emu10k1_pcm_efx_voices_mask = { +static const struct snd_kcontrol_new snd_emu10k1_pcm_efx_voices_mask = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Captured FX8010 Outputs", .info = snd_emu10k1_pcm_efx_voices_mask_info, diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 51736c2..ae09baa 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1530,7 +1530,7 @@ static int snd_es1373_rear_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ens1373_rear = +static const struct snd_kcontrol_new snd_ens1373_rear = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "AC97 2ch->4ch Copy Switch", @@ -1575,7 +1575,7 @@ static int snd_es1373_line_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new snd_ens1373_line = +static const struct snd_kcontrol_new snd_ens1373_line = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line In->Rear Out Switch", diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9913be8..bf491eb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1897,7 +1897,7 @@ static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new vmaster_mute_mode = { +static const struct snd_kcontrol_new vmaster_mute_mode = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Mute-LED Mode", .info = vmaster_mute_mode_info, @@ -2637,7 +2637,7 @@ static int spdif_share_sw_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new spdif_share_sw = { +static const struct snd_kcontrol_new spdif_share_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "IEC958 Default PCM Playback Switch", .info = snd_ctl_boolean_mono_info, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cf9bc042..59e0fed 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -354,7 +354,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new eld_bytes_ctl = { +static const struct snd_kcontrol_new eld_bytes_ctl = { .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "ELD", diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 3bfdc78..da5f37b 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -432,7 +432,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco return 0; } -static struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_status = +static const struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_status = { .access = (SNDRV_CTL_ELEM_ACCESS_READ), .iface = SNDRV_CTL_ELEM_IFACE_MIXER, diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 5cb587c..ec07136 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -719,7 +719,7 @@ static int snd_ice1712_ews88mt_input_sense_put(struct snd_kcontrol *kcontrol, st return ndata != data; } -static struct snd_kcontrol_new snd_ice1712_ews88mt_input_sense = { +static const struct snd_kcontrol_new snd_ice1712_ews88mt_input_sense = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Sensitivity Switch", .info = snd_ice1712_ewx_io_sense_info, @@ -728,7 +728,7 @@ static int snd_ice1712_ews88mt_input_sense_put(struct snd_kcontrol *kcontrol, st .count = 8, }; -static struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense = { +static const struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Output Sensitivity Switch", .info = snd_ice1712_ewx_io_sense_info, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index b4aa4c1..1d8612c 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -279,7 +279,7 @@ static int snd_ice1712_digmix_route_ac97_put(struct snd_kcontrol *kcontrol, stru return val != nval; } -static struct snd_kcontrol_new snd_ice1712_mixer_digmix_route_ac97 = { +static const struct snd_kcontrol_new snd_ice1712_mixer_digmix_route_ac97 = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Mixer To AC97", .info = snd_ice1712_digmix_route_ac97_info, @@ -1410,7 +1410,7 @@ static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struc .private_value = 10, }; -static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch = { +static const struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, SWITCH), .info = snd_ice1712_pro_mixer_switch_info, @@ -1432,7 +1432,7 @@ static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struc .tlv = { .p = db_scale_playback } }; -static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume = { +static const struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, VOLUME), .info = snd_ice1712_pro_mixer_volume_info, @@ -1630,7 +1630,7 @@ static int snd_ice1712_eeprom_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ice1712_eeprom = { +static const struct snd_kcontrol_new snd_ice1712_eeprom = { .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = "ICE1712 EEPROM", .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -1666,7 +1666,7 @@ static int snd_ice1712_spdif_default_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ice1712_spdif_default = +static const struct snd_kcontrol_new snd_ice1712_spdif_default = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), @@ -1717,7 +1717,7 @@ static int snd_ice1712_spdif_maskp_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ice1712_spdif_maskc = +static const struct snd_kcontrol_new snd_ice1712_spdif_maskc = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1726,7 +1726,7 @@ static int snd_ice1712_spdif_maskp_get(struct snd_kcontrol *kcontrol, .get = snd_ice1712_spdif_maskc_get, }; -static struct snd_kcontrol_new snd_ice1712_spdif_maskp = +static const struct snd_kcontrol_new snd_ice1712_spdif_maskp = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1753,7 +1753,7 @@ static int snd_ice1712_spdif_stream_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ice1712_spdif_stream = +static const struct snd_kcontrol_new snd_ice1712_spdif_stream = { .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE), @@ -1878,7 +1878,7 @@ static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ice1712_pro_internal_clock = { +static const struct snd_kcontrol_new snd_ice1712_pro_internal_clock = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Internal Clock", .info = snd_ice1712_pro_internal_clock_info, @@ -1943,7 +1943,7 @@ static int snd_ice1712_pro_internal_clock_default_put(struct snd_kcontrol *kcont return change; } -static struct snd_kcontrol_new snd_ice1712_pro_internal_clock_default = { +static const struct snd_kcontrol_new snd_ice1712_pro_internal_clock_default = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Internal Clock Default", .info = snd_ice1712_pro_internal_clock_default_info, @@ -1974,7 +1974,7 @@ static int snd_ice1712_pro_rate_locking_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ice1712_pro_rate_locking = { +static const struct snd_kcontrol_new snd_ice1712_pro_rate_locking = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Rate Locking", .info = snd_ice1712_pro_rate_locking_info, @@ -2005,7 +2005,7 @@ static int snd_ice1712_pro_rate_reset_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ice1712_pro_rate_reset = { +static const struct snd_kcontrol_new snd_ice1712_pro_rate_reset = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Rate Reset", .info = snd_ice1712_pro_rate_reset_info, @@ -2173,7 +2173,7 @@ static int snd_ice1712_pro_route_spdif_put(struct snd_kcontrol *kcontrol, .put = snd_ice1712_pro_route_analog_put, }; -static struct snd_kcontrol_new snd_ice1712_mixer_pro_spdif_route = { +static const struct snd_kcontrol_new snd_ice1712_mixer_pro_spdif_route = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route", .info = snd_ice1712_pro_route_info, @@ -2215,7 +2215,7 @@ static int snd_ice1712_pro_volume_rate_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ice1712_mixer_pro_volume_rate = { +static const struct snd_kcontrol_new snd_ice1712_mixer_pro_volume_rate = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Volume Rate", .info = snd_ice1712_pro_volume_rate_info, @@ -2248,7 +2248,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak = { +static const struct snd_kcontrol_new snd_ice1712_mixer_pro_peak = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index e5c52ed..cb9446e 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1598,7 +1598,7 @@ static int snd_vt1724_eeprom_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_vt1724_eeprom = { +static const struct snd_kcontrol_new snd_vt1724_eeprom = { .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = "ICE1724 EEPROM", .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -1711,7 +1711,7 @@ static int snd_vt1724_spdif_default_put(struct snd_kcontrol *kcontrol, return val != old; } -static struct snd_kcontrol_new snd_vt1724_spdif_default = +static const struct snd_kcontrol_new snd_vt1724_spdif_default = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), @@ -1743,7 +1743,7 @@ static int snd_vt1724_spdif_maskp_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_vt1724_spdif_maskc = +static const struct snd_kcontrol_new snd_vt1724_spdif_maskc = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1752,7 +1752,7 @@ static int snd_vt1724_spdif_maskp_get(struct snd_kcontrol *kcontrol, .get = snd_vt1724_spdif_maskc_get, }; -static struct snd_kcontrol_new snd_vt1724_spdif_maskp = +static const struct snd_kcontrol_new snd_vt1724_spdif_maskp = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1789,7 +1789,7 @@ static int snd_vt1724_spdif_sw_put(struct snd_kcontrol *kcontrol, return old != val; } -static struct snd_kcontrol_new snd_vt1724_spdif_switch = +static const struct snd_kcontrol_new snd_vt1724_spdif_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* FIXME: the following conflict with IEC958 Playback Route */ @@ -1964,7 +1964,7 @@ static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, return old_rate != new_rate; } -static struct snd_kcontrol_new snd_vt1724_pro_internal_clock = { +static const struct snd_kcontrol_new snd_vt1724_pro_internal_clock = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Internal Clock", .info = snd_vt1724_pro_internal_clock_info, @@ -1995,7 +1995,7 @@ static int snd_vt1724_pro_rate_locking_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_vt1724_pro_rate_locking = { +static const struct snd_kcontrol_new snd_vt1724_pro_rate_locking = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Rate Locking", .info = snd_vt1724_pro_rate_locking_info, @@ -2026,7 +2026,7 @@ static int snd_vt1724_pro_rate_reset_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_vt1724_pro_rate_reset = { +static const struct snd_kcontrol_new snd_vt1724_pro_rate_reset = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Multi Track Rate Reset", .info = snd_vt1724_pro_rate_reset_info, @@ -2151,7 +2151,7 @@ static int snd_vt1724_pro_route_spdif_put(struct snd_kcontrol *kcontrol, .put = snd_vt1724_pro_route_analog_put, }; -static struct snd_kcontrol_new snd_vt1724_mixer_pro_spdif_route = { +static const struct snd_kcontrol_new snd_vt1724_mixer_pro_spdif_route = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route", .info = snd_vt1724_pro_route_info, @@ -2187,7 +2187,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak = { +static const struct snd_kcontrol_new snd_vt1724_mixer_pro_peak = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index e7fe15d..cb25acf 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -645,7 +645,7 @@ static int lola_input_src_put(struct snd_kcontrol *kcontrol, return lola_set_src_config(chip, mask, true); } -static struct snd_kcontrol_new lola_input_src_mixer = { +static const struct snd_kcontrol_new lola_input_src_mixer = { .name = "Digital SRC Capture Switch", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = lola_input_src_info, diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index c0f0c34..f9c3e86 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -899,7 +899,7 @@ static int lx_control_playback_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new lx_control_playback_switch = { +static const struct snd_kcontrol_new lx_control_playback_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", .index = 0, diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 51e5349..4a4616a 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -448,7 +448,7 @@ static int mixart_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele return changed; } -static struct snd_kcontrol_new mixart_control_output_switch = { +static const struct snd_kcontrol_new mixart_control_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = mixart_sw_info, /* shared */ @@ -1024,7 +1024,7 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ return changed; } -static struct snd_kcontrol_new mixart_control_monitor_vol = { +static const struct snd_kcontrol_new mixart_control_monitor_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -1091,7 +1091,7 @@ static int mixart_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return (changed != 0); } -static struct snd_kcontrol_new mixart_control_monitor_sw = { +static const struct snd_kcontrol_new mixart_control_monitor_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitoring Switch", .info = mixart_sw_info, /* shared */ diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c index 6a56e53..8b4d028 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.c +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -744,7 +744,7 @@ static int hr222_mic_vol_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new hr222_control_mic_level = { +static const struct snd_kcontrol_new hr222_control_mic_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -794,7 +794,7 @@ static int hr222_mic_boost_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new hr222_control_mic_boost = { +static const struct snd_kcontrol_new hr222_control_mic_boost = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -836,7 +836,7 @@ static int hr222_phantom_power_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new hr222_phantom_power_switch = { +static const struct snd_kcontrol_new hr222_phantom_power_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Phantom Power Switch", .info = hr222_phantom_power_info, diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 63136c4..36875df 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -235,7 +235,7 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new pcxhr_control_output_switch = { +static const struct snd_kcontrol_new pcxhr_control_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", .info = pcxhr_sw_info, /* shared */ @@ -460,7 +460,7 @@ static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new pcxhr_control_pcm_switch = { +static const struct snd_kcontrol_new pcxhr_control_pcm_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Playback Switch", .count = PCXHR_PLAYBACK_STREAMS, @@ -509,7 +509,7 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new pcxhr_control_monitor_vol = { +static const struct snd_kcontrol_new pcxhr_control_monitor_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -562,7 +562,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol, return (changed != 0); } -static struct snd_kcontrol_new pcxhr_control_monitor_sw = { +static const struct snd_kcontrol_new pcxhr_control_monitor_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Monitoring Playback Switch", .info = pcxhr_sw_info, /* shared */ @@ -697,7 +697,7 @@ static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol, return ret; } -static struct snd_kcontrol_new pcxhr_control_audio_src = { +static const struct snd_kcontrol_new pcxhr_control_audio_src = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .info = pcxhr_audio_src_info, @@ -798,7 +798,7 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, return ret; } -static struct snd_kcontrol_new pcxhr_control_clock_type = { +static const struct snd_kcontrol_new pcxhr_control_clock_type = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Mode", .info = pcxhr_clock_type_info, @@ -842,7 +842,7 @@ static int pcxhr_clock_rate_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new pcxhr_control_clock_rate = { +static const struct snd_kcontrol_new pcxhr_control_clock_rate = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = "Clock Rates", @@ -1017,14 +1017,14 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol, return changed; } -static struct snd_kcontrol_new pcxhr_control_playback_iec958_mask = { +static const struct snd_kcontrol_new pcxhr_control_playback_iec958_mask = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), .info = pcxhr_iec958_info, .get = pcxhr_iec958_mask_get }; -static struct snd_kcontrol_new pcxhr_control_playback_iec958 = { +static const struct snd_kcontrol_new pcxhr_control_playback_iec958 = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = pcxhr_iec958_info, @@ -1033,14 +1033,14 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol, .private_value = 0 /* playback */ }; -static struct snd_kcontrol_new pcxhr_control_capture_iec958_mask = { +static const struct snd_kcontrol_new pcxhr_control_capture_iec958_mask = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK), .info = pcxhr_iec958_info, .get = pcxhr_iec958_mask_get }; -static struct snd_kcontrol_new pcxhr_control_capture_iec958 = { +static const struct snd_kcontrol_new pcxhr_control_capture_iec958 = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 92ad2d7..64d3b8e 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -2356,7 +2356,7 @@ static int snd_trident_spdif_control_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_spdif_control = +static const struct snd_kcontrol_new snd_trident_spdif_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), @@ -2419,7 +2419,7 @@ static int snd_trident_spdif_default_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_spdif_default = +static const struct snd_kcontrol_new snd_trident_spdif_default = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), @@ -2452,7 +2452,7 @@ static int snd_trident_spdif_mask_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_trident_spdif_mask = +static const struct snd_kcontrol_new snd_trident_spdif_mask = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -2514,7 +2514,7 @@ static int snd_trident_spdif_stream_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_spdif_stream = +static const struct snd_kcontrol_new snd_trident_spdif_stream = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -2564,7 +2564,7 @@ static int snd_trident_ac97_control_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_ac97_rear_control = +static const struct snd_kcontrol_new snd_trident_ac97_rear_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Rear Path", @@ -2622,7 +2622,7 @@ static int snd_trident_vol_control_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_vol_music_control = +static const struct snd_kcontrol_new snd_trident_vol_music_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Music Playback Volume", @@ -2633,7 +2633,7 @@ static int snd_trident_vol_control_put(struct snd_kcontrol *kcontrol, .tlv = { .p = db_scale_gvol }, }; -static struct snd_kcontrol_new snd_trident_vol_wave_control = +static const struct snd_kcontrol_new snd_trident_vol_wave_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Wave Playback Volume", @@ -2700,7 +2700,7 @@ static int snd_trident_pcm_vol_control_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_pcm_vol_control = +static const struct snd_kcontrol_new snd_trident_pcm_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Front Playback Volume", @@ -2764,7 +2764,7 @@ static int snd_trident_pcm_pan_control_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_pcm_pan_control = +static const struct snd_kcontrol_new snd_trident_pcm_pan_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Pan Playback Control", @@ -2821,7 +2821,7 @@ static int snd_trident_pcm_rvol_control_put(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(db_scale_crvol, -3175, 25, 1); -static struct snd_kcontrol_new snd_trident_pcm_rvol_control = +static const struct snd_kcontrol_new snd_trident_pcm_rvol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Reverb Playback Volume", @@ -2877,7 +2877,7 @@ static int snd_trident_pcm_cvol_control_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_trident_pcm_cvol_control = +static const struct snd_kcontrol_new snd_trident_pcm_cvol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PCM Chorus Playback Volume", diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 2d8c14e..d078e86 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1683,7 +1683,7 @@ static int snd_via8233_dxs3_spdif_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_via8233_dxs3_spdif_control = { +static const struct snd_kcontrol_new snd_via8233_dxs3_spdif_control = { .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_via8233_dxs3_spdif_info, @@ -1772,7 +1772,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1); -static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control = { +static const struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -1783,7 +1783,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol, .tlv = { .p = db_scale_dxs } }; -static struct snd_kcontrol_new snd_via8233_dxs_volume_control = { +static const struct snd_kcontrol_new snd_via8233_dxs_volume_control = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .device = 0, /* .subdevice set later */ diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index af83b3b..5911233 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -945,7 +945,7 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v return 0; } -static struct snd_kcontrol_new vx_control_input_level = { +static const struct snd_kcontrol_new vx_control_input_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -956,7 +956,7 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v .tlv = { .p = db_scale_mic }, }; -static struct snd_kcontrol_new vx_control_mic_level = { +static const struct snd_kcontrol_new vx_control_mic_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index ffee284..fe4ba46 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1316,7 +1316,7 @@ static int snd_ymfpci_spdif_default_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ymfpci_spdif_default = +static const struct snd_kcontrol_new snd_ymfpci_spdif_default = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), @@ -1344,7 +1344,7 @@ static int snd_ymfpci_spdif_mask_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ymfpci_spdif_mask = +static const struct snd_kcontrol_new snd_ymfpci_spdif_mask = { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1391,7 +1391,7 @@ static int snd_ymfpci_spdif_stream_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ymfpci_spdif_stream = +static const struct snd_kcontrol_new snd_ymfpci_spdif_stream = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1439,7 +1439,7 @@ static int snd_ymfpci_drec_source_put(struct snd_kcontrol *kcontrol, struct snd_ return reg != old_reg; } -static struct snd_kcontrol_new snd_ymfpci_drec_source = { +static const struct snd_kcontrol_new snd_ymfpci_drec_source = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Direct Recording Source", @@ -1609,7 +1609,7 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -static struct snd_kcontrol_new snd_ymfpci_dup4ch = { +static const struct snd_kcontrol_new snd_ymfpci_dup4ch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "4ch Duplication", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -1712,7 +1712,7 @@ static int snd_ymfpci_gpio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ return 0; } -static struct snd_kcontrol_new snd_ymfpci_rear_shared = { +static const struct snd_kcontrol_new snd_ymfpci_rear_shared = { .name = "Shared Rear/Line-In Switch", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = snd_ymfpci_gpio_sw_info, @@ -1776,7 +1776,7 @@ static int snd_ymfpci_pcm_vol_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_ymfpci_pcm_volume = { +static const struct snd_kcontrol_new snd_ymfpci_pcm_volume = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "PCM Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fix ES8328_I2C/SPI dependencies
by Arnd Bergmann 19 Feb '17

19 Feb '17

The two front-ends to the codec can now be selected individually, but fail to build when the bus support is missing: sound/built-in.o: In function `es8328_spi_probe': es8328-spi.c:(.text+0x125854): undefined reference to `__devm_regmap_init_spi' sound/built-in.o: In function `es8328_spi_driver_init': es8328-spi.c:(.init.text+0x3589): undefined reference to `__spi_register_driver' Related to this, the added dependency on SND_SOC_ES8328 breaks: warning: (SND_SOC_ALL_CODECS) selects SND_SOC_ES8328_I2C which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_ES8328 && I2C) This adds the respective Kconfig dependencies and changes SND_SOC_ES8328 to a hidden symbol that is selected implicitly by the two more specific options, as we do for some other codecs. We have to remove the 'depends on' for SND_SOC_IMX_ES8328 in the same step to avoid a recursive dependency. Fixes: aa00f2c8aff7 ("ASoC: Allow to select ES8328_I2C and ES8328_SPI directly") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 12 +++++++----- sound/soc/fsl/Kconfig | 2 +- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 37a4a43628b4..e49e9da7f1f6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -526,15 +526,17 @@ config SND_SOC_HDMI_CODEC select HDMI config SND_SOC_ES8328 - tristate "Everest Semi ES8328 CODEC" + tristate config SND_SOC_ES8328_I2C - depends on SND_SOC_ES8328 - tristate "I2C support for Everest Semi ES8328 CODEC" + tristate "Everest Semi ES8328 CODEC (I2C)" + depends on I2C + select SND_SOC_ES8328 config SND_SOC_ES8328_SPI - depends on SND_SOC_ES8328 - tristate "SPI support for Everest Semi ES8328 CODEC" + tristate "Everest Semi ES8328 CODEC (SPI)" + depends on SPI_MASTER + select SND_SOC_ES8328 config SND_SOC_GTM601 tristate 'GTM601 UMTS modem audio codec' diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0b914a1ca8d2..37f9b6201918 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -244,7 +244,7 @@ config SND_SOC_IMX_WM8962 config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" - depends on OF && (I2C || SPI) && SND_SOC_ES8328 + depends on OF && (I2C || SPI) select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_IMX_PCM_DMA -- 2.9.0

2 1

[alsa-devel] [PATCH -next] ASoC: sun4i-spdif: drop unnessary snd_soc_unregister_component()
by Wei Yongjun 19 Feb '17

19 Feb '17

From: Wei Yongjun <weiyongjun1(a)huawei.com> It's not necessary to unregister a component registered with devm_snd_soc_register_component(). Also removed pointness clk_disable_unprepare() from error path and snd_soc_unregister_platform() from the remove. Fixes: f8260afa444b ("ASoC: sunxi: Add support for the SPDIF block") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sunxi/sun4i-spdif.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index c03cd07..eaefd07 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -510,8 +510,7 @@ static int sun4i_spdif_probe(struct platform_device *pdev) host->spdif_clk = devm_clk_get(&pdev->dev, "spdif"); if (IS_ERR(host->spdif_clk)) { dev_err(&pdev->dev, "failed to get a spdif clock.\n"); - ret = PTR_ERR(host->spdif_clk); - goto err_disable_apb_clk; + return PTR_ERR(host->spdif_clk); } host->dma_params_tx.addr = res->start + quirks->reg_dac_txdata; @@ -525,7 +524,7 @@ static int sun4i_spdif_probe(struct platform_device *pdev) if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); - goto err_disable_apb_clk; + return ret; } if (!IS_ERR(host->rst)) reset_control_deassert(host->rst); @@ -534,7 +533,7 @@ static int sun4i_spdif_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(&pdev->dev, &sun4i_spdif_component, &sun4i_spdif_dai, 1); if (ret) - goto err_disable_apb_clk; + return ret; pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { @@ -552,9 +551,6 @@ static int sun4i_spdif_probe(struct platform_device *pdev) sun4i_spdif_runtime_suspend(&pdev->dev); err_unregister: pm_runtime_disable(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); -err_disable_apb_clk: - clk_disable_unprepare(host->apb_clk); return ret; } @@ -564,9 +560,6 @@ static int sun4i_spdif_remove(struct platform_device *pdev) if (!pm_runtime_status_suspended(&pdev->dev)) sun4i_spdif_runtime_suspend(&pdev->dev); - snd_soc_unregister_platform(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - return 0; }

3 2

[alsa-devel] [PATCH] ASLA: emu10k1: constify snd_emux_operators structure
by Bhumika Goyal 19 Feb '17

19 Feb '17

Declare snd_emux_operators structure as const as it is only copied into another structure. So, snd_emux_operators structures having this property can be made const. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/emu10k1/emu10k1_callback.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index d2c7ea3..aa2cc27 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -61,7 +61,7 @@ static struct snd_emux_voice *get_voice(struct snd_emux *emux, /* * set up operators */ -static struct snd_emux_operators emu10k1_ops = { +static const struct snd_emux_operators emu10k1_ops = { .owner = THIS_MODULE, .get_voice = get_voice, .prepare = start_voice, -- 1.9.1

3 2

[alsa-devel] [PATCH 1/1] alsa-plugins: enhancing configuration support for alsa plugins
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Mounesh Sutar <mounesh_sutar(a)mentor.com> Following plugins are made configurable in alsa-plugins: * oss * mix * usbstream * arcamav * speexdsp Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/configure.ac b/configure.ac index a55c735..190d935 100644 --- a/configure.ac +++ b/configure.ac @@ -20,6 +20,30 @@ PKG_CHECK_MODULES(ALSA, alsa >= 1.0.11) AC_CHECK_LIB(asound, snd_pcm_ioplug_create,, AC_ERROR([*** libasound has no external plugin SDK]), -ldl) +AC_ARG_ENABLE([oss], + AS_HELP_STRING([--disable-oss], [Disable building of OSS plugin])) +AS_IF([test "x$enable_oss" != "xno"], + [AM_CONDITIONAL(HAVE_OSS, true)], + [AM_CONDITIONAL(HAVE_OSS, false)]) + +AC_ARG_ENABLE([mix], + AS_HELP_STRING([--disable-mix], [Disable building of upmix and vdownmix plugins])) +AS_IF([test "x$enable_mix" != "xno"], + [AM_CONDITIONAL(HAVE_MIX, true)], + [AM_CONDITIONAL(HAVE_MIX, false)]) + +AC_ARG_ENABLE([usbstream], + AS_HELP_STRING([--disable-usbstream], [Disable building of usb_stream plugin])) +AS_IF([test "x$enable_usbstream" != "xno"], + [AM_CONDITIONAL(HAVE_USBSTREAM, true)], + [AM_CONDITIONAL(HAVE_USBSTREAM, false)]) + +AC_ARG_ENABLE([arcamav], + AS_HELP_STRING([--disable-arcamav], [Disable building of Arcam AV control plugin])) +AS_IF([test "x$enable_arcamav" != "xno"], + [AM_CONDITIONAL(HAVE_ARCAMAV, true)], + [AM_CONDITIONAL(HAVE_ARCAMAV, false)]) + AC_ARG_ENABLE([jack], AS_HELP_STRING([--disable-jack], [Disable building of JACK plugin])) @@ -108,7 +132,12 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) -PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +AC_ARG_ENABLE([speexdsp], + AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin])) + +if test "x$enable_speexdsp" != "xno"; then + PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2rc2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +fi AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes") AC_ARG_WITH([speex], -- 2.7.4

2 1

[alsa-devel] Help with midi player application
by amidi＠LEFerguson.com 17 Feb '17

17 Feb '17

I am hoping someone has a bit of patience to help. I apologize in advance if this is not appropriate use of this mailing list. I am trying to incorporate ALSA's sequencer within an application I am writing to play MIDI pieces, and provide controls (start at measure, tempo changes, etc), while output is sent to a piano player system with midi i/o (specifically it's a Pianodisc system). The result (if it ever works) will be open source. The good news is I can get it to work. The bad news is I am having some kind of timing or buffering issue, and am at a bit of a loss where to look. I am convinced the problem is in my code, or environment, as I can get pmidi (1.7.1) to work (including build it from source, so same compile environment). The actual application is in a QT5.8 application, so it is conceivable that some aspect of QT's environment is at issue. I have embedded the calls to the ALSA sequencer inside of a separate QThread so as to isolate it from QT's event loop. Since the code fails only in context, it is not practical to post the whole code. And yes I'm sure it is related to my code, I get that, am hoping for some trouble shooting ideas, or pointers to tools. The symptoms are on a fast, long piece play hangs for a long period of time (many seconds). I put into the loop queries as to queue info/state/etc, and what I see is the queue tick advancing correctly during this pause, but the events in the queue are steady at about 80%full and not emptying. I have set this up to run through qmidinet which outputs each notes as ALSA hands it off, and that pauses also (qmidinet is not the cause; writing direct to the midi port also pauses). My hope with this email is to get some suggestions how to find out what is happening when it pauses inappropriately. "It" here is whatever processes are draining the queue; the thread making the snd_seq calls continues to run, I can query the queue and see its status, etc. I have tried this with no self-written buffer management, just letting ALSA pause the thread; I also have it running managing the queue myself looking for it getting near full and waiting until it gets more empty (staying above the "room" size). Neither work, both do approximately the same, though trying to manage the buffer seems to work better and gives me event-count visibility in the queue. Allocating a much larger queue buffer (snd_seq_set_client_pool_output) delays the pause significantly but does not solve the problem. The queue is not full (but also is not empty) when the pauses occur. My initial thoughts were the tick in the events was incorrect and so waiting, but that is not the case as best I can tell (also, that would not cause a larger buffer to pause less often). Something is stopping the queue from processing outbound for longish periods. Essentially what I do is: snd_seq_open to get the handle send_seq_alloc_queue snd_seq_client and snd_seq_create_simple_port to get the client port snd_seq_connect_to to connect snd_seq_set_queue_tempo (initially with default snd_deq_start_queue snd_seq_drain_output (until ret = 0) Then I loop through all the midi events. Each event has: snd_seq_ev_schedule_tick called with parameter 2 = 1 for relative ticks. snd_seq_event_output snd_seq_drain_output (until ret = 0) (every 10 events) check queue events and if getting full pause for 10ms I have tried it with and without the drain after each send, neither one works, with the drain seems to work a bit better (and with the drain I can call the queue status and get more info). The environment is a HyperV x64 Ubuntu 16.04.01 install with QT5.8 built from source, and ALSA from the distro version (not built from source). The same thing occurs on a Raspberry Pi 3B (which is my eventual target) running Ubuntu 16.04 Mate. Both environments will run pmidi with the same song correctly. Yes, I know it's my code, not an ALSA bug. What I'm hoping is some suggestion as to how to get more visibility as to what is happening when it pauses. Some kind of trace or debug mode, hopefully without trying to build and manually instrument ALSA. Conversely.... maybe I am using ALSA inappropriate to send just to the sequencer and on to a midi device? I just wanted its scheduler to deal with timing, there is no near term plan to use a synthesizer or other connection instead. Any suggestions or insights appreciated. I started with too little hair, and fear even more has been pulled out now. Linwood Ferguson

2 5

[alsa-devel] [PATCH 5/6] pcm:extplug: refinement of masks in extplug
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Awais Belal <awais_belal(a)mentor.com> It should be possible to use empty mask format with extplug. The refinement of mask via extplug is now modified, to accept empty masks as well to work properly. Signed-off-by: Awais Belal <awais_belal(a)mentor.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index a04f826..1004f54 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -172,6 +172,8 @@ int snd_ext_parm_mask_refine(snd_mask_t *mask, struct snd_ext_parm *parm, int ty unsigned int i; parm += type; + if (!parm->active) + return 0; memset(&bits, 0, sizeof(bits)); for (i = 0; i < parm->num_list; i++) bits.bits[parm->list[i] / 32] |= 1U << (parm->list[i] % 32); -- 2.7.4

2 1

[alsa-devel] [PATCH 4/6] pcm: status dump fix timestamp formatting
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> nanosecond part formatted with %06 will give incorrect/confusing results: trigger_time: 154.9748287 trigger_time: 154.60109090 trigger_time: 154.110425257 time seems to run backwards... This patch converts to us before printing which gives the correct/expected result: trigger_time: 154.009748 trigger_time: 154.060109 trigger_time: 154.110425 Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 493e903..a16fd86 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -2194,9 +2194,10 @@ int snd_pcm_status_dump(snd_pcm_status_t *status, snd_output_t *out) assert(status); snd_output_printf(out, " state : %s\n", snd_pcm_state_name((snd_pcm_state_t) status->state)); snd_output_printf(out, " trigger_time: %ld.%06ld\n", - status->trigger_tstamp.tv_sec, status->trigger_tstamp.tv_nsec); + status->trigger_tstamp.tv_sec, + status->trigger_tstamp.tv_nsec / 1000); snd_output_printf(out, " tstamp : %ld.%06ld\n", - status->tstamp.tv_sec, status->tstamp.tv_nsec); + status->tstamp.tv_sec, status->tstamp.tv_nsec / 1000); snd_output_printf(out, " delay : %ld\n", (long)status->delay); snd_output_printf(out, " avail : %ld\n", (long)status->avail); snd_output_printf(out, " avail_max : %ld\n", (long)status->avail_max); -- 2.7.4

2 1

[alsa-devel] [PATCH 3/6] pcm:file: Enable file writing for capture path
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Timo Wischer <twischer(a)de.adit-jv.com> This commit reverts parts of commit 4081be0b87ab9fa53a8906e66bc240f18a7a9a54, because it is realy useful to use the file plugin in a capture path for debugging. Also it fixes the truncate issue mentioned in above commit. Additionally following MMAP access issue is considered: $ arecord -D teeraw -M -d5 arecord.wav Recording WAVE 'arecord.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm/pcm_file.c:358:(snd_pcm_file_write_bytes) write failed: Bad file descriptor ALSA lib pcm/pcm_file.c:358:(snd_pcm_file_write_bytes) write failed: Bad file descriptor arecord: pcm/pcm_file.c:397: snd_pcm_file_add_frames: Assertion `file->wbuf_used_bytes < file->wbuf_size_bytes' failed. Aborted by signal Aborted... Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 6d119d6..0363f84 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -544,6 +544,7 @@ static snd_pcm_sframes_t snd_pcm_file_writen(snd_pcm_t *pcm, void **bufs, snd_pc static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size) { snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas[pcm->channels]; snd_pcm_sframes_t n; n = _snd_pcm_readi(file->gen.slave, buffer, size); @@ -555,8 +556,10 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc __snd_pcm_unlock(pcm); if (n < 0) return n; - return n * 8 / pcm->frame_bits; + n = n * 8 / pcm->frame_bits; } + snd_pcm_areas_from_buf(pcm, areas, buffer); + snd_pcm_file_add_frames(pcm, areas, 0, n); return n; } @@ -564,6 +567,7 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size) { snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas[pcm->channels]; snd_pcm_sframes_t n; if (file->ifd >= 0) { @@ -572,6 +576,10 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm } n = _snd_pcm_readn(file->gen.slave, bufs, size); + if (n > 0) { + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_add_frames(pcm, areas, 0, n); + } return n; } @@ -635,7 +643,7 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) a->first = slave->sample_bits * channel; a->step = slave->frame_bits; } - if ((file->fd < 0) && (pcm->stream == SND_PCM_STREAM_PLAYBACK)) { + if (file->fd < 0) { err = snd_pcm_file_open_output_file(file); if (err < 0) { SYSERR("failed opening output file %s", file->fname); -- 2.7.4

2 1

[alsa-devel] [PATCH 2/6] pcm:direct: fix race on clearing timer events
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Andreas Pape <apape(a)de.adit-jv.com> snd_timer handling is racy: plugins clear timer queue if avail_min is not reached to force a sleep on timer. The race can happen if the expected event arrives in between the avail check and the clearing of pending events. If this race happens, the user will unnecessarily wait for one more timer event. On low latency/realtime streams this can lead to xruns and must be avoided. As a fix we recheck avail after having cleared poll events. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 364191b..8a75c42 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -515,10 +515,12 @@ int snd_pcm_direct_async(snd_pcm_t *pcm, int sig, pid_t pid) } /* empty the timer read queue */ -void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix) +int snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix) { + int changed = 0; if (dmix->timer_need_poll) { while (poll(&dmix->timer_fd, 1, 0) > 0) { + changed++; /* we don't need the value */ if (dmix->tread) { snd_timer_tread_t rbuf[4]; @@ -533,15 +535,17 @@ void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix) snd_timer_tread_t rbuf[4]; int len; while ((len = snd_timer_read(dmix->timer, rbuf, - sizeof(rbuf))) > 0 && + sizeof(rbuf))) > 0 + && (++changed) && len != sizeof(rbuf[0])) ; } else { snd_timer_read_t rbuf; while (snd_timer_read(dmix->timer, &rbuf, sizeof(rbuf)) > 0) - ; + changed++; } } + return changed; } int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) @@ -693,6 +697,8 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in int empty = 0; assert(pfds && nfds == 1 && revents); + +timer_changed: events = pfds[0].revents; if (events & POLLIN) { snd_pcm_uframes_t avail; @@ -720,7 +726,16 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in break; default: if (empty) { - snd_pcm_direct_clear_timer_queue(dmix); + /* here we have a race condition: + * if period event arrived after the avail_update call + * above we might clear this event with the following + * clear_timer_queue. + * There is no way to do this in atomic manner, so we + * need to recheck avail_update if we successfully + * cleared a poll event. + */ + if (snd_pcm_direct_clear_timer_queue(dmix)) + goto timer_changed; events &= ~(POLLOUT|POLLIN); /* additional check */ switch (__snd_pcm_state(pcm)) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index fba55fd..24e85f0 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -322,7 +322,7 @@ int snd_pcm_direct_munmap(snd_pcm_t *pcm); int snd_pcm_direct_prepare(snd_pcm_t *pcm); int snd_pcm_direct_resume(snd_pcm_t *pcm); int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix); -void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix); +int snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix); int snd_pcm_direct_set_timer_params(snd_pcm_direct_t *dmix); int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dmix, const char *client_name); -- 2.7.4

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[alsa-devel] [PATCH 1/6] pcm: dmix_rewind corrupts application pointer fix
by sutar.mounesh＠gmail.com 17 Feb '17

17 Feb '17

From: Timo Wischer <twischer(a)de.adit-jv.com> sometimes pulseaudio stops with the following assertion in libasound.so: alsa-lib-1.0.29/src/pcm/pcm.c:2761: snd_pcm_area_copy: Assertion `dst < src || dst >= src + bytes' failed. Application pointer is handled properly, in cases of rewind operations. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Ravikiran Polepalli <ravikiran_polepalli(a)mentor.com> Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index dd0356e..4b763e2 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -706,7 +706,7 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f { snd_pcm_direct_t *dmix = pcm->private_data; snd_pcm_uframes_t slave_appl_ptr, slave_size; - snd_pcm_uframes_t appl_ptr, size, transfer, result; + snd_pcm_uframes_t appl_ptr, size, transfer, result, frames_to_remix; int err; const snd_pcm_channel_area_t *src_areas, *dst_areas; @@ -717,6 +717,11 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f return err; } + /* (appl_ptr - last_appl_ptr) indicates the frames which are not already mixed + (last_appl_ptr - hw_ptr) indicates the frames which are already mixed but not played yet. + So they can be remixed. + */ + if (dmix->last_appl_ptr < dmix->appl_ptr) size = dmix->appl_ptr - dmix->last_appl_ptr; else @@ -729,6 +734,9 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f return size; result = size; + /* Always at this point last_appl_ptr == appl_ptr + So (appl_ptr - hw_ptr) indicates the frames which can be remixed + */ if (dmix->hw_ptr < dmix->appl_ptr) size = dmix->appl_ptr - dmix->hw_ptr; else @@ -741,9 +749,12 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f slave_size = dmix->slave_appl_ptr + (pcm->boundary - dmix->slave_hw_ptr); if (slave_size < size) size = slave_size; - frames -= size; - result += size; - + + /* frames which should be remixed will be saved + to also backward the appl pointer on success + */ + frames_to_remix = size; + /* add sample areas here */ src_areas = snd_pcm_mmap_areas(pcm); dst_areas = snd_pcm_mmap_areas(dmix->spcm); @@ -769,15 +780,15 @@ static snd_pcm_sframes_t snd_pcm_dmix_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f appl_ptr += transfer; appl_ptr %= pcm->buffer_size; } - dmix->last_appl_ptr -= frames; - dmix->last_appl_ptr %= pcm->boundary; - dmix->slave_appl_ptr -= frames; - dmix->slave_appl_ptr %= dmix->slave_boundary; dmix_up_sem(dmix); - snd_pcm_mmap_appl_backward(pcm, frames); + snd_pcm_mmap_appl_backward(pcm, frames_to_remix); + result += frames_to_remix; + /* At this point last_appl_ptr and appl_ptr has to indicate the + * position of the first not mixed frame + */ - return result + frames; + return result; } static snd_pcm_sframes_t snd_pcm_dmix_forwardable(snd_pcm_t *pcm) -- 2.7.4

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[alsa-devel] Applied "ASoC: nau8825: automatic BCLK and LRC divde in master mode" to the asoc tree
by Mark Brown 17 Feb '17

17 Feb '17

The patch ASoC: nau8825: automatic BCLK and LRC divde in master mode has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1e561f6166bacb9c12d6fa1d23df07999674573e Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Fri, 17 Feb 2017 09:55:33 +0800 Subject: [PATCH] ASoC: nau8825: automatic BCLK and LRC divde in master mode configurable LRC and BCLK divide. The driver will make configurations of LRC and BCLK automatically according to BCLK and FS information in master mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8825.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 4576f987a4a5..97fbeba9498f 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1231,7 +1231,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, osr; + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; nau8825_sema_acquire(nau8825, 3 * HZ); @@ -1261,6 +1261,24 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, osr_adc_sel[osr].clk_src << NAU8825_CLK_ADC_SRC_SFT); } + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, &ctrl_val); + if (ctrl_val & NAU8825_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / params_rate(params); + if (bclk_fs <= 32) + bclk_div = 2; + else if (bclk_fs <= 64) + bclk_div = 1; + else if (bclk_fs <= 128) + bclk_div = 0; + else + return -EINVAL; + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, + ((bclk_div + 1) << NAU8825_I2S_LRC_DIV_SFT) | bclk_div); + } + switch (params_width(params)) { case 16: val_len |= NAU8825_I2S_DL_16; -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: Intel: bxt: Add jack port initialize in bxt_rt298 machine
by jeeja.kp＠intel.com 17 Feb '17

17 Feb '17

From: Jeeja KP <jeeja.kp(a)intel.com> After the pcm jack is created, create and initialize the pin switch widget for each port. Pin switch is to enable/disable the pin when monitor is connected/disconnected. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> --- Change in v2: Add the SOB. sound/soc/intel/boards/bxt_rt298.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index d5f53a6..176c080 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -459,10 +459,12 @@ static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; + struct snd_soc_codec *codec = NULL; int err, i = 0; char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + codec = pcm->codec_dai->codec; snprintf(jack_name, sizeof(jack_name), "HDMI/DP, pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, @@ -480,7 +482,10 @@ static int bxt_card_late_probe(struct snd_soc_card *card) i++; } - return 0; + if (!codec) + return -EINVAL; + + return hdac_hdmi_jack_port_init(codec, &card->dapm); } -- 2.5.0

1 0

[alsa-devel] [PATCH v2] ASoC: hdac_hdmi: Enable MST audio
by jeeja.kp＠intel.com 17 Feb '17

17 Feb '17

From: Jeeja KP <jeeja.kp(a)intel.com> With the patch series "ASoC: hdmi: Preparatory work to support MST audio", the design is extended from pin to port mapping to handle MST usecases. A pin is MST capable if pipe is valid value (i.e. not -1) in the notify callback. If pipe = -1, default port is 0 else port = pipe. A MST capable pin widget is programmed by first sending a verb to select the device/port and then the actual verb is sent to program the widget. This patch series enables MST Audio by configuring MST capable pins and extended the jack handling to report event for ports. Now the jack events are based on port connect/disconnect. The design is also extended to support mapping of multiple port to a single converter. Now with the introduction of multiple ports with a single pin, a pin widget is created to enable/disable path based on device connection. Required machine driver changes are done to handle this change. Also use the ASoC jack instead of snd_jack framework to creates jacks in the machine driver. Note: This patch series has dependency on the series "ASoC: hdmi: Preparatory work to support MST audio" Change v2: - Fixed the SOB missing in the patch. - Removed already applied patches. Jeeja KP (1): ASoC: Intel: bxt: Add jack port initialize in bxt_rt298 machine sound/soc/intel/boards/bxt_rt298.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) -- 2.5.0

2 2

[alsa-devel] [PATCH 00/16] ASoC: hdac_hdmi: Enable MST audio
by jeeja.kp＠intel.com 17 Feb '17

17 Feb '17

From: Jeeja KP <jeeja.kp(a)intel.com> With the patch series "ASoC: hdmi: Preparatory work to support MST audio", the design is extended from pin to port mapping to handle MST usecases. A pin is MST capable if pipe is valid value (i.e. not -1) in the notify callback. If pipe = -1, default port is 0 else port = pipe. A MST capable pin widget is programmed by first sending a verb to select the device/port and then the actual verb is sent to program the widget. This patch series enables MST Audio by configuring MST capable pins and extended the jack handling to report event for ports. Now the jack events are based on port connect/disconnect. The design is also extended to support mapping of multiple port to a single converter. Now with the introduction of multiple ports with a single pin, a pin widget is created to enable/disable path based on device connection. Required machine driver changes are done to handle this change. Also use the ASoC jack instead of snd_jack framework to creates jacks in the machine driver. Note: This patch series has dependency on the series "ASoC: hdmi: Preparatory work to support MST audio" Jeeja KP (16): ASoC: hdac_hdmi: Add MST verb support ASoC: hdac_hdmi: Handle MST pin jack detection at boot/resume ASoc: hdac_hdmi: Configure pin verbs for MST ASoC: hdac_hdmi: Add support for multiple ports to a PCM ASoC: hdac_hdmi: Use ASoC jack instead of snd_jack ASoC: Intel: Skylake: Create ASoC jack for hdmi in rt286 machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in skl_nau88l25_max98357a machine ASoC: Intel: Skylake: Create ASoC jack for hdmi in nau88l25_ssm4567 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_rt298 machine ASoC: Intel: bxt: Create ASoC jack for hdmi in bxt_da7219_max98357 machine ASoC: hdac_hdmi: Add machine pin widget for each port ASoC: Intel: Skylake: Add jack port initialize in rt286 machine ASoC: Intel: Skylake: Add jack port initialize in nau88l25_max98357a machine ASoC: Intel: Skylake: Add jack port initialize in nau88l25_ssm4567 machine ASoC: Intel: bxt: Add jack port initialize in bxt_rt298 machine ASoC: Intel: bxt: Add jack port initialize in da7219_max98357a machine sound/soc/codecs/hdac_hdmi.c | 438 +++++++++++++++++++----- sound/soc/codecs/hdac_hdmi.h | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 26 +- sound/soc/intel/boards/bxt_rt298.c | 26 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 30 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 30 +- sound/soc/intel/boards/skl_rt286.c | 30 +- 7 files changed, 482 insertions(+), 103 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v3 0/7] Move dell-led to drivers/platform/x86
by Michał Kępień 17 Feb '17

17 Feb '17

This patch series moves the dell-led driver from the LED subsystem to the x86 platform driver subsystem. The original motivation behind this effort was to move all code using the dell-smbios module to the x86 platform driver subsystem. While I was investigating the possibilities to do that, it quickly emerged that dell-led can and in fact should be moved to the x86 platform driver subsystem in its entirety. dell-led consists of two major parts: - the part exposing a microphone mute LED interface, introduced in db6d8cc00773 ("dell-led: add mic mute led interface"); this interface is used by sound/pci/hda/dell_wmi_helper.c; while the original implementation used a WMI interface, it was changed to use dell-smbios in cf0d7ea33596 ("dell-led: use dell_smbios_find_token() for finding mic DMI tokens") and 0c41a08e131d ("dell-led: use dell_smbios_send_request() for performing SMBIOS calls"), - the part handling an activity LED present in Dell Latitude 2100 netbooks, introduced in 72dcd8d08aca ("leds: Add Dell Business Class Netbook LED driver"); it binds to a specific WMI GUID and then registers a LED device which is controlled using WMI (i.e. it is essentially a WMI driver). Patches 1 and 2 clean up the microphone mute LED interface to minimize the amount of code moved around. Patch 3 updates a variable name in sound/pci/hda/dell_wmi_helper.c so that it better matches that variable's role. Patch 4 moves the microphone mute LED interface to drivers/platform/x86/dell-laptop.c, effectively causing sound/pci/hda/dell_wmi_helper.c to depend on CONFIG_DELL_LAPTOP instead of CONFIG_LEDS_DELL_NETBOOKS. Patch 5 reverts dell-led to the state it was in after its initial commit 72dcd8d08aca ("leds: Add Dell Business Class Netbook LED driver") by removing all remnants of the microphone mute LED handling code. Patch 6 moves all that is left of dell-led (i.e. the activity LED part, as originally implemented), to a new module which is placed in drivers/platform/x86/dell-wmi-led.c. Patch 7 fixes coding style issues in drivers/platform/x86/dell-wmi-led.c to make sure it gets a clean start in the x86 platform driver subsystem. As all patches except patch 3 in this series affect the LED subsystem, the series is based on linux-leds/for-next. Changes from v2: - Add patch 7 fixing coding style issues (feedback from Joe and Andy taken into account). - Add leading zero to constants in patch 4. - Move dell-led.h include one line higher in patch 4. Changes from v1: - Squash patches 2-4 from v1 into a single patch (#2 in v2). - Add patch 3. - Fix subject pattern in patch 4. - Slight commit message adjustments, including fixing a typo ("COFIG_LEDS_DELL_NETBOOKS") in patch 6. - Remove the name of the module's source file from the header comment in drivers/platform/x86/dell-wmi-led.c to avoid the need to update it in the future. drivers/leds/Kconfig | 9 -- drivers/leds/Makefile | 1 - drivers/platform/x86/Kconfig | 8 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/dell-laptop.c | 28 ++++ .../dell-led.c => platform/x86/dell-wmi-led.c} | 141 +++++---------------- include/linux/dell-led.h | 6 +- sound/pci/hda/dell_wmi_helper.c | 30 ++--- 8 files changed, 86 insertions(+), 138 deletions(-) rename drivers/{leds/dell-led.c => platform/x86/dell-wmi-led.c} (56%) -- 2.11.1

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[alsa-devel] [asoc:topic/intel 83/98] sound/soc/codecs/hdac_hdmi.c:1884:2-21: iterator with update on line 1885 (fwd)
by Julia Lawall 16 Feb '17

16 Feb '17

I don't think that what is done in lines 1884 and 1885 is the right approach. port is just a local variable that points to the different list elements and allows list_for_each_entry to get from one element to the other. Setting port to NULL doesn't change the list, and it makes it impossible for list_for_each_entry to find the next element. julia ---------- Forwarded message ---------- Date: Fri, 17 Feb 2017 05:00:49 +0800 From: kbuild test robot <fengguang.wu(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: [asoc:topic/intel 83/98] sound/soc/codecs/hdac_hdmi.c:1884:2-21: iterator with update on line 1885 tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: e3efb2ad834b50cb9c8625155e3e2674f5bc443b commit: e0e5d3e5a53b3bc354c18030b78b7ebcb33e004b [83/98] ASoC: hdac_hdmi: Add support for multiple ports to a PCM :::::: branch date: 2 hours ago :::::: commit date: 2 hours ago >> sound/soc/codecs/hdac_hdmi.c:1884:2-21: iterator with update on line 1885 git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout e0e5d3e5a53b3bc354c18030b78b7ebcb33e004b vim +1884 sound/soc/codecs/hdac_hdmi.c 18382ead Subhransu S. Prusty 2015-11-10 1878 4a3478de Jeeja KP 2016-02-12 1879 list_for_each_entry_safe(pcm, pcm_next, &hdmi->pcm_list, head) { 4a3478de Jeeja KP 2016-02-12 1880 pcm->cvt = NULL; e0e5d3e5 Jeeja KP 2017-02-07 1881 if (list_empty(&pcm->port_list)) e0e5d3e5 Jeeja KP 2017-02-07 1882 continue; e0e5d3e5 Jeeja KP 2017-02-07 1883 e0e5d3e5 Jeeja KP 2017-02-07 @1884 list_for_each_entry(port, &pcm->port_list, head) e0e5d3e5 Jeeja KP 2017-02-07 @1885 port = NULL; e0e5d3e5 Jeeja KP 2017-02-07 1886 4a3478de Jeeja KP 2016-02-12 1887 list_del(&pcm->head); 4a3478de Jeeja KP 2016-02-12 1888 kfree(pcm); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/intel 79/98] sound/soc/codecs/hdac_hdmi.c:1419:5-8: ERROR: invalid reference to the index variable of the iterator on line 1397
by Julia Lawall 16 Feb '17

16 Feb '17

Lists don't end with NULL, They end with a dummy element. So pin in line 1419 is always true. You probably need some other way of detecting how the loop was exited. julia ---------- Forwarded message ---------- Date: Fri, 17 Feb 2017 04:31:06 +0800 From: kbuild test robot <fengguang.wu(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: [asoc:topic/intel 79/98] sound/soc/codecs/hdac_hdmi.c:1419:5-8: ERROR: invalid reference to the index variable of the iterator on line 1397 tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: e3efb2ad834b50cb9c8625155e3e2674f5bc443b commit: 2acd8309a3a4e6dc04e72d2db0716825095c02d6 [79/98] ASoC: hdac_hdmi: Add support to handle MST capable pin :::::: branch date: 2 hours ago :::::: commit date: 8 days ago >> sound/soc/codecs/hdac_hdmi.c:1419:5-8: ERROR: invalid reference to the index variable of the iterator on line 1397 git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 2acd8309a3a4e6dc04e72d2db0716825095c02d6 vim +1419 sound/soc/codecs/hdac_hdmi.c b8a54545 Subhransu S. Prusty 2016-02-12 1391 SNDRV_CTL_POWER_D0) b8a54545 Subhransu S. Prusty 2016-02-12 1392 return; b8a54545 Subhransu S. Prusty 2016-02-12 1393 b8a54545 Subhransu S. Prusty 2016-02-12 1394 if (atomic_read(&edev->hdac.in_pm)) b8a54545 Subhransu S. Prusty 2016-02-12 1395 return; b8a54545 Subhransu S. Prusty 2016-02-12 1396 b8a54545 Subhransu S. Prusty 2016-02-12 @1397 list_for_each_entry(pin, &hdmi->pin_list, head) { 754695f9 Jeeja KP 2017-02-06 1398 if (pin->nid != pin_nid) 754695f9 Jeeja KP 2017-02-06 1399 continue; 754695f9 Jeeja KP 2017-02-06 1400 754695f9 Jeeja KP 2017-02-06 1401 /* In case of non MST pin, pipe is -1 */ 754695f9 Jeeja KP 2017-02-06 1402 if (pipe == -1) { 2acd8309 Jeeja KP 2017-02-06 1403 pin->mst_capable = false; 754695f9 Jeeja KP 2017-02-06 1404 /* if not MST, default is port[0] */ 754695f9 Jeeja KP 2017-02-06 1405 hport = &pin->ports[0]; 2acd8309 Jeeja KP 2017-02-06 1406 goto out; 2acd8309 Jeeja KP 2017-02-06 1407 } else { 2acd8309 Jeeja KP 2017-02-06 1408 for (i = 0; i < pin->num_ports; i++) { 2acd8309 Jeeja KP 2017-02-06 1409 pin->mst_capable = true; 2acd8309 Jeeja KP 2017-02-06 1410 if (pin->ports[i].id == pipe) { 2acd8309 Jeeja KP 2017-02-06 1411 hport = &pin->ports[i]; 2acd8309 Jeeja KP 2017-02-06 1412 goto out; 2acd8309 Jeeja KP 2017-02-06 1413 } 2acd8309 Jeeja KP 2017-02-06 1414 } 754695f9 Jeeja KP 2017-02-06 1415 } b8a54545 Subhransu S. Prusty 2016-02-12 1416 } 754695f9 Jeeja KP 2017-02-06 1417 2acd8309 Jeeja KP 2017-02-06 1418 out: 2acd8309 Jeeja KP 2017-02-06 @1419 if (pin && hport) 754695f9 Jeeja KP 2017-02-06 1420 hdac_hdmi_present_sense(pin, hport); b8a54545 Subhransu S. Prusty 2016-02-12 1421 } b8a54545 Subhransu S. Prusty 2016-02-12 1422 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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