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[alsa-devel] Applied "ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static" to the asoc tree
by Mark Brown 11 Jul '17

11 Jul '17

The patch ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static 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 fc2b74460d96ec28110a581d6e58d15c27304d16 Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Tue, 11 Jul 2017 15:08:55 +0100 Subject: [PATCH] ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static Don't populate array hdmi_codec_eld_spk_alloc_bits on the stack but make it static. Makes the object code smaller by over 260 bytes: Before: text data bss dec hex filename 10882 3384 64 14330 37fa sound/soc/codecs/hdmi-codec.o After: text data bss dec hex filename 10557 3440 64 14061 36ed sound/soc/codecs/hdmi-codec.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7686a80861f1..509ab513b4b2 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -326,7 +326,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) { int i; - const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { + static const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { [0] = FL | FR, [1] = LFE, [2] = FC, [3] = RL | RR, [4] = RC, [5] = FLC | FRC, [6] = RLC | RRC, }; -- 2.13.2

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[alsa-devel] [PATCH] ASoC: hdmi-codec: make const array hdmi_codec_eld_spk_alloc_bits static
by Colin King 11 Jul '17

11 Jul '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate array hdmi_codec_eld_spk_alloc_bits on the stack but make it static. Makes the object code smaller by over 260 bytes: Before: text data bss dec hex filename 10882 3384 64 14330 37fa sound/soc/codecs/hdmi-codec.o After: text data bss dec hex filename 10557 3440 64 14061 36ed sound/soc/codecs/hdmi-codec.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7686a80861f1..509ab513b4b2 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -326,7 +326,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, static unsigned long hdmi_codec_spk_mask_from_alloc(int spk_alloc) { int i; - const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { + static const unsigned long hdmi_codec_eld_spk_alloc_bits[] = { [0] = FL | FR, [1] = LFE, [2] = FC, [3] = RL | RR, [4] = RC, [5] = FLC | FRC, [6] = RLC | RRC, }; -- 2.11.0

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[alsa-devel] New codec device ID for ALC1220
by Kailang 11 Jul '17

11 Jul '17

Hi Takashi, ALC1220 has another device ID. Attach patch was add 0x1168 device ID support. Many Thanks. BR, Kailang

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[alsa-devel] [PATCH][-next] ASoC: rt5663: add in missing loop counter to avoid infinite loop
by Colin King 10 Jul '17

10 Jul '17

From: Colin Ian King <colin.king(a)canonical.com> The loop counter 'count' is missing an increment, so we could end up with an infinite loop. Add in a pre-increment to count to fix this. Detected by CoverityScan, CID#1450120 ("Logically Dead Code") Fixes: 1a425dbf1a10 ("ASoC: rt5663: Update the calibration funciton") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 82edbcf39caa..5b3c50c641d8 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3044,7 +3044,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) else break; - if (count > 200) + if (++count > 200) break; } -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rt5514: Move the auto disable DSP function to set_bias_level()
by Oder Chiou 10 Jul '17

10 Jul '17

Move the auto disable DSP function from the widget event "DAPM Pre" to the set_bias_level() function to avoid the unexpected behavior. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 49 +++++++++++++++++------------------------------ 1 file changed, 18 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1b6796c..a71e630 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -471,35 +471,6 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } -static int rt5514_pre_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 rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - /** - * If the DSP is enabled in start of recording, the DSP - * should be disabled, and sync back to normal recording - * settings to make sure recording properly. - */ - if (rt5514->dsp_enabled) { - rt5514->dsp_enabled = 0; - regmap_multi_reg_write(rt5514->i2c_regmap, - rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); - regcache_mark_dirty(rt5514->regmap); - regcache_sync(rt5514->regmap); - } - break; - - default: - return 0; - } - - return 0; -} - static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -607,8 +578,6 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), - - SND_SOC_DAPM_PRE("DAPM Pre", rt5514_pre_event), }; static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { @@ -975,6 +944,24 @@ static int rt5514_set_bias_level(struct snd_soc_codec *codec, } break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* + * If the DSP is enabled in start of recording, the DSP + * should be disabled, and sync back to normal recording + * settings to make sure recording properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regmap_multi_reg_write(rt5514->i2c_regmap, + rt5514_i2c_patch, + ARRAY_SIZE(rt5514_i2c_patch)); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->regmap); + } + } + break; + default: break; } -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: rt5663: Add the manual offset field to compensate the DC offset
by Oder Chiou 10 Jul '17

10 Jul '17

The patch adds the manual offset field in the devicetree to compensate the DC offset that will be different between the PCB layout. It only can be measured by the real production. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5663.txt | 5 ++ include/sound/rt5663.h | 20 ++++++++ sound/soc/codecs/rt5663.c | 55 ++++++++++++++++++++-- sound/soc/codecs/rt5663.h | 2 + 4 files changed, 77 insertions(+), 5 deletions(-) create mode 100644 include/sound/rt5663.h diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 70eaeae..f7c7cc5 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -12,6 +12,11 @@ Required properties: Optional properties: +- "realtek,dc_offset_l_manual" +- "realtek,dc_offset_r_manual" + Based on the different PCB layout, add the manual offset value to + compensate the DC offset for each L and R channel. + Pins on the device (for linking into audio routes) for RT5663: * IN1P diff --git a/include/sound/rt5663.h b/include/sound/rt5663.h new file mode 100644 index 0000000..be6f1b2 --- /dev/null +++ b/include/sound/rt5663.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5663.h -- Platform data for RT5663 + * + * Copyright 2017 Realtek Semiconductor Corp. + * + * 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 __LINUX_SND_RT5663_H +#define __LINUX_SND_RT5663_H + +struct rt5663_platform_data { + unsigned int dc_offset_l_manual; + unsigned int dc_offset_r_manual; +}; + +#endif + diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 82edbcf..52ecca4 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -40,6 +40,7 @@ enum { struct rt5663_priv { struct snd_soc_codec *codec; + struct rt5663_platform_data pdata; struct regmap *regmap; struct delayed_work jack_detect_work; struct snd_soc_jack *hs_jack; @@ -57,6 +58,10 @@ struct rt5663_priv { int jack_type; }; +static const struct reg_sequence rt5663_patch_list[] = { + { 0x002a, 0x8080 }, +}; + static const struct reg_default rt5663_v2_reg[] = { { 0x0000, 0x0000 }, { 0x0001, 0xc8c8 }, @@ -476,7 +481,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0023, 0x0039 }, { 0x0026, 0xc0c0 }, { 0x0029, 0x8080 }, - { 0x002a, 0xa0a0 }, + { 0x002a, 0x8080 }, { 0x002c, 0x000c }, { 0x002d, 0x0000 }, { 0x0040, 0x0808 }, @@ -1958,15 +1963,11 @@ static const struct snd_kcontrol_new rt5663_adda_r_mix[] = { static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, RT5663_M_DAC_L1_STO_L_SHIFT, 1, 1), - SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, - RT5663_M_DAC_R1_STO_L_SHIFT, 1, 1), }; static const struct snd_kcontrol_new rt5663_sto1_dac_r_mix[] = { SOC_DAPM_SINGLE("DAC L Switch", RT5663_STO_DAC_MIXER, RT5663_M_DAC_L1_STO_R_SHIFT, 1, 1), - SOC_DAPM_SINGLE("DAC R Switch", RT5663_STO_DAC_MIXER, - RT5663_M_DAC_R1_STO_R_SHIFT, 1, 1), }; /* Out Switch */ @@ -3126,9 +3127,20 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) usleep_range(10000, 10005); } +static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) +{ + device_property_read_u32(dev, "realtek,dc_offset_l_manual", + &rt5663->pdata.dc_offset_l_manual); + device_property_read_u32(dev, "realtek,dc_offset_r_manual", + &rt5663->pdata.dc_offset_r_manual); + + return 0; +} + static int rt5663_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt5663_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5663_priv *rt5663; int ret; unsigned int val; @@ -3142,6 +3154,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5663); + if (pdata) + rt5663->pdata = *pdata; + else + rt5663_parse_dp(rt5663, &i2c->dev); + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); if (IS_ERR(regmap)) { ret = PTR_ERR(regmap); @@ -3190,6 +3207,34 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_write(rt5663->regmap, RT5663_RESET, 0); dev_dbg(&i2c->dev, "calibrate done\n"); + switch (rt5663->codec_ver) { + case CODEC_VER_1: + break; + case CODEC_VER_0: + ret = regmap_register_patch(rt5663->regmap, rt5663_patch_list, + ARRAY_SIZE(rt5663_patch_list)); + if (ret != 0) + dev_warn(&i2c->dev, + "Failed to apply regmap patch: %d\n", ret); + break; + default: + dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); + } + + if (rt5663->pdata.dc_offset_l_manual) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_2, + rt5663->pdata.dc_offset_l_manual >> 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_3, + rt5663->pdata.dc_offset_l_manual & 0xffff); + } + + if (rt5663->pdata.dc_offset_r_manual) { + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_5, + rt5663->pdata.dc_offset_r_manual >> 16); + regmap_write(rt5663->regmap, RT5663_MIC_DECRO_6, + rt5663->pdata.dc_offset_r_manual & 0xffff); + } + /* GPIO1 as IRQ */ regmap_update_bits(rt5663->regmap, RT5663_GPIO_1, RT5663_GP1_PIN_MASK, RT5663_GP1_PIN_IRQ); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index 4621812..c5a9b69 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -12,6 +12,8 @@ #ifndef __RT5663_H__ #define __RT5663_H__ +#include <sound/rt5663.h> + /* Info */ #define RT5663_RESET 0x0000 #define RT5663_VENDOR_ID 0x00fd -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix default dma_buffer_size
by Subhransu S. Prusty 10 Jul '17

10 Jul '17

If the dma_buffer_size is not defined in topology, fix it to 2ms default value to make backward compatible. Fixes: f6e6ab1d16ec ("ASoC: Intel: Skylake: Fix dma buffer size calculation") Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index eca85827dbd2..fb2f1f603f3c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -540,6 +540,14 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, cpr_mconfig->gtw_cfg.dma_buffer_size = mconfig->dma_buffer_size * dma_io_buf; + /* fallback to 2ms default value */ + if (!cpr_mconfig->gtw_cfg.dma_buffer_size) { + if (mconfig->hw_conn_type == SKL_CONN_SOURCE) + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs; + else + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs; + } + cpr_mconfig->cpr_feature_mask = 0; cpr_mconfig->gtw_cfg.config_length = 0; -- 1.9.1

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[alsa-devel] ALSA BLE MIDI
by happy musicmaker 10 Jul '17

10 Jul '17

Is there development for Bluetooth (BLE) MIDI under Linux ? More and more devices now support BLE MIDI. I have an BLE USB adapter that works fine with many MIDI BLE devices but the devices don't shop up in ALSA (Ubuntu 17.04). It seems Linux only supports audio devices.

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[alsa-devel] MEDION sound devices informations
by fulgor fulgor 10 Jul '17

10 Jul '17

I have run the script alsa-info.SH www.alsa-project.org/db/?f=6fb68d0201f421b41fe65677fde2ec5b1c9d4429 thanks for all your work

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[alsa-devel] [PATCH] ALSA: pcm: Simplify check for dma_mmap_coherent() availability
by Takashi Iwai 10 Jul '17

10 Jul '17

We check the availability of dma_mmap_coherent() in hw_support_mmap() but with an ugly ifdef of lots of arch-checks. Now we have a nice CONFIG_ARCH_NO_COHERENT_DMA_MMAP kconfig, and this can be used together with CONFIG_HAS_DMA check for a cleaner and more comprehensive check. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b26c7e61d600..2a730f38e375 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -238,10 +238,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) { if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - /* check architectures that return -EINVAL from dma_mmap_coherent() */ - /* FIXME: this should be some global flag */ -#if defined(CONFIG_C6X) || defined(CONFIG_FRV) || defined(CONFIG_MN10300) ||\ - defined(CONFIG_PARISC) || defined(CONFIG_XTENSA) + /* architecture supports dma_mmap_coherent()? */ +#if defined(CONFIG_ARCH_NO_COHERENT_DMA_MMAP) || !defined(CONFIG_HAS_DMA) if (!substream->ops->mmap && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return false; -- 2.13.2

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[alsa-devel] [PATCH] ALSA: pcm: Protect call to dma_mmap_coherent() by check for HAS_DMA
by Geert Uytterhoeven 10 Jul '17

10 Jul '17

If NO_DMA=y: sound/core/pcm_native.o: In function `snd_pcm_lib_default_mmap': pcm_native.c:(.text+0x144c): undefined reference to `bad_dma_ops' pcm_native.c:(.text+0x1474): undefined reference to `dma_common_mmap' Add a check for HAS_DMA to fix this. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b3d5bed750297ec2..e540a22aa3d8988a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3502,7 +3502,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, } #endif /* CONFIG_GENERIC_ALLOCATOR */ #ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ - if (!substream->ops->page && + if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, area, -- 2.7.4

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[alsa-devel] arecord cannot interrupt streaming via CTRL-C when using voc_pcm_wirte()
by Wischer, Timo (ADITG/ESB) 10 Jul '17

10 Jul '17

Hi all, please merge following patch: >From fb58aa47b8715924667a0a4383ce9a3615378413 Mon Sep 17 00:00:00 2001 From: Srikanth Krishnakar <Srikanth_Krishnakar(a)mentor.com> Date: Mon, 19 Dec 2016 14:04:39 +0530 Subject: alsa-utils: interrupt streaming via signal in voc_pcm_write arecord (alsa-utils v1.1.2) cannot interrupt streaming via CTRL-C. Fixed the issue by properly handling 'in_aborting' flag in appropriate functions. Signed-off-by: Anant Agrawal <Anant_Agrawal(a)mentor.com> Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Signed-off-by: Srikanth Krishnakar <Srikanth_Krishnakar(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 2da7dda..cdaa1a8 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2161,7 +2161,7 @@ static ssize_t voc_pcm_write(u_char *data, size_t count) ssize_t result = count, r; size_t size; - while (count > 0) { + while (count > 0 && !in_aborting) { size = count; if (size > chunk_bytes - buffer_pos) size = chunk_bytes - buffer_pos; -- 2.7.4 Best regards Timo Wischer Advanced Driver Information Technology GmbH Software Group I (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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[alsa-devel] [PATCH v8 0/5] Add platform clock for BayTrail platforms
by Pierre-Louis Bossart 10 Jul '17

10 Jul '17

These patches specifically enable the audio MCLK required by Baytrail CR devices. It is the remaining part of a bigger set of patches (already merged in Mark Brown's tree) that enable sound for Baytrail CR devices (especially Asus T100TAF) [1]. They include the clock driver and clock enabling in the pmc_atom code (along with moving of the non-architectural pmc_atom driver code into drivers/platform/x86 as suggested by Thomas Gleixner [2]). This move includes a new header in include/linux/platform_data/x86/. While there is an agreement that the definitions for PMC clocks are not really platform data this location is seen as a good-enough compromise with an agreement between Darren Hart and Andy Shevchenko [3] [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111704.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-October/113936.ht… [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-December/115892.h… Changes from v7: - Replaced clk-readl/writel by regular readl/writel - Fixed two nitpick comments (newline and driver name string) - Fixed [5:0] to [0..5] in comment (missed in v6) Changes from v6: - Addressed comments from Stephen Boyd, Andy Shevchenko and Vinod Koul - Renamed clk-byt-plt to clk-pmc-atom - Rebase to 4.10-rc3 (broonie/for-next) - Added helper functions to make error handling more compact - Removed module-related code since it's not possible/supported - Removed useless tests on remove - PMC_CLK_OFFSET move to clk-pcm-atom (more logical) - Split typo correction and makefile fix in separate patches - small fixes (spaces, defaults in switches, alphabetical order, use of clk_writel/readl and no test on write, simpler error handling, use of while(i--), unsigned loop variables, kernel doc fields, format-patch -M -C -D, use of PLATFORM_DEVID_NONE) Changes from v5: - fix build error reported by kbuild test robot - split the clk driver code from x86 platform changes Changes from v4: - move the pmc_atom driver from arch/x86/platform/atom to drivers/platform/x86 Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create Irina Tirdea (3): 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 Pierre-Louis Bossart (2): clk: Make x86/ conditional on CONFIG_COMMON_CLK platform/x86: fix typo in comment arch/x86/Kconfig | 4 - arch/x86/platform/atom/Makefile | 1 - drivers/acpi/acpi_lpss.c | 2 +- drivers/clk/Makefile | 2 + drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-pmc-atom.c | 371 +++++++++++++++++++++ drivers/platform/x86/Kconfig | 5 + drivers/platform/x86/Makefile | 1 + .../atom => drivers/platform/x86}/pmc_atom.c | 88 ++++- include/linux/platform_data/x86/clk-pmc-atom.h | 44 +++ .../linux/platform_data/x86}/pmc_atom.h | 2 +- 11 files changed, 506 insertions(+), 15 deletions(-) create mode 100644 drivers/clk/x86/clk-pmc-atom.c rename {arch/x86/platform/atom => drivers/platform/x86}/pmc_atom.c (87%) create mode 100644 include/linux/platform_data/x86/clk-pmc-atom.h rename {arch/x86/include/asm => include/linux/platform_data/x86}/pmc_atom.h (98%) -- 2.7.4

4 11

[alsa-devel] [PATCH v2] ASoC: rt5514: Move the auto disable DSP function to set_bias_level()
by Oder Chiou 10 Jul '17

10 Jul '17

Move the auto disable DSP function from the widget event "DAPM Pre" to the set_bias_level() function to avoid the unexpected behavior. While the SPI was recording using the DAI "rt5514-dsp-cpu-dai", the event "DAPM Pre" would be manipulated even the DAPM widgets were not connected, and it also would power off the DSP that cause the DSP buffered data lost. So the auto disable DSP function should move to the set_bias_level() function to make sure that it is only manipulated by the DAPM widgets are connected. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 49 +++++++++++++++++------------------------------ 1 file changed, 18 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1b6796c..a71e630 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -471,35 +471,6 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } -static int rt5514_pre_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 rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - /** - * If the DSP is enabled in start of recording, the DSP - * should be disabled, and sync back to normal recording - * settings to make sure recording properly. - */ - if (rt5514->dsp_enabled) { - rt5514->dsp_enabled = 0; - regmap_multi_reg_write(rt5514->i2c_regmap, - rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); - regcache_mark_dirty(rt5514->regmap); - regcache_sync(rt5514->regmap); - } - break; - - default: - return 0; - } - - return 0; -} - static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -607,8 +578,6 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), - - SND_SOC_DAPM_PRE("DAPM Pre", rt5514_pre_event), }; static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { @@ -975,6 +944,24 @@ static int rt5514_set_bias_level(struct snd_soc_codec *codec, } break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* + * If the DSP is enabled in start of recording, the DSP + * should be disabled, and sync back to normal recording + * settings to make sure recording properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regmap_multi_reg_write(rt5514->i2c_regmap, + rt5514_i2c_patch, + ARRAY_SIZE(rt5514_i2c_patch)); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->regmap); + } + } + break; + default: break; } -- 2.7.4

1 0

[alsa-devel] [PATCH v2 0/9] add support for Sama5d2 audio PLLs and enable ClassD
by Quentin Schulz 10 Jul '17

10 Jul '17

This patch series adds support for the audio PLLs and enables ClassD that can be found in ATMEL Sama5d2 SoC. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, there are multiple children clocks that could technically change the rate of audio_pll (via gck). With the rate locking introduced in Jerome Brunet's patch series[5], the first consumer to enable the clock will be the one definitely setting the rate of the clock. Without the rate locking, the last consumer to set the rate will be able to mess with the rate. Since audio IPs are most likely to request the same rate, we enforce that the only clks able to modify gck rate are those of audio IPs. To remain consistent, we deny other clocks to be children of audio_pll. Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html [5] http://www.spinics.net/lists/linux-clk/msg17927.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Quentin Schulz (7): clk: at91: clk-generated: remove useless divisor loop clk: at91: add audio plls to the compatible list in DT binding clk: at91: add audio pll clock drivers clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate ASoC: atmel-classd: remove aclk clock from DT binding ASoC: atmel-classd: remove aclk clock .../devicetree/bindings/clock/at91-clock.txt | 10 + .../devicetree/bindings/sound/atmel-classd.txt | 9 +- arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 ++ arch/arm/boot/dts/sama5d2.dtsi | 39 +++- arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 2 + drivers/clk/at91/clk-audio-pll-pad.c | 206 ++++++++++++++++++ drivers/clk/at91/clk-audio-pll-pmc.c | 174 +++++++++++++++ drivers/clk/at91/clk-audio-pll.c | 239 +++++++++++++++++++++ drivers/clk/at91/clk-generated.c | 86 ++++++-- include/linux/clk/at91_pmc.h | 25 +++ sound/soc/atmel/atmel-classd.c | 47 ++-- 12 files changed, 798 insertions(+), 59 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll-pad.c create mode 100644 drivers/clk/at91/clk-audio-pll-pmc.c create mode 100644 drivers/clk/at91/clk-audio-pll.c -- 2.11.0

6 15

[alsa-devel] [PATCH v2] ASoC: hdmi-codec: ELD control corresponds to the PCM stream
by Kuninori Morimoto 07 Jul '17

07 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current hdmi-codec driver is using hdmi_controls for "ELD" control. But, hdmi-codec driver might be used from many HDMIs. Thus, we need to correspond device number, otherwise we will receive below error. xxx: control x:x:x:ELD:x is already present This patch registers ELD control in .pcm_new by using .device = rtd->pcm->device to corresponding to PCM stream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [Takashi: use snd_ctl_new1()/snd_ctl_add()] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1 -> v2 - change title - ASoC: hdmi-codec: hdmi_codec_priv includes snd_kcontrol_new + ASoC: hdmi-codec: ELD control corresponds to the PCM stream - hdmi_eld_ctl is now local variable - Author should be Takashi ? sound/soc/codecs/hdmi-codec.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 22ed0dc..7686a80 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -399,18 +399,6 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, return 0; } - -static const struct snd_kcontrol_new hdmi_controls[] = { - { - .access = SNDRV_CTL_ELEM_ACCESS_READ | - SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "ELD", - .info = hdmi_eld_ctl_info, - .get = hdmi_eld_ctl_get, - }, -}; - static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -668,6 +656,16 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai_driver *drv = dai->driver; struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct snd_kcontrol *kctl; + struct snd_kcontrol_new hdmi_eld_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, + .device = rtd->pcm->device, + }; int ret; dev_dbg(dai->dev, "%s()\n", __func__); @@ -686,7 +684,12 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, hcp->chmap_info->chmap = hdmi_codec_stereo_chmaps; hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; - return 0; + /* add ELD ctl with the device number corresponding to the PCM stream */ + kctl = snd_ctl_new1(&hdmi_eld_ctl, dai->component); + if (!kctl) + return -ENOMEM; + + return snd_ctl_add(rtd->card->snd_card, kctl); } static struct snd_soc_dai_driver hdmi_i2s_dai = { @@ -732,8 +735,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static struct snd_soc_codec_driver hdmi_codec = { .component_driver = { - .controls = hdmi_controls, - .num_controls = ARRAY_SIZE(hdmi_controls), .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, -- 1.9.1

4 3

[alsa-devel] [PATCH 1/3] ASoC: rt5663: Update the calibration funciton
by Oder Chiou 07 Jul '17

07 Jul '17

The patch updates the calibration funciton for getting the correct DC offset of headphone output. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 100 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 79 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index a33202a..18b3b52 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2986,47 +2986,75 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) { int value, count; - regmap_write(rt5663->regmap, RT5663_RC_CLK, 0x0280); + regmap_write(rt5663->regmap, RT5663_RESET, 0x0000); + msleep(20); + regmap_write(rt5663->regmap, RT5663_ANA_BIAS_CUR_4, 0x00a1); + regmap_write(rt5663->regmap, RT5663_RC_CLK, 0x0380); regmap_write(rt5663->regmap, RT5663_GLB_CLK, 0x8000); - regmap_write(rt5663->regmap, RT5663_DIG_MISC, 0x8001); + regmap_write(rt5663->regmap, RT5663_ADDA_CLK_1, 0x1000); regmap_write(rt5663->regmap, RT5663_VREF_RECMIX, 0x0032); - regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa2be); - msleep(20); - regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf2be); - regmap_write(rt5663->regmap, RT5663_PWR_DIG_2, 0x8400); - regmap_write(rt5663->regmap, RT5663_CHOP_ADC, 0x3000); - regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x003b); - regmap_write(rt5663->regmap, RT5663_PWR_DIG_1, 0x8df8); - regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x0003); - regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c); - regmap_write(rt5663->regmap, RT5663_ADDA_CLK_1, 0x1111); + regmap_write(rt5663->regmap, RT5663_HP_IMP_SEN_19, 0x000c); + regmap_write(rt5663->regmap, RT5663_DUMMY_1, 0x0324); + regmap_write(rt5663->regmap, RT5663_DIG_MISC, 0x8001); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xa23b); + msleep(30); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0xf23b); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x8000); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x0008); regmap_write(rt5663->regmap, RT5663_PRE_DIV_GATING_1, 0xffff); regmap_write(rt5663->regmap, RT5663_PRE_DIV_GATING_2, 0xffff); + regmap_write(rt5663->regmap, RT5663_CBJ_1, 0x8c10); + regmap_write(rt5663->regmap, RT5663_IL_CMD_2, 0x00c1); + regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_1, 0xb880); + regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_2, 0x4110); + regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_2, 0x4118); + + count = 0; + while (true) { + regmap_read(rt5663->regmap, RT5663_INT_ST_2, &value); + if (!(value & 0x80)) + usleep_range(10000, 10005); + else + break; + + if (count > 200) + break; + } + + regmap_write(rt5663->regmap, RT5663_HP_IMP_SEN_19, 0x0000); regmap_write(rt5663->regmap, RT5663_DEPOP_2, 0x3003); + regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x0038); regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x003b); + regmap_write(rt5663->regmap, RT5663_PWR_DIG_2, 0x8400); + regmap_write(rt5663->regmap, RT5663_PWR_DIG_1, 0x8df8); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x8003); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c); regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32); - regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_2, 0x1371); regmap_write(rt5663->regmap, RT5663_DACREF_LDO, 0x3b0b); - regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x2080); + regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x0000); regmap_write(rt5663->regmap, RT5663_BYPASS_STO_DAC, 0x000c); - regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xabba); + regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xafaa); regmap_write(rt5663->regmap, RT5663_CHARGE_PUMP_1, 0x2224); regmap_write(rt5663->regmap, RT5663_HP_OUT_EN, 0x8088); regmap_write(rt5663->regmap, RT5663_STO_DRE_9, 0x0017); regmap_write(rt5663->regmap, RT5663_STO_DRE_10, 0x0017); regmap_write(rt5663->regmap, RT5663_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5663->regmap, RT5663_CHOP_ADC, 0x3000); regmap_write(rt5663->regmap, RT5663_RECMIX, 0x0005); regmap_write(rt5663->regmap, RT5663_ADDA_RST, 0xc000); regmap_write(rt5663->regmap, RT5663_STO1_HPF_ADJ1, 0x3320); regmap_write(rt5663->regmap, RT5663_HP_CALIB_2, 0x00c9); regmap_write(rt5663->regmap, RT5663_DUMMY_1, 0x004c); - regmap_write(rt5663->regmap, RT5663_ANA_BIAS_CUR_1, 0x7766); - regmap_write(rt5663->regmap, RT5663_BIAS_CUR_8, 0x4702); - msleep(200); + regmap_write(rt5663->regmap, RT5663_ANA_BIAS_CUR_1, 0x1111); + regmap_write(rt5663->regmap, RT5663_BIAS_CUR_8, 0x4402); + regmap_write(rt5663->regmap, RT5663_CHARGE_PUMP_2, 0x3311); regmap_write(rt5663->regmap, RT5663_HP_CALIB_1, 0x0069); - regmap_write(rt5663->regmap, RT5663_HP_CALIB_3, 0x06c2); - regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0x7b00); - regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0xfb00); + regmap_write(rt5663->regmap, RT5663_HP_CALIB_3, 0x06ce); + regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0x6800); + regmap_write(rt5663->regmap, RT5663_CHARGE_PUMP_2, 0x1100); + regmap_write(rt5663->regmap, RT5663_HP_CALIB_7, 0x0057); + regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0xe800); + count = 0; while (true) { regmap_read(rt5663->regmap, RT5663_HP_CALIB_1_1, &value); @@ -3039,6 +3067,36 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) return; count++; } + + regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0x6200); + regmap_write(rt5663->regmap, RT5663_HP_CALIB_7, 0x0059); + regmap_write(rt5663->regmap, RT5663_HP_CALIB_1_1, 0xe200); + + count = 0; + while (true) { + regmap_read(rt5663->regmap, RT5663_HP_CALIB_1_1, &value); + if (value & 0x8000) + usleep_range(10000, 10005); + else + break; + + if (count > 200) + return; + count++; + } + + regmap_write(rt5663->regmap, RT5663_EM_JACK_TYPE_1, 0xb8e0); + usleep_range(10000, 10005); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_1, 0x003b); + usleep_range(10000, 10005); + regmap_write(rt5663->regmap, RT5663_PWR_DIG_1, 0x0000); + usleep_range(10000, 10005); + regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x000b); + usleep_range(10000, 10005); + regmap_write(rt5663->regmap, RT5663_DEPOP_1, 0x0008); + usleep_range(10000, 10005); + regmap_write(rt5663->regmap, RT5663_PWR_ANLG_2, 0x0000); + usleep_range(10000, 10005); } static int rt5663_i2c_probe(struct i2c_client *i2c, -- 2.7.4

2 5

[alsa-devel] [PATCH] ASoC: spear: fix error return code in spdif_in_probe()
by Gustavo A. R. Silva 07 Jul '17

07 Jul '17

platform_get_irq() returns an error code, but the spdif_in driver ignores it and always returns -EINVAL. This is not correct, and prevents -EPROBE_DEFER from being propagated properly. Notice that platform_get_irq() no longer returns 0 on error: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… Print error message and propagate the return value of platform_get_irq on failure. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/spear/spdif_in.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 977a078..7f32527 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -223,8 +223,10 @@ static int spdif_in_probe(struct platform_device *pdev) host->io_base = io_base; host->irq = platform_get_irq(pdev, 0); - if (host->irq < 0) - return -EINVAL; + if (host->irq < 0) { + dev_warn(&pdev->dev, "failed to get IRQ: %d\n", host->irq); + return host->irq; + } host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: kirkwood-i2s: fix error return code in kirkwood_i2s_dev_probe()
by Gustavo A. R. Silva 07 Jul '17

07 Jul '17

platform_get_irq() returns an error code, but the kirkwood-i2s driver ignores it and always returns -ENXIO. This is not correct, and prevents -EPROBE_DEFER from being propagated properly. Notice that platform_get_irq() no longer returns 0 on error. Print and propagate the return value of platform_get_irq on failure. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/kirkwood/kirkwood-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3a36d60..3b09f71 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -550,9 +550,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->io); priv->irq = platform_get_irq(pdev, 0); - if (priv->irq <= 0) { - dev_err(&pdev->dev, "platform_get_irq failed\n"); - return -ENXIO; + if (priv->irq < 0) { + dev_err(&pdev->dev, "platform_get_irq failed: %d\n", priv->irq); + return priv->irq; } if (np) { -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: twl6040: fix error return code in twl6040_probe()
by Gustavo A. R. Silva 07 Jul '17

07 Jul '17

platform_get_irq() returns an error code, but the twl6040 driver ignores it and always returns -EINVAL. This is not correct, and prevents -EPROBE_DEFER from being propagated properly. Notice that platform_get_irq() no longer returns 0 on error. Print and propagate the return value of platform_get_irq on failure. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2b6ad09..96cf94a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1123,8 +1123,8 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->plug_irq = platform_get_irq(pdev, 0); if (priv->plug_irq < 0) { - dev_err(codec->dev, "invalid irq\n"); - return -EINVAL; + dev_err(codec->dev, "invalid irq: %d\n", priv->plug_irq); + return priv->plug_irq; } INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work); -- 2.5.0

2 1

[alsa-devel] [PATCH] ASoC: imx-ssi: add check on platform_get_irq return value
by Gustavo A. R. Silva 07 Jul '17

07 Jul '17

Check return value from call to platform_get_irq(), so in case of failure print error message and propagate the return value. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/soc/fsl/imx-ssi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index b95132e..0679061 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -527,6 +527,10 @@ static int imx_ssi_probe(struct platform_device *pdev) } ssi->irq = platform_get_irq(pdev, 0); + if (ssi->irq < 0) { + dev_err(&pdev->dev, "Failed to get IRQ: %d\n", ssi->irq); + return ssi->irq; + } ssi->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi->clk)) { -- 2.5.0

3 2

[alsa-devel] [PATCH] ALSA: hda/realtek - change the location for one of two front microphones
by Hui Wang 07 Jul '17

07 Jul '17

On this Lenovo machine, there are two front mics, and both of them are assigned the same name "Mic", but pulseaudio can't support two mics with the same name, as a workaround, we change the location for one of them, then the driver will assign "Front Mic" and "Mic" for them. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.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 cebf59e..a6c3ce0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5179,6 +5179,7 @@ enum { ALC233_FIXUP_ASUS_MIC_NO_PRESENCE, ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, ALC233_FIXUP_LENOVO_MULTI_CODECS, + ALC294_FIXUP_LENOVO_MIC_LOCATION, }; static const struct hda_fixup alc269_fixups[] = { @@ -5962,6 +5963,18 @@ enum { .type = HDA_FIXUP_FUNC, .v.func = alc233_alc662_fixup_lenovo_dual_codecs, }, + [ALC294_FIXUP_LENOVO_MIC_LOCATION] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* Change the mic location from front to right, otherwise there are + two front mics with the same name, pulseaudio can't handle them. + This is just a temporary workaround, after applying this fixup, + there will be one "Front Mic" and one "Mic" in this machine. + */ + { 0x1a, 0x04a19040 }, + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6143,6 +6156,7 @@ 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, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 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), -- 1.9.1

2 3

[alsa-devel] [PATCH 2/2] soc: codecs: rt5645: fixed coding style issuses
by Chris Gorman 07 Jul '17

07 Jul '17

fixed coding style issues 6 warnings and 1 error left two warnings alone because I was unsure about changing the logic and felt the 80 + character line necessary sound/soc/codecs/rt5645.c:702: WARNING: else is not generally useful after a break or return sound/soc/codecs/rt5645.c:3024: WARNING: line over 80 characters Signed-off-by: Chris Gorman <chrisjohgorman(a)bell.net> --- sound/soc/codecs/rt5645.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 87844a4..7c14317 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3021,7 +3021,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2, @@ -3117,7 +3117,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); if (!dapm->card->instantiated) { /* Power up necessary bits for JD if dapm is - not ready yet */ + * not ready yet + */ regmap_update_bits(rt5645->regmap, RT5645_PWR_ANLG1, RT5645_PWR_MB | RT5645_PWR_VREF2, RT5645_PWR_MB | RT5645_PWR_VREF2); @@ -3143,9 +3144,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) if (val == 1 || val == 2) { rt5645->jack_type = SND_JACK_HEADSET; - if (rt5645->en_button_func) { + if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, true); - } } else { snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); @@ -3262,10 +3262,11 @@ static void rt5645_jack_detect_work(struct work_struct *work) report = SND_JACK_HEADSET; btn_type = rt5645_button_detect(rt5645->codec); /* rt5650 can report three kinds of button behavior, - one click, double click and hold. However, - currently we will report button pressed/released - event. So all the three button behaviors are - treated as button pressed. */ + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ switch (btn_type) { case 0x8000: case 0x4000: @@ -3527,7 +3528,7 @@ static const struct regmap_config rt5650_regmap = { }; static const struct regmap_config temp_regmap = { - .name="nocache", + .name = "nocache", .reg_bits = 8, .val_bits = 16, .use_single_rw = true, -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: msnd: Optimize / harden DSP and MIDI loops
by Takashi Iwai 06 Jul '17

06 Jul '17

The ISA msnd drivers have loops fetching the ring-buffer head, tail and size values inside the loops. Such codes are inefficient and fragile. This patch optimizes it, and also adds the sanity check to avoid the endless loops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196131 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196133 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/msnd/msnd_midi.c | 30 +++++++++++++++--------------- sound/isa/msnd/msnd_pinnacle.c | 23 ++++++++++++----------- 2 files changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index 912b5a9ccbab..013d8d1170fe 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -120,24 +120,24 @@ void snd_msndmidi_input_read(void *mpuv) unsigned long flags; struct snd_msndmidi *mpu = mpuv; void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + u16 head, tail, size; spin_lock_irqsave(&mpu->input_lock, flags); - while (readw(mpu->dev->MIDQ + JQS_wTail) != - readw(mpu->dev->MIDQ + JQS_wHead)) { - u16 wTmp, val; - val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); - - if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, - &mpu->mode)) - snd_rawmidi_receive(mpu->substream_input, - (unsigned char *)&val, 1); - - wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; - if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) - writew(0, mpu->dev->MIDQ + JQS_wHead); - else - writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + head = readw(mpu->dev->MIDQ + JQS_wHead); + tail = readw(mpu->dev->MIDQ + JQS_wTail); + size = readw(mpu->dev->MIDQ + JQS_wSize); + if (head > size || tail > size) + goto out; + while (head != tail) { + unsigned char val = readw(pwMIDQData + 2 * head); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, &val, 1); + if (++head > size) + head = 0; + writew(head, mpu->dev->MIDQ + JQS_wHead); } + out: spin_unlock_irqrestore(&mpu->input_lock, flags); } EXPORT_SYMBOL(snd_msndmidi_input_read); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index ad4897337df5..fc4fb1904aef 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -170,23 +170,24 @@ static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) { struct snd_msnd *chip = dev_id; void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + u16 head, tail, size; /* Send ack to DSP */ /* inb(chip->io + HP_RXL); */ /* Evaluate queued DSP messages */ - while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { - u16 wTmp; - - snd_msnd_eval_dsp_msg(chip, - readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); - - wTmp = readw(chip->DSPQ + JQS_wHead) + 1; - if (wTmp > readw(chip->DSPQ + JQS_wSize)) - writew(0, chip->DSPQ + JQS_wHead); - else - writew(wTmp, chip->DSPQ + JQS_wHead); + head = readw(chip->DSPQ + JQS_wHead); + tail = readw(chip->DSPQ + JQS_wTail); + size = readw(chip->DSPQ + JQS_wSize); + if (head > size || tail > size) + goto out; + while (head != tail) { + snd_msnd_eval_dsp_msg(chip, readw(pwDSPQData + 2 * head)); + if (++head > size) + head = 0; + writew(head, chip->DSPQ + JQS_wHead); } + out: /* Send ack to DSP */ inb(chip->io + HP_RXL); return IRQ_HANDLED; -- 2.13.2

1 0

[alsa-devel] A problem about two microphones
by Hui Wang 06 Jul '17

06 Jul '17

Recently we met a problem, on a lenovo machine, it has two microphones(one is microphone, the other is headset-mic), and both of them are located in the front panel. When the driver assigns names for them (via the function hda_get_autocfg_input_label()), they are assigned the same name "Mic" since they have the same location. So there are two "Mic", but the pulseaudio can't handle two "Mic", in the pulseaudio, it can only handle multiple inputs/microphones with different names, like "Mic" and "Front Mic". If two microphones have the same name, we can only see one of them through pulseaudio. I don't know how to solve this problem, the simplest way to workaround it I can figure out is to change one of the microphone's location, let them have differnt location, then they have differnet names. Pin Default 0x02a19040: [Jack] Mic at Ext Front Changed to: Pin Default 0x04a19040: [Jack] Mic at Ext Right Do you have any ideas to solve this problem? The alsa-info.txt for this machine is at: http://paste.ubuntu.com/24967758/ Thanks, Hui.

2 2

[alsa-devel] [PATCH 00/18] treewide: Move storage class before return type
by Joe Perches 06 Jul '17

06 Jul '17

Move the inline/asmlinkage keywords before the return types Add a checkpatch test for this too. Joe Perches (18): checkpatch: improve the STORAGE_CLASS test ARM: KVM: Move asmlinkage before type ARM: HP Jornada 7XX: Move inline before return type CRIS: gpio: Move inline before return type FRV: tlbflush: Move asmlinkage before return type ia64: Move inline before return type ia64: sn: pci: Move inline before type m68k: coldfire: Move inline before return type MIPS: SMP: Move asmlinkage before return type sh: Move inline before return type x86/efi: Move asmlinkage before return type drivers: s390: Move static and inline before return type drivers: tty: serial: Move inline before return type USB: serial: safe_serial: Move __inline__ before return type video: fbdev: intelfb: Move inline before return type video: fbdev: omap: Move inline before return type ARM: samsung: usb-ohci: Move inline before return type ALSA: opl4: Move inline before return type arch/arm/include/asm/kvm_hyp.h | 8 ++++---- arch/arm/mach-sa1100/jornada720_ssp.c | 2 +- arch/cris/arch-v10/drivers/gpio.c | 4 ++-- arch/frv/include/asm/tlbflush.h | 8 ++++---- arch/ia64/kernel/mca.c | 2 +- arch/ia64/sn/pci/pcibr/pcibr_ate.c | 2 +- arch/ia64/sn/pci/tioce_provider.c | 4 ++-- arch/m68k/coldfire/intc-simr.c | 4 ++-- arch/mips/include/asm/smp.h | 2 +- arch/sh/mm/cache-sh5.c | 2 +- arch/x86/include/asm/efi.h | 4 ++-- drivers/s390/net/ctcm_main.c | 2 +- drivers/s390/net/qeth_l3_main.c | 2 +- drivers/tty/serial/ioc3_serial.c | 4 ++-- drivers/tty/serial/ioc4_serial.c | 4 ++-- drivers/usb/serial/safe_serial.c | 2 +- drivers/video/fbdev/intelfb/intelfbdrv.c | 2 +- drivers/video/fbdev/omap/lcdc.c | 6 +++--- include/linux/platform_data/usb-ohci-s3c2410.h | 2 +- scripts/checkpatch.pl | 12 ++++++++++-- sound/drivers/opl4/opl4_lib.c | 2 +- 21 files changed, 44 insertions(+), 36 deletions(-) -- 2.10.0.rc2.1.g053435c

3 3

[alsa-devel] [PATCH 0/3] mfd: move include files out of include/linux/i2c
by Wolfram Sang 06 Jul '17

06 Jul '17

It doesn't make sense to use include/linux/i2c for client drivers which may in fact rather be hwmon or input or whatever devices. As a result, I want to deprecate include/linux/i2c for good. This series moves the include files to a better location, for the mfd subsystem to include/linux/mfd. Because of twl.h, this touches quite some subsystems, but I still think it is worth it. I prefer the series to go upstream via the subsystem tree; if you prefer that I take it via I2C, just let me know. No runtime testing because of no HW, but buildbot is happy with this series at least. A branch can be found here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/platform_data Thanks and kind regards, Wolfram Wolfram Sang (3): mfd: dm355evm_msp: move header file out of I2C realm mfd: tps65010: move header file out of I2C realm mfd: twl: move header file out of I2C realm arch/arm/mach-omap1/board-h2-mmc.c | 2 +- arch/arm/mach-omap1/board-h2.c | 2 +- arch/arm/mach-omap1/board-h3-mmc.c | 2 +- arch/arm/mach-omap1/board-h3.c | 2 +- arch/arm/mach-omap1/board-osk.c | 2 +- arch/arm/mach-omap2/common.h | 2 +- arch/arm/mach-omap2/omap_twl.c | 2 +- arch/arm/mach-s3c24xx/mach-osiris-dvs.c | 2 +- arch/arm/mach-s3c24xx/mach-osiris.c | 2 +- drivers/gpio/gpio-twl4030.c | 2 +- drivers/iio/adc/twl4030-madc.c | 2 +- drivers/iio/adc/twl6030-gpadc.c | 2 +- drivers/input/keyboard/twl4030_keypad.c | 2 +- drivers/input/misc/dm355evm_keys.c | 2 +- drivers/input/misc/twl4030-pwrbutton.c | 2 +- drivers/input/misc/twl4030-vibra.c | 2 +- drivers/mfd/dm355evm_msp.c | 2 +- drivers/mfd/tps65010.c | 2 +- drivers/mfd/twl-core.c | 6 +++--- drivers/mfd/twl4030-audio.c | 2 +- drivers/mfd/twl4030-irq.c | 2 +- drivers/mfd/twl4030-power.c | 2 +- drivers/mfd/twl6030-irq.c | 2 +- drivers/phy/phy-twl4030-usb.c | 2 +- drivers/power/supply/twl4030_charger.c | 2 +- drivers/pwm/pwm-twl-led.c | 2 +- drivers/pwm/pwm-twl.c | 2 +- drivers/regulator/twl-regulator.c | 2 +- drivers/regulator/twl6030-regulator.c | 2 +- drivers/rtc/rtc-dm355evm.c | 2 +- drivers/rtc/rtc-twl.c | 2 +- drivers/usb/host/ohci-omap.c | 2 +- drivers/usb/phy/phy-isp1301-omap.c | 2 +- drivers/usb/phy/phy-twl6030-usb.c | 2 +- drivers/video/backlight/pandora_bl.c | 2 +- drivers/video/fbdev/omap/lcd_h3.c | 2 +- drivers/watchdog/twl4030_wdt.c | 2 +- include/linux/{i2c => mfd}/dm355evm_msp.h | 0 include/linux/{i2c => mfd}/tps65010.h | 2 +- include/linux/{i2c => mfd}/twl.h | 0 sound/soc/codecs/twl4030.c | 2 +- 41 files changed, 41 insertions(+), 41 deletions(-) rename include/linux/{i2c => mfd}/dm355evm_msp.h (100%) rename include/linux/{i2c => mfd}/tps65010.h (99%) rename include/linux/{i2c => mfd}/twl.h (100%) -- 2.11.0

11 15

[alsa-devel] [PATCH 0/3] ASoC: Add I2S support for Allwinner H3 SoCs
by codekipper＠gmail.com 06 Jul '17

06 Jul '17

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, please find attached a series of patches to bring i2s support to the Allwinner H3 SoC. This has been tested with the following setups: A20 Olimex EVB connected to a pcm5102 Orange Pi 2 connected to a uda1380 Orange Pi 2 hdmi audio playback Pine 64 connected to the audio DAC board To get i2s working some additional patches are required which will be delivered later. For now they have been pushed here https://github.com/codekipper/linux-sunxi/commits/sunxi-audio-h3 Thanks in advance, CK Marcus Cooper (3): ASoC: sun4i-i2s: Add more quirks for newer SoCs ASoC: sun4i-i2s: Get startup to call set_fmt ASoC: sun4i-i2s: Add support for H3 .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + sound/soc/sunxi/sun4i-i2s.c | 391 ++++++++++++++++++++- 2 files changed, 376 insertions(+), 17 deletions(-) -- 2.13.2

4 7

[alsa-devel] [PATCH] ASoC: hdmi-codec: hdmi_codec_priv includes snd_kcontrol_new
by Kuninori Morimoto 06 Jul '17

06 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current hdmi-codec driver is using hdmi_controls for "ELD" control. But, hdmi-codec driver might be used from many HDMIs. In such case, they will use same "ELD" name and kernel will indicate below error. xxx: control x:x:x:ELD:x is already present hdmi_controls will be registered in soc_probe_component(), and we can replace it by component driver probe function. This patch registers current hdmi_controls in new hdmi_probe(). If hdmi-codec is used from only 1 device, it will use "ELD" as control name (We can keep compatibility). If hdmi-codec is used from many devices, it will use "ELD.x" as control name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/hdmi-codec.c | 60 +++++++++++++++++++++++++++++++++---------- 1 file changed, 46 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 22ed0dc..5835a90 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -276,6 +276,7 @@ struct hdmi_codec_cea_spk_alloc { .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, }; +#define ELD_NAME_SIZE 8 struct hdmi_codec_priv { struct hdmi_codec_pdata hcd; struct snd_soc_dai_driver *daidrv; @@ -284,7 +285,10 @@ struct hdmi_codec_priv { struct snd_pcm_substream *current_stream; uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; + struct snd_kcontrol_new control; unsigned int chmap_idx; + int id; + char eld_name[ELD_NAME_SIZE]; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { @@ -399,18 +403,6 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, return 0; } - -static const struct snd_kcontrol_new hdmi_controls[] = { - { - .access = SNDRV_CTL_ELEM_ACCESS_READ | - SNDRV_CTL_ELEM_ACCESS_VOLATILE, - .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "ELD", - .info = hdmi_eld_ctl_info, - .get = hdmi_eld_ctl_get, - }, -}; - static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -718,6 +710,45 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, .pcm_new = hdmi_codec_pcm_new, }; +static int hdmi_last_id = 0; +static int hdmi_probe(struct snd_soc_component *component) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + static struct snd_soc_component *component_1st = NULL; + char *name = "ELD"; + + /* + * It will use "ELD" if hdmi_codec was used from only 1 device. + * It will use "ELD.x" if hdmi_codec was used from many devices. + * Then, first "ELD" will be replaced to "ELD.0" + */ + snprintf(hcp->eld_name, ELD_NAME_SIZE, "ELD.%d", hcp->id); + + if (hcp->id == 0) + component_1st = component; + + if (hdmi_last_id > 1) { + name = hcp->eld_name; + + if (component_1st) { + struct hdmi_codec_priv *hcp_1st; + + /* replaced 1st "ELD" to "ELD.0" */ + hcp_1st = snd_soc_component_get_drvdata(component_1st); + hcp_1st->control.name = hcp_1st->eld_name; + } + } + + hcp->control.access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE; + hcp->control.iface = SNDRV_CTL_ELEM_IFACE_PCM; + hcp->control.name = name; + hcp->control.info = hdmi_eld_ctl_info; + hcp->control.get = hdmi_eld_ctl_get; + + return snd_soc_add_component_controls(component, &hcp->control, 1); +} + static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, struct device_node *endpoint) { @@ -732,8 +763,7 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static struct snd_soc_codec_driver hdmi_codec = { .component_driver = { - .controls = hdmi_controls, - .num_controls = ARRAY_SIZE(hdmi_controls), + .probe = hdmi_probe, .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, @@ -768,6 +798,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) if (!hcp) return -ENOMEM; + hcp->id = hdmi_last_id; hcp->hcd = *hcd; mutex_init(&hcp->current_stream_lock); @@ -795,6 +826,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) } dev_set_drvdata(dev, hcp); + hdmi_last_id++; return 0; } -- 1.9.1

3 4

[alsa-devel] [RFC PATCH] aplay: Add samples argument for playing/recording a given number of samples
by Ion-Horia Petrisor 05 Jul '17

05 Jul '17

-s --samples allows aplay to be used for playback/capture a given numbeer of samplese by channel Signed-off-by: Ion-Horia Petrisor <ion-horia.petrisor(a)nxp.com> --- This is just an RFC, we need to clarify the aplay raw format bug from: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-July/122509.html aplay/aplay.c | 34 +++++++++++++++++++++++++++++----- 1 file changed, 29 insertions(+), 5 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 00af662..6ee0866 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -100,6 +100,7 @@ static struct { unsigned int rate; } hwparams, rhwparams; static int timelimit = 0; +static int sampleslimit = 0; static int quiet_mode = 0; static int file_type = FORMAT_DEFAULT; static int open_mode = 0; @@ -212,6 +213,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" "-f, --format=FORMAT sample format (case insensitive)\n" "-r, --rate=# sample rate\n" "-d, --duration=# interrupt after # seconds\n" +"-s, --samples=# interrupt after # samples per channel\n" "-M, --mmap mmap stream\n" "-N, --nonblock nonblocking mode\n" "-F, --period-time=# distance between interrupts is # microseconds\n" @@ -466,8 +468,9 @@ static long parse_long(const char *str, int *err) int main(int argc, char *argv[]) { + int duration_or_sample = 0; int option_index; - static const char short_options[] = "hnlLD:qt:c:f:r:d:MNF:A:R:T:B:vV:IPCi" + static const char short_options[] = "hnlLD:qt:c:f:r:d:s:MNF:A:R:T:B:vV:IPCi" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -485,6 +488,7 @@ int main(int argc, char *argv[]) {"format", 1, 0, 'f'}, {"rate", 1, 0, 'r'}, {"duration", 1, 0 ,'d'}, + {"samples", 1, 0, 's'}, {"mmap", 0, 0, 'M'}, {"nonblock", 0, 0, 'N'}, {"period-time", 1, 0, 'F'}, @@ -640,11 +644,28 @@ int main(int argc, char *argv[]) } break; case 'd': + if (duration_or_sample) { + error(_("duration argument '%s' and samples argument cannot be used together"), optarg); + return 1; + } timelimit = parse_long(optarg, &err); if (err < 0) { error(_("invalid duration argument '%s'"), optarg); return 1; } + duration_or_sample = 1; + break; + case 's': + if (duration_or_sample) { + error(_("samples argument '%s' and duration argument cannot be used together"), optarg); + return 1; + } + sampleslimit = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid samples argument '%s'"), optarg); + return 1; + } + duration_or_sample = 1; break; case 'N': nonblock = 1; @@ -2477,9 +2498,12 @@ static off64_t calc_count(void) { off64_t count; - if (timelimit == 0) { - count = pbrec_count; - } else { + if (timelimit == 0) + if (sampleslimit == 0) + count = pbrec_count; + else + count = snd_pcm_format_size(hwparams.format, sampleslimit * hwparams.channels); + else { count = snd_pcm_format_size(hwparams.format, hwparams.rate * hwparams.channels); count *= (off64_t)timelimit; } @@ -3125,7 +3149,7 @@ static void capture(char *orig_name) /* repeat the loop when format is raw without timelimit or * requested counts of data are recorded */ - } while ((file_type == FORMAT_RAW && !timelimit) || count > 0); + } while (((file_type == FORMAT_RAW && !timelimit) && !sampleslimit) || count > 0); } static void playbackv_go(int* fds, unsigned int channels, size_t loaded, off64_t count, int rtype, char **names) -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: samsung: i2s: Supported high resolution rates
by Jaechul Lee 05 Jul '17

05 Jul '17

This driver can support more frequencies over 96KHz. There are no reasons to limit the frequency range below 96KHz. If codecs/amps or something else can't support high resolution rates, the constraints would be set rates properly because each drivers have its own limits. Signed-off-by: Jaechul Lee <jcsing.lee(a)samsung.com> --- sound/soc/samsung/i2s.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index af3ba4d4ccc5..fc34af6a2c1e 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -50,6 +50,7 @@ struct samsung_i2s_variant_regs { struct samsung_i2s_dai_data { u32 quirks; + unsigned int rates; const struct samsung_i2s_variant_regs *i2s_variant_regs; }; @@ -1076,20 +1077,25 @@ static const struct snd_soc_component_driver samsung_i2s_component = { .name = "samsung-i2s", }; -#define SAMSUNG_I2S_RATES SNDRV_PCM_RATE_8000_96000 - #define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE) static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { + const struct samsung_i2s_dai_data *i2s_dai_data; struct i2s_dai *i2s; i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) return NULL; + if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) + i2s_dai_data = of_device_get_match_data(&pdev->dev); + else + i2s_dai_data = (struct samsung_i2s_dai_data *) + platform_get_device_id(pdev)->driver_data; + i2s->pdev = pdev; i2s->pri_dai = NULL; i2s->sec_dai = NULL; @@ -1101,13 +1107,13 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.resume = i2s_resume; i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; - i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; + i2s->i2s_dai_drv.playback.rates = i2s_dai_data->rates; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; if (!sec) { i2s->i2s_dai_drv.capture.channels_min = 1; i2s->i2s_dai_drv.capture.channels_max = 2; - i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; + i2s->i2s_dai_drv.capture.rates = i2s_dai_data->rates; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; } return i2s; @@ -1452,29 +1458,34 @@ static const struct samsung_i2s_variant_regs i2sv5_i2s1_regs = { static const struct samsung_i2s_dai_data i2sv3_dai_type = { .quirks = QUIRK_NO_MUXPSR, + .rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_IDMA, + .rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv6_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA, + .rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv6_regs, }; static const struct samsung_i2s_dai_data i2sv7_dai_type = { .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM, + .rates = SNDRV_PCM_RATE_8000_192000, .i2s_variant_regs = &i2sv7_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type_i2s1 = { .quirks = QUIRK_PRI_6CHAN | QUIRK_NEED_RSTCLR, + .rates = SNDRV_PCM_RATE_8000_96000, .i2s_variant_regs = &i2sv5_i2s1_regs, }; -- 2.11.0

4 4

[alsa-devel] ioctl(fd, SNDRV_CTL_IOCTL_ELEM_REPLACE, ... not working from userspace ?
by Fulup Ar Foll 04 Jul '17

04 Jul '17

Hi, I try to rename existing sound control from userspace. The goal is to create a generic abstraction layer for AutomotiveGradeLinux audio server While I found two hidden methods to access to low level functionalities from user space, non of them seems to accecp replace call. I isolated the code in two basic samples https://github.com/fulup-bzh/alsa-ctl-replace - First one uses a directt IOTCL model - Second one leverage private sndctl control handle. Any idea would be more than welcome. Fulup

3 4

[alsa-devel] [PATCH] ALSA: hda - Fix unbalance of i915 module refcount
by Takashi Iwai 04 Jul '17

04 Jul '17

The commit dba9b7b6ca1a ("ALSA: hda - Fix doubly initialization of i915 component") contained a typo that leads to the unbalance of i915 module reference. The value to be checked is not chip->driver_type but chip->driver_caps. Fixes: dba9b7b6ca1a ("ALSA: hda - Fix doubly initialization of i915 component") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196219 Reported-by: Martin Peres <martin.peres(a)free.fr> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 03e34edc8f24..5ae8ddab6412 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1385,7 +1385,7 @@ static int azx_free(struct azx *chip) if (hda->need_i915_power) snd_hdac_display_power(bus, false); } - if (chip->driver_type & AZX_DCAPS_I915_COMPONENT) + if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); kfree(hda); -- 2.13.2

1 0

[alsa-devel] [GIT PULL] ASoC updates for v4.13
by Mark Brown 04 Jul '17

04 Jul '17

The following changes since commit 6f7da290413ba713f0cdd9ff1a2a9bb129ef4f6c: Linux 4.12 (2017-07-02 16:07:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.13 for you to fetch changes up to b821d298f8864dc3c3d42643812df43289159507: Merge remote-tracking branch 'asoc/topic/inte' into asoc-next (2017-07-03 16:51:38 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.13 The big news with this release is the of-graph card, this provides a replacement for simple-card that is much more flexibile and scalable, allowing many more systems to use a generic sound card than was possible before: - The of-graph card, finally merged after a long and dedicated effort by Morimoto-san. - New widget types intended mainly for use with DSPs. - New drivers for Allwinner V3s SoCs, Ensonic ES8316, several classes of x86 machine, Rockchip PDM controllers, STM32 I2S and S/PDIF controllers and ZTE AUD96P22 CODECs. ---------------------------------------------------------------- Adam Thomson (4): ASoC: da7213: Update driver to use device_property* FW functions ASoC: da7218: Fix incorrect usage of bitwise '&' operator for SRM check ASoC: da7219: Fix HP detection procedure for all MCLK frequencies ASoC: Intel: bxt: Move codec sysclk config to codec_init function Andy Shevchenko (5): ASoC: omap-mcbsp: Use sysfs_match_string() helper ASoC: Intel: byt-max98090: Add GPIO ACPI mapping table ASoC: rt5677: Introduce proper table for ACPI enumeration ASoC: rt5677: Move platform code to board file ASoC: Intel: bdw-rt5677: Switch to devm_acpi_dev_add_driver_gpios() Arnd Bergmann (3): ASoC: rt5514: fix gcc-7 warning ASoC: rt5665: fix gcc-7 warning ASoC: es8316: add I2C dependency Arvind Yadav (1): ASoC: rsnd: constify dev_pm_ops structures. Axel Lin (1): ASoC: ak4613: Fix out of bounds array access for ak4613_iface Baoyou Xie (2): ASoC: add bindings for ZTE zx-aud96p22 audio codec ASoC: zx_aud96p22: add ZTE ZX AUD96P22 codec driver Bard Liao (11): ASoC: rt5665: add ADC STO2 ASRC support ASoC: rt5645: read jd1_1 status for jd detection ASoC: rt5645: rename jd_invert flag in platform data ASoC: rt5645: add inv_jd1_1 flag ASoC: rt5670: fix wrong audio route ASoC: rt5670: remove duplicate route. ASoC: rt5670: move set_sysclk to codec level ASoC: rt5651: remove unexisting Muxes ASoC: rt5645: enable speaker protection features ASoC: rt5670: fix incompatible pointer type of set_sysclk ASoC: rt5665: calibration should be done before jack detection Brian Austin (2): ASoC: cs35l35: Add Boost Inductor Calculation ASoC: cs35l35: Add DT handling for Inductor Charles Keepax (3): ASoC: cs35l35: Correctly handle 0 for bst_ipk ASoC: cs4271: Remove unnecessary additional variable definition ASoC: cs35l35: Add additional delay for reset Christophe Jaillet (1): ASoC: rockchip: Fix an error handling in 'rockchip_i2s_probe' Colin Ian King (6): ASoC: Intel: sst: fix spelling mistake: "allocationf" -> "allocation" ASoC: hdmi-codec: fix spelling mistake: "deteced" -> "detected" ASoC: audio-graph-card: fix spelling mistake: "missmatch" -> "mismatch" ASoC: max9867: make array ni_div static const ASoC: ak4642: make arrays fs_list and ps_list static const ASoC: rsnd: make arrays path and cmd_case static const Dan Carpenter (1): ASoC: stm32: sai: remove some stray tabs Daniel Drake (2): ASoC: add es8316 codec driver ASoC: Intel: add machine driver for BYT/CHT + ES8316 Dharageswari R (3): ASoC: Intel: Skylake: Add mic-select module type ASoC: Intel: Skylake: Add enum control for mic selection ASoC: Intel: Boards: Add 4-channel DMIC fixup. Douglas Anderson (1): ASoC: Intel: sst: Delete sst_shim_regs64; saved regs are never used Geert Uytterhoeven (3): ASoC: rt5665: Fix uninitialized warning in rt5665_i2s_pin_event() ASoC: fsi: Move inline fsi_stream_is_play() before use ASoC: rsnd: Document optional reset properties Guneshwor Singh (1): ASoC: Intel: Skylake: Add sram address to sst_addr structure Hans de Goede (1): ASoC: rt5645: del btn_check_timer on remove Harsha Priya N (3): ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927 ASoC: Intel: Convert all sst_codecs data definition to c99 style ASoC: Intel: Add Kabylake RT5663+RT5514+MAX98927 machine driver entry Icenowy Zheng (4): ASoC: sun8i-codec-analog: split out mbias ASoC: sun8i-codec-analog: prepare a mixer control/widget/route set for V3s ASoC: sun8i-codec-analog: add support for V3s SoC ASoC: sun4i-codec: Add support for V3s codec John Hsu (6): ASoC: nau8824: TDM support ASoC: nau8825: default value for property ASoC: nau8825: fix jack type detection issue after resume ASoC: nau8825: make crosstalk function optional ASoC: nau8825: debug message of crosstalk bypass ASoC: nau8825: change crosstalk-bypass property to bool type John Stultz (1): drm: adv7511_audio: Add .get_dai_id callback to map port number to dai id Julian Scheel (1): ASoC: simple-card: Fix misleading error message Kuninori Morimoto (83): ASoC: ak4613: disable asymmetric audio interface format of-graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support ASoC: rsnd: move "renesas,rsrc-card" to "simple-scu-audio-card" on Document ASoC: rsnd: add missing clocks/clock-names on Document ASoC: rsnd: remove un-used OUF_SRCO/I macro ASoC: rsnd: remove mismatch explain comment from src.c ASoC: rsnd: clkout-lr-asynchronous is AUDIO_CLKOUT related property ASoC: rsnd: remove error pointer for adg->clkout[i] ASoC: rsnd: need to call nolock_stop if nolock_start was failed ASoC: rsnd: check src mod pointer for rsnd_mod_id() ASoC: simple-card: tidyup return method from probe() ASoC: simple-scu-card: tidyup return method from probe() ASoC: audio-graph-card: tidyup return method from probe() ASoC: simple-scu-card: cleanup documentation ASoC: rsnd: add support graph base DT phase 1 ASoC: rsnd: add support graph base DT phase 2 ASoC: rsnd: add HDMI output support ASoC: add snd_soc_get_dai_id() function ASoC: simple-card-utils: support snd_soc_get_dai_id() ASoC: hdmi-codec: remove multi detection support ASoC: hdmi-codec: add .get_dai_id support ASoC: simple-card-utils: remove strict limitation of bit/frame master ASoC: add audio-graph-scu-card document ASoC: add audio-graph-scu-card support ASoC: audio-graph-scu-card: tidyup return method from probe() ASoC: rsnd: remove unused rsnd_ssi_non_ops ASoC: rsnd: remove unused rsnd_dai_path_info ASoC: simple-scu-card: tidyup "Sampling Rate Conversion" ASoC: audio-graph-scu-card: add missing MIX binding example ASoC: simple-card-utils: share same dev_dbg() for Card Name ASoC: simple-card-utils: share same dev_dbg() for DAI format ASoC: simple-card-utils: share same dev_dbg() for Dai Name ASoC: simple-card-utils: share same dev_dbg() for sysclk ASoC: hdmi-codec: remove unused ratec ASoC: simple-card: remove duplicate parameter from asoc_simple_card_parse_of() ASoC: simple-scu-card: remove duplicate paramera from asoc_simple_card_parse_of ASoC: rsnd: update pointer more accurate ASoC: rsnd: cleanup pointer related code ASoC: rsnd: don't call free_irq() on Parent SSI ASoC: rsnd: add detail explanation of L/R conversion timing ASoC: rsnd: control kctrl items acceptance anytime/runtime ASoC: simple_card_utils: add asoc_simple_card_clk_xxx() ASoC: simple-card: use asoc_simple_card_clk_xxx() ASoC: simple-scu-card: use asoc_simple_card_clk_xxx() ASoC: audio-graph-scu-card: use asoc_simple_card_clk_xxx() ASoC: audio-graph-card: use asoc_simple_card_clk_xxx() ASoC: simple_card_utils: add EXPORT_SYMBOL_GPL() for asoc_simple_card_clk_xxx() ASoC: rsnd: reduce confusable macro parameter magic ASoC: ak4613: add missing 64000 in ak4613_dai_hw_params() ASoC: simple-card-utils: add asoc_simple_card_of_parse_tdm() ASoC: simple-card: use asoc_simple_card_of_parse_tdm() ASoC: simple-scu-card: use asoc_simple_card_of_parse_tdm() ASoC: audio-graph-card: use asoc_simple_card_of_parse_tdm() ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_tdm() ASoC: simple-card-utils: add asoc_simple_card_convert_fixup() ASoC: simple-scu-card: use asoc_simple_card_convert_fixup() ASoC: audio-graph-scu-card: use asoc_simple_card_convert_fixup() ASoC: simple-card-utils: add asoc_simple_card_of_parse_routing() ASoC: simple-card: use asoc_simple_card_of_parse_routing() ASoC: simple-scu-card: use asoc_simple_card_of_parse_routing() ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_routing() ASoC: rsnd: rename "slots" related variable ASoC: rsnd: add rsnd_adg_clk_query() ASoC: rsnd: add rsnd_ssi_clk_query() ASoC: rsnd: add rsnd_soc_hw_rule/constraint() ASoC: ak4613: add hw_constraint rule for Sampling Rate ASoC: simple-card-utils: add asoc_simple_card_of_parse_widgets() ASoC: simple-card: use asoc_simple_card_of_parse_widgets() ASoC: fsl: mpc5200_dma: remove unused psc_dma ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC ASoC: rsnd: fixup unsigned expression compared with zero: main_rate ASoC: audio-graph-card: tidyup asoc_simple_card_canonicalize_cpu() parameter ASoC: audio-graph-scu-card: tidyup asoc_simple_card_canonicalize_cpu() parameter ASoC: audio-graph-scu-card: support 2nd codec endpoint on DT drm: dw-hdmi-i2s: add .get_dai_id callback for ALSA SoC Liam Girdwood (8): ASoC: topology: Allow bespoke configuration post widget creation ASoC: topology: rephrase deferred binding warning. ASoC: topology: Dont free template strings whilst they are in use. ASoC: topology: Fix potential build issues with undeclared structs ASoC: topology: Allow bespoke configuration post widget creation ASoC: topology: Fix usage of SND_SOC_TPLG_INDEX_ALL during load ASoC: topology: show index in debug when adding DAPM routes ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs. Mark Brown (37): Merge branch 'topic/dai-id' of git://git.kernel.org/.../broonie/sound into asoc-of-graph Merge branch 'topic/of-graph' of git://git.kernel.org/.../broonie/sound into asoc-hdmi Merge branch 'topic/of-graph' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'fix/rcar' of git://git.kernel.org/.../broonie/sound into asoc-rcar ASoC: Back out post commit widget creation changes Merge branch 'topic/simple' of git://git.kernel.org/.../broonie/sound into asoc-of-graph Merge branch 'topic/rcar' of git://git.kernel.org/.../broonie/sound into asoc-of-graph ASoC: rockchip: Fix build Merge branch 'topic/rt5677' of git://git.kernel.org/.../broonie/sound into asoc-intel 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/ak4613', 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/cs35l35' and 'asoc/fix/da7219' into asoc-linus Merge remote-tracking branches 'asoc/fix/rockchip', 'asoc/fix/rt5645', 'asoc/fix/rt5663', 'asoc/fix/rt5670' and 'asoc/fix/zte' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/atmel', 'asoc/topic/cs35l34' and 'asoc/topic/cs35l35' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4271', 'asoc/topic/cs53l30', 'asoc/topic/da7213' and 'asoc/topic/da7218' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/drm', 'asoc/topic/dwc' and 'asoc/topic/es8316' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/hdmi', 'asoc/topic/hisi' and 'asoc/topic/max9867' into asoc-next Merge remote-tracking branches 'asoc/topic/msm8916', 'asoc/topic/mtk', 'asoc/topic/nau8824', 'asoc/topic/nau8825' and 'asoc/topic/of-graph' into asoc-next Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/rockchip' and 'asoc/topic/rt5514' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5614', 'asoc/topic/rt5645', 'asoc/topic/rt5651', 'asoc/topic/rt5665' and 'asoc/topic/rt5670' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000' and 'asoc/topic/sh' into asoc-next Merge remote-tracking branches 'asoc/topic/stm32', 'asoc/topic/sunxi', 'asoc/topic/tlv320dac31xx', 'asoc/topic/topology' and 'asoc/topic/wm-adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/zx-aud96p22' into asoc-next 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/ak4613', 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/cs35l35' and 'asoc/fix/da7219' into asoc-linus Merge remote-tracking branches 'asoc/fix/rockchip', 'asoc/fix/rt5645', 'asoc/fix/rt5663', 'asoc/fix/rt5670' and 'asoc/fix/zte' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/atmel', 'asoc/topic/cs35l34' and 'asoc/topic/cs35l35' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4271', 'asoc/topic/cs53l30', 'asoc/topic/da7213' and 'asoc/topic/da7218' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/drm', 'asoc/topic/dwc' and 'asoc/topic/es8316' into asoc-next Merge remote-tracking branch 'asoc/topic/inte' into asoc-next Matthias Kaehlcke (2): ASoC: cht_bsw_max98090_ti: Remove unused function cht_get_codec_dai() ASoC: Intel: sst: Remove unused function sst_restore_shim64() Mengdong Lin (1): ASoC: Drop invalid DMI fields when setting card long name from DMI info Naveen M (7): ASoC: Intel: Convert atom machine data to C99 style ASoC: Intel: Convert skl machine data to C99 style ASoC: Intel: Create a helper to search for matching machine ASoC: Move quirk to identify correct machine driver ASoC: Improve machine driver selection based on quirk data ASoC: Intel: Add Kabylake Realtek Maxim machine driver ASoC: Intel: Add Kabylake RT5663+MAX98927 machine driver entry Oder Chiou (4): ASoC: rt5514: Add more width and channels support in the TDM mode ASoC: rt5514: Add ACPI match ID ASoC: rt5663: Fix the IRQ issue ASoC: rt5663: Check the JD status in the resume function Peter Rosin (1): ASoC: atmel: tse850: fix off-by-one in the "ANA" enumeration count Peter Ujfalusi (1): ASoC: davinci-mcasp: Support for one channel (mono) audio Pierre-Louis Bossart (2): ASoC: Intel: cht_bsw_rt5672: 19.2MHz clock for Baytrail platforms ASoC: codecs: rt5670: fix jd mode for Lenovo Miix 2 10 Ramesh Babu (2): ASoC: Intel: Skylake: Add deep buffer support ASoC: Intel: Skylake: Fix dma buffer size calculation Richard Fitzgerald (3): ASoC: wm_adsp: Fix type warning in sprintf ASoC: wm_adsp: Remove unused member of struct wm_coeff_ctl_ops ASoC: wm_adsp: Fix typo in algorithm list warning message Richard Leitner (1): ASoC: sgtl5000: add avc support Ryo Kodama (1): ASoC: ak4613: Improve counting DAI number Satish Babu Patakokila (1): ASoC: compress: Derive substream from stream based on direction Shawn Guo (4): ASoC: zx-i2s: fix harsh noise with 16-bit audio ASoC: zx-i2s: flip I2S master/slave mode ASoC: audio-graph-card: update bindings for amplifier support ASoC: audio-graph-card: add widgets and routing for external amplifier support Shreyas NC (1): ASoC: Intel: Skylake: Support for multiple data blocks Subhransu S. Prusty (2): ASoC: Intel: Skylake: Fix typo for token d0i3 caps ASoC: skl_rt286: Add deepbuffer dai link Sugar Zhang (3): ASoC: rockchip: add support for pdm controller ASoC: rockchip: add bindings for spdif controller ASoC: rockchip: i2s: add a delay before i2s clear Sylwester Nawrocki (1): ASoC: samsung: Odroid DT binding documentation corrections Takashi Iwai (11): ASoC: atmel: Use IS_ENABLED() ASoC: intel: byt: Constify hw_constraints ASoC: intel: skl: Constify hw_constraints ASoC: cs35l34: Constify hw_constraints ASoC: cs53l30: Constify hw_constraints ASoC: mediatek: Constify hw_constraints ASoC: samsung: Constify hw_constraints ASoC: intel: bxt: Constify hw_constraints ASoC: intel: Use kvzalloc() for suspend buffers ASoC: rt5645: Add jack detection workaround for GPD Win ASoC: rt5645: Add quirk override by module option Takashi Sakamoto (3): ASoC: codecs: msm8916: fix invalid cast to bool type ASoC: hisilicon: localize functions without external linkage ASoC: intel: atom: localize variable without external linkage Tomas Vilda (1): ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec Vijendar Mukunda (1): ASoC: dwc: Added a quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to dwc driver Vinod Koul (4): ASoC: Intel: Skylake: Add debugfs support ASoC: Intel: Skylake: Debugfs facility to dump module config ASoC: Intel: Skylake: explicitly add the headers sst-dsp.h ASoC: Intel: Skylake: Remove driver debugfs exit Vunny Sodhi (1): ASoC: Intel: Skylake: Add support to read firmware registers Wei Yongjun (1): ASoC: rt5665: make local symbol rt5665_i2c_driver static Wu Fengguang (1): ASoC: fix semicolon.cocci warnings olivier moysan (14): ASoC: stm32: Document STM32 I2S bindings ASoC: stm32: Add I2S driver ASoC: stm32: Add full duplex support to i2s ASoC: stm32: sai: fix DT example ASoC: stm32: sai: typo fixes ASoC: stm32: sai: remove spurious trace ASoC: stm32: sai: change stop sequence ASoC: stm32: sai: fix clock management ASoC: stm32: sai: manage master clock ASoC: stm32: add h7 support for sai ASoC: stm32: sai: add h7 support ASoC: stm32: change configuration flag ASoC: stm32: Add DT bindings for SPDIFRX interface ASoC: stm32: Add SPDIFRX support zhangjun (1): ASoC: rockchip: i2s: add other configurable formats .../bindings/display/bridge/adi,adv7511.txt | 8 + .../bindings/display/bridge/renesas,dw-hdmi.txt | 9 +- .../devicetree/bindings/sound/audio-graph-card.txt | 129 +++ .../bindings/sound/audio-graph-scu-card.txt | 122 +++ .../devicetree/bindings/sound/cs35l35.txt | 3 + .../devicetree/bindings/sound/nau8825.txt | 3 + .../devicetree/bindings/sound/renesas,rsnd.txt | 37 +- .../devicetree/bindings/sound/rockchip,pdm.txt | 39 + .../devicetree/bindings/sound/rockchip-spdif.txt | 2 + .../devicetree/bindings/sound/samsung,odroid.txt | 15 +- .../devicetree/bindings/sound/simple-scu-card.txt | 67 +- .../devicetree/bindings/sound/st,stm32-i2s.txt | 62 ++ .../devicetree/bindings/sound/st,stm32-sai.txt | 41 +- .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 ++ .../devicetree/bindings/sound/sun4i-codec.txt | 11 +- .../bindings/sound/sun8i-codec-analog.txt | 1 + .../devicetree/bindings/sound/zte,zx-aud96p22.txt | 24 + Documentation/sound/soc/dapm.rst | 18 + drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 22 + .../gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 21 + drivers/of/base.c | 62 +- include/linux/of_graph.h | 21 + include/sound/cs35l35.h | 2 + include/sound/designware_i2s.h | 1 + include/sound/hdmi-codec.h | 9 + include/sound/rt5645.h | 6 +- include/sound/simple_card_utils.h | 43 +- include/sound/soc-dapm.h | 7 + include/sound/soc-topology.h | 5 + include/sound/soc.h | 3 + include/uapi/sound/asoc.h | 10 +- include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/atmel/atmel-pcm.h | 6 +- sound/soc/atmel/tse850-pcm5142.c | 4 +- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ak4613.c | 95 +- sound/soc/codecs/ak4642.c | 4 +- sound/soc/codecs/cs35l34.c | 4 +- sound/soc/codecs/cs35l35.c | 94 +- sound/soc/codecs/cs35l35.h | 6 + sound/soc/codecs/cs4271.c | 2 - sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/da7213.c | 37 +- sound/soc/codecs/da7218.c | 2 +- sound/soc/codecs/da7219-aad.c | 31 +- sound/soc/codecs/da7219.c | 53 +- sound/soc/codecs/da7219.h | 5 +- sound/soc/codecs/es8316.c | 637 +++++++++++++ sound/soc/codecs/es8316.h | 129 +++ sound/soc/codecs/hdmi-codec.c | 92 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 6 +- sound/soc/codecs/nau8824.c | 52 ++ sound/soc/codecs/nau8824.h | 12 + sound/soc/codecs/nau8825.c | 76 +- sound/soc/codecs/nau8825.h | 1 + sound/soc/codecs/rt5514.c | 37 +- sound/soc/codecs/rt5514.h | 6 + sound/soc/codecs/rt5645.c | 97 +- sound/soc/codecs/rt5651.c | 44 +- sound/soc/codecs/rt5663.c | 4 +- sound/soc/codecs/rt5663.h | 4 + sound/soc/codecs/rt5665.c | 158 ++-- sound/soc/codecs/rt5670.c | 25 +- sound/soc/codecs/rt5677.c | 32 +- sound/soc/codecs/sgtl5000.c | 89 ++ sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/codecs/wm_adsp.c | 6 +- sound/soc/codecs/zx_aud96p22.c | 403 +++++++++ sound/soc/davinci/davinci-mcasp.c | 12 +- sound/soc/dwc/dwc-i2s.c | 6 + sound/soc/fsl/mpc5200_dma.c | 1 - sound/soc/generic/Kconfig | 17 + sound/soc/generic/Makefile | 4 + sound/soc/generic/audio-graph-card.c | 338 +++++++ sound/soc/generic/audio-graph-scu-card.c | 411 +++++++++ sound/soc/generic/simple-card-utils.c | 198 +++- sound/soc/generic/simple-card.c | 69 +- sound/soc/generic/simple-scu-card.c | 55 +- sound/soc/hisilicon/hi6210-i2s.c | 11 +- sound/soc/intel/Kconfig | 42 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst.c | 59 +- sound/soc/intel/atom/sst/sst.h | 30 +- sound/soc/intel/atom/sst/sst_acpi.c | 232 +++-- sound/soc/intel/boards/Makefile | 6 + sound/soc/intel/boards/bdw-rt5677.c | 26 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 26 +- sound/soc/intel/boards/bxt_rt298.c | 19 +- sound/soc/intel/boards/byt-max98090.c | 24 +- sound/soc/intel/boards/bytcht_es8316.c | 300 +++++++ sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 - sound/soc/intel/boards/cht_bsw_rt5672.c | 89 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 687 ++++++++++++++ .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 640 +++++++++++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 16 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +- sound/soc/intel/boards/skl_rt286.c | 30 +- sound/soc/intel/common/sst-acpi.h | 23 + sound/soc/intel/common/sst-dsp-priv.h | 4 + sound/soc/intel/common/sst-match-acpi.c | 47 +- sound/soc/intel/skylake/Makefile | 4 + sound/soc/intel/skylake/bxt-sst.c | 4 + sound/soc/intel/skylake/skl-debug.c | 261 ++++++ sound/soc/intel/skylake/skl-messages.c | 31 +- sound/soc/intel/skylake/skl-pcm.c | 6 +- sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 187 +++- sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl.c | 114 ++- sound/soc/intel/skylake/skl.h | 23 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/omap/mcbsp.c | 12 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s.c | 41 +- sound/soc/rockchip/rockchip_i2s.h | 3 + sound/soc/rockchip/rockchip_pdm.c | 516 +++++++++++ sound/soc/rockchip/rockchip_pdm.h | 83 ++ sound/soc/rockchip/rockchip_spdif.c | 4 + sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/sh/Kconfig | 2 +- sound/soc/sh/fsi.c | 13 +- sound/soc/sh/rcar/adg.c | 61 +- sound/soc/sh/rcar/cmd.c | 7 +- sound/soc/sh/rcar/core.c | 494 +++++++--- sound/soc/sh/rcar/ctu.c | 6 + sound/soc/sh/rcar/dma.c | 32 +- sound/soc/sh/rcar/dvc.c | 12 +- sound/soc/sh/rcar/gen.c | 2 + sound/soc/sh/rcar/rsnd.h | 68 +- sound/soc/sh/rcar/src.c | 20 +- sound/soc/sh/rcar/ssi.c | 241 +++-- sound/soc/sh/rcar/ssiu.c | 37 + sound/soc/soc-compress.c | 6 +- sound/soc/soc-core.c | 86 +- sound/soc/soc-topology.c | 37 +- sound/soc/stm/Kconfig | 29 +- sound/soc/stm/Makefile | 12 +- sound/soc/stm/stm32_i2s.c | 946 +++++++++++++++++++ sound/soc/stm/stm32_sai.c | 15 +- sound/soc/stm/stm32_sai.h | 73 +- sound/soc/stm/stm32_sai_sub.c | 143 ++- sound/soc/stm/stm32_spdifrx.c | 998 +++++++++++++++++++++ sound/soc/sunxi/sun4i-codec.c | 63 ++ sound/soc/sunxi/sun8i-codec-analog.c | 145 ++- sound/soc/zte/zx-i2s.c | 15 +- 152 files changed, 10306 insertions(+), 1191 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h create mode 100644 sound/soc/codecs/zx_aud96p22.c create mode 100644 sound/soc/generic/audio-graph-card.c create mode 100644 sound/soc/generic/audio-graph-scu-card.c create mode 100644 sound/soc/intel/boards/bytcht_es8316.c create mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c create mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c create mode 100644 sound/soc/intel/skylake/skl-debug.c create mode 100644 sound/soc/rockchip/rockchip_pdm.c create mode 100644 sound/soc/rockchip/rockchip_pdm.h create mode 100644 sound/soc/stm/stm32_i2s.c create mode 100644 sound/soc/stm/stm32_spdifrx.c

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[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: explicitly add the headers sst-dsp.h
by Guneshwor Singh 03 Jul '17

03 Jul '17

From: Vinod Koul <vinod.koul(a)intel.com> Commit bdd0384a5ada ("ASoC: Intel: Skylake: Add support to read firmware registers") introduced firmware register read so added sst-dsp-priv.h but missed adding sst-dsp.h as that leads to below compiler warning: In file included from sound/soc/intel/skylake/skl-debug.c:23:0: >> sound/soc/intel/skylake/../common/sst-dsp-priv.h:63:42: warning: 'struct sst_pdata' declared inside parameter list [enabled by default] int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); ^ >> sound/soc/intel/skylake/../common/sst-dsp-priv.h:63:42: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default] So add the missing header. Fixes: bdd0384a5ada ("ASoC: Intel: Skylake: Add support to read firmware registers") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/skylake/skl-debug.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index 75497b1fda45..34d06abbf796 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -20,6 +20,7 @@ #include "skl-sst-ipc.h" #include "skl-tplg-interface.h" #include "skl-topology.h" +#include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" #define MOD_BUF PAGE_SIZE -- 2.13.0

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Re: [alsa-devel] ioctl(fd, SNDRV_CTL_IOCTL_ELEM_REPLACE, ... not working from userspace ?
by Clemens Ladisch 03 Jul '17

03 Jul '17

Fulup Ar Foll wrote: > Looking at this code I do not understand why the replace iotcl is refused. if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_USER)) { ret = -EINVAL; Kernel drivers do not expect that their own controls change. Regards, Clemens

1 0

[alsa-devel] [PATCH 0/6] add support for Sama5d2 audio PLL and enable ClassD
by Quentin Schulz 03 Jul '17

03 Jul '17

This patch series adds support for the audio PLL and enables ClassD that can be found in ATMEL Sama5d2 SoCs. There are two audio PLLs (PMC and PAD) that shares the same parent (FRAC). FRAC can output between 620 and 700MHz and only multiply the rate of its parent. The two audio PLLs then divide the FRAC rate to best match the asked rate. I basically took an old patch series posted by Nicolas on December, 6th 2016[1][2][3] and the comments Boris did on the first version[4] Nicolas sent on July, 15th 2015. I also fixed the function used to compute the divisors, removed useless spinlocks and added a range to the audio frac PLL to stay within vendor's supported range. Clocks that are children of gclk (generated-clk) are now able to propagate rate to the audio PLL clocks when needed. However, since there is no rate locking, we enforce that classd_gclk be the only child of audio_pll_pmc so we avoid multiple children modifiying parent's rate, and classd driver modifying parent's rate which would impact the rate of each of parent's children. Thanks, Quentin [1] https://patchwork.kernel.org/patch/9462351/ [2] https://patchwork.kernel.org/patch/9462347/ [3] https://patchwork.kernel.org/patch/9462349/ [4] https://www.spinics.net/lists/arm-kernel/msg436120.html Cyrille Pitchen (2): ARM: dts: at91: sama5d2: add classd nodes ARM: dts: at91: sama5d2_xplained: add pin muxing and enable classd Nicolas Ferre (1): clk: at91: add audio pll clock driver Quentin Schulz (3): clk: at91: clk-generated: remove useless divisor loop clk: at91: clk-generated: create function to find best_diff clk: at91: clk-generated: make gclk determine audio_pll rate .../devicetree/bindings/clock/at91-clock.txt | 10 + arch/arm/boot/dts/at91-sama5d2_xplained.dts | 16 ++ arch/arm/boot/dts/sama5d2.dtsi | 39 +++- arch/arm/mach-at91/Kconfig | 4 + drivers/clk/at91/Makefile | 2 + drivers/clk/at91/clk-audio-pll-pad.c | 204 ++++++++++++++++++ drivers/clk/at91/clk-audio-pll-pmc.c | 175 +++++++++++++++ drivers/clk/at91/clk-audio-pll.c | 237 +++++++++++++++++++++ drivers/clk/at91/clk-generated.c | 79 +++++-- include/linux/clk/at91_pmc.h | 25 +++ sound/soc/atmel/atmel-classd.c | 20 +- 11 files changed, 772 insertions(+), 39 deletions(-) create mode 100644 drivers/clk/at91/clk-audio-pll-pad.c create mode 100644 drivers/clk/at91/clk-audio-pll-pmc.c create mode 100644 drivers/clk/at91/clk-audio-pll.c -- 2.11.0

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[alsa-devel] Issue on Linux 4.12-rc7 on iMX6 based board
by gianluca 03 Jul '17

03 Jul '17

Hello list, I have some issue using the sgtl5000 on two custom boards based with iMX6. One board has iMX6DL and the other has iMX6QP. The issue is I have a (dramatically high) strange value on mixer settings (looks like a int32 limit or similar) This is the output of amixer: > # amixer > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch pswitch-joined > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 0 [2147483647%] [-51.50dB] [on] > Front Right: Playback 0 [2147483647%] [-51.50dB] [on] > Simple mixer control 'Headphone Mux',0 > Capabilities: enum > Items: 'DAC' 'LINE_IN' > Item0: 'DAC' > Simple mixer control 'Headphone Playback ZC',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 192 > Mono: > Front Left: Playback 1 [2147483647%] > Front Right: Playback 0 [2147483647%] > Simple mixer control 'Lineout',0 > Capabilities: pvolume pswitch pswitch-joined > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 18 [2147483647%] [-6.50dB] [on] > Front Right: Playback 18 [2147483647%] [-6.50dB] [on] > Simple mixer control 'Mic',0 > Capabilities: volume volume-joined > Playback channels: Mono > Capture channels: Mono > Limits: 0 - 3 > Mono: 0 [2147483647%] [0.00dB] > Simple mixer control 'Capture',0 > Capabilities: cvolume > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [2147483647%] > Front Right: Capture 0 [2147483647%] > Simple mixer control 'Capture Attenuate Switch (-6dB)',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Capture Mux',0 > Capabilities: enum > Items: 'MIC_IN' 'LINE_IN' > Item0: 'MIC_IN' > Simple mixer control 'Capture ZC',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [on] > In attachment there is a screenshot of alsamixer All audio stuff are not hear at all.... :-( After the output of the sgtl5000 I have a audio-booster and it is set at the maximum level. I have another board based on the iMX28, SGTL5000 and the same audio booster and everything is working good. So I can suppose something related the initialization of the codec. Any help? Best Regards, Gianluca Renzi -- Eurek s.r.l. | Electronic Engineering | http://www.eurek.it via Celletta 8/B, 40026 Imola, Italy | Phone: +39-(0)542-609120 p.iva 00690621206 - c.f. 04020030377 | Fax: +39-(0)542-609212

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[alsa-devel] [PATCH] aplay: Fix header for raw format
by Ion-Horia Petrisor 03 Jul '17

03 Jul '17

Raw format has no header, so use 0 when calling playback_go. Signed-off-by: Ion-Horia Petrisor <ion-horia.petrisor(a)nxp.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 00af662..fe274ed 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2831,7 +2831,7 @@ static void playback(char *name) /* should be raw data */ init_raw_data(); pbrec_count = calc_count(); - playback_go(fd, dta, pbrec_count, FORMAT_RAW, name); + playback_go(fd, 0, pbrec_count, FORMAT_RAW, name); } __end: if (fd != 0) -- 2.7.4

3 6

[alsa-devel] [PATCH] ALSA: hda/realtek - Remove GPIO_MASK
by Kai-Heng Feng 03 Jul '17

03 Jul '17

GPIO_MASK was introduced in commit "df694daa3c01 [ALSA] hda-codec - Add the support of ALC262,ALC883,ALC885,ALC861" but it's not being used. So we can simply remove it. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c79491a1cd47..425a4abd8ad0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -41,9 +41,6 @@ /* keep halting ALC5505 DSP, for power saving */ #define HALT_REALTEK_ALC5505 -/* for GPIO Poll */ -#define GPIO_MASK 0x03 - /* extra amp-initialization sequence types */ enum { ALC_INIT_NONE, -- 2.13.2

2 1

[alsa-devel] I sent hardware informations with the script about a MEDION laptop with Intel devices
by fulgor fulgor 03 Jul '17

03 Jul '17

Is there any news about a module for this laptop ? fulgor

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix typo
by Shih-Yuan Lee (FourDollars) 03 Jul '17

03 Jul '17

The PIN number for Dell headset mode of ALC3271 is wrong. Signed-off-by: Shih-Yuan Lee (FourDollars) <sylee(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c39bc49..d0cf1f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6553,7 +6553,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, - {0x13, 0xb8a60140}, + {0x13, 0xb8a61140}, {0x17, 0x90170110}), {} }; -- 2.7.4

2 1

[alsa-devel] [v2 0/4] Add debugfs support for skylake driver
by Guneshwor Singh 03 Jul '17

03 Jul '17

This series introduces debugfs support for skylake driver. It helps in debugging by providing module configurations which is loaded from topology. It also allows to read the firmware registers which are a part of SRAM memory area accessible by host CPU through a memory window. Guneshwor Singh (1): ASoC: Intel: common: Add sram address to sst_addr structure Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add support to read firmware registers Vinod Koul (2): ASoC: Intel: Skylake: Add debugfs support ASoC: Intel: Skylake: Debugfs facility to dump module config Changes in v2: - Re-ordered patch and squashed platform changes Guneshwor Singh (1): ASoC: Intel: common: Add sram address to sst_addr structure Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add support to read firmware registers Vinod Koul (2): ASoC: Intel: Skylake: Add debugfs support ASoC: Intel: Skylake: Debugfs facility to dump module config sound/soc/intel/common/sst-dsp-priv.h | 4 + sound/soc/intel/skylake/Makefile | 4 + sound/soc/intel/skylake/bxt-sst.c | 4 + sound/soc/intel/skylake/skl-debug.c | 269 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 5 + sound/soc/intel/skylake/skl.h | 26 ++++ 8 files changed, 320 insertions(+) create mode 100644 sound/soc/intel/skylake/skl-debug.c -- 2.13.0

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[alsa-devel] [asoc:topic/intel 37/40] sound/soc/intel/skylake/skl-debug.c:53:1-6: WARNING: invalid free of devm_ allocated data (fwd)
by Julia Lawall 03 Jul '17

03 Jul '17

This doesn't show enough of the context to see the allocation, but please check. thanks, julia ---------- Forwarded message ---------- Date: Sat, 1 Jul 2017 02:40:21 +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 37/40] sound/soc/intel/skylake/skl-debug.c:53:1-6: WARNING: invalid free of devm_ allocated data tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: bdd0384a5ada8bb5745e5f29c10a5ba88827efad commit: 5cdf6c09ca9de3f037ba2d770206f3374459602d [37/40] ASoC: Intel: Skylake: Add debugfs support :::::: branch date: 6 hours ago :::::: commit date: 6 hours ago >> sound/soc/intel/skylake/skl-debug.c:53:1-6: WARNING: invalid free of devm_ allocated data git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 5cdf6c09ca9de3f037ba2d770206f3374459602d vim +53 sound/soc/intel/skylake/skl-debug.c 5cdf6c09 Vinod Koul 2017-06-30 37 skl->platform->component.debugfs_root); 5cdf6c09 Vinod Koul 2017-06-30 38 if (IS_ERR(d->fs) || !d->fs) { 5cdf6c09 Vinod Koul 2017-06-30 39 dev_err(&skl->pci->dev, "debugfs root creation failed\n"); 5cdf6c09 Vinod Koul 2017-06-30 40 return NULL; 5cdf6c09 Vinod Koul 2017-06-30 41 } 5cdf6c09 Vinod Koul 2017-06-30 42 5cdf6c09 Vinod Koul 2017-06-30 43 d->skl = skl; 5cdf6c09 Vinod Koul 2017-06-30 44 d->dev = &skl->pci->dev; 5cdf6c09 Vinod Koul 2017-06-30 45 5cdf6c09 Vinod Koul 2017-06-30 46 return d; 5cdf6c09 Vinod Koul 2017-06-30 47 } 5cdf6c09 Vinod Koul 2017-06-30 48 5cdf6c09 Vinod Koul 2017-06-30 49 void skl_debugfs_exit(struct skl_debug *d) 5cdf6c09 Vinod Koul 2017-06-30 50 { 5cdf6c09 Vinod Koul 2017-06-30 51 debugfs_remove_recursive(d->fs); 5cdf6c09 Vinod Koul 2017-06-30 52 5cdf6c09 Vinod Koul 2017-06-30 @53 kfree(d); 5cdf6c09 Vinod Koul 2017-06-30 54 5cdf6c09 Vinod Koul 2017-06-30 55 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH v2] ALSA: pcm: add a documentation for tracepoints
by Takashi Sakamoto 02 Jul '17

02 Jul '17

In PCM interface/protocol for userspace, parameters of runtime for PCM substream is decided by an interaction between applications and ALSA PCM core. In former commits, some tracepoints were added to probe a part of the interaction. This commit adds a documentation about the interaction and the tracepoints. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/tracepoints.rst | 172 ++++++++++++++++++++++++++++ 2 files changed, 173 insertions(+) create mode 100644 Documentation/sound/designs/tracepoints.rst diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 04dcdae3e4f2..f0749943ccb2 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -9,6 +9,7 @@ Designs and Implementations compress-offload timestamping jack-controls + tracepoints procfile powersave oss-emulation diff --git a/Documentation/sound/designs/tracepoints.rst b/Documentation/sound/designs/tracepoints.rst new file mode 100644 index 000000000000..d2e481c6da37 --- /dev/null +++ b/Documentation/sound/designs/tracepoints.rst @@ -0,0 +1,172 @@ +=================== +Tracepoints in ALSA +=================== + +2017/07/02 +Takasahi Sakamoto + +Tracepoints in ALSA PCM core +============================ + +ALSA PCM core registers ``snd_pcm`` subsystem to kernel tracepoint system. +This subsystem includes two categories of tracepoints; for state of PCM buffer +and for processing of PCM hardware parameters. These tracepoints are available +when corresponding kernel configurations are enabled. When ``CONFIG_SND_DEBUG`` +is enabled, the latter tracepoints are available. When additional +``SND_PCM_XRUN_DEBUG`` is enabled too, the former trace points are enabled. + +Tracepoints for state of PCM buffer +------------------------------------ + +This category includes four tracepoints; ``hwptr``, ``applptr``, ``xrun`` and +``hw_ptr_error``. + +Tracepoints for processing of PCM hardware parameters +----------------------------------------------------- + +This category includes two tracepoints; ``hw_mask_param`` and +``hw_interval_param``. + +In a design of ALSA PCM core, data transmission is abstracted as PCM substream. +Applications manage PCM substream to maintain data transmission for PCM frames. +Before starting the data transmission, applications need to configure PCM +substream. In this procedure, PCM hardware parameters are decided by +interaction between applications and ALSA PCM core. Once decided, runtime of +the PCM substream keeps the parameters. + +The parameters are described in :c:type:`struct snd_pcm_hw_params`. This +structure includes several types of parameters. Applications set preferable +value to these parameters, then execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE +or SNDRV_PCM_IOCTL_HW_PARAMS. The former is used just for refining available +set of parameters. The latter is used for an actual decision of the parameters. + +The :c:type:`struct snd_pcm_hw_params` structure has below members: + +``flags`` + Configurable. ALSA PCM core and some drivers handle this flag to select + convenient parameters or change their behaviour. +``masks`` + Configurable. This type of parameter is described in + :c:type:`struct snd_mask` and represent mask values. As of PCM protocol + v2.0.13, three types are defined. + + - SNDRV_PCM_HW_PARAM_ACCESS + - SNDRV_PCM_HW_PARAM_FORMAT + - SNDRV_PCM_HW_PARAM_SUBFORMAT +``intervals`` + Configurable. This type of parameter is described in + :c:type:`struct snd_interval` and represent values with a range. As of + PCM protocol v2.0.13, twelve types are defined. + + - SNDRV_PCM_HW_PARAM_SAMPLE_BITS + - SNDRV_PCM_HW_PARAM_FRAME_BITS + - SNDRV_PCM_HW_PARAM_CHANNELS + - SNDRV_PCM_HW_PARAM_RATE + - SNDRV_PCM_HW_PARAM_PERIOD_TIME + - SNDRV_PCM_HW_PARAM_PERIOD_SIZE + - SNDRV_PCM_HW_PARAM_PERIOD_BYTES + - SNDRV_PCM_HW_PARAM_PERIODS + - SNDRV_PCM_HW_PARAM_BUFFER_TIME + - SNDRV_PCM_HW_PARAM_BUFFER_SIZE + - SNDRV_PCM_HW_PARAM_BUFFER_BYTES + - SNDRV_PCM_HW_PARAM_TICK_TIME +``rmask`` + Configurable. This is evaluated at ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE only. Applications can select which + mask/interval parameter can be changed by ALSA PCM core. For + SNDRV_PCM_IOCTL_HW_PARAMS, this mask is ignored and all of parameters + are going to be changed. +``cmask`` + Read-only. After returning from ioctl(2), buffer in user space for + :c:type:`struct snd_pcm_hw_params` includes result of each operation. + This mask represents which mask/interval parameter is actually changed. +``info`` + Read-only. This represents hardware/driver capabilities as bit flags + with SNDRV_PCM_INFO_XXX. Typically, applications execute ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE to retrieve this flag, then decide candidates + of parameters and execute ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS to + configure PCM substream. +``msbits`` + Read-only. This value represents available bit width in MSB side of + a PCM sample. When a parameter of SNDRV_PCM_HW_PARAM_SAMPLE_BITS was + decided as a fixed number, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_num`` + Read-only. This value represents numerator of sampling rate in fraction + notation. Basically, when a parameter of SNDRV_PCM_HW_PARAM_RATE was + decided as a single value, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_den`` + Read-only. This value represents denominator of sampling rate in + fraction notation. Basically, when a parameter of + SNDRV_PCM_HW_PARAM_RATE was decided as a single value, this value is + also calculated according to it. Else, zero. But this behaviour depends + on implementations in driver side. +``fifo_size`` + Read-only. This value represents the size of FIFO in serial sound + interface of hardware. Basically, each driver can assigns a proper + value to this parameter but some drivers intentionally set zero with + a care of hardware design or data transmission protocol. + +ALSA PCM core handles buffer of :c:type:`struct snd_pcm_hw_params` when +applications execute ioctl(2) with SNDRV_PCM_HW_REFINE or SNDRV_PCM_HW_PARAMS. +Parameters in the buffer are changed according to +:c:type:`struct snd_pcm_hardware` and rules of constraints in the runtime. The +structure describes capabilities of handled hardware. The rules describes +dependencies on which a parameter is decided according to several parameters. +A rule has a callback function, and drivers can register arbitrary functions +to compute the target parameter. ALSA PCM core registers some rules to the +runtime as a default. + +Each driver can join in the interaction as long as it prepared for two stuffs +in a callback of :c:type:`struct snd_pcm_ops.open`. + +1. In the callback, drivers are expected to change a member of + :c:type:`struct snd_pcm_hardware` type in the runtime, according to + capacities of corresponding hardware. +2. In the same callback, drivers are also expected to register additional rules + of constraints into the runtime when several parameters have dependencies + due to hardware design. + +The driver can refers to result of the interaction in a callback of +:c:type:`struct snd_pcm_ops.hw_params`, however it should not change the +content. + +Tracepoints in this category are designed to trace changes of the +mask/interval parameters. When ALSA PCM core changes them, ``hw_mask_param`` or +``hw_interval_param`` event is probed according to type of the changed parameter. + +ALSA PCM core also has a pretty print format for each of the tracepoints. Below +is an example for ``hw_mask_param``. + +:: + + hw_mask_param: pcmC0D0p 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044 + + +Below is an example for ``hw_interval_param``. + +:: + + hw_interval_param: pcmC0D0p 000/023 BUFFER_SIZE 0 0 [0 4294967295] 0 1 [0 4294967295] + +The first three fields are common. They represent name of ALSA PCM character +device, rules of constraint and name of the changed parameter, in order. The +field for rules of constraint consists of two sub-fields; index of applied rule +and total number of rules added to the runtime. As an exception, the index 000 +means that the parameter is changed by ALSA PCM core, regardless of the rules. + +The rest of field represent state of the parameter before/after changing. These +fields are different according to type of the parameter. For parameters of mask +type, the fields represent hexadecimal dump of content of the parameter. For +parameters of interval type, the fields represent values of each member of +``empty``, ``integer``, ``openmin``, ``min``, ``max``, ``openmax`` in +:c:type:`struct snd_interval` in this order. + +Tracepoints in drivers +====================== + +Some drivers have tracepoints for developers' convenience. For them, please +refer to each documentation or implementation. -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: atmel: ac97c: fix error return code in atmel_ac97c_probe()
by Gustavo A. R. Silva 01 Jul '17

01 Jul '17

platform_get_irq() returns an error code, but the ac97c driver ignores it and always returns -ENXIO. This is not correct, and prevents -EPROBE_DEFER from being propagated properly. Notice that platform_get_irq() no longer returns 0 on error. Print and propagate the return value of platform_get_irq on failure. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/atmel/ac97c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 65e6948..9d2c9d9 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -774,8 +774,8 @@ static int atmel_ac97c_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_dbg(&pdev->dev, "could not get irq\n"); - return -ENXIO; + dev_dbg(&pdev->dev, "could not get irq: %d\n", irq); + return irq; } pclk = clk_get(&pdev->dev, "ac97_clk"); -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: x86: fix error return code in hdmi_lpe_audio_probe()
by Gustavo A. R. Silva 30 Jun '17

30 Jun '17

platform_get_irq() returns an error code, but the sata_rcar driver ignores it and always returns -ENODEV. This is not correct, and prevents -EPROBE_DEFER from being propagated properly. Also, notice that platform_get_irq() no longer returns 0 on error. Print and propagate the return value of platform_get_irq on failure. Signed-off-by: Gustavo A. R. Silva <garsilva(a)embeddedor.com> --- sound/x86/intel_hdmi_audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index c19efc9..a095150 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1758,8 +1758,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* get resources */ irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "Could not get irq resource\n"); - return -ENODEV; + dev_err(&pdev->dev, "Could not get irq resource: %d\n", irq); + return irq; } res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); -- 2.5.0

2 7

[alsa-devel] [PATCH alsa-lib v2 0/6] pcm: hw: optimization for v2.0.14 of PCM protocol/interface
by Takashi Sakamoto 30 Jun '17

30 Jun '17

Hi, This patchset is a revised version of my previous one: [PATCH alsa-lib 00/12] pcm: hw: optimization for v2.0.14 of PCM protocol/interface http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122392.html Changes: - add a helper function, 'query_status_and_control_data()' to update appl_ptr in user space. - rename 'query_state()' to 'query_status_data()' - reduce two patches. - aggregate some patches for querying control data. I did quick mesurement by execute aplay on x86 machine with. 1. With Intel HDA driver with usual configuration. ``` poll([{fd=4, events=POLLOUT|POLLERR|POLLNVAL}], 1, -1) = 1 ([{fd=4, revents=POLLOUT}]) ioctl(4, SNDRV_PCM_IOCTL_HWSYNC, 0) = 0 read(3, "a\23IH\205\177z\257\276]\346\237\313\220L\5\337V\363\351\216\17\346s\22\245-yb\312\37D"..., 16384) = 16384 ioctl(4, SNDRV_PCM_IOCTL_HWSYNC, 0) = 0 poll([{fd=4, events=POLLOUT|POLLERR|POLLNVAL}], 1, -1) = 1 ([{fd=4, revents=POLLOUT}]) ``` 2. With Intel HDA driver with 'sync_ptr_ioctl' option to hw PCM plugin. ``` poll([{fd=4, events=POLLOUT|POLLERR|POLLNVAL}], 1, -1) = 1 ([{fd=4, revents=POLLOUT}]) ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 read(3, "Rz\vB\v|A\275\356`\201)\335i3\r}\21Q^\30\324'\r'\375l\235&\3330\2"..., 16384) = 16384 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55fe03539f80) = 0 poll([{fd=4, events=POLLOUT|POLLERR|POLLNVAL}], 1, -1) = 1 ([{fd=4, revents=POLLOUT}]) ``` 3. With Fireworks driver with a patch for snd-firewire-lib to add SNDRV_PCM_INFO_SYNC_APPLPTR support[1]. ``` poll([{fd=4, events=POLLOUT|POLLERR|POLLNVAL}], 1, -1) = 1 ([{fd=4, revents=POLLOUT}]) ioctl(4, SNDRV_PCM_IOCTL_HWSYNC, 0) = 0 ioctl(4, SNDRV_PCM_IOCTL_SYNC_PTR, 0x55b647ec9c00) = 0 read(3, "|\n*\336\322\252*&\377\312\372&\276\333.\207\21\343\362\3\5\363\201\266\3421lZ;\\\22*"..., 24576) = 24576 ioctl(4, SNDRV_PCM_IOCTL_HWSYNC, 0) = 0 poll([{fd=4, events=POLLOUT|POLLERR|POLLNVAL}], 1, -1) = 1 ([{fd=4, revents=POLLOUT}]) ``` We can see this optimization reduces some ioctl calls in case 3. This is due to status data mapping. [1] Below. diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8c2765..04fb3df 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -160,7 +160,8 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_APPLPTR; /* SNDRV_PCM_INFO_BATCH */ hw->periods_min = 2; Takashi Sakamoto (6): pcm: hw: add a helper function to query status/control data pcm: hw: add a helper function just to query status data pcm: hw: add a helper function to request hwsync without side-effects pcm: hw: add a helper function to issue appl_ptr without side-effects pcm: hw: add a helper function to issue avail_min without side-effects pcm: hw: remove superfluous code to call of SNDRV_PCM_IOCTL_SYNC_PTR in snd_pcm_hw_forward() src/pcm/pcm_hw.c | 116 +++++++++++++++++++++++++++++++++++++++++-------------- 1 file changed, 86 insertions(+), 30 deletions(-) -- 2.11.0

2 7

[alsa-devel] [PATCH alsa-lib] pcm: hw: minor refactoring for initialization of control data
by Takashi Sakamoto 30 Jun '17

30 Jun '17

At failure of control data mapping, alsa-lib goes to fallback mode. In this mode, a buffer is kept in user space and executes ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR for the buffer to synchronize the control data. In current implementation, no helper function is used for initialize the control data. This commit use an proper helper function instead of a direct call of ioctl(2). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/pcm/pcm_hw.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index e0931577..f03bc589 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -957,11 +957,9 @@ static int map_status_and_control_data(snd_pcm_t *pcm, bool force_fallback) snd_pcm_set_appl_ptr(pcm, &hw->mmap_control->appl_ptr, hw->fd, SNDRV_PCM_MMAP_OFFSET_CONTROL); if (hw->mmap_control_fallbacked) { - if (ioctl(hw->fd, SNDRV_PCM_IOCTL_SYNC_PTR, hw->sync_ptr) < 0) { - err = -errno; - SYSMSG("SNDRV_PCM_IOCTL_SYNC_PTR failed (%i)", err); + err = sync_ptr1(hw, 0); + if (err < 0) return err; - } } return 0; -- 2.11.0

2 1

[alsa-devel] [PATCH] alsalib: topology: Add support for new widget types
by Liam Girdwood 30 Jun '17

30 Jun '17

Add topology support for new DSP widget types. This allows the new widgets to be added to the driver and firmware DAPM graphs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Kernel part now upstream in Mark's ASoC for-next branch. include/sound/asoc.h | 10 +++++++++- src/topology/dapm.c | 8 ++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) diff --git a/include/sound/asoc.h b/include/sound/asoc.h index 082c5429..0f5d9f9a 100644 --- a/include/sound/asoc.h +++ b/include/sound/asoc.h @@ -70,7 +70,15 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_SCHEDULER 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_ENCODER 22 +#define SND_SOC_TPLG_DAPM_DECODER 23 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DECODER /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 6af750b9..66892a66 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -38,6 +38,14 @@ static const struct map_elem widget_map[] = { {"dai_in", SND_SOC_TPLG_DAPM_DAI_IN}, {"dai_out", SND_SOC_TPLG_DAPM_DAI_OUT}, {"dai_link", SND_SOC_TPLG_DAPM_DAI_LINK}, + {"buffer", SND_SOC_TPLG_DAPM_BUFFER}, + {"scheduler", SND_SOC_TPLG_DAPM_SCHEDULER}, + {"effect", SND_SOC_TPLG_DAPM_EFFECT}, + {"siggen", SND_SOC_TPLG_DAPM_SIGGEN}, + {"src", SND_SOC_TPLG_DAPM_SRC}, + {"asrc", SND_SOC_TPLG_DAPM_ASRC}, + {"encoder", SND_SOC_TPLG_DAPM_ENCODER}, + {"decoder", SND_SOC_TPLG_DAPM_DECODER}, }; static int lookup_widget(const char *w) -- 2.11.0

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[alsa-devel] [asoc:topic/intel 40/40] sound/soc/intel/skylake/../common/sst-dsp-priv.h:63:42: warning: 'struct sst_pdata' declared inside parameter list will not be visible outside of this definition or declaration
by kbuild test robot 30 Jun '17

30 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: bdd0384a5ada8bb5745e5f29c10a5ba88827efad commit: bdd0384a5ada8bb5745e5f29c10a5ba88827efad [40/40] ASoC: Intel: Skylake: Add support to read firmware registers config: x86_64-randconfig-x015-201726 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout bdd0384a5ada8bb5745e5f29c10a5ba88827efad # save the attached .config to linux build tree make ARCH=x86_64 All warnings (new ones prefixed by >>): In file included from sound/soc/intel/skylake/skl-debug.c:23:0: >> sound/soc/intel/skylake/../common/sst-dsp-priv.h:63:42: warning: 'struct sst_pdata' declared inside parameter list will not be visible outside of this definition or declaration int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); ^~~~~~~~~ vim +63 sound/soc/intel/skylake/../common/sst-dsp-priv.h a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 47 u32 (*read)(void __iomem *addr, u32 offset); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 48 void (*write64)(void __iomem *addr, u32 offset, u64 value); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 49 u64 (*read64)(void __iomem *addr, u32 offset); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 50 a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 51 /* DSP I/DRAM IO */ a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 52 void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src, a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 53 size_t bytes); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 54 void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src, a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 55 size_t bytes); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 56 a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 57 void (*dump)(struct sst_dsp *); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 58 a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 59 /* IRQ handlers */ a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 60 irqreturn_t (*irq_handler)(int irq, void *context); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 61 a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 62 /* SST init and free */ a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 @63 int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 64 void (*free)(struct sst_dsp *sst); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 65 a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 66 /* FW module parser/loader */ a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 67 int (*parse_fw)(struct sst_fw *sst_fw); a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 68 }; a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 69 a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 70 /* a4b12990 sound/soc/intel/sst-dsp-priv.h Mark Brown 2014-03-12 71 * Audio DSP memory offsets and addresses. :::::: The code at line 63 was first introduced by commit :::::: a4b12990b68079290ab62799035afe175b4bdc23 Merge remote-tracking branches 'asoc/topic/ml26124', 'asoc/topic/of', 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/rcar' into asoc-next :::::: TO: Mark Brown <broonie(a)linaro.org> :::::: CC: Mark Brown <broonie(a)linaro.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] [asoc:topic/drm 2/2] drivers/gpu//drm/bridge/adv7511/adv7511_audio.c:210:2: error: unknown field 'get_dai_id' specified in initializer
by kbuild test robot 30 Jun '17

30 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/drm head: 7204e97685634813d8456f1900b7f38fa7701e60 commit: 7204e97685634813d8456f1900b7f38fa7701e60 [2/2] drm: adv7511_audio: Add .get_dai_id callback to map port number to dai id config: x86_64-randconfig-x004-201726 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 7204e97685634813d8456f1900b7f38fa7701e60 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): >> drivers/gpu//drm/bridge/adv7511/adv7511_audio.c:210:2: error: unknown field 'get_dai_id' specified in initializer .get_dai_id = adv7511_hdmi_i2s_get_dai_id, ^ >> drivers/gpu//drm/bridge/adv7511/adv7511_audio.c:210:16: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] .get_dai_id = adv7511_hdmi_i2s_get_dai_id, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ drivers/gpu//drm/bridge/adv7511/adv7511_audio.c:210:16: note: (near initialization for 'adv7511_codec_ops.hw_params') cc1: some warnings being treated as errors vim +/get_dai_id +210 drivers/gpu//drm/bridge/adv7511/adv7511_audio.c 204 } 205 206 static const struct hdmi_codec_ops adv7511_codec_ops = { 207 .hw_params = adv7511_hdmi_hw_params, 208 .audio_shutdown = audio_shutdown, 209 .audio_startup = audio_startup, > 210 .get_dai_id = adv7511_hdmi_i2s_get_dai_id, 211 }; 212 213 static struct hdmi_codec_pdata codec_data = { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v3 0/4] [PATCH v3 0/4] Add debugfs support for skylake driver
by Guneshwor Singh 30 Jun '17

30 Jun '17

This series introduces debugfs support for skylake driver. It helps in debugging by providing module configurations which is loaded from topology. It also allows to read the firmware registers which are a part of SRAM memory area accessible by host CPU through a memory window. Changes in v3: - Change debugfs parent from debugfs root to ASoC platform component Changes in v2: - Re-ordered patch and squashed platform changes Guneshwor Singh (1): ASoC: Intel: Skylake: Add sram address to sst_addr structure Vinod Koul (2): ASoC: Intel: Skylake: Add debugfs support ASoC: Intel: Skylake: Debugfs facility to dump module config Vunny Sodhi (1): ASoC: Intel: Skylake: Add support to read firmware registers sound/soc/intel/common/sst-dsp-priv.h | 4 + sound/soc/intel/skylake/Makefile | 4 + sound/soc/intel/skylake/bxt-sst.c | 4 + sound/soc/intel/skylake/skl-debug.c | 268 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 6 +- sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 2 + sound/soc/intel/skylake/skl.h | 26 ++++ 9 files changed, 321 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/skylake/skl-debug.c -- 2.13.0

3 6

[alsa-devel] [PATCH 0/5] Add debugfs support for skylake driver
by Guneshwor Singh 30 Jun '17

30 Jun '17

This series introduces debugfs support for skylake driver. It helps in debugging by providing module configurations which is loaded from topology. It also allows to read the firmware registers which are a part of SRAM memory area accessible by host CPU through a memory window. Guneshwor Singh (2): ASoC: Intel: bxtn: Assign sram address to sst structure ASoC: Intel: Skylake: Assign sram address to sst structure Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add support to read firmware registers Vinod Koul (2): ASoC: Intel: Skylake: Add debugfs support ASoC: Intel: Skylake: Debugfs facility to dump module config sound/soc/intel/common/sst-dsp-priv.h | 4 + sound/soc/intel/skylake/Makefile | 4 + sound/soc/intel/skylake/bxt-sst.c | 4 + sound/soc/intel/skylake/skl-debug.c | 269 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 5 + sound/soc/intel/skylake/skl.h | 26 ++++ 8 files changed, 320 insertions(+) create mode 100644 sound/soc/intel/skylake/skl-debug.c -- 2.13.0

3 7

[alsa-devel] [PATCH v3 0/4] Add debugfs support for skylake driver
by Guneshwor Singh 30 Jun '17

30 Jun '17

This series introduces debugfs support for skylake driver. It helps in debugging by providing module configurations which is loaded from topology. It also allows to read the firmware registers which are a part of SRAM memory area accessible by host CPU through a memory window. Changes in v3: - Change debugfs parent from debugfs root to ASoC platform component Changes in v2: - Re-ordered patch and squashed platform changes Guneshwor Singh (1): ASoC: Intel: Skylake: Add sram address to sst_addr structure Sodhi, VunnyX (1): ASoC: Intel: Skylake: Add support to read firmware registers Vinod Koul (2): ASoC: Intel: Skylake: Add debugfs support ASoC: Intel: Skylake: Debugfs facility to dump module config sound/soc/intel/common/sst-dsp-priv.h | 4 + sound/soc/intel/skylake/Makefile | 4 + sound/soc/intel/skylake/bxt-sst.c | 4 + sound/soc/intel/skylake/skl-debug.c | 268 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 6 +- sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 2 + sound/soc/intel/skylake/skl.h | 26 ++++ 9 files changed, 321 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/skylake/skl-debug.c -- 2.13.0

3 7

[alsa-devel] [PATCH] drm: adv7511_audio: Add .get_dai_id callback to map port number to dai id
by John Stultz 30 Jun '17

30 Jun '17

ALSA SoC needs to know connected DAI ID for probing. Using the new audio-card-graph approach, ports/endpoints are used to describe how the links are connected. Unfortunately, since ports/endpoints are used as well for video linkages, there are some issues mixing the port ids to the two (video and audio) namespaces. To solve this issue, this patch adds new .get_dai_id callback on hdmi_codec_ops. The will assume that HDMI audio out will be connected to reg = <2>. This will then be remapped to the ALSA SoC side will as DAI 0. Allowing the adv7511's hdmi audio support to be used with the audio-card-graph. Credit to Kuninori Morimoto who's patch to dw-hdmi-i2s-audio.c was what this was mostly copy-pasted from. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Archit Taneja <architt(a)codeaurora.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Linux-ALSA <alsa-devel(a)alsa-project.org> Cc: dri-devel(a)lists.freedesktop.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- .../bindings/display/bridge/adi,adv7511.txt | 8 ++++++++ drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 22 ++++++++++++++++++++++ 2 files changed, 30 insertions(+) diff --git a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt index 00ea670..06668bc 100644 --- a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt +++ b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.txt @@ -78,6 +78,7 @@ graph bindings specified in Documentation/devicetree/bindings/graph.txt. remote endpoint phandle should be a reference to a valid mipi_dsi_host device node. - Video port 1 for the HDMI output +- Audio port 2 for the HDMI audio input Example @@ -112,5 +113,12 @@ Example remote-endpoint = <&hdmi_connector_in>; }; }; + + port@2 { + reg = <2>; + codec_endpoint: endpoint { + remote-endpoint = <&i2s0_cpu_endpoint>; + }; + }; }; }; diff --git a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c index cf92ebf..67469c2 100644 --- a/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c +++ b/drivers/gpu/drm/bridge/adv7511/adv7511_audio.c @@ -11,6 +11,7 @@ #include <sound/hdmi-codec.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <linux/of_graph.h> #include "adv7511.h" @@ -182,10 +183,31 @@ static void audio_shutdown(struct device *dev, void *data) { } +static int adv7511_hdmi_i2s_get_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + struct of_endpoint of_ep; + int ret; + + ret = of_graph_parse_endpoint(endpoint, &of_ep); + if (ret < 0) + return ret; + + /* + * HDMI sound should be located as reg = <2> + * Then, it is sound port 0 + */ + if (of_ep.port == 2) + return 0; + + return -EINVAL; +} + static const struct hdmi_codec_ops adv7511_codec_ops = { .hw_params = adv7511_hdmi_hw_params, .audio_shutdown = audio_shutdown, .audio_startup = audio_startup, + .get_dai_id = adv7511_hdmi_i2s_get_dai_id, }; static struct hdmi_codec_pdata codec_data = { -- 2.7.4

2 5

[alsa-devel] [PATCH] ASoC: ak4642: make arrays fs_list and ps_list static const
by Colin King 30 Jun '17

30 Jun '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the arrays fs_list and ps_list on the stack but make them static const. Makes the object code smaller: Before: text data bss dec hex filename 12084 4888 64 17036 428c sound/soc/codecs/ak4642.o After: text data bss dec hex filename 11883 5032 64 16979 4253 sound/soc/codecs/ak4642.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/ak4642.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 23ab9646c351..66de8a2013a6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -433,7 +433,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int ak4642_set_mcko(struct snd_soc_codec *codec, u32 frequency) { - u32 fs_list[] = { + static const u32 fs_list[] = { [0] = 8000, [1] = 12000, [2] = 16000, @@ -447,7 +447,7 @@ static int ak4642_set_mcko(struct snd_soc_codec *codec, [14] = 29400, [15] = 44100, }; - u32 ps_list[] = { + static const u32 ps_list[] = { [0] = 256, [1] = 128, [2] = 64, -- 2.11.0

2 1

[alsa-devel] [PATCHv4] ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.
by Liam Girdwood 30 Jun '17

30 Jun '17

Add some DAPM widget types to better support the construction of DAPM graphs within DSPs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Chnages V3. o Go back to the original widget type plan and replace pipeline with scheduler widget. Changes V2. o No need for multiple types since core does not care. Changes v1. o Added some documentation. o Split codec widget into encoder and decoder to avoid confusion with HW CODECS. Documentation/sound/soc/dapm.rst | 18 ++++++++++++++++++ include/sound/soc-dapm.h | 7 +++++++ include/uapi/sound/asoc.h | 10 +++++++++- sound/soc/soc-topology.c | 8 ++++++++ 4 files changed, 42 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index a27f42befa4d..8e44107933ab 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -105,6 +105,24 @@ Pre Special PRE widget (exec before all others) Post Special POST widget (exec after all others) +Buffer + Inter widget audio data buffer within a DSP. +Scheduler + DSP internal scheduler that schedules component/pipeline processing + work. +Effect + Widget that performs an audio processing effect. +SRC + Sample Rate Converter within DSP or CODEC +ASRC + Asynchronous Sample Rate Converter within DSP or CODEC +Encoder + Widget that encodes audio data from one format (usually PCM) to another + usually more compressed format. +Decoder + Widget that decodes audio data from a compressed format to an + uncompressed format like PCM. + (Widgets are defined in include/sound/soc-dapm.h) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..344b96c206a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,13 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_buffer, /* DSP/CODEC internal buffer */ + snd_soc_dapm_scheduler, /* DSP/CODEC internal scheduler */ + snd_soc_dapm_effect, /* DSP/CODEC effect component */ + snd_soc_dapm_src, /* DSP/CODEC SRC component */ + snd_soc_dapm_asrc, /* DSP/CODEC ASRC component */ + snd_soc_dapm_encoder, /* FW/SW audio encoder component */ + snd_soc_dapm_decoder, /* FW/SW audio decoder component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..78014ec56357 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,15 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_SCHEDULER 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_ENCODER 22 +#define SND_SOC_TPLG_DAPM_DECODER 23 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DECODER /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 12e189701924..10797d74a700 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,14 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_BUFFER, snd_soc_dapm_buffer}, + {SND_SOC_TPLG_DAPM_SCHEDULER, snd_soc_dapm_scheduler}, + {SND_SOC_TPLG_DAPM_EFFECT, snd_soc_dapm_effect}, + {SND_SOC_TPLG_DAPM_SIGGEN, snd_soc_dapm_siggen}, + {SND_SOC_TPLG_DAPM_SRC, snd_soc_dapm_src}, + {SND_SOC_TPLG_DAPM_ASRC, snd_soc_dapm_asrc}, + {SND_SOC_TPLG_DAPM_ENCODER, snd_soc_dapm_encoder}, + {SND_SOC_TPLG_DAPM_DECODER, snd_soc_dapm_decoder}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- 2.11.0

2 5

[alsa-devel] [PATCH] ASoC: rt5670: fix incompatible pointer type of set_sysclk
by Bard Liao 30 Jun '17

30 Jun '17

The first parameter is codec not dai. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Sorry for missing it in the "move set_sysclk to codec level" patch --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 7fa63ad..64756dc 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2442,7 +2442,7 @@ static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int rt5670_set_codec_sysclk(struct snd_soc_dai *dai, int clk_id, +static int rt5670_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: nau8825: change crosstalk-bypass property to bool type" to the asoc tree
by Mark Brown 30 Jun '17

30 Jun '17

The patch ASoC: nau8825: change crosstalk-bypass property to bool type 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 fc3ba81a5adac413312019413c91b1e6a5d8d1fa Mon Sep 17 00:00:00 2001 From: John Hsu <supercraig0719(a)gmail.com> Date: Thu, 29 Jun 2017 11:41:30 +0800 Subject: [PATCH] ASoC: nau8825: change crosstalk-bypass property to bool type The property type of "nuvoton,crosstalk-bypass" changes to boolean. The document is updated as well. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/nau8825.txt | 3 +++ sound/soc/codecs/nau8825.c | 6 ++---- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index d3374231c871..2f5e973285a6 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,6 +69,8 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,crosstalk-bypass: make crosstalk function bypass if set. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock @@ -96,6 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-bypass; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 80bae481e75d..46a30eaa7ace 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2506,10 +2506,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", - &nau8825->xtalk_bypass); - if (ret) - nau8825->xtalk_bypass = 1; + nau8825->xtalk_bypass = device_property_read_bool(dev, + "nuvoton,crosstalk-bypass"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- 2.13.2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: constify dev_pm_ops structures.
by Arvind Yadav 30 Jun '17

30 Jun '17

dev_pm_ops are not supposed to change at runtime. All functions working with dev_pm_ops provided by <linux/device.h> work with const dev_pm_ops. So mark the non-const structs as const. File size before: text data bss dec hex filename 8172 920 0 9092 2384 sound/soc/sh/rcar/core.o File size After adding 'const': text data bss dec hex filename 8364 728 0 9092 2384 sound/soc/sh/rcar/core.o Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1744015..e327436 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1281,7 +1281,7 @@ static int rsnd_resume(struct device *dev) return 0; } -static struct dev_pm_ops rsnd_pm_ops = { +static const struct dev_pm_ops rsnd_pm_ops = { .suspend = rsnd_suspend, .resume = rsnd_resume, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5665: calibration should be done before jack detection
by Bard Liao 30 Jun '17

30 Jun '17

We will set some volatile registers in jack detection function. But those volatile registers will be clear in rt5665_calibrate function because we set cache bypass and reset codec in rt5665_calibrate function. This patch add a flag to make sure that rt5665_calibrate is done before starting jack detection. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 7420010..370ed54d 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -70,6 +70,7 @@ struct rt5665_priv { int jack_type; int irq_work_delay_time; unsigned int sar_adc_value; + bool calibration_done; }; static const struct reg_default rt5665_reg[] = { @@ -1305,6 +1306,11 @@ static void rt5665_jack_detect_handler(struct work_struct *work) usleep_range(10000, 15000); } + while (!rt5665->calibration_done) { + pr_debug("%s calibration not ready\n", __func__); + usleep_range(10000, 15000); + } + mutex_lock(&rt5665->calibrate_mutex); val = snd_soc_read(rt5665->codec, RT5665_AJD1_CTRL) & 0x0010; @@ -4695,6 +4701,7 @@ static void rt5665_calibrate(struct rt5665_priv *rt5665) regmap_write(rt5665->regmap, RT5665_ASRC_8, 0x0120); out_unlock: + rt5665->calibration_done = true; mutex_unlock(&rt5665->calibrate_mutex); } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: audio-graph-card: support external amplifier with DAPM widget
by Shawn Guo 30 Jun '17

30 Jun '17

From: Shawn Guo <shawn.guo(a)linaro.org> It's basically the result of discussion around how external amplifier controlled by GPIO should be supported by audio card, that uses generic machine driver like audio-graph-card. The patch series updates audio-graph-card bindings and driver to support DAPM widget and audio routing, and then based on that, creates an output driver widget with event to control the amplifier via GPIO. [1] https://www.spinics.net/lists/arm-kernel/msg589717.html Shawn Guo (2): ASoC: audio-graph-card: update bindings for amplifier support ASoC: audio-graph-card: add widgets and routing for external amplifier support .../devicetree/bindings/sound/audio-graph-card.txt | 5 +++ sound/soc/generic/audio-graph-card.c | 48 +++++++++++++++++++++- 2 files changed, 52 insertions(+), 1 deletion(-) -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: rsnd: make arrays path and cmd_case static const
by Colin King 30 Jun '17

30 Jun '17

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the arrays path and cmd_case on the stack but make them static const. Makes the object code smaller: Before: text data bss dec hex filename 2673 624 0 3297 ce1 sound/soc/sh/rcar/cmd.o After: text data bss dec hex filename 2398 768 0 3166 c5e sound/soc/sh/rcar/cmd.o Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/cmd.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 9a136d86e2a9..f1d4fb566892 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -31,7 +31,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); struct device *dev = rsnd_priv_to_dev(priv); u32 data; - u32 path[] = { + static const u32 path[] = { [1] = 1 << 0, [5] = 1 << 8, [6] = 1 << 12, @@ -71,7 +71,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, } else { struct rsnd_mod *src = rsnd_io_to_mod_src(io); - u8 cmd_case[] = { + static const u8 cmd_case[] = { [0] = 0x3, [1] = 0x3, [2] = 0x4, -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Add quirk override by module option
by Takashi Iwai 30 Jun '17

30 Jun '17

For making the development easier, add quirk module option to override the platform data setup. For example, a platform with inverted jack detection with jd_mode=2, pass the value 0x21 (0x1 = inv_jd1_1, 0x20 = jd_mode=2). It overrides the whole pdata fields, so pass it carefully. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/rt5645.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 308c22f5909a..22771e6ab3fa 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -34,6 +34,17 @@ #include "rl6231.h" #include "rt5645.h" +#define QUIRK_INV_JD1_1(q) ((q) & 1) +#define QUIRK_LEVEL_IRQ(q) (((q) >> 1) & 1) +#define QUIRK_IN2_DIFF(q) (((q) >> 2) & 1) +#define QUIRK_JD_MODE(q) (((q) >> 4) & 7) +#define QUIRK_DMIC1_DATA_PIN(q) (((q) >> 8) & 3) +#define QUIRK_DMIC2_DATA_PIN(q) (((q) >> 12) & 3) + +static unsigned int quirk = -1; +module_param(quirk, uint, 0444); +MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override"); + #define RT5645_DEVICE_ID 0x6308 #define RT5650_DEVICE_ID 0x6419 @@ -3679,6 +3690,15 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, else if (dmi_check_system(dmi_platform_gpd_win)) rt5645->pdata = gpd_win_platform_data; + if (quirk != -1) { + rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); + rt5645->pdata.level_trigger_irq = QUIRK_LEVEL_IRQ(quirk); + rt5645->pdata.inv_jd1_1 = QUIRK_INV_JD1_1(quirk); + rt5645->pdata.jd_mode = QUIRK_JD_MODE(quirk); + rt5645->pdata.dmic1_data_pin = QUIRK_DMIC1_DATA_PIN(quirk); + rt5645->pdata.dmic2_data_pin = QUIRK_DMIC2_DATA_PIN(quirk); + } + rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect", GPIOD_IN); -- 2.13.2

3 2

[alsa-devel] [asoc:topic/intel 33/35] sound/soc/soc-core.c:1961:2-3: Unneeded semicolon
by kbuild test robot 30 Jun '17

30 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8cffb5034ee033fa292cb1ab7a082b57ce2c740b commit: 98faf436ee05503e258ee934a9de8d8a3472b40f [33/35] ASoC: Drop invalid DMI fields when setting card long name from DMI info coccinelle warnings: (new ones prefixed by >>) >> sound/soc/soc-core.c:1961:2-3: Unneeded semicolon Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH] ASoC: rt5670: Fix arguments to set_sysclk
by Subhransu S. Prusty 30 Jun '17

30 Jun '17

set_sysclk expects pointer to snd_soc_codec. Fix it. Fixes: 6c28ce3c425e ("ASoC: rt5670: move set_sysclk to codec level") Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> CC: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 10fec7cdf1a1..0ec7985ed306 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2442,7 +2442,7 @@ static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int rt5670_set_codec_sysclk(struct snd_soc_dai *dai, int clk_id, +static int rt5670_set_codec_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: Drop invalid DMI fields when setting card long name from DMI info
by mengdong.lin＠linux.intel.com 30 Jun '17

30 Jun '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Sometimes DMI fields may be invalid and so can't give useful vendor, product or board info, such as "Type2 - Board Manufacturer" or "Type1 - TBD by OEM". Including such invalid DMI fileds may create silly card long name. So this patch creates a black list of invalid strings. And if a DMI field contains any string in this list, it will be excluded from the card long name. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cfa9cf1476f2..7107bc3a8a45 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -69,6 +69,21 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +/* If a DMI filed contain strings in this blacklist (e.g. + * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken + * as invalid and dropped when setting the card long name from DMI info. + */ +static const char * const dmi_blacklist[] = { + "To be filled by OEM", + "TBD by OEM", + "Default String", + "Default string", + "Board Manufacturer", + "Board Vendor Name", + "Board Product Name", + NULL, /* terminator */ +}; + /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -1934,6 +1949,22 @@ static void cleanup_dmi_name(char *name) name[j] = '\0'; } +/* Check if a DMI field is valid, i.e. not containing any string + * in the black list. + */ +static int is_dmi_valid(const char *field) +{ + int i = 0; + + while (dmi_blacklist[i]) { + if (strstr(field, dmi_blacklist[i])) + return 0; + i++; + }; + + return 1; +} + /** * snd_soc_set_dmi_name() - Register DMI names to card * @card: The card to register DMI names @@ -1976,17 +2007,18 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) /* make up dmi long name as: vendor.product.version.board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); - if (!vendor) { + if (!vendor || !is_dmi_valid(vendor)) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } + snprintf(card->dmi_longname, sizeof(card->snd_card->longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); product = dmi_get_system_info(DMI_PRODUCT_NAME); - if (product) { + if (product && is_dmi_valid(product)) { len = strlen(card->dmi_longname); snprintf(card->dmi_longname + len, longname_buf_size - len, @@ -2000,7 +2032,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); - if (product_version) { + if (product_version && is_dmi_valid(product_version)) { len = strlen(card->dmi_longname); snprintf(card->dmi_longname + len, longname_buf_size - len, @@ -2013,7 +2045,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) } board = dmi_get_system_info(DMI_BOARD_NAME); - if (board) { + if (board && is_dmi_valid(board)) { len = strlen(card->dmi_longname); snprintf(card->dmi_longname + len, longname_buf_size - len, -- 2.11.0

2 1

[alsa-devel] [PATCH alsa-lib] pcm: hw: fix to initialize function local variable
by Takashi Sakamoto 30 Jun '17

30 Jun '17

This commit is to fix below warning. pcm_hw.c: In function ‘snd1_pcm_hw_open_fd’: pcm_hw.c:955:33: warning: ‘mmap_control’ may be used uninitialized in this function [-Wmaybe-uninitialized] if (mmap_control == MAP_FAILED || mmap_control == NULL) { ^ pcm_hw.c:946:31: note: ‘mmap_control’ was declared here struct snd_pcm_mmap_control *mmap_control; ^~~~~~~~~~~~ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/pcm/pcm_hw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 64188b22..e0931577 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -904,6 +904,7 @@ static bool map_control_data(snd_pcm_hw_t *hw, struct snd_pcm_mmap_control *mmap_control; bool fallbacked; + mmap_control = MAP_FAILED; if (!force_fallback) { mmap_control = mmap(NULL, page_align(sizeof(*mmap_control)), PROT_READ|PROT_WRITE, MAP_FILE|MAP_SHARED, -- 2.11.0

2 1

[alsa-devel] [PATCH 0/8] Add ASoC support for AMD Stoney APUs
by Alex Deucher 30 Jun '17

30 Jun '17

This patch set updates the AMD GPU and Audio CoProcessor (ACP) audio drivers and the designware i2s driver for Stoney (ST). ST is an APU similar to Carrizo (CZ) which already has ACP audio support. The i2s controller and ACP audio DMA engine are part of the GPU and both need updating so I would like to upstream the whole patch set via one tree if possible. The current code is based on drm-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. The entire patch set can be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=drm-next-4.13-acp Thanks! Alex Akshu Agrawal (1): ASoC: AMD: Add machine driver for cz rt5650 Vijendar Mukunda (7): drm/amd/amdgpu: Added asic_type as ACP DMA driver platform data ASoC: dwc: Added a quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to dwc driver drm/amd/amdgpu: Added a dwc quirk for Stoney platform ASoC: AMD: added condition checks for CZ specific code ASoC: AMD: DMA driver changes for Stoney Platform ASoC: AMD: Buffer related changes for Stoney drm/amd/amdgpu: Disable ACP Power Gating for Stoney platform drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 54 +++++--- include/sound/designware_i2s.h | 1 + sound/soc/amd/Kconfig | 7 ++ sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-pcm-dma.c | 215 ++++++++++++++++++++++++-------- sound/soc/amd/acp-rt5645.c | 210 +++++++++++++++++++++++++++++++ sound/soc/amd/acp.h | 9 ++ sound/soc/dwc/dwc-i2s.c | 6 + 8 files changed, 432 insertions(+), 72 deletions(-) create mode 100644 sound/soc/amd/acp-rt5645.c -- 2.5.5

5 18

Re: [alsa-devel] Add codecs support for ALC215/ALC285/ALC289
by Takashi Iwai 30 Jun '17

30 Jun '17

On Fri, 30 Jun 2017 10:08:00 +0200, Kailang wrote: > > Hi Takashi, > > Attach patch was support our new codecs for ALC215 ALC285 ALC289. > Old 0x0285(ALC285) device ID was relpace by this. Applied, thanks. Takashi

1 0

Re: [alsa-devel] Remove device ID of ALC285
by Takashi Iwai 30 Jun '17

30 Jun '17

On Fri, 30 Jun 2017 09:39:15 +0200, Kailang wrote: > > Hi Takashi, > > 0x0285 Device ID was not used. > It just assigned for alias of ALC293 as backup. > So, remove it first. > Because our new codec will use this device ID. Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH alsa-lib 00/12] pcm: hw: optimization for v2.0.14 of PCM protocol/interface
by Takashi Sakamoto 30 Jun '17

30 Jun '17

Hi, This patchset is an optimization for version 2.0.14 of ALSA PCM protocol/interface. In this version, for some devices/drivers, applications are disallowed to map control data of runtime of PCM substream. On the other hand, mapping status data is still available. In current implementation of alsa-lib, execution of ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR is still done without enough care of the change. For example, even if status data is successfully mapped, the ioctl is executed to query the status data. This is inefficient. This patchset is to arrange the execution. Takashi Sakamoto (12): pcm: hw: fix to initialize function local variable pcm: hw: add a helper function just to query status/control data of PCM substream pcm: hw: use heler function to query status/control data after reading/writing PCM frames pcm: hw: use helper function to query status/control data after HW_PARAMS call pcm: hw: use helper function to query status/control data after PREPARE/RESET call pcm: hw: use helper function to query status/control data after REWIND/FORWARD call pcm: hw: use helper function to query status/control data for calculation of available space on PCM buffer pcm: hw: add a helper function to request hwsync without side-effects pcm: hw: add a helper function to issue appl_ptr without sub-effects pcm: hw: add a helper function to issue avail_min without sub-effects pcm: hw: remove superfluous code to call of SNDRV_PCM_IOCTL_SYNC_PTR in snd_pcm_hw_forward() pcm: hw: minor refactoring for initialization of control data src/pcm/pcm_hw.c | 109 ++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 75 insertions(+), 34 deletions(-) -- 2.11.0

1 13

[alsa-devel] [asoc:topic/drm 1/1] drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:88:21: error: storage size of 'of_ep' isn't known
by kbuild test robot 30 Jun '17

30 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/drm head: e3839bd6f56a291f00a4c3737eb15ca0344a82a9 commit: e3839bd6f56a291f00a4c3737eb15ca0344a82a9 [1/1] drm: dw-hdmi-i2s: add .get_dai_id callback for ALSA SoC config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout e3839bd6f56a291f00a4c3737eb15ca0344a82a9 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:85:42: warning: 'struct snd_soc_component' declared inside parameter list will not be visible outside of this definition or declaration static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, ^~~~~~~~~~~~~~~~~ drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c: In function 'dw_hdmi_i2s_get_dai_id': >> drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:88:21: error: storage size of 'of_ep' isn't known struct of_endpoint of_ep; ^~~~~ >> drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:91:8: error: implicit declaration of function 'of_graph_parse_endpoint' [-Werror=implicit-function-declaration] ret = of_graph_parse_endpoint(endpoint, &of_ep); ^~~~~~~~~~~~~~~~~~~~~~~ drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:88:21: warning: unused variable 'of_ep' [-Wunused-variable] struct of_endpoint of_ep; ^~~~~ drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c: At top level: >> drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:108:2: error: unknown field 'get_dai_id' specified in initializer .get_dai_id = dw_hdmi_i2s_get_dai_id, ^ >> drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:108:16: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] .get_dai_id = dw_hdmi_i2s_get_dai_id, ^~~~~~~~~~~~~~~~~~~~~~ drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c:108:16: note: (near initialization for 'dw_hdmi_i2s_ops.digital_mute') cc1: some warnings being treated as errors vim +88 drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c 79 80 dw_hdmi_audio_disable(hdmi); 81 82 hdmi_write(audio, HDMI_AUD_CONF0_SW_RESET, HDMI_AUD_CONF0); 83 } 84 > 85 static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, 86 struct device_node *endpoint) 87 { > 88 struct of_endpoint of_ep; 89 int ret; 90 > 91 ret = of_graph_parse_endpoint(endpoint, &of_ep); 92 if (ret < 0) 93 return ret; 94 95 /* 96 * HDMI sound should be located as reg = <2> 97 * Then, it is sound port 0 98 */ 99 if (of_ep.port == 2) 100 return 0; 101 102 return -EINVAL; 103 } 104 105 static struct hdmi_codec_ops dw_hdmi_i2s_ops = { 106 .hw_params = dw_hdmi_i2s_hw_params, 107 .audio_shutdown = dw_hdmi_i2s_audio_shutdown, > 108 .get_dai_id = dw_hdmi_i2s_get_dai_id, 109 }; 110 111 static int snd_dw_hdmi_probe(struct platform_device *pdev) --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [asoc:topic/drm 1/1] drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:88:21: warning: unused variable 'of_ep'
by kbuild test robot 30 Jun '17

30 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/drm head: e3839bd6f56a291f00a4c3737eb15ca0344a82a9 commit: e3839bd6f56a291f00a4c3737eb15ca0344a82a9 [1/1] drm: dw-hdmi-i2s: add .get_dai_id callback for ALSA SoC config: tile-allmodconfig (attached as .config) compiler: tilegx-linux-gcc (GCC) 4.6.2 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout e3839bd6f56a291f00a4c3737eb15ca0344a82a9 # save the attached .config to linux build tree make.cross ARCH=tile All warnings (new ones prefixed by >>): drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:86:14: warning: 'struct snd_soc_component' declared inside parameter list [enabled by default] drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:86:14: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default] drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c: In function 'dw_hdmi_i2s_get_dai_id': drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:88:21: error: storage size of 'of_ep' isn't known drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:91:2: error: implicit declaration of function 'of_graph_parse_endpoint' >> drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:88:21: warning: unused variable 'of_ep' drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c: At top level: drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:108:2: error: unknown field 'get_dai_id' specified in initializer drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:108:2: warning: initialization from incompatible pointer type [enabled by default] drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c:108:2: warning: (near initialization for 'dw_hdmi_i2s_ops.digital_mute') [enabled by default] cc1: some warnings being treated as errors vim +/of_ep +88 drivers/gpu//drm/bridge/synopsys/dw-hdmi-i2s-audio.c 80 dw_hdmi_audio_disable(hdmi); 81 82 hdmi_write(audio, HDMI_AUD_CONF0_SW_RESET, HDMI_AUD_CONF0); 83 } 84 85 static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, > 86 struct device_node *endpoint) 87 { > 88 struct of_endpoint of_ep; 89 int ret; 90 91 ret = of_graph_parse_endpoint(endpoint, &of_ep); --- 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: constify attribute_group structures.
by Arvind Yadav 29 Jun '17

29 Jun '17

attribute_groups are not supposed to change at runtime. All functions working with attribute_groups provided by <linux/device.h> work with const attribute_group. So mark the non-const structs as const. File size before: text data bss dec hex filename 995 608 0 1603 643 sound/pci/hda/hda_sysfs.o File size After adding 'const': text data bss dec hex filename 1035 544 0 1579 62b sound/pci/hda/hda_sysfs.o Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/pci/hda/hda_sysfs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 9739fce..9b7efec 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -761,7 +761,7 @@ int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf) NULL }; -static struct attribute_group hda_dev_attr_group = { +static const struct attribute_group hda_dev_attr_group = { .attrs = hda_dev_attrs, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: pcm: constify attribute_group structures.
by Arvind Yadav 29 Jun '17

29 Jun '17

attribute_groups are not supposed to change at runtime. All functions working with attribute_groups provided by <linux/device.h> work with const attribute_group. So mark the non-const structs as const. File size before: text data bss dec hex filename 9781 240 8 10029 272d sound/core/pcm.o File size After adding 'const': text data bss dec hex filename 9813 176 8 9997 270d sound/core/pcm.o Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/core/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8e980aa..ee3e8db0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1056,7 +1056,7 @@ static ssize_t show_pcm_class(struct device *dev, NULL }; -static struct attribute_group pcm_dev_attr_group = { +static const struct attribute_group pcm_dev_attr_group = { .attrs = pcm_dev_attrs, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix doubly initialization of i915 component
by Takashi Iwai 29 Jun '17

29 Jun '17

In the commit fcc88d91cd36 ("ALSA: hda - Bind with i915 component before codec binding"), the binding with i915 audio component is moved to be performed always at probing the controller. This fixed the potential problems on IVB, but now it brought another issue on HSW and BDW. These two platforms give two individual HD-audio controllers, one for the analog codec on PCH and another for HDMI over gfx. Since I decided to take a lazy path to check only AZX_DRIVER_PCH type in the commit above, now both controllers try to bind with i915, and you see a kernel WARNING. This patch tries to address it again properly. Now a new DCAPS bit, AZX_DCAPS_I915_COMPONENT, is introduced for indicating the binding with i915 component in addition to the existing I915_POWERWELL bit flag. Each PCI entry has to give this new flag if it requires the binding with i915 component. For HSW/BDW PCH (i.e. the ones defined by AZX_DCAPS_INTEL_PCH) doesn't contain AZX_DCAPS_I915_COMPONENT bit while others have it. While we're at it, add parentheses around the bit flag check for avoiding possible compiler warnings, too. The bug was spotted by Intel CI tests. Fixes: fcc88d91cd36 ("ALSA: hda - Bind with i915 component before codec binding") Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196219 Reported-by: Martin Peres <martin.peres(a)free.fr> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.h | 6 +++++- sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++------------------ 2 files changed, 27 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 35a9ab2cac46..a68e75b00ea3 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -32,7 +32,11 @@ #define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ #define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ #define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ -/* 13 unused */ +#ifdef CONFIG_SND_HDA_I915 +#define AZX_DCAPS_I915_COMPONENT (1 << 13) /* bind with i915 gfx */ +#else +#define AZX_DCAPS_I915_COMPONENT 0 /* NOP */ +#endif /* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a157582b8f2c..03e34edc8f24 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -293,38 +293,43 @@ enum { (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_SNOOP_TYPE(SCH)) -/* PCH up to IVB; no runtime PM */ +/* PCH up to IVB; no runtime PM; bind with i915 gfx */ #define AZX_DCAPS_INTEL_PCH_NOPM \ - (AZX_DCAPS_INTEL_PCH_BASE) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT) /* PCH for HSW/BDW; with runtime PM */ +/* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ - AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ + AZX_DCAPS_I915_POWERWELL | AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ - AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ + AZX_DCAPS_I915_POWERWELL | AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_BAYTRAIL \ - (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT |\ + AZX_DCAPS_I915_POWERWELL) #define AZX_DCAPS_INTEL_BRASWELL \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_I915_COMPONENT | AZX_DCAPS_I915_POWERWELL) #define AZX_DCAPS_INTEL_SKYLAKE \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT |\ AZX_DCAPS_I915_POWERWELL) #define AZX_DCAPS_INTEL_BROXTON \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT |\ AZX_DCAPS_I915_POWERWELL) /* quirks for ATI SB / AMD Hudson */ @@ -1008,7 +1013,7 @@ static int azx_suspend(struct device *dev) if (chip->msi) pci_disable_msi(chip->pci); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) && hda->need_i915_power) snd_hdac_display_power(bus, false); @@ -1114,7 +1119,7 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) && hda->need_i915_power) snd_hdac_display_power(azx_bus(chip), false); @@ -1380,8 +1385,7 @@ static int azx_free(struct azx *chip) if (hda->need_i915_power) snd_hdac_display_power(bus, false); } - if (chip->driver_type == AZX_DRIVER_PCH || - (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)) + if (chip->driver_type & AZX_DCAPS_I915_COMPONENT) snd_hdac_i915_exit(bus); kfree(hda); @@ -2199,8 +2203,7 @@ static int azx_probe_continue(struct azx *chip) hda->probe_continued = 1; /* bind with i915 if needed */ - if (chip->driver_type == AZX_DRIVER_PCH || - (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)) { + if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT) { err = snd_hdac_i915_init(bus); if (err < 0) { /* if the controller is bound only with HDMI/DP @@ -2214,7 +2217,8 @@ static int azx_probe_continue(struct azx *chip) goto out_free; } else { /* don't bother any longer */ - chip->driver_caps &= ~AZX_DCAPS_I915_POWERWELL; + chip->driver_caps &= + ~(AZX_DCAPS_I915_COMPONENT | AZX_DCAPS_I915_POWERWELL); } } } @@ -2279,7 +2283,7 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_autosuspend(&pci->dev); out_free: - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + if ((chip->driver_caps & AZX_DCAPS_I915_POWERWELL) && !hda->need_i915_power) snd_hdac_display_power(bus, false); -- 2.13.2

1 0

[alsa-devel] Make snd_card_free() a no-op again for NULL ptr argument
by Delio Brignoli 29 Jun '17

29 Jun '17

Hello Takashi, Commit “ALSA: Use standard device refcount for card accounting” <https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/s…> changed the behaviour of snd_card_free() in case of a NULL ptr argument. Before f24640648186b59bd39f8cd640d631bdb61a3197 snd_card_free() was a no-op in case of a NULL ptr argument as I would expect for a *_free() function. Currently passing a NULL ptr results in a NULL ptr dereference. I believe the previous behaviour is worth restoring. If you agree I’ll follow up with a patch. Thank you. — Delio

2 2

Re: [alsa-devel] Add headset mode for new Dell machine
by Takashi Iwai 29 Jun '17

29 Jun '17

On Thu, 29 Jun 2017 09:43:22 +0200, Kailang wrote: > > Hi Takashi, > > Attach patch was for Add headset mode for new Dell machine. > Many Thanks. Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH 0/4] ASoC: rt5645: GPD Win jack support
by Takashi Iwai 29 Jun '17

29 Jun '17

Hi, this is a patchset containing the fixes for the inverted jack detection on RT5645 codec and a quirk for GPD Win device. Possibly the newer one, GPD Pocket, would work with the same quirk. First three are from Bard, and the last one is the suggested by Hans. You may notice the ugliness there, but unfortunately this is the only way to identify the machine. Blame BIOS programmer. thanks, Takashi === Bard Liao (3): ASoC: rt5645: read jd1_1 status for jd detection ASoC: rt5645: rename jd_invert flag in platform data ASoC: rt5645: add inv_jd1_1 flag Takashi Iwai (1): ASoC: rt5645: Add jack detection workaround for GPD Win include/sound/rt5645.h | 6 ++-- sound/soc/codecs/rt5645.c | 72 +++++++++++++++++++++++++++++------------------ 2 files changed, 49 insertions(+), 29 deletions(-) -- 2.13.2

5 12

[alsa-devel] [PATCH] ASoC: Drop invalid DMI fields when setting card long name from DMI info
by mengdong.lin＠linux.intel.com 29 Jun '17

29 Jun '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Sometimes DMI fields may be invalid and so can't give useful vendor, product or board info, such as "Type2 - Board Manufacturer" or "Type1 - TBD by OEM". Including such invalid DMI fileds may create silly card long name. So this patch creates a black list of invalid strings. And if a DMI field contains any string in this list, it will be excluded from the card long name. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cfa9cf1476f2..419b788fd801 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -69,6 +69,20 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +/* If a DMI filed contain strings in this blacklist (e.g. + * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken + * as invalid and dropped when setting the card long name from DMI info. + */ +static const char * const dmi_blacklist[] = { + "To be filled by OEM", + "TBD by OEM", + "Default String", + "Board Manufacturer", + "Board Vendor Name", + "Board Product Name", + NULL, /* terminator */ +}; + /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -1934,6 +1948,22 @@ static void cleanup_dmi_name(char *name) name[j] = '\0'; } +/* Check if a DMI field is valid, i.e. not containing any string + * in the black list. + */ +static int is_dmi_valid(const char *field) +{ + int i = 0; + + while (dmi_blacklist[i]) { + if (strstr(field, dmi_blacklist[i])) + return 0; + i++; + }; + + return 1; +} + /** * snd_soc_set_dmi_name() - Register DMI names to card * @card: The card to register DMI names @@ -1976,17 +2006,18 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) /* make up dmi long name as: vendor.product.version.board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); - if (!vendor) { + if (!vendor || !is_dmi_valid(vendor)) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } + snprintf(card->dmi_longname, sizeof(card->snd_card->longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); product = dmi_get_system_info(DMI_PRODUCT_NAME); - if (product) { + if (product && is_dmi_valid(product)) { len = strlen(card->dmi_longname); snprintf(card->dmi_longname + len, longname_buf_size - len, @@ -2000,7 +2031,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); - if (product_version) { + if (product_version && is_dmi_valid(product_version)) { len = strlen(card->dmi_longname); snprintf(card->dmi_longname + len, longname_buf_size - len, @@ -2013,7 +2044,7 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) } board = dmi_get_system_info(DMI_BOARD_NAME); - if (board) { + if (board && is_dmi_valid(board)) { len = strlen(card->dmi_longname); snprintf(card->dmi_longname + len, longname_buf_size - len, -- 2.11.0

4 3

[alsa-devel] [PATCH] ASoC: nau8825: change crosstalk-bypass property to bool type
by John Hsu 29 Jun '17

29 Jun '17

The property type of "nuvoton,crosstalk-bypass" changes to boolean. The document is updated as well. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- Documentation/devicetree/bindings/sound/nau8825.txt | 3 +++ sound/soc/codecs/nau8825.c | 8 ++++---- 2 files changed, 7 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index d337423..2f5e973 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,6 +69,8 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,crosstalk-bypass: make crosstalk function bypass if set. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock @@ -96,6 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-bypass; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index a8c7a55..46a30ea 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2440,6 +2440,8 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); + dev_dbg(dev, "crosstalk-bypass: %d\n", + nau8825->xtalk_bypass); } static int nau8825_read_device_properties(struct device *dev, @@ -2504,10 +2506,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", - &nau8825->xtalk_bypass); - if (ret) - nau8825->xtalk_bypass = 1; + nau8825->xtalk_bypass = device_property_read_bool(dev, + "nuvoton,crosstalk-bypass"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- 2.6.4

1 0

[alsa-devel] [asoc:topic/rt5670 3/3] sound/soc//codecs/rt5670.c:2775:2: warning: initialization from incompatible pointer type
by kbuild test robot 29 Jun '17

29 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5670 head: 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 commit: 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 [3/3] ASoC: rt5670: move set_sysclk to codec level config: xtensa-allmodconfig (attached as .config) compiler: xtensa-linux-gcc (GCC) 4.9.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 # save the attached .config to linux build tree make.cross ARCH=xtensa All warnings (new ones prefixed by >>): sound/soc//codecs/rt5670.c: In function 'rt5670_set_codec_sysclk': sound/soc//codecs/rt5670.c:2448:57: error: 'codec' undeclared (first use in this function) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); ^ sound/soc//codecs/rt5670.c:2448:57: note: each undeclared identifier is reported only once for each function it appears in sound/soc//codecs/rt5670.c: At top level: >> sound/soc//codecs/rt5670.c:2775:2: warning: initialization from incompatible pointer type .set_sysclk = rt5670_set_codec_sysclk, ^ sound/soc//codecs/rt5670.c:2775:2: warning: (near initialization for 'soc_codec_dev_rt5670.set_sysclk') vim +2775 sound/soc//codecs/rt5670.c 2759 .channels_min = 1, 2760 .channels_max = 2, 2761 .rates = RT5670_STEREO_RATES, 2762 .formats = RT5670_FORMATS, 2763 }, 2764 .ops = &rt5670_aif_dai_ops, 2765 }, 2766 }; 2767 2768 static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { 2769 .probe = rt5670_probe, 2770 .remove = rt5670_remove, 2771 .suspend = rt5670_suspend, 2772 .resume = rt5670_resume, 2773 .set_bias_level = rt5670_set_bias_level, 2774 .idle_bias_off = true, > 2775 .set_sysclk = rt5670_set_codec_sysclk, 2776 .component_driver = { 2777 .controls = rt5670_snd_controls, 2778 .num_controls = ARRAY_SIZE(rt5670_snd_controls), 2779 .dapm_widgets = rt5670_dapm_widgets, 2780 .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), 2781 .dapm_routes = rt5670_dapm_routes, 2782 .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), 2783 }, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: nau8825: devicetree for crosstalk bypass
by John Hsu 29 Jun '17

29 Jun '17

Update devicetree document for crosstalk function bypass. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- Documentation/devicetree/bindings/sound/nau8825.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index d337423..bf4451b 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,6 +69,8 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,crosstalk-bypass: make crosstalk function bypass if set 1. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock @@ -96,6 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-bypass = <1>; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; -- 2.6.4

3 2

[alsa-devel] [asoc:topic/rt5670 3/3] sound/soc/codecs/rt5670.c:2448:57: error: 'codec' undeclared
by kbuild test robot 29 Jun '17

29 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5670 head: 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 commit: 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 [3/3] ASoC: rt5670: move set_sysclk to codec level config: x86_64-randconfig-x002-201726 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 6c28ce3c425e32d372c7c6ee98d3c3711f13ad69 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/codecs/rt5670.c: In function 'rt5670_set_codec_sysclk': >> sound/soc/codecs/rt5670.c:2448:57: error: 'codec' undeclared (first use in this function) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); ^~~~~ sound/soc/codecs/rt5670.c:2448:57: note: each undeclared identifier is reported only once for each function it appears in sound/soc/codecs/rt5670.c: At top level: >> sound/soc/codecs/rt5670.c:2775:16: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] .set_sysclk = rt5670_set_codec_sysclk, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/rt5670.c:2775:16: note: (near initialization for 'soc_codec_dev_rt5670.set_sysclk') cc1: some warnings being treated as errors vim +/codec +2448 sound/soc/codecs/rt5670.c 5e8351de Bard Liao 2014-06-30 2442 return 0; 5e8351de Bard Liao 2014-06-30 2443 } 5e8351de Bard Liao 2014-06-30 2444 6c28ce3c Bard Liao 2017-06-28 2445 static int rt5670_set_codec_sysclk(struct snd_soc_dai *dai, int clk_id, 6c28ce3c Bard Liao 2017-06-28 2446 int source, unsigned int freq, int dir) 5e8351de Bard Liao 2014-06-30 2447 { 5e8351de Bard Liao 2014-06-30 @2448 struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); 5e8351de Bard Liao 2014-06-30 2449 unsigned int reg_val = 0; 5e8351de Bard Liao 2014-06-30 2450 5e8351de Bard Liao 2014-06-30 2451 switch (clk_id) { :::::: The code at line 2448 was first introduced by commit :::::: 5e8351de740d9eff26cc146a6591a4e7517496b0 ASoC: add RT5670 CODEC driver :::::: TO: Bard Liao <bardliao(a)realtek.com> :::::: CC: Mark Brown <broonie(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 0/5] ASoC: add HDMI sound support for Codec
by Kuninori Morimoto 28 Jun '17

28 Jun '17

Hi Mark, Archit These supports HDMI sound for Codec. Now, SYNOPSYS dw-hdmi-i2s-audio driver is supporting HDMI I2S and is using hdmi-codec driver. Now, ALSA SoC supports OF-graph DT style, this means V4L2/ALSA SoC can share same DT bindings. But 1 issue is that ALSA SoC side needs to adjust port ID <-> Sound DAI matching. 1) - 4) are for these purpose. 1) - 4) : ASoC side patch for hdmi-codec 5) : V4L2 side patch for SYNOPSYS dw-hdmi Kuninori Morimoto (5): 1) ASoC: add snd_soc_get_dai_id() function 2) ASoC: simple-card-utils: support snd_soc_get_dai_id() 3) ASoC: hdmi-codec: remove multi detection support 4) ASoC: hdmi-codec: add .get_dai_id support 5) drm: dw-hdmi-i2s: add .get_dai_id callback for ALSA SoC .../bindings/display/bridge/renesas,dw-hdmi.txt | 9 ++- .../gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 21 +++++ include/sound/hdmi-codec.h | 9 +++ include/sound/soc.h | 3 + sound/soc/codecs/hdmi-codec.c | 91 +++------------------- sound/soc/generic/simple-card-utils.c | 9 +++ sound/soc/soc-core.c | 37 +++++++++ 7 files changed, 98 insertions(+), 81 deletions(-) -- 1.9.1

3 16

[alsa-devel] [PATCH] ASoC: rt5645: del btn_check_timer on remove
by Hans de Goede 28 Jun '17

28 Jun '17

The timer may still be running when rt5645_i2c_remove() gets called, call del_timer_sync() to make sure it is stopped. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d8e7487482c2..e6a3ad17437c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3915,6 +3915,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); + del_timer_sync(&rt5645->btn_check_timer); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.13.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5670: fix wrong audio route
by Bard Liao 28 Jun '17

28 Jun '17

When we select "IF1_ADC4" for IF1 ADC Muxes, it is actually connected to "TxDP_ADC" widget. This patch fixes the audio route and remove the unexisting "IF1_ADC4" widget. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index e27c5a4..d95d2e6 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1717,7 +1717,6 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -2086,13 +2085,13 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, - { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "TxDP_ADC" }, { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, - { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "TxDP_ADC" }, { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, -- 2.7.4

2 5

[alsa-devel] [PATCH] ASoC: rt5651: remove unexisting Muxes
by Bard Liao 28 Jun '17

28 Jun '17

These MUXes are unexisting. So, remove them. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 44 +------------------------------------------- 1 file changed, 1 insertion(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f5d3415..db05b60 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, RT5651_R_MUTE_SFT, 1, 1); -/* INL/R source */ -static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl_enum, RT5651_INL1_INR1_VOL, - RT5651_INL_SEL_SFT, rt5651_inl_src); - -static const struct snd_kcontrol_new rt5651_inl1_mux = - SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); - -static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr1_enum, RT5651_INL1_INR1_VOL, - RT5651_INR_SEL_SFT, rt5651_inr1_src); - -static const struct snd_kcontrol_new rt5651_inr1_mux = - SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); - -static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl2_enum, RT5651_INL2_INR2_VOL, - RT5651_INL_SEL_SFT, rt5651_inl2_src); - -static const struct snd_kcontrol_new rt5651_inl2_mux = - SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); - -static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr2_enum, RT5651_INL2_INR2_VOL, - RT5651_INR_SEL_SFT, rt5651_inr2_src); - -static const struct snd_kcontrol_new rt5651_inr2_mux = - SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); - - /* Stereo ADC source */ static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; @@ -955,11 +917,7 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), + /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: rt5645: enable speaker protection features
by Bard Liao 28 Jun '17

28 Jun '17

This patch is uploaded for enabling the speaker protection features of the audio codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * modify register value by regmap_update_bits instead of regmap_register_patch --- sound/soc/codecs/rt5645.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 162044d..627e801 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -59,7 +59,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x1c, 0xfd70}, {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, @@ -3730,6 +3730,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); } + regmap_update_bits(rt5645->regmap, RT5645_CLSD_OUT_CTRL, 0xc0, 0xc0); + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); -- 2.7.4

2 1

[alsa-devel] [PATCH v13 00/14] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 28 Jun '17

28 Jun '17

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds a mux for the capture source and the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (14): sun4i-codec: Add MIC2 Pre-Amplifier, Mic2 input, Mic2 routes. sun4i-codec: Add Mic Playback Volume. sun4i-codec: Add support for extra controls to struct sun4i_codec_quirks and use them. sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume. sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls. sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch. sun4i-codec: Add FM Playback Volume. sun4i-codec: Add FM Left, FM Right, FM Playback Switch. sun4i-codec: Add Line Playback Volume. sun4i-codec: Add Line Boost Volume. sun4i-codec: Add Line Right, Line Left, Line Playback Switch. sun4i-codec: Add Differential Line Source. sun4i-codec: Add Left Capture Select, Right Capture Select. sun4i-codec: Add Capture Volume. sound/soc/sunxi/sun4i-codec.c | 233 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 210 insertions(+), 23 deletions(-) -- 2.1.4

5 21

[alsa-devel] [PATCH] ASoC: arizona: Add DAC Volume Limit controls
by Charles Keepax 28 Jun '17

28 Jun '17

Add controls that allow the limiting of the volume coming from the DACs on the Arizona class devices, note that wm8998 does not have this feature. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs47l24.c | 8 ++++++++ sound/soc/codecs/wm5102.c | 17 +++++++++++++++++ sound/soc/codecs/wm5110.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm8997.c | 14 ++++++++++++++ 4 files changed, 59 insertions(+) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 47e6fddef92b..a972e63bdda5 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -78,6 +78,7 @@ static int cs47l24_adsp_power_ev(struct snd_soc_dapm_widget *w, static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -203,6 +204,13 @@ SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1fe358e6be61..7a295146524b 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -42,6 +42,7 @@ struct wm5102_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -915,6 +916,22 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_2L, + ARIZONA_DAC_VOLUME_LIMIT_2R, ARIZONA_OUT2L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("EPOUT Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_3L, + ARIZONA_DAC_VOLUME_LIMIT_3R, ARIZONA_OUT3L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_5L, + ARIZONA_DAC_VOLUME_LIMIT_5R, ARIZONA_OUT5L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), SOC_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 1bc942152eff..f7cf9f7dffa6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -575,6 +575,7 @@ static int wm5110_in_ev(struct snd_soc_dapm_widget *w, static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -855,6 +856,25 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_2L, + ARIZONA_DAC_VOLUME_LIMIT_2R, ARIZONA_OUT2L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_3L, + ARIZONA_DAC_VOLUME_LIMIT_3R, ARIZONA_OUT3L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_5L, + ARIZONA_DAC_VOLUME_LIMIT_5R, ARIZONA_OUT5L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT2 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_6L, + ARIZONA_DAC_VOLUME_LIMIT_6R, ARIZONA_OUT6L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 49401a8aae64..753a1c6a1484 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -40,6 +40,7 @@ struct wm8997_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -287,6 +288,19 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("EPOUT Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_3L, + ARIZONA_DAC_VOLUME_LIMIT_3R, ARIZONA_OUT3L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_5L, + ARIZONA_DAC_VOLUME_LIMIT_5R, ARIZONA_OUT5L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), SOC_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), -- 2.11.0

2 2

[alsa-devel] [PATCH 0/4] ALSA: HD-audio HDMI binding fixes
by Takashi Iwai 28 Jun '17

28 Jun '17

Hi, this is a patch serires to address the issue recently reported by Intel gfx people. The first patch is the critical one, and it'll go to for-linus branch, possibly for 4.12, while others are some following works and cleanups. Takashi === Takashi Iwai (4): ALSA: hda - Fix endless loop of codec configure ALSA: hda - Skip card registration when no codec is found ALSA: hda - Bind with i915 component before codec binding ALSA: hda - Minor code refactoring for Intel HDMI codec parsers sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_controller.c | 11 ++++- sound/pci/hda/hda_intel.c | 4 +- sound/pci/hda/patch_hdmi.c | 94 +++++++++++++++--------------------------- 4 files changed, 47 insertions(+), 64 deletions(-) -- 2.13.2

1 5

[alsa-devel] [PATCH] ASoC: stm32: sai: remove some stray tabs
by Dan Carpenter 28 Jun '17

28 Jun '17

This line was accidentally indented too far. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ba3fdc777ed8..90d439613899 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -631,7 +631,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div); mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); - cr1 = SAI_XCR1_MCKDIV_SET(div); + cr1 = SAI_XCR1_MCKDIV_SET(div); ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n");

1 0

[alsa-devel] [PATCH] ALSA: Fix forgotten dependency fix for tristate OSS sequencer kconfig
by Takashi Iwai 28 Jun '17

28 Jun '17

In the commit 3d774d5ef066 ("ALSA: seq: Allow the tristate build of OSS emulation") we changed CONFIG_SND_SEQUENCER_OSS to tristate, but a couple of places were forgotten, namely, opl3 and emux Makefile. These contain the line like snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o and this doesn't work any longer as expected because snd-opl3-synth can be built-in while CONFIG_SND_SEQUENCER_OSS=m. This patch fixes these places to build properly for the new kconfig dependency. In the end, we had to use ifneq() to satisfy the requirement. It's a bit ugly, but lesser evil. Fixes: 3d774d5ef066 ("ALSA: seq: Allow the tristate build of OSS emulation") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/opl3/Makefile | 4 +++- sound/synth/emux/Makefile | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 7f2c2a10c4e5..d72b1e7b51c4 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -5,7 +5,9 @@ snd-opl3-lib-objs := opl3_lib.o opl3_synth.o snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o -snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o +ifneq ($(CONFIG_SND_SEQUENCER_OSS),) +snd-opl3-synth-y += opl3_oss.o +endif obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile index 4599108452fd..d1bac923eb1b 100644 --- a/sound/synth/emux/Makefile +++ b/sound/synth/emux/Makefile @@ -6,6 +6,8 @@ snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ emux_effect.o emux_hwdep.o soundfont.o snd-emux-synth-$(CONFIG_SND_PROC_FS) += emux_proc.o -snd-emux-synth-$(CONFIG_SND_SEQUENCER_OSS) += emux_oss.o +ifneq ($(CONFIG_SND_SEQUENCER_OSS),) +snd-emux-synth-y += emux_oss.o +endif obj-$(CONFIG_SND_SYNTH_EMUX) += snd-emux-synth.o -- 2.13.2

1 0

[alsa-devel] ALSA core info race condition
by b_lkasam＠codeaurora.org 28 Jun '17

28 Jun '17

hi ALSA team, there is a race condition in below API when accessing list API. In file sound/core/info.c: Added below patch to avoid list access of same parent node by two threads at same time causing list_debug crash. diff --git a/sound/core/info.c b/sound/core/info.c index b5158b5..c1fd671 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -747,8 +747,11 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent) INIT_LIST_HEAD(&entry->children); INIT_LIST_HEAD(&entry->list); entry->parent = parent; - if (parent) + if (parent) { + mutex_lock(&parent->access); list_add_tail(&entry->list, &parent->children); + mutex_unlock(&parent->access); + } return entry; } Please check above logic looks fine, and help comment accordingly. Thanks Kasam

2 3

[alsa-devel] [PATCH] ALSA: hda - set input_path bitmap to zero after moving it to new place
by Hui Wang 28 Jun '17

28 Jun '17

Recently we met a problem, the codec has valid adcs and input pins, and they can form valid input paths, but the driver does not build valid controls for them like "Mic boost", "Capture Volume" and "Capture Switch". Through debugging, I found the driver needs to shrink the invalid adcs and input paths for this machine, so it will move the whole column bitmap value to the previous column, after moving it, the driver forgets to set the original column bitmap value to zero, as a result, the driver will invalidate the path whose index value is the original colume bitmap value. After executing this function, all valid input paths are invalidated by a mistake, there are no any valid input paths, so the driver won't build controls for them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- the alsa-info.txt of this machine (without this patch) is pasted here: http://paste.ubuntu.com/24967758/ sound/pci/hda/hda_generic.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 557ecfc..28e265a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3216,6 +3216,7 @@ static int check_dyn_adc_switch(struct hda_codec *codec) spec->input_paths[i][nums]); spec->input_paths[i][nums] = spec->input_paths[i][n]; + spec->input_paths[i][n] = 0; } } nums++; -- 1.9.1

2 1

[alsa-devel] [PATCH v3 0/3] Add support for es8323 and hdmi audio to the firefly-rk3288
by Romain Perier 27 Jun '17

27 Jun '17

This set of patches enables support for the es8323 codecs on the firefly-rk3288. It adds the required DT definitions to link rockchip-i2s to the es8388 analog output. As the support for HDMI audio is also built-in in the corresponding ASoC machine driver, HDMI audio output is also enabled. Changes in v3: - Synced DT with the one of the Rock2, the properties have been lexically re-ordered in the right order, I have splitted the headphone pinctrl in two: phone_ctl and hp_det Changes in v2: - Adapted DT for the firefly-rk3288 to the ASoC machine driver rk3288-hdmi-analog - Added patch 02/03 to this series - Re-wrote this cover letter Romain Perier (3): SoC: es8328-i2c: Add compatible for ES8323 arm: dts: rockchip: Don't set otp-gpio pinctrl by default in rk3288.dtsi arm: dts: Add support for ES8323 to the Firefly-RK3288 Documentation/devicetree/bindings/sound/es8328.txt | 5 ++- arch/arm/boot/dts/rk3288-evb.dtsi | 4 +++ arch/arm/boot/dts/rk3288-firefly.dtsi | 38 ++++++++++++++++++++++ arch/arm/boot/dts/rk3288-veyron.dtsi | 4 +++ arch/arm/boot/dts/rk3288.dtsi | 4 --- sound/soc/codecs/es8328-i2c.c | 2 ++ 6 files changed, 52 insertions(+), 5 deletions(-) -- 2.11.0

3 20

[alsa-devel] [PATCH 0/2] ASoC: stm32: add SPDIFRX driver
by olivier moysan 27 Jun '17

27 Jun '17

This patch-set handles the SPDIFRX on STM32 platforms. The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with IEC-60958 and IEC-61937 standards. SPDIFRX uses two DMA channels: - one DMA channel for S/PDIF data stream. - one DMA channel for control flow (channel status and user data bits). DMA is used for control flow instead of interrupts to get better performances, as channel status and user data are retrieved byte per byte. Note: This patch-set depends on "change SAI configuration flag" patch, which introduces a change on Kconfig file. olivier moysan (2): dt-bindings: Document the STM32 SPDIFRX interface ASoC: stm32: Add SPDIFRX support .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 ++ sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_spdifrx.c | 998 +++++++++++++++++++++ sound/soc/stm/stm32_spdifrx.h | 224 +++++ 5 files changed, 1292 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 sound/soc/stm/stm32_spdifrx.c create mode 100644 sound/soc/stm/stm32_spdifrx.h -- 1.9.1

3 6

[alsa-devel] [PATCH v3 0/2] ALSA: Add PCM USER_PVERSION ioctl
by Takashi Iwai 27 Jun '17

27 Jun '17

Hi, this is a revised patchset that is applied on top of the current for-next branch, in order to optimize the appl_ptr sync on x86-64, adapting the alsa-lib side change by Sakamoto-san's patchset. Basically it adds a new ioctl for receiving the supported PCM protocol version from user-space side, and change the mmap code to allow only disabling the PCM status record depending on the supported protocol version. The former patch is identical with the previous version, only the latter one differs. The corresponding alsa-lib patch will follow. thanks, Takashi === Takashi Iwai (2): ALSA: pcm: Add an ioctl to specify the supported protocol version ALSA: pcm: Disable only control mmap for explicit appl_ptr sync include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 3 ++- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 29 ++++++++++++++++++++++++----- 4 files changed, 28 insertions(+), 6 deletions(-) -- 2.13.2

2 5

[alsa-devel] [PATCH v2 alsa-lib 0/6] pcm: hw: maintain fallback mode for status/control data separately
by Takashi Sakamoto 27 Jun '17

27 Jun '17

Hi, This patchset is a revised version for a part of my previous RFC: [alsa-devel] [alsa-lib][RFC][PATCH 0/9] pcm: handle status/control mapping independently http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122130.html As I described, we found an advantage to handle results of page mapping for status/control data separately in a discussion about ALSA PCM protocol v2.0.14. In this version, when kernel-land drivers support SNDRV_PCM_INFO_SYNC_APPLPTR for a certain design of hardware, userspace applications are expected to issue current position of applptr for several merits. This should be performed even if applications are programmed to run with mmap operation for PCM buffer. In the above discussion, the way to take applications to perform it is one of important issues. On platforms without enough cache coherency support such as ARM, this doesn't matter. On the platforms, ALSA PCM core already disallows any of mappings for the status/control data of runtime of PCM substream. For this case, userland stuffs were already programmed to go to fallback mode, in which ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR is used to issue/query the status/control data. Even if handling PCM frames on PCM buffer by mmap operation, the position of applptr is issues into ALSA PCM core by a call of ioctl(2) with the command. As a solution for old stuffs in userland, the above mechanism is used for cache coherent architecture such as x86. A patch was already merged into kernel land[1]. This patch changes behaviours of ALSA PCM core to disallow userland to map the status/control data according to drivers' support for SNDRV_PCM_INFO_SYNC_APPLPTR. However, disallowing mappings for both of status/control data has a slight overhead to increase calls of ioctl(2), than usual operation on x86. As I reported[2]. Furthermore, it's clear that current proposals make no sense for drivers because 'struct snd_pcm_ops.ack' in driver implementation is not called expectedly due to a commit[4]. Even if the applptr is issued by ioctl(2), the callback is skipped as long as mapped status includes the updated value for the applptr. In the above discussion, some solutions were suggested[5]. One of them is to enable map of the status data and disable map of the control data. This idea can archive both of the aim of v2.0.14 protocol and reduction of calls for status data, for newer userland stuffs. This patchset for alsa-lib is a preparation for the idea. In current implementation of this library, results of mapping operation for status/control data are not handled separately. This patchset is a similar to code refactoring, thus this includes no essential behaviour changes against released kernel lands. [1] ALSA: pcm: Add the explicit appl_ptr sync support https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… [2] ALSA: pcm: add 'applptr' event of tracepoint https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… [3] [alsa-devel] [PATCH 0/3] ALSA: Add the explicit appl_ptr sync support http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122092.html [4] ALSA: pcm: Skip ack callback without actual appl_ptr update https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… [5] [alsa-devel] [PATCH 0/3] ALSA: Add the explicit appl_ptr sync support http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122093.html Takashi Sakamoto (6): pcm: hw: add helper functions to map/unmap status/control data for runtime of PCM substream pcm: hw: add an arrangement for initialization of appl_ptr/avail_min pcm: hw: deallocate fallback buffer when trials of unmapping finished pcm: hw: allocate fallback buffer in advance of trials of mapping pcm: hw: maintain fallback mode for status data mapping pcm: hw: maintain fallback mode for control data mapping independently src/pcm/pcm_hw.c | 192 +++++++++++++++++++++++++++++++++---------------------- 1 file changed, 115 insertions(+), 77 deletions(-) -- 2.11.0

2 12

[alsa-devel] [PATCH alsa-lib v2] pcm: hw: Call USER_PVERSION ioctl at open
by Takashi Iwai 27 Jun '17

27 Jun '17

Up from the new PCM protocol 2.0.14, user-space can inform the protocol version it supports to kernel, so that the kernel may switch its behavior depending on it. Add this ioctl call in the PCM hw plugin at opening. The patch contains also the addition of SNDRV_PCM_INFO_SYNC_APPLPTR carried from the upstream kernel commit 42f945970af9 ("ALSA: pcm: Add the explicit appl_ptr sync support"), as well as the trivial change (an addition of comma) to sync with the kernel asound.h. Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Tested-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/asound.h | 6 ++++-- src/pcm/pcm_hw.c | 10 ++++++++++ 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index fb8d7d7ef8ad..1949923a40bf 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -111,7 +111,7 @@ enum { SNDRV_HWDEP_IFACE_FW_FIREFACE, /* RME Fireface series */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_FIREFACE, + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_FIREFACE }; struct snd_hwdep_info { @@ -152,7 +152,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 14) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -268,6 +268,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_MMAP_VALID 0x00000002 /* period data are valid during transfer */ #define SNDRV_PCM_INFO_DOUBLE 0x00000004 /* Double buffering needed for PCM start/stop */ #define SNDRV_PCM_INFO_BATCH 0x00000010 /* double buffering */ +#define SNDRV_PCM_INFO_SYNC_APPLPTR 0x00000020 /* need the explicit sync of appl_ptr update */ #define SNDRV_PCM_INFO_INTERLEAVED 0x00000100 /* channels are interleaved */ #define SNDRV_PCM_INFO_NONINTERLEAVED 0x00000200 /* channels are not interleaved */ #define SNDRV_PCM_INFO_COMPLEX 0x00000400 /* complex frame organization (mmap only) */ @@ -563,6 +564,7 @@ enum { #define SNDRV_PCM_IOCTL_INFO _IOR('A', 0x01, struct snd_pcm_info) #define SNDRV_PCM_IOCTL_TSTAMP _IOW('A', 0x02, int) #define SNDRV_PCM_IOCTL_TTSTAMP _IOW('A', 0x03, int) +#define SNDRV_PCM_IOCTL_USER_PVERSION _IOW('A', 0x04, int) #define SNDRV_PCM_IOCTL_HW_REFINE _IOWR('A', 0x10, struct snd_pcm_hw_params) #define SNDRV_PCM_IOCTL_HW_PARAMS _IOWR('A', 0x11, struct snd_pcm_hw_params) #define SNDRV_PCM_IOCTL_HW_FREE _IO('A', 0x12) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 5573fce2d43d..64188b2258d6 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1494,6 +1494,16 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, int fd, if (SNDRV_PROTOCOL_INCOMPATIBLE(ver, SNDRV_PCM_VERSION_MAX)) return -SND_ERROR_INCOMPATIBLE_VERSION; + if (SNDRV_PROTOCOL_VERSION(2, 0, 14) <= ver) { + /* inform the protocol version we're supporting */ + unsigned int user_ver = SNDRV_PCM_VERSION; + if (ioctl(fd, SNDRV_PCM_IOCTL_USER_PVERSION, &user_ver) < 0) { + ret = -errno; + SNDMSG("USER_PVERSION failed\n"); + return ret; + } + } + #if defined(HAVE_CLOCK_GETTIME) && defined(CLOCK_MONOTONIC) if (SNDRV_PROTOCOL_VERSION(2, 0, 9) <= ver) { struct timespec timespec; -- 2.13.2

1 0

[alsa-devel] remove dma_alloc_noncoherent
by Christoph Hellwig 26 Jun '17

26 Jun '17

For many years we've had the dma_alloc_attrs API that is more flexible than dma_alloc_noncoherent. This series moves the remaining users over to the attrs API.

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[alsa-devel] [PATCH] Fix bug in arecord --max-file-time where the file size could overflow 32 bits
by Gilliland, Scott M 26 Jun '17

26 Jun '17

From: Scott Gilliland <scott.gilliland(a)gatech.edu> Fix bug in arecord --max-file-time where the file size could overflow 32 bits Signed-off-by: Scott Gilliland <scott.gilliland(a)gatech.edu> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index f793c82..00af662 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3027,7 +3027,7 @@ static void capture(char *orig_name) if (count == 0) count = LLONG_MAX; /* compute the number of bytes per file */ - max_file_size = max_file_time * + max_file_size = (long long) max_file_time * snd_pcm_format_size(hwparams.format, hwparams.rate * hwparams.channels); /* WAVE-file should be even (I'm not sure), but wasting one byte -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: pcm: add a documentation for tracepoints
by Takashi Sakamoto 26 Jun '17

26 Jun '17

In PCM interface/protocol for userspace, parameters of runtime for PCM substream is decided by an interaction between applications and ALSA PCM core. In former commits, some tracepoints were added to probe a part of the interaction. This commit adds a documentation about the interaction and the tracepoints. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/tracepoints.rst | 177 ++++++++++++++++++++++++++++ 2 files changed, 178 insertions(+) create mode 100644 Documentation/sound/designs/tracepoints.rst diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 04dcdae3e4f2..f0749943ccb2 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -9,6 +9,7 @@ Designs and Implementations compress-offload timestamping jack-controls + tracepoints procfile powersave oss-emulation diff --git a/Documentation/sound/designs/tracepoints.rst b/Documentation/sound/designs/tracepoints.rst new file mode 100644 index 000000000000..b45e9f765451 --- /dev/null +++ b/Documentation/sound/designs/tracepoints.rst @@ -0,0 +1,177 @@ +=================== +Tracepoints in ALSA +=================== + +2017/06/26 +Takasahi Sakamoto + +Tracepoints in ALSA PCM core +============================ + +ALSA PCM core registers ``snd_pcm`` subsystem to kernel tracepoint system. +This subsystem includes two categories of tracepoints; for state of PCM buffer +and for processing of PCM hardware parameters. These tracepoints are available +when corresponding kernel configurations are enabled. When ``CONFIG_SND_DEBUG`` +is enabled, the latter tracepoints are available. When additional +``SND_PCM_XRUN_DEBUG`` is enabled too, the former trace points are enabled. + +Tracepoints for state of PCM buffer +------------------------------------ + +This category includes four tracepoints; ``hwptr``, ``applptr``, ``xrun`` and +``hw_ptr_error``. + +Tracepoints for processing of PCM hardware parameters +----------------------------------------------------- + +This category includes two tracepoints; ``hw_mask_param`` and +``hw_interval_param``. + +In a design of ALSA PCM core, data transmission is abstracted as PCM substream. +Applications manage PCM substream to maintain data transmission for PCM frames. +Before stating the data transmission, applications need to configure PCM +substream. In this procedure, PCM hardware parameters are decided by +interaction between applications and ALSA PCM core. Once decided, runtime of +the PCM substream keeps the parameters. + +The hardware parameters are described in :c:type:`struct snd_pcm_hw_params`. +This structure includes several types of parameters. Applications set preferable +value to these parameters, then execute ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE +or SNDRV_PCM_IOCTL_HW_PARAMS. The former is used just for querying available +set of parameters. The latter is used for an actual decision of the parameters. + +The :c:type:`struct snd_pcm_hw_params` structure has below members: + +``flags`` + Configurable. ALSA PCM core and some drivers handle this flag to select + convenient parameters or change their behaviour. As of PCM protocol + v2.0.13, three flags are defined. + + - SNDRV_PCM_HW_PARAMS_NORESAMPLE + - SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER + - SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP +``masks`` + Configurable. This type of parameter is described in + :c:type:`struct snd_mask` and represent mask values. As of PCM protocol + v2.0.13, three types are defined. + + - SNDRV_PCM_HW_PARAM_ACCESS + - SNDRV_PCM_HW_PARAM_FORMAT + - SNDRV_PCM_HW_PARAM_SUBFORMAT +``intervals`` + Configurable. This type of parameter is described in + :c:type:`struct snd_interval` and represent values with a range. As of + PCM protocol v2.0.13, twelve types are defined. + + - SNDRV_PCM_HW_PARAM_SAMPLE_BITS + - SNDRV_PCM_HW_PARAM_FRAME_BITS + - SNDRV_PCM_HW_PARAM_CHANNELS + - SNDRV_PCM_HW_PARAM_RATE + - SNDRV_PCM_HW_PARAM_PERIOD_TIME + - SNDRV_PCM_HW_PARAM_PERIOD_SIZE + - SNDRV_PCM_HW_PARAM_PERIOD_BYTES + - SNDRV_PCM_HW_PARAM_PERIODS + - SNDRV_PCM_HW_PARAM_BUFFER_TIME + - SNDRV_PCM_HW_PARAM_BUFFER_SIZE + - SNDRV_PCM_HW_PARAM_BUFFER_BYTES + - SNDRV_PCM_HW_PARAM_TICK_TIME +``rmask`` + Configurable. This is evaluated at ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE only. Applications can select which + mask/interval parameter should be queried to ALSA PCM core. For + SNDRV_PCM_IOCTL_HW_PARAMS, this mask makes no sense and all of + parameters are going to be changed. +``cmask`` + Read-only. After returning from ioctl(2), buffer in user space for + :c:type:`struct snd_pcm_hw_params` includes result. This mask represents + which mask/interval parameter is actually changed. +``info`` + Read-only. This represents hardware/driver capabilities as bit flags + with SNDRV_PCM_INFO_XXX. Typically, applications execute ioctl(2) with + SNDRV_PCM_IOCTL_HW_REFINE to retrieve this flag, then decide candidates + of parameters and execute ioctl(2) with SNDRV_PCM_IOCTL_HW_PARAMS to + configure PCM substream. +``msbits`` + Read-only. This value represents available bit width in MSB side of + a PCM sample. When a parameter of SNDRV_PCM_HW_PARAM_SAMPLE_BITS was + decided as a fixed number, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_num`` + Read-only. This value represents numerator of sampling rate in fraction + notation. Basically, when a parameter of SNDRV_PCM_HW_PARAM_RATE was + decided as a fixed number, this value is also calculated according to + it. Else, zero. But this behaviour depends on implementations in driver + side. +``rate_den`` + Read-only. This value represents denominator of sampling rate in + fraction notation. Basically, when a parameter of + SNDRV_PCM_HW_PARAM_RATE was decided as a fixed number, this value is + also calculated according to it. Else, zero. But this behaviour depends + on implementations in driver side. +``fifo_size`` + Read-only. This value represents the size of FIFO in serial sound + interface of hardware. Basically, each driver can assigns a proper + value to this parameter but some drivers intentionally set zero with + a care of hardware design or data transmission protocol. + +ALSA PCM core handles buffer of :c:type:`struct snd_pcm_hw_params` when +applications execute ioctl(2) with SNDRV_PCM_HW_REFINE or SNDRV_PCM_HW_PARAMS. +Pamraeters in the buffer are changed according to +:c:type:`struct snd_pcm_hardware` and rules of constraints in the runtime. The +structure describes capabilities of handled hardware. The rules describes +dependencies on which a parameter is decided according to several parameters. +A rule has a callback function, and drivers can register arbitrary functions +to compute the target parameter. ALSA PCM core registers some rules to the +runtime as a default. + +Each driver can join in the interaction as long as it prepared for two stuffs +in a callback of :c:type:`struct snd_pcm_ops.open`. + +1. In the callback, drivers are expected to change a member of + :c:type:`struct snd_pcm_hardware` type in the runtime, according to + capacities of corresponding hardware. +2. In the same callback, drivers are also expected to register additional rules + of constraints into the runtime when several parameters have dependencies + due to hardware design. + +The driver can refers to result of the interaction in a callback of +:c:type:`struct snd_pcm_ops.hw_params`, however it should not change the +content. + +Tracepoints in this category are designed to trace changes of the +mask/interval parameters. When ALSA PCM core changes them, ``hw_mask_param`` or +``hw_interval_param`` event is probed according to type of the changed parameter. + +ALSA PCM core also has a pretty print format for each of the tracepoints. Below +is an example for ``hw_mask_param``. + +:: + + hw_mask_param: pcmC0D0p 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044 + + +Below is an example for ``hw_interval_param``. + +:: + + hw_interval_param: pcmC0D0p 000/023 BUFFER_SIZE 0 0 [0 4294967295] 0 1 [0 4294967295] + +The first three fields are common. They represent ALSA PCM character device, +rules of constraint and name of the changed parameter, in order. The field for +rules of constraint consists of two sub-fields; index of applied rule and +total number of rules added to the runtime. The index 000 means that the +parameter is changed by ALSA PCM core regardless of the rules. + +The rest of field represent state of the parameter before/after changing. These +fields are different according to type of the parameter. For parameter of mask +type, the fields represent hexadecimal dump of content of the parameter. For +parameter of interval type, the fields represent values of each member of +``empty``, ``integer``, ``openmin``, ``min``, ``max``, ``openmax`` in +:c:type:`struct snd_interval`. + +Tracepoints in drivers +====================== + +Some drivers have tracepoints for developers' convenience. For them, please +refer to each documentation or implementation. -- 2.11.0

2 1

[alsa-devel] [PATCH][sound-next] ASoC: zx_aud96p22: make aud96p22_dt_ids static
by Colin King 26 Jun '17

26 Jun '17

From: Colin Ian King <colin.king(a)canonical.com> The structure aud96p22_dt_ids does not need to be in global scope, so make it static. Cleans up sparse warning: "warning: symbol 'aud96p22_dt_ids' was not declared. Should it be static?" Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/zx_aud96p22.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c index 032fb7cf6cbd..bc23ce1a30a1 100644 --- a/sound/soc/codecs/zx_aud96p22.c +++ b/sound/soc/codecs/zx_aud96p22.c @@ -382,7 +382,7 @@ static int aud96p22_i2c_remove(struct i2c_client *i2c) return 0; } -const struct of_device_id aud96p22_dt_ids[] = { +static const struct of_device_id aud96p22_dt_ids[] = { { .compatible = "zte,zx-aud96p22", }, { } }; -- 2.11.0

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - There is no loopback mixer in the ALC234/274/294
by Hui Wang 26 Jun '17

26 Jun '17

We have a Lenovo machine with the codec ALC294 on it, without the patch, the Node 0x0b is regarded as the loopback mixer, but the Node 0x0b in this codec is "Vendor Defined Widget" instead of the audio mixer, please see the log below: Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 And I have consulted with Realtek, so far the ALC234/274/294 all don't have loopback mixer. Cc: Kailang Yang <kailang(a)realtek.com> 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 c4a1b49..bc00a37 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6696,6 +6696,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0274: case 0x10ec0294: spec->codec_variant = ALC269_TYPE_ALC294; + spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ break; case 0x10ec0700: -- 2.7.4

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[alsa-devel] [RFC 0/x v2] ASoC: replace platform to component
by Kuninori Morimoto 26 Jun '17

26 Jun '17

Hi Mark, Lars-Peter These are v2 of platform replace patch-set. Big difference is that from version, rtd can have component list. As you know, current rtd has platform, codec, and this patch replace rtd->platform into component list. We can replace codec in same style in the future. This patch-set adds new snd_soc_rtdcom_lookup(). So, current struct snd_soc_platform *platform = rtd->platform; will be replaced into struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRIVER_NAME); 1 rtd doesn't detect same component multiple times, I guess. Thus, lookup by driver name should be OK. Current framework is listing component to card (it was originally codec) by component_dev_list. And it is used in soc-pcm.c list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { ... } But it is not correct I think. card can have multiple runtimes, and runtime can have multiple components. Thus, I think we can do below case ? Card -+-> runtime1 -> component1 + component2 +-> runtime2 -> component1 + component2 +-> runtime3 -> component1 + component3 Current list_for_each_entry() will call component1/2/3, but runtime1 needs component1/2 only, for example. It is connecting all components to card by component_dev_list. OTOH, all runtimes are listed to card by rtd_list. This patch lists component to runtime. I think we can replace component_dev_list into for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; ... } 1) - 5) : prepare to replace 6) - 8) : sample of platform -> component replace 9) : remove platform Kuninori Morimoto (9): 1) ASoC: soc-core: add component remove/unregister_exp/lookup functions 2) ASoC: soc-devres: use expanded unregister on devm register component 3) ASoC: soc-core: add snd_soc_rtdcom_xxx() 4) ASoC: snd_soc_component_driver has snd_pcm_ops 5) ASoC: snd_soc_component_driver has snd_compr_ops 6) ASoC: replace platform to component on rsnd 7) ASoC: replace platform to component on soc-utils 8) ASoC: replace platform to component on soc-generic-dmaengine-pcm 9) ASoC: remove platform related things Best regards --- Kuninori Morimoto

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[alsa-devel] additional high input device for snd-dice (A&H Zed R16)
by Allan Klinbail 26 Jun '17

26 Jun '17

Hi, I noticed on the changelog that some devices are able to access all channels without doing any .asoundrc funkiness. Form 4.6 kernel onwards - Add support for previously unavailable higher PCM channels on certain devices with high channel count, notably Focusrite Saffire PRO 40, Focusrite Liquid Saffire 56, and TC Electronic Studio Konnekt 48. These devices spread the PCM and MIDI channels across 2 tx + 2 rx IEEE 1394 channels instead of just 1 tx + 1 rx IEEE 1394 channel, as most other devices do. Pretty sure I've tested with 4.6 and wasn't able to achieve this with my device. But can test again with 4.8 if I am wrong. My device is an Allen and Heath Zed R16.. While latency performance in the way ALSA works is not satisfactory for my usage others may wish to use this device with ALSA for simplicity. (remove the need for FFADO entirely as there is no software mixer for this device and remove the need for any MIDI bridge) Currently under ALSA in 44.1k or 48k the device has 16 inputs and 16 outputs on the first firewire channel, and then and additional 10 ins and 10 outs on the second. If this is any help, here is the vendor info, as used by FFADO { # Allen and Heath Zed R16. Information from Brendan Pike. vendorid = 0x000004C4; modelid = 0x00000000; vendorname = "Allen and Heath"; modelname = "Zed R16"; driver = "DICE"; mixer = "Generic_Dice_EAP"; } On that latency topic- just voicing my opinion and I don't expect change. When the paradigm of using periods/frames and buffers is common across all OS and driver platforms, I don't believe that the driver should force developers to implement a new method, even if there are better ways of doing things. That might be okay for small open source projects, but bigger commercial applications would not be likely to maintain separate configuration options for Linux, they are more likely to drop Linux altogether if forced to adopt I also know that Takashi S, does not agree but as FFADO devs have agreed to keep the driver in FFADO I don't mind..

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[alsa-devel] [PATCH] ASoC: nau8825: debug message of crosstalk bypass
by John Hsu 26 Jun '17

26 Jun '17

Add debug message for crosstalk function bypass. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index a8c7a55..80bae48 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2440,6 +2440,8 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); + dev_dbg(dev, "crosstalk-bypass: %d\n", + nau8825->xtalk_bypass); } static int nau8825_read_device_properties(struct device *dev, -- 2.6.4

1 0

Re: [alsa-devel] Fix bug in arecord --max-file-time where the file size could overflow 32 bits
by John Sauter 24 Jun '17

24 Jun '17

> Date: Fri, 23 Jun 2017 18:35:03 +0000 > From: "Gilliland, Scott M" <scott.gilliland(a)gatech.edu> > To: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> > Subject: [alsa-devel] [PATCH] Fix bug in arecord --max-file-time > where > the file size could overflow 32 bits > Message-ID: > <CY4PR07MB3544F3467606CB0401C3F0918DD80(a)CY4PR07MB3544.namprd07. > prod.outlook.com> > > Content-Type: text/plain; charset="iso-8859-1" > > From: Scott Gilliland <scott.gilliland(a)gatech.edu> > > Fix bug in arecord --max-file-time where the file size could overflow > 32 bits > > Signed-off-by: Scott Gilliland <scott.gilliland(a)gatech.edu> > --- > aplay/aplay.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/aplay/aplay.c b/aplay/aplay.c > index f793c82..00af662 100644 > --- a/aplay/aplay.c > +++ b/aplay/aplay.c > @@ -3027,7 +3027,7 @@ static void capture(char *orig_name) > if (count == 0) > count = LLONG_MAX; > /* compute the number of bytes per file */ > - max_file_size = max_file_time * > + max_file_size = (long long) max_file_time * > snd_pcm_format_size(hwparams.format, > hwparams.rate * > hwparams.channels); > /* WAVE-file should be even (I'm not sure), but wasting one > byte > -- > 2.11.0 Thank you for fixing this bug, which I introduced in 2010 through failing to understand the type conversion rules of C. Acked-by: John Sauter <John_Sauter(a)systemeyescomputerstore.com> -- PGP fingerprint E24A D25B E5FE 4914 A603 49EC 7030 3EA1 9A0B 511E

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[alsa-devel] [PATCH 0/2] Lenovo tablet fixes
by Pierre-Louis Bossart 23 Jun '17

23 Jun '17

2 patches to enable rt5670 on Baytrail and fix jack detection. There are likely follow-up patches or UCM updates needed since some users report additional problems with headphones and subwoofers [1] but this is already working for a number of people. [1] Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96691 Pierre-Louis Bossart (2): ASoC: Intel: cht_bsw_rt5672: 19.2MHz clock for Baytrail platforms ASoC: codecs: rt5670: fix jd mode for Lenovo Miix 2 10 sound/soc/codecs/rt5670.c | 9 ++++ sound/soc/intel/boards/cht_bsw_rt5672.c | 89 +++++++++++++++++++++++++++++---- 2 files changed, 89 insertions(+), 9 deletions(-) -- 2.9.3

1 2

[alsa-devel] Bug in ALSA-utils/arecord with --max-file-time
by Gilliland, Scott M 23 Jun '17

23 Jun '17

Hi All, I seem to have found a minor bug in aplay, within alsa-utils 1.1.4. It's possible that I'm the first person who's ever hit it. This appears to be the correct place to report it; if not, I humbly apologize, and ask for direction on where to submit this. In aplay.c, around line 3030, we do this calculation: max_file_size = max_file_time * snd_pcm_format_size(hwparams.format, hwparams.rate * hwparams.channels); We do this to convert max_file_time from the '--max-file-time' parameter (an int) into max_file_size, a file length in bytes (a long long). Since snd_pcm_format_size() may return something 32-bit, this might overflow 32-bits when we do the multiplication, wasting the extra bytes of max_file_size. On my test case on an armv7l system, a 1800-second file at 192 KHz recorded in S24_3LE doesn't overflow, but recording in S32_LE does. I accidentally found an edge case. In case anyone wants to know, the failure mode is that arecord will repeatedly create 44 byte .wav files very quickly, since these are larger than the negative max file size. This continues until you have a directory full of thousands of 44-byte files. Hilarious. A simple change to cast max_file_time to a long long fixes this in my test case. There might be some cleaner solution, but this at least worked for me. max_file_size = (long long) max_file_time * snd_pcm_format_size(hwparams.format, hwparams.rate * hwparams.channels); Hopefully someone can vet this and carry it over to git. I've attached a patch file as well. -- Additionally, while working to report this, I noticed that the bug tracker link on the sidebar of https://www.alsa-project.org/main/index.php/Main_Page is a dead link. -Scott -- Scott Gilliland imtc.gatech.edu/people/scott-gilliland

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Re: [alsa-devel] [PATCH] ASoC: rt5645: enable speaker protection features
by Mark Brown 23 Jun '17

23 Jun '17

On Fri, Jun 23, 2017 at 01:55:29PM +0000, Bard Liao wrote: > > On Fri, Jun 23, 2017 at 10:24:57AM +0800, Bard Liao wrote: > > You shouldn't be changing the register defaults away from the physical > > defaults, just do it with a normal register write. Changing the > > defaults will break cache resyncing. > Agree. The reason I did it is the register will also be modified by > regmap_register_patch. If it goes in via a normal write it'll still get restored, just as part of the regular cache writeout on resume.

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[alsa-devel] Query regarding possibility of PM QoS unvote/vote when substream Pause/Resume called
by bgoswami＠codeaurora.org 23 Jun '17

23 Jun '17

Hi, Current ALSA core implementation has PM QoS vote/unvote calls in snd_pcm_hw_params() and snd_pcm_hw_free(). However, this might have an implication in terms of power, as PM QoS vote would still be valid for that substream, even when the substream is paused. Especially for those cases, where, to avoid missing the QoS, one or more CPU cores might be prevented from entering into lower power mode (or power-collapse), as it might take longer to bring those cores out of power collapse. This might be specially more prominent when the period size is smaller. So, in an effort to optimize power (by whatever smaller amount), I was wondering, if it is better to remove the PM QoS vote (pm_qos_remove_request) when Pause (SNDRV_PCM_TRIGGER_PAUSE_PUSH) is called from user space (using IOCTL) for a particular substream. Similarly, PM QoS vote can be added (pm_qos_add_request) back, when user space (using IOCTL) calls Resume (SNDRV_PCM_TRIGGER_PAUSE_RELEASE) for that same substream. I am looking for opinions from subject matter experts, to see if the proposed approach might have any unintended side-effects or any known limitation that I might be missing. Thanks and Regards, Banajit Goswami The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH v2 0/3] ALSA: Add the explicit appl_ptr sync support
by Takashi Iwai 23 Jun '17

23 Jun '17

Hi, this is a revised patchset for supporting the explicit appl_ptr update. Since the protocol version from user-space needs to be stored locally for each opened file, the check of status/control mmap was slightly changed as well. But the basic strategy is same as the previous patchset. Note that this patchset doesn't solve the issue that appl_ptr update isn't properly notified to ack ops as Sakamoto-san pointed out previously. We'll still need to sort it out. But the hack I introduced here doesn't basically conflict with it, and it'll even make it easier to solve later, by the new USER_PVERSION ioctl. thanks, Takashi === Takashi Iwai (3): ALSA: pcm: Add the explicit appl_ptr sync support ALSA: pcm: Add an ioctl to specify the supported protocol version ALSA: pcm: Limit the appl_ptr sync workaround only for old user-space include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 4 +++- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 34 ++++++++++++++++++++++++++++++++-- 4 files changed, 37 insertions(+), 3 deletions(-) -- 2.13.1

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[alsa-devel] [PATCH alsa-lib 0/2] pcm: minor code cleanup
by Takashi Sakamoto 23 Jun '17

23 Jun '17

Hi, This includes two patches from my previous RFC for minor code cleanup: [alsa-devel] [alsa-lib][RFC][PATCH 0/9] pcm: handle status/control mapping independently http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122130.html Takashi Sakamoto (2): pcm: obsolete 'mmap_emulation' parameter of snd_pcm_hw_open_fd() pcm: minor code cleanup for ioctl call src/pcm/pcm_direct.c | 2 +- src/pcm/pcm_hw.c | 9 +++------ src/pcm/pcm_local.h | 3 ++- 3 files changed, 6 insertions(+), 8 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2 0/2] Add ZTE zx-aud96p22 audio codec driver
by Shawn Guo 23 Jun '17

23 Jun '17

From: Shawn Guo <shawn.guo(a)linaro.org> It adds the ASoC driver for zx-aud96p22 audio codec found on ZTE ZX family SoCs. Changes for v2: - Follow document control-names.rst to name controls in a standard way, so that user space mixer UIs know what to do with them. - Drop the GPIO support for an optional external amplifier, as it belongs to machine driver rather than codec driver. - Support overall power bit with a DAPM supply widget and attach it to both DAC and ADC components. - Instead of reset DAC/ADC data path in startup/shutdown hooks, handle it with DAPM event, so that startup/shutdown hooks get dropped completely. Baoyou Xie (2): ASoC: add bindings for ZTE zx-aud96p22 audio codec ASoC: zx_aud96p22: add ZTE ZX AUD96P22 codec driver .../devicetree/bindings/sound/zte,zx-aud96p22.txt | 24 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/zx_aud96p22.c | 403 +++++++++++++++++++++ 4 files changed, 434 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt create mode 100644 sound/soc/codecs/zx_aud96p22.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: default value for property
by John Hsu 23 Jun '17

23 Jun '17

Assign default value for codec private data when property not given. If without those default value and property, the codec will work abnormally. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 53 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 41 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 97fbeba..c00b86d 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2429,6 +2429,7 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) static int nau8825_read_device_properties(struct device *dev, struct nau8825 *nau8825) { + int ret; nau8825->jkdet_enable = device_property_read_bool(dev, "nuvoton,jkdet-enable"); @@ -2436,30 +2437,58 @@ static int nau8825_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8825->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); - device_property_read_u32(dev, "nuvoton,jkdet-polarity", + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8825->jkdet_polarity); - device_property_read_u32(dev, "nuvoton,micbias-voltage", + if (ret) + nau8825->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", &nau8825->micbias_voltage); - device_property_read_u32(dev, "nuvoton,vref-impedance", + if (ret) + nau8825->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", &nau8825->vref_impedance); - device_property_read_u32(dev, "nuvoton,sar-threshold-num", + if (ret) + nau8825->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num", &nau8825->sar_threshold_num); - device_property_read_u32_array(dev, "nuvoton,sar-threshold", + if (ret) + nau8825->sar_threshold_num = 4; + ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold", nau8825->sar_threshold, nau8825->sar_threshold_num); - device_property_read_u32(dev, "nuvoton,sar-hysteresis", + if (ret) { + nau8825->sar_threshold[0] = 0x08; + nau8825->sar_threshold[1] = 0x12; + nau8825->sar_threshold[2] = 0x26; + nau8825->sar_threshold[3] = 0x73; + } + ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis", &nau8825->sar_hysteresis); - device_property_read_u32(dev, "nuvoton,sar-voltage", + if (ret) + nau8825->sar_hysteresis = 0; + ret = device_property_read_u32(dev, "nuvoton,sar-voltage", &nau8825->sar_voltage); - device_property_read_u32(dev, "nuvoton,sar-compare-time", + if (ret) + nau8825->sar_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,sar-compare-time", &nau8825->sar_compare_time); - device_property_read_u32(dev, "nuvoton,sar-sampling-time", + if (ret) + nau8825->sar_compare_time = 1; + ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time", &nau8825->sar_sampling_time); - device_property_read_u32(dev, "nuvoton,short-key-debounce", + if (ret) + nau8825->sar_sampling_time = 1; + ret = device_property_read_u32(dev, "nuvoton,short-key-debounce", &nau8825->key_debounce); - device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + if (ret) + nau8825->key_debounce = 3; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", &nau8825->jack_insert_debounce); - device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + if (ret) + nau8825->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", &nau8825->jack_eject_debounce); + if (ret) + nau8825->jack_eject_debounce = 0; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- 2.6.4

2 1

[alsa-devel] ASoC: nau8825: fix jack type detection issue after resume
by John Hsu 23 Jun '17

23 Jun '17

Fix the issue that mic type detection error after resume. The patch help to fix the issue as follows: Google issue 37973093: CTIA/OMTP jack type detection failure after resume Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index fd307f5..fb9f1d4 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1612,7 +1612,6 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_sync(dapm); break; case 2: - case 3: dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); type = SND_JACK_HEADSET; @@ -1632,6 +1631,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); break; + case 3: + /* detect error case */ + dev_err(nau8825->dev, "detection error; disable mic function\n"); + type = SND_JACK_HEADPHONE; + break; } /* Leaving HPOL/R grounded after jack insert by default. They will be @@ -2328,6 +2332,13 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + /* Reset the configuration of jack type for detection */ + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_HSD_CTRL, 0xf, 0xf); /* Cancel and reset cross talk detection funciton */ nau8825_xtalk_cancel(nau8825); /* Turn off all interruptions before system shutdown. Keep the @@ -2351,6 +2362,10 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec) disable_irq(nau8825->irq); snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8825->dapm, "SAR"); + snd_soc_dapm_disable_pin(nau8825->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8825->dapm); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); -- 2.6.4

3 3

[alsa-devel] ASoC: nau8825: make crosstalk function optional
by John Hsu 23 Jun '17

23 Jun '17

Make crosstalk functoin optional. The jack detection can speed up without crosstalk detection. Let the decision of function usage to platform design. The patch helps the issue concern as follows: Google issue 35574278: Chell_headphone pop back from S3 Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 6 +++++- sound/soc/codecs/nau8825.h | 1 + 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index ff827bb..fd307f5 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1682,7 +1682,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->high_imped) { + if (!nau8825->xtalk_bypass && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -2498,6 +2498,10 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", + &nau8825->xtalk_bypass); + if (ret) + nau8825->xtalk_bypass = 1; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 514fd13..8aee5c86 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,6 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int xtalk_bypass; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: tidyup asoc_simple_card_canonicalize_cpu() parameter
by Kuninori Morimoto 23 Jun '17

23 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's DAI links, not Card links. This patch fixup it. Otherwise, audio-graph-card can't handle CPU component correctly if CPU has mult-DAIs and Card uses only one of them Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 885b405..ee752f62 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -100,7 +100,6 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; - struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); struct device_node *rcpu_ep = of_graph_get_remote_endpoint(codec_ep); @@ -162,7 +161,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, dai_link->init = asoc_graph_card_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, - card->num_links == 1); + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); dai_link_of_err: of_node_put(cpu_ep); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: tidyup asoc_simple_card_canonicalize_cpu() parameter
by Kuninori Morimoto 23 Jun '17

23 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's DAI links, not Card links. This patch fixup it. Otherwise, audio-graph-card can't handle CPU component correctly if CPU has mult-DAIs and Card uses only one of them Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 05934b2..061c7a6 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -125,7 +125,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, /* card->num_links includes Codec */ asoc_simple_card_canonicalize_cpu(dai_link, - (card->num_links - 1) == 1); + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); } else { /* FE is dummy */ dai_link->cpu_of_node = NULL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: support 2nd codec endpoint on DT
by Kuninori Morimoto 23 Jun '17

23 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> audio-graph-scu-card can handle below connection which is mainly for sound mixing purpose. +----------+ +-------+ | CPU0--+--|-->| Codec | | | | +-------+ | CPU1--+ | +----------+ From OF-graph point of view, it should have CPU0 <-> Codec, and CPU1 <-> Codec on DT. But current driver doesn't care about 2nd connection of Codec, because it is dummy from DPCM point of view. This patch can care 2nd Codec connection, and it should be supported from OF-graph point of view. It still have backward compatibility. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/audio-graph-scu-card.txt | 9 +++++-- sound/soc/generic/audio-graph-scu-card.c | 28 +++++++++++++++++++--- 2 files changed, 32 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b63c559..8b8afe9 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -90,9 +90,12 @@ Example 2. 2 CPU 1 Codec (Mixing) ... port { - codec_endpoint: endpoint { + codec_endpoint0: endpoint { remote-endpoint = <&cpu_endpoint0>; }; + codec_endpoint1: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; }; }; @@ -101,7 +104,7 @@ Example 2. 2 CPU 1 Codec (Mixing) ports { cpu_port0: port { cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint>; + remote-endpoint = <&codec_endpoint0>; dai-format = "left_j"; ... @@ -109,6 +112,8 @@ Example 2. 2 CPU 1 Codec (Mixing) }; cpu_port1: port { cpu_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint1>; + dai-format = "left_j"; ... }; diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 061c7a6..dcd2df3 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -183,6 +183,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *cpu_ep; struct device_node *codec_ep; struct device_node *rcpu_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; unsigned int daifmt = 0; int dai_idx, ret; int rc, codec; @@ -235,6 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } dai_idx = 0; + codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* * To listup valid sounds continuously, @@ -245,15 +248,22 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (codec) { - if (!codec_ep) + if (!codec_port) continue; + if (codec_port_old == codec_port) + continue; + + codec_port_old = codec_port; + /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); if (ret < 0) @@ -284,22 +294,34 @@ static int asoc_graph_get_dais_count(struct device *dev) struct device_node *cpu_port; struct device_node *cpu_ep; struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; int count = 0; int rc; + codec_port_old = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (cpu_ep) count++; - if (codec_ep) - count++; + + if (!codec_port) + continue; + + if (codec_port_old == codec_port) + continue; + + count++; + codec_port_old = codec_port; } return count; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5645: enable speaker protection features
by Bard Liao 23 Jun '17

23 Jun '17

This patch is uploaded for enabling the speaker protection features of the audio codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 162044d..9344af2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -59,10 +59,11 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x1c, 0xfd70}, {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, + {RT5645_CLSD_OUT_CTRL, 0xa0e8}, {RT5645_ASRC_4, 0x0120}, }; @@ -171,7 +172,7 @@ static const struct reg_default rt5645_reg[] = { { 0x9c, 0x0aea }, { 0x9d, 0x000c }, { 0x9e, 0x0400 }, - { 0xa0, 0xa0a8 }, + { 0xa0, 0xa0e8 }, { 0xa1, 0x0059 }, { 0xa2, 0x0001 }, { 0xae, 0x6000 }, @@ -328,7 +329,7 @@ static const struct reg_default rt5650_reg[] = { { 0x9c, 0x0aea }, { 0x9d, 0x000c }, { 0x9e, 0x0400 }, - { 0xa0, 0xa0a8 }, + { 0xa0, 0xa0e8 }, { 0xa1, 0x0059 }, { 0xa2, 0x0001 }, { 0xae, 0x6000 }, -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: Add the explicit appl_ptr sync support
by Takashi Iwai 23 Jun '17

23 Jun '17

Hi, this is a patchset for the issues we've discussed recently lengthily, the explicit sync of appl_ptr by disabling the control mmap. The first patch achieves it for a driver setting the new INFO flag. The second patch is for a generic ABI extension I wanted to implement for long time. It allows user-space reporting its supporting protocol version, so that the kernel can behave differently. It's especially useful for extending the ABI struct to use the reserved bits. The third patch is the optimization by the second patch; it simply skips the previous workaround when the user-space declares itself being new enough to support the feature. The corresponding alsa-lib patchset will follow in another thread. thanks, Takashi === Takashi Iwai (3): ALSA: pcm: Add the explicit appl_ptr sync support ALSA: pcm: Add an ioctl to specify the supported protocol version ALSA: pcm: Limit the appl_ptr sync workaround only for old user-space include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 4 +++- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 28 ++++++++++++++++++++++++++++ 4 files changed, 33 insertions(+), 1 deletion(-) -- 2.13.1

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[alsa-devel] Constant headphone noise on Dell XPS 13 9365
by Felipe Contreras 23 Jun '17

23 Jun '17

Hi, Since I bought this machine I've always heard audio noise when I plug the headphones. This laptop has Intel HDA with the Realtec codec. This has happened in all the Linux versions I've tried, and I've also tried HDA Analyzer and changed every value I could find and nothing helps. The only value that seems to have an effect is "Headphone Playback Switch"; if I mute it I don't hear the noise any more, but I don't hear any audio either. Lowering the volume to zero doesn't help either, the noise remains at the same audible level. I'm running 4.11.3 at the moment. Any ideas? Cheers. -- Felipe Contreras

2 4

[alsa-devel] [alsa-lib][RFC][PATCH 0/9] pcm: handle status/control mapping independently
by Takashi Sakamoto 22 Jun '17

22 Jun '17

Hi, This RFC patchset is for recent discussion about new version of PCM protocol/interface (2.0.14). This is accumulated on the below patchset. http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122066.html In the discussion, we found an advantage to handle page mapping for status/control data of runtime of PCM substream independently. Aim of this patchset is a proof-of-concept for alsa-lib. I tested this patchset with two machines based on x86/armv7a architectures. This looks work well without any regression. But I've never tested a case in which mapping is failed for one of the status/control data. For your information. Takashi Sakamoto (9): pcm: obsolete 'mmap_emulation' parameter of snd_pcm_hw_open_fd() pcm: minor code refactoring for ioctl call pcm: handle status/control mapping independently pcm: add a helper function to query status of PCM substream pcm: add a helper function to query hwptr pcm: add a helper function to query applptr pcm: add a helper function to issue avail_min pcm: add a helper function to issue applptr pcm: code refactoring to use helper functions src/pcm/pcm_direct.c | 2 +- src/pcm/pcm_hw.c | 324 +++++++++++++++++++++++++++++++++------------------ src/pcm/pcm_local.h | 3 +- 3 files changed, 214 insertions(+), 115 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: max9867: make array ni_div static const
by Colin King 22 Jun '17

22 Jun '17

From: Colin Ian King <colin.king(a)canonical.com> The array ni_div does not need to be in global scope and is not modified, so make it static const. Cleans up sparse warning: "symbol 'ni_div' was not declared. Should it be static?" Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/max9867.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 0247edc9c84e..2a40a69a7513 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -132,7 +132,7 @@ enum rates { pcm_rate_48, max_pcm_rate, }; -struct ni_div_rates { +static const struct ni_div_rates { u32 mclk; u16 ni[max_pcm_rate]; } ni_div[] = { -- 2.11.0

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[alsa-devel] [PATCH] ASoC: es8316: add I2C dependency
by Arnd Bergmann 22 Jun '17

22 Jun '17

Without CONFIG_I2C, we get a build failure: sound/soc/codecs/es8316.c:633:1: error: data definition has no type or storage class [-Werror] sound/soc/codecs/es8316.c:633:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/es8316.c:633:1: error: parameter names (without types) in function declaration [-Werror] sound/soc/codecs/es8316.c:623:26: error: 'es8316_i2c_driver' defined but not used [-Werror=unused-variable] This adds the required Kconfig dependency. Fixes: b8b88b70875a ("ASoC: add es8316 codec driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c6286e5ba511..f0f794186186 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -546,6 +546,7 @@ config SND_SOC_ES7134 config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" + depends on I2C config SND_SOC_ES8328 tristate -- 2.9.0

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[alsa-devel] [PATCH 3/3] ASoC: pxa: add COMPILE_TEST on SND_MMP_SOC
by Kuninori Morimoto 22 Jun '17

22 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It doesn't use asm header. We can add COMPILE_TEST Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e..6feb34b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -9,7 +9,7 @@ config SND_PXA2XX_SOC config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" - depends on ARCH_MMP + depends on ARCH_MMP || COMPILE_TEST select MMP_SRAM select SND_SOC_GENERIC_DMAENGINE_PCM select SND_ARM -- 1.9.1

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[alsa-devel] Alsa-lib: Dmix slowptr latency behaviour
by Matin Momin 22 Jun '17

22 Jun '17

Hi, On my embedded device, I am using a push mode simple alsa pipeline with following command > arecord -D"dsp" -r48000 -c2 -fs32_le | aplay -D"dmix1" where the asound conf looks this ------------------------------------------------------------------- # This is the actual HW device pcm.dsp { type hw card 1 rate 48000 channels 2 format S32_LE } pcm.dmix1 { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0666 slave.pcm "dsp" slowptr true } ------------------------------------------------------------------- I measured the pipeline delay by using Audacity on my PC, and measuring the difference in spike of audio recorded on the PC in a silent environment (1st spike will be from the actual sound and second will be from the output of my embedded device). Here I get a delay of around 245 msecs. Now, in the same configuration if I disable slowptr in dmix1, the delay comes down to 160 msecs, which corresponds exactly to buffer time (85 msecs). I want to know whether this is expected or is it an aberration? I have measured the userland time for the data to travel from alsa-lib/src/pcm/pcm_hw.c: snd_pcm_hw_readi() to alsa-lib/src/pcm/pcm_dmix.c:mix_areas() (By adding incremental numbers to each period recorded by arecord and seeing how much time they take to reach dmix). I cannot find any difference in absolute time difference with both slowptr true and false. Also, the critical hw_ptr and appl_ptr variables in pcm_hw and pcm_dmix also reflect same values for slowptr true and false. Is the above mentioned method to measure delay correct? If I don't see a delay here, where else should I look? The alsa-lib version is 1.1.0 and kernel is 3.8.13. Thanks in advance. Regards, Matin

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[alsa-devel] [PATCH] ASoC: nau8825: make crosstalk function optional
by John Hsu 22 Jun '17

22 Jun '17

Make crosstalk functoin optional. The jack detection can speed up without crosstalk detection. Let the decision of function usage to platform design. The patch helps the issue concern as follows: Google issue 35574278: Chell_headphone pop back from S3 There is a concern as follows: cras getting blocked for 2 seconds (worst-case 3 seconds) As I understand, ChromeOS expects resume finishes in 1 seconds. Video/Audio playing after 3 seconds of resume seems against the spec. If we really have to make the choice I would choose pop noise instead of waiting for 3 seconds. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 6 +++++- sound/soc/codecs/nau8825.h | 1 + 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c00b86d..588805c 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1682,7 +1682,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->high_imped) { + if (!nau8825->xtalk_bypass && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -2489,6 +2489,10 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", + &nau8825->xtalk_bypass); + if (ret) + nau8825->xtalk_bypass = 1; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 514fd13..8aee5c86 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,6 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int xtalk_bypass; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: nau8825: fix jack type detection issue after resume
by John Hsu 22 Jun '17

22 Jun '17

Fix the issue that mic type detection error after resume. The microphone type detection procedure will recognize testing signal on JKSLV pin, but before the procedure, JKSLV already had supply voltage, that results in the failure. Therefore, the patch turns off the power and reset the jack type configuration before suspend. Then redo the jack detection procedure after resume. The patch help to fix the issue as follows: Google issue 37973093: CTIA/OMTP jack type detection failure after resume Reported Issue Chrome OS Version : ChromeOS R59-9460.13.0 Type of hardware : DVT sample What steps will reproduce the problem? (1 Play a music (2 Insert a headphones (3 Close laptop lid 3 sec then open it What is the expected output? The music is normal in the headphones. What do you see instead? Singer voice in the music is not clear. How frequently does this problem reproduce? Always What is the impact to the user, and is there a workaround? If so, what is it? Re-insert the headset or close the laptop lid and then open it again can be repaired. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 588805c..a8c7a55 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1612,7 +1612,6 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_sync(dapm); break; case 2: - case 3: dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); type = SND_JACK_HEADSET; @@ -1632,6 +1631,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); break; + case 3: + /* detect error case */ + dev_err(nau8825->dev, "detection error; disable mic function\n"); + type = SND_JACK_HEADPHONE; + break; } /* Leaving HPOL/R grounded after jack insert by default. They will be @@ -2328,6 +2332,13 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + /* Reset the configuration of jack type for detection */ + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_HSD_CTRL, 0xf, 0xf); /* Cancel and reset cross talk detection funciton */ nau8825_xtalk_cancel(nau8825); /* Turn off all interruptions before system shutdown. Keep the @@ -2351,6 +2362,10 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec) disable_irq(nau8825->irq); snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8825->dapm, "SAR"); + snd_soc_dapm_disable_pin(nau8825->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8825->dapm); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); -- 2.6.4

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[alsa-devel] ASoC: nau8825: default value for property
by John Hsu 22 Jun '17

22 Jun '17

Assign default value for codec private data when property not given. If without those default value and property, the codec will work abnormally. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 62 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 50 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 97fbeba..ff827bb 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2429,37 +2429,75 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) static int nau8825_read_device_properties(struct device *dev, struct nau8825 *nau8825) { + int ret; nau8825->jkdet_enable = device_property_read_bool(dev, "nuvoton,jkdet-enable"); + if (nau8825->jkdet_enable != true && + nau8825->jkdet_enable != false) + nau8825->jkdet_enable = 1; nau8825->jkdet_pull_enable = device_property_read_bool(dev, "nuvoton,jkdet-pull-enable"); + if (nau8825->jkdet_pull_enable != true && + nau8825->jkdet_pull_enable != false) + nau8825->jkdet_pull_enable = 1; nau8825->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); - device_property_read_u32(dev, "nuvoton,jkdet-polarity", + if (nau8825->jkdet_pull_up != true && + nau8825->jkdet_pull_up != false) + nau8825->jkdet_pull_up = 1; + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8825->jkdet_polarity); - device_property_read_u32(dev, "nuvoton,micbias-voltage", + if (ret) + nau8825->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", &nau8825->micbias_voltage); - device_property_read_u32(dev, "nuvoton,vref-impedance", + if (ret) + nau8825->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", &nau8825->vref_impedance); - device_property_read_u32(dev, "nuvoton,sar-threshold-num", + if (ret) + nau8825->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num", &nau8825->sar_threshold_num); - device_property_read_u32_array(dev, "nuvoton,sar-threshold", + if (ret) + nau8825->sar_threshold_num = 4; + ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold", nau8825->sar_threshold, nau8825->sar_threshold_num); - device_property_read_u32(dev, "nuvoton,sar-hysteresis", + if (ret) { + nau8825->sar_threshold[0] = 0x08; + nau8825->sar_threshold[1] = 0x12; + nau8825->sar_threshold[2] = 0x26; + nau8825->sar_threshold[3] = 0x73; + } + ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis", &nau8825->sar_hysteresis); - device_property_read_u32(dev, "nuvoton,sar-voltage", + if (ret) + nau8825->sar_hysteresis = 0; + ret = device_property_read_u32(dev, "nuvoton,sar-voltage", &nau8825->sar_voltage); - device_property_read_u32(dev, "nuvoton,sar-compare-time", + if (ret) + nau8825->sar_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,sar-compare-time", &nau8825->sar_compare_time); - device_property_read_u32(dev, "nuvoton,sar-sampling-time", + if (ret) + nau8825->sar_compare_time = 1; + ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time", &nau8825->sar_sampling_time); - device_property_read_u32(dev, "nuvoton,short-key-debounce", + if (ret) + nau8825->sar_sampling_time = 1; + ret = device_property_read_u32(dev, "nuvoton,short-key-debounce", &nau8825->key_debounce); - device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + if (ret) + nau8825->key_debounce = 3; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", &nau8825->jack_insert_debounce); - device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + if (ret) + nau8825->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", &nau8825->jack_eject_debounce); + if (ret) + nau8825->jack_eject_debounce = 0; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- 2.6.4

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[alsa-devel] [RFC][PATCH 0/x] ASoC: replace platform to component
by Kuninori Morimoto 22 Jun '17

22 Jun '17

Hi Mark, Lars-Peter Now, we can replace "platform" to "component". It will be large patch-set. To avoid many posting big-patch-set, I will post few of then as example. 1) - 2) : prepare for last missing parts for replace 3) - 4) : example of replace patch 5) : remove platform Kuninori Morimoto (xx): 1) ASoC: snd_soc_component_driver has snd_pcm_ops 2) ASoC: snd_soc_component_driver has snd_compr_ops 3) ASoC: replace platform to component on sh/rcar 4) ASoC: replace platform to component on sst-xxx 5) ASoC: remove platform related things Big change of this replace are 1) snd_soc_lookup_platform() will be snd_soc_lookup_component() 2) debugfs_create_file("platform", xxx) will be debugfs_create_file("component", xxx) 3) remove platform_list 1st Note is that it is still using rtd->platform_com style. I guess it will be removed or replaced in the future, but it is not this patch-set purpose. 2nd Note is that, as you can see, above example is merging few components into 1 component. Almost all cases, 1 component is for register DAI, 1 component is for register platform. I think we can merge these into 1 component. I have 1 concern. Current snd_soc_unregister_component() is checking "dev" only. This means, 1 device : 1 component can only be correctly handled. I don't know detail, but below case, maybe works. snd_soc_register_component(dev, driver_A); snd_soc_register_component(dev, driver_B); snd_soc_unregister_component(dev); snd_soc_unregister_component(dev); I think we want to have "driver" parameter on unregister, like snd_soc_unregister_component(dev, driver_A); snd_soc_unregister_component(dev, driver_B); For example, tegra sound driver is registering its component, and snd_dmaengine_pcm_register() which register its component. One headache is that snd_dmaengine_pcm_unregister() is using snd_soc_lookup_platform() which will be replaced to snd_soc_lookup_component(). And its parameter is only "dev". We want to have below snd_soc_lookup_component(dev, driver); snd_soc_unregister_component(dev, driver); Can I replace current unregister before "replace platform" patch set ? (= many drivers are using it) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/3] ASoC: add/use asoc_simple_card_of_canonicalize_cpu()
by Kuninori Morimoto 22 Jun '17

22 Jun '17

Hi Mark These fixes audio graph cards DAI counting bug. Simple Card side doesn't have this issue, but Audio Graph Card side has it. Kuninori Morimoto (3): ASoC: simple-card-utils: add asoc_simple_card_of_canonicalize_cpu() ASoC: audio-graph-card: use asoc_simple_card_of_canonicalize_cpu() ASoC: audio-graph-scu-card: use asoc_simple_card_of_canonicalize_cpu() include/sound/simple_card_utils.h | 1 + sound/soc/generic/audio-graph-card.c | 4 +--- sound/soc/generic/audio-graph-scu-card.c | 4 +--- sound/soc/generic/simple-card-utils.c | 20 ++++++++++++++++++++ 4 files changed, 23 insertions(+), 6 deletions(-) -- 1.9.1

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[alsa-devel] [asoc:topic/pxa 2/2] /tmp/ccrgbnNE.s:1411: Error: unknown mnemonic `ldmia' -- `ldmia x26, {r0, r1, r2, r3}'
by kbuild test robot 21 Jun '17

21 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/pxa head: d89e69c0f985392e708b54c92b322163da820cb9 commit: d89e69c0f985392e708b54c92b322163da820cb9 [2/2] ASoC: pxa: add COMPILE_TEST on SND_MMP_SOC config: arm64-allmodconfig (attached as .config) compiler: aarch64-linux-gnu-gcc (Debian 6.1.1-9) 6.1.1 20160705 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d89e69c0f985392e708b54c92b322163da820cb9 # save the attached .config to linux build tree make.cross ARCH=arm64 All errors (new ones prefixed by >>): /tmp/ccrgbnNE.s: Assembler messages: >> /tmp/ccrgbnNE.s:1411: Error: unknown mnemonic `ldmia' -- `ldmia x26,{r0,r1,r2,r3}' >> /tmp/ccrgbnNE.s:1412: Error: unknown mnemonic `stmia' -- `stmia x19!,{r0,r1,r2,r3}' >> /tmp/ccrgbnNE.s:1618: Error: unknown mnemonic `ldmia' -- `ldmia x19!,{r0,r1,r2,r3}' >> /tmp/ccrgbnNE.s:1619: Error: unknown mnemonic `stmia' -- `stmia x26,{r0,r1,r2,r3}' --- 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 0/2] Add ZTE ZTE zx-aud96p22 audio codec driver
by Shawn Guo 21 Jun '17

21 Jun '17

From: Shawn Guo <shawn.guo(a)linaro.org> This is a pick-up of Baoyou's zx-aud96p22 audio codec driver submission [1], with quite some significant rework as below. - Instead of hard-coding the GPIO used to control external amplifier, use generic devm_gpiod_get_optional() call to retrieve the GPIO defined in device tree. - Rather than directly using I2C bus calls, use REGMAP_I2C interface which is more common for accessing codec register via I2C bus. - Use snd_kcontrol_new infrastructural for Volume and Mute control. - Use DAPM widget and route for various power bits control. - A lot of code cleanups [1] https://lkml.org/lkml/2017/2/15/170 Baoyou Xie (2): ASoC: add bindings for ZTE zx-aud96p22 audio codec ASoC: zx_aud96p22: add ZTE ZX AUD96P22 codec driver .../devicetree/bindings/sound/zte,zx-aud96p22.txt | 28 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/zx_aud96p22.c | 425 +++++++++++++++++++++ 4 files changed, 460 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt create mode 100644 sound/soc/codecs/zx_aud96p22.c -- 1.9.1

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[alsa-devel] [asoc:topic/pxa 2/2] sound/soc/pxa/mmp-pcm.c:130:3: error: implicit declaration of function '__phys_to_pfn'
by kbuild test robot 21 Jun '17

21 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/pxa head: d89e69c0f985392e708b54c92b322163da820cb9 commit: d89e69c0f985392e708b54c92b322163da820cb9 [2/2] ASoC: pxa: add COMPILE_TEST on SND_MMP_SOC config: ia64-allyesconfig (attached as .config) compiler: ia64-linux-gcc (GCC) 6.2.0 reproduce: wget https://raw.githubusercontent.com/01org/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d89e69c0f985392e708b54c92b322163da820cb9 # save the attached .config to linux build tree make.cross ARCH=ia64 All errors (new ones prefixed by >>): sound/soc/pxa/mmp-pcm.c: In function 'mmp_pcm_mmap': >> sound/soc/pxa/mmp-pcm.c:130:3: error: implicit declaration of function '__phys_to_pfn' [-Werror=implicit-function-declaration] __phys_to_pfn(runtime->dma_addr) + off, ^~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/__phys_to_pfn +130 sound/soc/pxa/mmp-pcm.c 7a824e214 Zhangfei Gao 2012-06-11 114 ac581e60d Lars-Peter Clausen 2013-03-22 115 dma_data.dma_res = r; ac581e60d Lars-Peter Clausen 2013-03-22 116 dma_data.ssp_id = cpu_dai->id; 7a824e214 Zhangfei Gao 2012-06-11 117 7c1c1d4a7 Lars-Peter Clausen 2013-04-15 118 return snd_dmaengine_pcm_open_request_chan(substream, filter, 7c1c1d4a7 Lars-Peter Clausen 2013-04-15 119 &dma_data); 7a824e214 Zhangfei Gao 2012-06-11 120 } 7a824e214 Zhangfei Gao 2012-06-11 121 7a824e214 Zhangfei Gao 2012-06-11 122 static int mmp_pcm_mmap(struct snd_pcm_substream *substream, 7a824e214 Zhangfei Gao 2012-06-11 123 struct vm_area_struct *vma) 7a824e214 Zhangfei Gao 2012-06-11 124 { 7a824e214 Zhangfei Gao 2012-06-11 125 struct snd_pcm_runtime *runtime = substream->runtime; 7a824e214 Zhangfei Gao 2012-06-11 126 unsigned long off = vma->vm_pgoff; 7a824e214 Zhangfei Gao 2012-06-11 127 7a824e214 Zhangfei Gao 2012-06-11 128 vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); 7a824e214 Zhangfei Gao 2012-06-11 129 return remap_pfn_range(vma, vma->vm_start, 7a824e214 Zhangfei Gao 2012-06-11 @130 __phys_to_pfn(runtime->dma_addr) + off, 7a824e214 Zhangfei Gao 2012-06-11 131 vma->vm_end - vma->vm_start, vma->vm_page_prot); 7a824e214 Zhangfei Gao 2012-06-11 132 } 7a824e214 Zhangfei Gao 2012-06-11 133 60e10d2fb Lars-Peter Clausen 2013-05-14 134 static struct snd_pcm_ops mmp_pcm_ops = { 7a824e214 Zhangfei Gao 2012-06-11 135 .open = mmp_pcm_open, 7c1c1d4a7 Lars-Peter Clausen 2013-04-15 136 .close = snd_dmaengine_pcm_close_release_chan, 7a824e214 Zhangfei Gao 2012-06-11 137 .ioctl = snd_pcm_lib_ioctl, 7a824e214 Zhangfei Gao 2012-06-11 138 .hw_params = mmp_pcm_hw_params, :::::: The code at line 130 was first introduced by commit :::::: 7a824e214e25a49442fe868dac0af8a904b24f58 ASoC: mmp: add audio dma support :::::: TO: Zhangfei Gao <zhangfei.gao(a)marvell.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.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] [PATCH 1/3] ASoC: fsl: mpc5200_dma: remove unused psc_dma
by Kuninori Morimoto 21 Jun '17

21 Jun '17

linux/sound/soc/fsl/mpc5200_dma.c:305:18: warning: unused variable \ psc_dma’ [-Wunused-variable] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/fsl/mpc5200_dma.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 93885d9..cd8ccaba 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -302,7 +302,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); size_t size = psc_dma_hardware.buffer_bytes_max; int rc; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC
by Kuninori Morimoto 21 Jun '17

21 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It doesn't use asm header. We can add COMPILE_TEST Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/pxa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 823b5a2..960744e 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA + depends on ARCH_PXA || COMPILE_TEST select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: stm32: add SPDIFRX driver
by olivier moysan 21 Jun '17

21 Jun '17

This patch-set handles the SPDIFRX on STM32 platforms. The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with IEC-60958 and IEC-61937 standards. SPDIFRX uses two DMA channels: - one DMA channel for S/PDIF data stream. - one DMA channel for control flow (channel status and user data bits). DMA is used for control flow instead of interrupts to get better performances, as channel status and user data are retrieved byte per byte. v1 -> v2: - fix merge issue olivier moysan (2): ASoC: stm32: Add DT bindings for SPDIFRX interface ASoC: stm32: Add SPDIFRX support .../devicetree/bindings/sound/st,stm32-spdifrx.txt | 56 ++ sound/soc/stm/Kconfig | 10 + sound/soc/stm/Makefile | 4 + sound/soc/stm/stm32_spdifrx.c | 998 +++++++++++++++++++++ 4 files changed, 1068 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 sound/soc/stm/stm32_spdifrx.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: fixup unsigned expression compared with zero: main_rate
by Kuninori Morimoto 21 Jun '17

21 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch fixes this WARNING sound/soc/sh/rcar/ssi.c:285:5-14: WARNING: Unsigned expression\ compared with zero: main_rate < 0 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Julia I think this patch solves reported issue, but my compiler and smatch doesn't indicate it. Thus, I couldn't confirm it. sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4892c0a..409bac3 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -763,7 +763,7 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, struct snd_interval *baseline, struct snd_interval *iv) { struct snd_interval p; - int rate; + unsigned int rate; int i; snd_interval_any(&p); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9428d4e..99c5761 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -680,7 +680,7 @@ void rsnd_ssi_parse_hdmi_connection(struct rsnd_priv *priv, void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); -int rsnd_ssi_clk_query(struct rsnd_priv *priv, +unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, int param1, int param2, int *idx); /* diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index c8956c3..46feddd 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -208,14 +208,14 @@ u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io) return 0; } -int rsnd_ssi_clk_query(struct rsnd_priv *priv, +unsigned int rsnd_ssi_clk_query(struct rsnd_priv *priv, int param1, int param2, int *idx) { int ssi_clk_mul_table[] = { 1, 2, 4, 8, 16, 6, 12, }; int j, ret; - int main_rate; + unsigned int main_rate; for (j = 0; j < ARRAY_SIZE(ssi_clk_mul_table); j++) { @@ -245,7 +245,7 @@ int rsnd_ssi_clk_query(struct rsnd_priv *priv, return main_rate; } - return -EINVAL; + return 0; } static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, @@ -282,7 +282,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, } main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx); - if (main_rate < 0) { + if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); return -EIO; } -- 1.9.1

2 1

[alsa-devel] [PATCH alsa-lib 0/4] Add the support for explicit sync of appl_ptr
by Takashi Iwai 21 Jun '17

21 Jun '17

Hi, this is a patchset for alsa-lib corresponding to the patchset I've sent for kernel. The first two patches do the actual fixes, while the last two are merely cleanups / optimizations. thanks, Takashi === Takashi Iwai (4): pcm: hw: Add the support for explicit sync of appl_ptr pcm: hw: Call USER_PVERSION ioctl at open pcm: hw: Remove superfluous call of snd_pcm_set_appl_ptr() pcm: hw: Reduce redundant sync_ptr() calls include/sound/asound.h | 4 +++- src/pcm/pcm_hw.c | 47 ++++++++++++++++++++++++++++++++--------------- 2 files changed, 35 insertions(+), 16 deletions(-) -- 2.13.1

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Re: [alsa-devel] Default procedure for sespend and resume state
by Takashi Iwai 21 Jun '17

21 Jun '17

On Wed, 21 Jun 2017 09:13:23 +0200, Kailang wrote: > > Hi Takashi, > > This patch will be more stable during sespend and resume state. It's "suspend" :) The code changes look OK, but please be more specific what problem this patch is trying to address. I suppose here "stability" meaning the pop-noise issue or such, not about the functionality of suspend/resume? thanks, Takashi

1 0

Re: [alsa-devel] Dell USB audio driver workaround
by Takashi Iwai 21 Jun '17

21 Jun '17

On Wed, 06 Apr 2016 08:29:12 +0200, Kailang wrote: > > Hi Takashi, > > Dell had USB dock. > Maybe firmware issue, the master volume always keep at 0x57. > Attach patch will fix it to keep at Max volume. > > Could you help us to update it? Well, I don't get how this patch works.. > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > index 4f85757..84d9a4e 100644 > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -846,6 +846,12 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, > { > struct snd_usb_audio *chip = cval->head.mixer->chip; > switch (chip->usb_id) { > + case USB_ID(0x0bda, 0x4014): /* Dell workaround */ > + if (strstr(kctl->id.name, "Playback Volume") != NULL) { > + cval->min = 0x0000; > + cval->max = 0x0000; > + } > + break; So, this means that this value has to be fixed to 0? If yes, can't we simply remove this volume instead? thanks, Takashi

4 16

Re: [alsa-devel] Support headset mode for ALC274
by Takashi Iwai 20 Jun '17

20 Jun '17

On Tue, 20 Jun 2017 10:50:57 +0200, Kailang wrote: > > Hi Takashi, > > Attach patch is support headset mode for ALC234/ALC274/ALC294. > > Many Thanks. Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH alsa-lib 1/2] USB-Audio: fix dsnoop args for Audiophile USB card
by Alexander Tsoy 20 Jun '17

20 Jun '17

Fixes: a9b129955659 ("USB-Audio.conf: fix definition for M-Audio AudioP...") Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- src/conf/cards/USB-Audio.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index e365f297..f72ef121 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -90,7 +90,7 @@ USB-Audio."Audiophile USB (tm)".pcm.default { type plug slave.pcm { @func concat - strings [ "dsnoop:DEVICE=1,CARD=" $CARD ] + strings [ "dsnoop:DEV=1,CARD=" $CARD ] } } } -- 2.13.0

2 3

[alsa-devel] [PATCH 1/2] ALSA: line6: add support for POD HD500X
by Hans P. Möller Ebner 20 Jun '17

20 Jun '17

Add support for the Line6 POD HD500X multi effect processor for playback and capture (in/out audio) through USB. Signed-off-by: Hans P. Moller <hmoller(a)uc.cl> --- sound/usb/line6/podhd.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 6ab23e5..e389f92 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -3,6 +3,7 @@ * * Copyright (C) 2011 Stefan Hajnoczi <stefanha(a)gmail.com> * Copyright (C) 2015 Andrej Krutak <dev(a)andree.sk> + * Copyright (C) 2015 Hans P. Moller <hmoller(a)uc.cl> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License as @@ -37,7 +38,8 @@ enum { LINE6_PODHD500_0, LINE6_PODHD500_1, LINE6_PODX3, - LINE6_PODX3LIVE + LINE6_PODX3LIVE, + LINE6_PODHD500X }; struct usb_line6_podhd { @@ -372,6 +374,7 @@ static const struct usb_device_id podhd_id_table[] = { { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 }, { LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 }, { LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE }, + { LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X }, {} }; @@ -446,6 +449,18 @@ static const struct line6_properties podhd_properties_table[] = { .ep_audio_r = 0x86, .ep_audio_w = 0x02, }, + [LINE6_PODHD500X] = { + .id = "PODHD500X", + .name = "POD HD500X", + .capabilities = LINE6_CAP_CONTROL + | LINE6_CAP_PCM | LINE6_CAP_HWMON, + .altsetting = 1, + .ep_ctrl_r = 0x81, + .ep_ctrl_w = 0x01, + .ctrl_if = 1, + .ep_audio_r = 0x86, + .ep_audio_w = 0x02, + }, }; /* -- 2.7.4

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[alsa-devel] [RESEND] topology: Add parsing for rates from conf
by Guneshwor Singh 20 Jun '17

20 Jun '17

In alsa-lib topology parser, rate_min and rate_max are parsed currently. Add support to parse rates also. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- include/topology.h | 1 + src/topology/pcm.c | 71 +++++++++++++++++++++++++++++++++++++++++++++++ src/topology/tplg_local.h | 20 +++++++++++++ 3 files changed, 92 insertions(+) diff --git a/include/topology.h b/include/topology.h index ccb3a004..42d23762 100644 --- a/include/topology.h +++ b/include/topology.h @@ -534,6 +534,7 @@ extern "C" { * SectionPCMCapabilities."name" { * * formats "S24_LE,S16_LE" # Supported formats + * rates "48000" # Supported rates * rate_min "48000" # Max supported sample rate * rate_max "48000" # Min supported sample rate * channels_min "2" # Min number of channels diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 0f4deb4f..bb63142e 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -20,6 +20,26 @@ #include "list.h" #include "tplg_local.h" +#define RATE(v) [SND_PCM_RATE_##v] = #v + +static const char *const snd_pcm_rate_names[] = { + RATE(5512), + RATE(8000), + RATE(11025), + RATE(16000), + RATE(22050), + RATE(32000), + RATE(44100), + RATE(48000), + RATE(64000), + RATE(88200), + RATE(96000), + RATE(176400), + RATE(192000), + RATE(CONTINUOUS), + RATE(KNOT), +}; + struct tplg_elem *lookup_pcm_dai_stream(struct list_head *base, const char* id) { struct list_head *pos; @@ -309,6 +329,42 @@ static int split_format(struct snd_soc_tplg_stream_caps *caps, char *str) return 0; } +static int get_rate_value(const char* name) +{ + int rate; + for (rate = 0; rate <= SND_PCM_RATE_LAST; rate++) { + if (snd_pcm_rate_names[rate] && + strcasecmp(name, snd_pcm_rate_names[rate]) == 0) { + return rate; + } + } + + return SND_PCM_RATE_UNKNOWN; +} + +static int split_rate(struct snd_soc_tplg_stream_caps *caps, char *str) +{ + char *s = NULL; + snd_pcm_rates_t rate; + int i = 0; + + s = strtok(str, ","); + while (s) { + rate = get_rate_value(s); + + if (rate == SND_PCM_RATE_UNKNOWN) { + SNDERR("error: unsupported stream rate %s\n", s); + return -EINVAL; + } + + caps->rates |= 1 << rate; + s = strtok(NULL, ", "); + i++; + } + + return 0; +} + /* Parse pcm stream capabilities */ int tplg_parse_stream_caps(snd_tplg_t *tplg, snd_config_t *cfg, void *private ATTRIBUTE_UNUSED) @@ -360,6 +416,21 @@ int tplg_parse_stream_caps(snd_tplg_t *tplg, continue; } + if (strcmp(id, "rates") == 0) { + s = strdup(val); + if (!s) + return -ENOMEM; + + err = split_rate(sc, s); + free(s); + + if (err < 0) + return err; + + tplg_dbg("\t\t%s: %s\n", id, val); + continue; + } + if (strcmp(id, "rate_min") == 0) { sc->rate_min = atoi(val); tplg_dbg("\t\t%s: %d\n", id, sc->rate_min); diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index 60af0177..af599145 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -37,6 +37,26 @@ struct tplg_ref; struct tplg_elem; +typedef enum _snd_pcm_rates { + SND_PCM_RATE_UNKNOWN = -1, + SND_PCM_RATE_5512 = 0, + SND_PCM_RATE_8000, + SND_PCM_RATE_11025, + SND_PCM_RATE_16000, + SND_PCM_RATE_22050, + SND_PCM_RATE_32000, + SND_PCM_RATE_44100, + SND_PCM_RATE_48000, + SND_PCM_RATE_64000, + SND_PCM_RATE_88200, + SND_PCM_RATE_96000, + SND_PCM_RATE_176400, + SND_PCM_RATE_192000, + SND_PCM_RATE_CONTINUOUS = 30, + SND_PCM_RATE_KNOT = 31, + SND_PCM_RATE_LAST = SND_PCM_RATE_KNOT, +} snd_pcm_rates_t; + struct snd_tplg { /* opaque vendor data */ -- 2.13.0

2 1

[alsa-devel] [PATCH 0/2] ALSA: add const to snd_ratnum structures
by Bhumika Goyal 20 Jun '17

20 Jun '17

Declare snd_ratnum structures as const as their reference is only stored in the rate field of a snd_pcm_hw_constraint_ratnums structure. This field is of type const, so snd_ratnum structures can be made const. Bhumika Goyal (2): ALSA: sound/isa: constify snd_ratnum structures ALSA: sound/pci: add const to snd_ratnum structures sound/isa/gus/gus_pcm.c | 2 +- sound/isa/sb/sb8_main.c | 2 +- sound/pci/bt87x.c | 2 +- sound/pci/ens1370.c | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH][next] ASoC: rsnd: make main_rate signed to check for -ve error return codes
by Colin King 20 Jun '17

20 Jun '17

From: Colin Ian King <colin.king(a)canonical.com> The comparison of main_rate < 0 has no effect because main_rate is an unsigned int. Make it signed so that any -ve error returns from the call to rsnd_clk_query can be detected. Detected by CoverityScan, CID#1446152 ("Unsigned compared against 0") Fixes: ef4cf5d6a143e0 ("ASoC: rsnd: add rsnd_ssi_clk_query()") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index c8956c3484dd..d220512918f9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -258,7 +258,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, struct rsnd_mod *ssi_parent_mod = rsnd_io_to_mod_ssip(io); int chan = rsnd_runtime_channel_for_ssi(io); int idx, ret; - unsigned int main_rate; + int main_rate; unsigned int rate = rsnd_io_is_play(io) ? rsnd_src_get_out_rate(priv, io) : rsnd_src_get_in_rate(priv, io); -- 2.11.0

2 1

[alsa-devel] [PATCH 1/1] ASoC: stm32: change configuration flag
by olivier moysan 19 Jun '17

19 Jun '17

Use a specific flag for SAI and I2S interfaces, instead of common flag. Signed-off-by: olivier moysan <olivier.moysan(a)st.com> --- sound/soc/stm/Kconfig | 19 ++++++++++++++++--- sound/soc/stm/Makefile | 6 +++--- 2 files changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/stm/Kconfig b/sound/soc/stm/Kconfig index a6372de..23600a5 100644 --- a/sound/soc/stm/Kconfig +++ b/sound/soc/stm/Kconfig @@ -1,8 +1,21 @@ -menuconfig SND_SOC_STM32 - tristate "STMicroelectronics STM32 SOC audio support" +menu "STMicroelectronics STM32 SOC audio support" + +config SND_SOC_STM32_SAI + tristate "STM32 SAI interface (Serial Audio Interface) support" depends on ARCH_STM32 || COMPILE_TEST depends on SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help - Say Y if you want to enable ASoC support for STM32 + Say Y if you want to enable SAI for STM32 + +config SND_SOC_STM32_I2S + tristate "STM32 I2S interface (SPI/I2S block) support" + depends on ARCH_STM32 || COMPILE_TEST + depends on SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y if you want to enable I2S for STM32 + +endmenu diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile index 8251931..4140c67 100644 --- a/sound/soc/stm/Makefile +++ b/sound/soc/stm/Makefile @@ -1,10 +1,10 @@ # SAI snd-soc-stm32-sai-sub-objs := stm32_sai_sub.o -obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai-sub.o +obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai-sub.o snd-soc-stm32-sai-objs := stm32_sai.o -obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-sai.o +obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai.o # I2S snd-soc-stm32-i2s-objs := stm32_i2s.o -obj-$(CONFIG_SND_SOC_STM32) += snd-soc-stm32-i2s.o +obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o -- 1.9.1

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[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Skylake: Add deep buffer support
by Subhransu S. Prusty 19 Jun '17

19 Jun '17

Now that we can configure DSP dma buffer size, the device can support deep buffer playback. DSP fetches large buffer and can result fewer wakes, which helps in power reduction. So this series first adds a token in topology to get dma_buffer_size and sends to firmware after conversion. Also ads a deepbuffer device to use this support. v1->v2 Remove an unrelated change. Ramesh Babu (2): ASoC: Intel: Skylake: Add deep buffer support ASoC: Intel: Skylake: Fix dma buffer size calculation Subhransu S. Prusty (1): ASoC: skl_rt286: Add deepbuffer dai link include/uapi/sound/snd_sst_tokens.h | 5 ++++- sound/soc/intel/boards/skl_rt286.c | 18 ++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 29 +++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.c | 4 ++++ sound/soc/intel/skylake/skl-topology.h | 1 + 5 files changed, 52 insertions(+), 5 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Add deep buffer support
by Subhransu S. Prusty 19 Jun '17

19 Jun '17

Now that we can configure DSP dma buffer size, the device can support deep buffer playback. DSP fetches large buffer and can result fewer wakes, which helps in power reduction. So this series first adds a token in topology to get dma_buffer_size and sends to firmware after conversion. Also ads a deepbuffer device to use this support. Ramesh Babu (2): ASoC: Intel: Skylake: Add deep buffer support ASoC: Intel: Skylake: Fix dma buffer size calculation Subhransu S. Prusty (1): ASoC: skl_rt286: Add deepbuffer dai link include/uapi/sound/snd_sst_tokens.h | 7 +++++-- sound/soc/intel/boards/skl_rt286.c | 18 ++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 29 +++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.c | 4 ++++ sound/soc/intel/skylake/skl-topology.h | 1 + 5 files changed, 53 insertions(+), 6 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH 0/2] ASoC: da7219: Resolve HP detect issues relating to MCLK
by Adam Thomson 19 Jun '17

19 Jun '17

The patch set resolves issues in the HP detect feature, relating to MCLK, where the feature can misdetect headphones as lineout. Adam Thomson (2): ASoC: da7219: Fix HP detection procedure for all MCLK frequencies ASoC: Intel: bxt: Move codec sysclk config to codec_init function sound/soc/codecs/da7219-aad.c | 31 ++++++++++++---- sound/soc/codecs/da7219.c | 53 ++++++++++++++++++--------- sound/soc/codecs/da7219.h | 5 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 14 ++++--- 4 files changed, 73 insertions(+), 30 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: compress: Derive substream from stream based on direction
by bgoswami＠codeaurora.org 19 Jun '17

19 Jun '17

From: Satish Babu Patakokila <sbpata(a)codeaurora.org> Currently compress driver hardcodes direction as playback to get substream from the stream. This results in getting the incorrect substream for compressed capture usecase. To fix this, remove the hardcoding and derive substream based on the stream direction. Signed-off-by: Satish Babu Patakokila <sbpata(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- sound/soc/soc-compress.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index bfd71b8..206f36b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -81,7 +81,8 @@ static int soc_compr_open(struct snd_compr_stream *cstream) static int soc_compr_open_fe(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; + struct snd_pcm_substream *fe_substream = + fe->pcm->streams[cstream->direction].substream; struct snd_soc_platform *platform = fe->platform; struct snd_soc_dai *cpu_dai = fe->cpu_dai; struct snd_soc_dpcm *dpcm; @@ -467,7 +468,8 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; + struct snd_pcm_substream *fe_substream = + fe->pcm->streams[cstream->direction].substream; struct snd_soc_platform *platform = fe->platform; struct snd_soc_dai *cpu_dai = fe->cpu_dai; int ret = 0, stream; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 2

[alsa-devel] [PATCH] ASoC: zx-i2s: flip I2S master/slave mode
by Shawn Guo 19 Jun '17

19 Jun '17

From: Shawn Guo <shawn.guo(a)linaro.org> The SND_SOC_DAIFMT_MASTER bits are defined to specify the master/slave mode for Codec, not I2S. So the I2S master/slave mode should be flipped according to SND_SOC_DAIFMT_MASTER bits. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/zte/zx-i2s.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index dd124d5acf01..16195ee776e5 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -203,13 +203,15 @@ static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - i2s->master = 1; - val |= ZX_I2S_TIMING_MAST; - break; - case SND_SOC_DAIFMT_CBS_CFS: + /* Codec is master, and I2S is slave. */ i2s->master = 0; val |= ZX_I2S_TIMING_SLAVE; break; + case SND_SOC_DAIFMT_CBS_CFS: + /* Codec is slave, and I2S is master. */ + i2s->master = 1; + val |= ZX_I2S_TIMING_MAST; + break; default: dev_err(cpu_dai->dev, "Unknown master/slave format\n"); return -EINVAL; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix potential race at unregistration and unsol events
by Takashi Iwai 19 Jun '17

19 Jun '17

When the codec device is unregistered / freed, it may release the resource while being used in an unsolicited event like the jack detection work. This leads to use-after-free. The fix here is to unregister the device at first, i.e. removing the codec from the list, then flushing the pending works to assure that all unsol events are gone. After this point, we're free from accessing the codec via unsol events, thus can release the resources gracefully. The issue was spotted originally by Intel CI, but it couldn't be reproduced reliably by its nature. So let's hope this fix really addresses the whole issues. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196045 Reported-by: Martin Peres <martin.peres(a)free.fr> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_bus.c | 1 + sound/hda/hdac_device.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 0e81ea89a596..714a51721a31 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -212,5 +212,6 @@ void snd_hdac_bus_remove_device(struct hdac_bus *bus, bus->caddr_tbl[codec->addr] = NULL; clear_bit(codec->addr, &bus->codec_powered); bus->num_codecs--; + flush_work(&bus->unsol_work); } EXPORT_SYMBOL_GPL(snd_hdac_bus_remove_device); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 03c9872c31cf..19deb306facb 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -159,6 +159,7 @@ void snd_hdac_device_unregister(struct hdac_device *codec) if (device_is_registered(&codec->dev)) { hda_widget_sysfs_exit(codec); device_del(&codec->dev); + snd_hdac_bus_remove_device(codec->bus, codec); } } EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); -- 2.13.1

1 0

[alsa-devel] [PATCH v3] ALSA: firewire-lib: Fix stall of process context at packet error
by Takashi Sakamoto 19 Jun '17

19 Jun '17

At Linux v3.5, packet processing can be done in process context of ALSA PCM application as well as software IRQ context for OHCI 1394. Below is an example of the callgraph (some calls are omitted). ioctl(2) with e.g. HWSYNC (sound/core/pcm_native.c) ->snd_pcm_common_ioctl1() ->snd_pcm_hwsync() ->snd_pcm_stream_lock_irq (sound/core/pcm_lib.c) ->snd_pcm_update_hw_ptr() ->snd_pcm_udpate_hw_ptr0() ->struct snd_pcm_ops.pointer() (sound/firewire/*) = Each handler on drivers in ALSA firewire stack (sound/firewire/amdtp-stream.c) ->amdtp_stream_pcm_pointer() (drivers/firewire/core-iso.c) ->fw_iso_context_flush_completions() ->struct fw_card_driver.flush_iso_completion() (drivers/firewire/ohci.c) = flush_iso_completions() ->struct fw_iso_context.callback.sc (sound/firewire/amdtp-stream.c) = in_stream_callback() or out_stream_callback() ->... ->snd_pcm_stream_unlock_irq When packet queueing error occurs or detecting invalid packets in 'in_stream_callback()' or 'out_stream_callback()', 'snd_pcm_stop_xrun()' is called on local CPU with disabled IRQ. (sound/firewire/amdtp-stream.c) in_stream_callback() or out_stream_callback() ->amdtp_stream_pcm_abort() ->snd_pcm_stop_xrun() ->snd_pcm_stream_lock_irqsave() ->snd_pcm_stop() ->snd_pcm_stream_unlock_irqrestore() The process is stalled on the CPU due to attempt to acquire recursive lock. [ 562.630853] INFO: rcu_sched detected stalls on CPUs/tasks: [ 562.630861] 2-...: (1 GPs behind) idle=37d/140000000000000/0 softirq=38323/38323 fqs=7140 [ 562.630862] (detected by 3, t=15002 jiffies, g=21036, c=21035, q=5933) [ 562.630866] Task dump for CPU 2: [ 562.630867] alsa-source-OXF R running task 0 6619 1 0x00000008 [ 562.630870] Call Trace: [ 562.630876] ? vt_console_print+0x79/0x3e0 [ 562.630880] ? msg_print_text+0x9d/0x100 [ 562.630883] ? up+0x32/0x50 [ 562.630885] ? irq_work_queue+0x8d/0xa0 [ 562.630886] ? console_unlock+0x2b6/0x4b0 [ 562.630888] ? vprintk_emit+0x312/0x4a0 [ 562.630892] ? dev_vprintk_emit+0xbf/0x230 [ 562.630895] ? do_sys_poll+0x37a/0x550 [ 562.630897] ? dev_printk_emit+0x4e/0x70 [ 562.630900] ? __dev_printk+0x3c/0x80 [ 562.630903] ? _raw_spin_lock+0x20/0x30 [ 562.630909] ? snd_pcm_stream_lock+0x31/0x50 [snd_pcm] [ 562.630914] ? _snd_pcm_stream_lock_irqsave+0x2e/0x40 [snd_pcm] [ 562.630918] ? snd_pcm_stop_xrun+0x16/0x70 [snd_pcm] [ 562.630922] ? in_stream_callback+0x3e6/0x450 [snd_firewire_lib] [ 562.630925] ? handle_ir_packet_per_buffer+0x8e/0x1a0 [firewire_ohci] [ 562.630928] ? ohci_flush_iso_completions+0xa3/0x130 [firewire_ohci] [ 562.630932] ? fw_iso_context_flush_completions+0x15/0x20 [firewire_core] [ 562.630935] ? amdtp_stream_pcm_pointer+0x2d/0x40 [snd_firewire_lib] [ 562.630938] ? pcm_capture_pointer+0x19/0x20 [snd_oxfw] [ 562.630943] ? snd_pcm_update_hw_ptr0+0x47/0x3d0 [snd_pcm] [ 562.630945] ? poll_select_copy_remaining+0x150/0x150 [ 562.630947] ? poll_select_copy_remaining+0x150/0x150 [ 562.630952] ? snd_pcm_update_hw_ptr+0x10/0x20 [snd_pcm] [ 562.630956] ? snd_pcm_hwsync+0x45/0xb0 [snd_pcm] [ 562.630960] ? snd_pcm_common_ioctl1+0x1ff/0xc90 [snd_pcm] [ 562.630962] ? futex_wake+0x90/0x170 [ 562.630966] ? snd_pcm_capture_ioctl1+0x136/0x260 [snd_pcm] [ 562.630970] ? snd_pcm_capture_ioctl+0x27/0x40 [snd_pcm] [ 562.630972] ? do_vfs_ioctl+0xa3/0x610 [ 562.630974] ? vfs_read+0x11b/0x130 [ 562.630976] ? SyS_ioctl+0x79/0x90 [ 562.630978] ? entry_SYSCALL_64_fastpath+0x1e/0xad This commit fixes the above bug. This assumes two cases: 1. Any error is detected in software IRQ context of OHCI 1394 context. In this case, PCM substream should be aborted in packet handler. On the other hand, it should not be done in any process context. TO distinguish these two context, use 'in_interrupt()' macro. 2. Any error is detect in process context of ALSA PCM application. In this case, PCM substream should not be aborted in packet handler because PCM substream lock is acquired. The task to abort PCM substream should be done in ALSA PCM core. For this purpose, SNDRV_PCM_POS_XRUN is returned at 'struct snd_pcm_ops.pointer()'. Suggested-by: Clemens Ladisch <clemens(a)ladisch.de> Fixes: e9148dddc3c7("ALSA: firewire-lib: flush completed packets when reading PCM position") Cc: <stable(a)vger.kernel.org> # 4.9+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 8 ++++++-- sound/firewire/amdtp-stream.h | 2 +- 2 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 9678bc75dc5b..3fc581a5ad62 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -701,7 +701,9 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, cycle = increment_cycle_count(cycle, 1); if (s->handle_packet(s, 0, cycle, i) < 0) { s->packet_index = -1; - amdtp_stream_pcm_abort(s); + if (in_interrupt()) + amdtp_stream_pcm_abort(s); + WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); return; } } @@ -753,7 +755,9 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, /* Queueing error or detecting invalid payload. */ if (i < packets) { s->packet_index = -1; - amdtp_stream_pcm_abort(s); + if (in_interrupt()) + amdtp_stream_pcm_abort(s); + WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); return; } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 6d613f2eb612..ed6eafd10992 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -135,7 +135,7 @@ struct amdtp_stream { /* For a PCM substream processing. */ struct snd_pcm_substream *pcm; struct tasklet_struct period_tasklet; - unsigned int pcm_buffer_pointer; + snd_pcm_uframes_t pcm_buffer_pointer; unsigned int pcm_period_pointer; /* To wait for first packet. */ -- 2.11.0

2 2

[alsa-devel] [PATCH v3] ALSA: core: let low-level driver or userspace disable rewinds
by Subhransu S. Prusty 19 Jun '17

19 Jun '17

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add new hw_params flag to explicitly tell driver that rewinds will never be used. This can be used by low-level driver to optimize DMA operations and reduce power consumption. Use this flag only when data written in ring buffer will never be invalidated, e.g. any update of appl_ptr is final. Note that the update of appl_ptr include both a read/write data operation as well as snd_pcm_forward() whose behavior is not modified. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ramesh Babu <ramesh.babu(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v2 -> v3 return error for rewind operation if no_rewinds set. include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 8 ++++++++ 3 files changed, 10 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 79fedf517070..c1e2b87cd409 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -368,6 +368,7 @@ struct snd_pcm_runtime { unsigned int rate_num; unsigned int rate_den; unsigned int no_period_wakeup: 1; + unsigned int no_rewinds:1; /* -- SW params -- */ int tstamp_mode; /* mmap timestamp is updated */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index fd41697cb4d3..c697ff90450d 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -365,6 +365,7 @@ struct snd_pcm_info { #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ #define SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER (1<<1) /* export buffer */ #define SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP (1<<2) /* disable period wakeups */ +#define SNDRV_PCM_HW_PARAMS_NO_REWINDS (1<<3) /* disable rewinds */ struct snd_interval { unsigned int min, max; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index bf5d0f2acfb9..6769f4751fa0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -554,6 +554,8 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->no_period_wakeup = (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + runtime->no_rewinds = + (params->flags & SNDRV_PCM_HW_PARAMS_NO_REWINDS) ? 1 : 0; bits = snd_pcm_format_physical_width(runtime->format); runtime->sample_bits = bits; @@ -2521,6 +2523,9 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst if (frames == 0) return 0; + if (runtime->no_rewinds) + return -ENODEV; + snd_pcm_stream_lock_irq(substream); ret = do_pcm_hwsync(substream); if (!ret) @@ -2539,6 +2544,9 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr if (frames == 0) return 0; + if (runtime->no_rewinds) + return -ENODEV; + snd_pcm_stream_lock_irq(substream); ret = do_pcm_hwsync(substream); if (!ret) -- 1.9.1

4 5

[alsa-devel] [RFC][PATCH] ALSA: pcm: introduce ACK_APPLPTR flag and bump up interface version to v2.0.14
by Takashi Sakamoto 19 Jun '17

19 Jun '17

Hi, This patchset includes commits for both kernel/user land for an issue addressed in this message thread[1]. In a development period for v4.13, it's clear that there's a type of devices which demand userspace applications to change their behaviour. These devices can take care of application-side pointer on PCM buffer for their data transmission. As a result, these devices achieve some advantages. For these devices, drivers need to get feedback about current position of the pointer from applications when they queue/dequeue PCM frames on the buffer. ALSA PCM core partly assist the above design. The core can call a handler in driver side when applications operate PCM frames by ioctl(2) with SNDRV_PCM_IOCTL_[READ|WRITE][N|I] command. However, when operating for mapped PCM buffer, it can't assist applications for the above design. In this scenario, applications should call ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR after handle any PCM frames on the buffer. This patchset bumps up interface version with new info flag to notify the above change to user land developers. This patchset is written with below ideas: 1. Stuffs in user land can tell their assists for the above design to kernel land by using hw_param flag. When they don't assist, they see on hw_params data that mmap operation is not supported. 2. Drivers can acknowledge the above flag. When registering appropriate PCM rules in advance, they can calculate values for each parameters in both cases that stuffs in user land can/cannot assist it. This is useful for packet-oriented drivers to judge utilizing any interrupt context for packetization. 3. This is device-specific issue, independently of platform architecture. On x86/ppc/alpha platforms, status/control data of PCM substream is still available in process' VMA of application via page frame mapping. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/121438.html For kernel land: Takashi Sakamoto (1): ALSA: pcm: introduce ACK_APPLPTR flag and bump up interface version to 2.0.14 include/uapi/sound/asound.h | 11 ++++++++++- sound/core/pcm_native.c | 25 +++++++++++++++++++++++++ 2 files changed, 35 insertions(+), 1 deletion(-) For user land: Takashi Sakamoto (2): ALSA: pcm: introduce ACK_APPLPTR flag and bump up interface version to 2.0.14 pcm: add support for interface version 2.0.14 include/sound/asound.h | 11 ++++++++++- src/pcm/pcm_hw.c | 43 +++++++++++++++++++++++++++++++++++++------ 2 files changed, 47 insertions(+), 7 deletions(-) -- 2.11.0

2 12

[alsa-devel] [PATCH] ALSA: pcm: Fix possible inconsistent appl_ptr update via mmap
by Takashi Iwai 19 Jun '17

19 Jun '17

The ALSA PCM core refers to the appl_ptr value stored on the mmapped page that is shared between kernel and user-space. Although the reference is performed in the PCM stream lock, it doesn't guarantee the atomic access when the value gets updated concurrently from the user-space on another CPU. In most of codes, this is no big problem, but still there are a few places that may result in slight inconsistencies because they access runtime->control->appl_ptr multiple times; that is, the second read might be a different value from the first value. It can be even backward or jumping, as we have no control for it. Hence, the calculation may give an unexpected value. Luckily, there is no security vulnerability by that, as far as I've checked. But still we should address it. This patch tries to reduce such possible cases. The fix is simple -- we just read once, store it to a local variable and use it for the rest calculations. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 631cd598ba6c..cda258d91490 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -65,15 +65,16 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram if (runtime->silence_size < runtime->boundary) { snd_pcm_sframes_t noise_dist, n; - if (runtime->silence_start != runtime->control->appl_ptr) { - n = runtime->control->appl_ptr - runtime->silence_start; + snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr; + if (runtime->silence_start != appl_ptr) { + n = appl_ptr - runtime->silence_start; if (n < 0) n += runtime->boundary; if ((snd_pcm_uframes_t)n < runtime->silence_filled) runtime->silence_filled -= n; else runtime->silence_filled = 0; - runtime->silence_start = runtime->control->appl_ptr; + runtime->silence_start = appl_ptr; } if (runtime->silence_filled >= runtime->buffer_size) return; @@ -2224,7 +2225,9 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, continue; /* draining */ } frames = size > avail ? avail : size; - cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size; + appl_ptr = runtime->control->appl_ptr; + appl_ofs = appl_ptr % runtime->buffer_size; + cont = runtime->buffer_size - appl_ofs; if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { @@ -2232,8 +2235,6 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); return -EINVAL; } - appl_ptr = runtime->control->appl_ptr; - appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); err = writer(substream, appl_ofs, data, offset, frames, transfer); -- 2.13.1

3 6

[alsa-devel] [PATCH 0/7] ASoC: stm32: sai: misc fixes
by olivier moysan 19 Jun '17

19 Jun '17

This patch-set gathers various fixes on STM32 SAI driver. olivier moysan (7): dt-bindings: stm32: sai: fix DT example ASoC: stm32: change SAI configuration flag ASoC: stm32: sai: typo fixes ASoC: stm32: sai: remove spurious trace ASoC: stm32: sai: change stop sequence ASoC: stm32: sai: fix clock management ASoC: stm32: sai: manage master clock .../devicetree/bindings/sound/st,stm32-sai.txt | 25 ++++------ sound/soc/stm/Kconfig | 10 ++-- sound/soc/stm/Makefile | 4 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai.h | 1 - sound/soc/stm/stm32_sai_sub.c | 57 +++++++++++++--------- 6 files changed, 54 insertions(+), 45 deletions(-) -- 1.9.1

2 16

[alsa-devel] MIDI Clock issue when aconnect multiple times to same device.
by Allan Klinbail 19 Jun '17

19 Jun '17

Hi All, I have noticed an issue with the handling of MIDI clock when making multiple connections to and from the same device.. Specifically with EMAGIC AMT-8 and Unitor 8 Mk2 (I can't confirm it is happening for other devices). The scenario is MIDI Port 1 is receiving MIDI clock along with note information for devices on ports 2 & 3 .. (in this scenario sending at 120 BPM) The MIDI clock signal is automatically distributed by the device to all other ports. When connecting Input 1 to output 2 (to send MIDI from the first external device to the second).. The MIDI clock appears to be retransmitted a second time.. the device on port 2 then believes it is receiving a clock signal at approximately 168 BPM. This signal is not stable When connecting input 1 to output 3 the MIDI clock seems to be distributed another time to the EMAGIC .. which is now sending out an even more unstable clock signal at approximate 189 BPM.. It is possible to filter physical inputs from receiving MIDI Clock or outputs from sending. This does not solve this issue (Filtering the input stops the clock altogether, filtering the output stops it from being received by the desired device) However it is not possible to filter the messaging going to to the unit from a connection established with aconnect (or using tools like qjackctl). I am not sure if it is possible to fix the driver for this issue,especially if the MIDI clock source is external. However, I thought it best to alert you to this situation. thanks Allan

1 0

[alsa-devel] [PATCH] oss: Fix a sleep-in-atomic bug in midi_outc
by Jia-Ju Bai 19 Jun '17

19 Jun '17

The driver may sleep under a spin lock, and the function call path is: midi_outc (acquire the lock by spin_lock_irqsave) oss_broken_sleep_on schedule_timeout --> may sleep To fix it, the lock is released before oss_broken_sleep_on, and the lock is acquired again after this function. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- sound/oss/sequencer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index f19da4b..3d95d752 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1211,7 +1211,9 @@ static void midi_outc(int dev, unsigned char data) spin_lock_irqsave(&lock,flags); while (n && !midi_devs[dev]->outputc(dev, data)) { + spin_unlock_irqrestore(&lock, flags); oss_broken_sleep_on(&seq_sleeper, HZ/25); + spin_lock_irqsave(&lock, flags); n--; } spin_unlock_irqrestore(&lock,flags); -- 1.7.9.5

1 0

[alsa-devel] [PATCH 2] ALSA: pcm: Fix possible inconsistent appl_ptr update via mmap
by Takashi Iwai 19 Jun '17

19 Jun '17

The ALSA PCM core refers to the appl_ptr value stored on the mmapped page that is shared between kernel and user-space. Although the reference is performed in the PCM stream lock, it doesn't guarantee the atomic access when the value gets updated concurrently from the user-space on another CPU. In most of codes, this is no big problem, but still there are a few places that may result in slight inconsistencies because they access runtime->control->appl_ptr multiple times; that is, the second read might be a different value from the first value. It can be even backward or jumping, as we have no control for it. Hence, the calculation may give an unexpected value. Luckily, there is no security vulnerability by that, as far as I've checked. But still we should address it. This patch tries to reduce such possible cases. The fix is simple -- we just read once, store it to a local variable and use it for the rest calculations. The READ_ONCE() macro is used for it in order to avoid the ill-effect by possible compiler optimizations. Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Use proper READ_ONCE() macro. sound/core/pcm_lib.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 461c21f21caf..415346b84ff8 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -65,15 +65,16 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram if (runtime->silence_size < runtime->boundary) { snd_pcm_sframes_t noise_dist, n; - if (runtime->silence_start != runtime->control->appl_ptr) { - n = runtime->control->appl_ptr - runtime->silence_start; + snd_pcm_uframes_t appl_ptr = READ_ONCE(runtime->control->appl_ptr); + if (runtime->silence_start != appl_ptr) { + n = appl_ptr - runtime->silence_start; if (n < 0) n += runtime->boundary; if ((snd_pcm_uframes_t)n < runtime->silence_filled) runtime->silence_filled -= n; else runtime->silence_filled = 0; - runtime->silence_start = runtime->control->appl_ptr; + runtime->silence_start = appl_ptr; } if (runtime->silence_filled >= runtime->buffer_size) return; @@ -2203,7 +2204,9 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, continue; /* draining */ } frames = size > avail ? avail : size; - cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size; + appl_ptr = READ_ONCE(runtime->control->appl_ptr); + appl_ofs = appl_ptr % runtime->buffer_size; + cont = runtime->buffer_size - appl_ofs; if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { @@ -2211,8 +2214,6 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); return -EINVAL; } - appl_ptr = runtime->control->appl_ptr; - appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); err = writer(substream, appl_ofs, data, offset, frames, transfer); -- 2.13.1

1 0

[alsa-devel] [PATCH v1 0/2] Patches to add support pdm controller for rk3328
by Sugar Zhang 18 Jun '17

18 Jun '17

These patches to add support pdm for rk3328 socs. Sugar Zhang (2): ASoC: rockchip: add support for pdm controller arm64: dts: rk3328: add pdm node .../devicetree/bindings/sound/rockchip,pdm.txt | 39 ++ arch/arm64/boot/dts/rockchip/rk3328.dtsi | 55 +++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_pdm.c | 516 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.h | 83 ++++ 6 files changed, 704 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 sound/soc/rockchip/rockchip_pdm.c create mode 100644 sound/soc/rockchip/rockchip_pdm.h -- 1.9.1

3 3

[alsa-devel] [PATCH 0/3] ALSA: firewire: use .ack callback of PCM operation for packet processing
by Takashi Sakamoto 18 Jun '17

18 Jun '17

Hi, In recent work for ALSA PCM core, driver-side implementation for 'struct snd_pcm_ops.ack' is called when appl_ptr in intermediate PCM buffer is changed, except for a case that mapped page frame is used for the buffer and userspace applications doesn't voluntarily notify the change. In packet-oriented drivers, this callback is useful for the timing to queue packets for data transmission with burstness. If things works well, reduction of latency between PCM frame queueing and actual data transmission. This patchset is for the purpose, additionally including minor code refactoring. Major use case of ALSA PCM interface is memory mapped I/O and this patchset adds no advantage in this case due to design of the .ack callback, as I described. Takashi Sakamoto (2): ALSA: firewire: process packets in 'struct snd_pcm_ops.ack' callback ALSA: fireface: constify ALSA specific operations sound/firewire/amdtp-stream.c | 19 ++++++++++ sound/firewire/amdtp-stream.h | 1 + sound/firewire/bebob/bebob_pcm.c | 16 +++++++++ sound/firewire/dice/dice-pcm.c | 18 ++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 16 +++++++++ sound/firewire/fireface/ff-midi.c | 22 ++++++------ sound/firewire/fireface/ff-pcm.c | 62 +++++++++++++++++++------------- sound/firewire/fireworks/fireworks_pcm.c | 16 +++++++++ sound/firewire/motu/motu-pcm.c | 16 +++++++++ sound/firewire/oxfw/oxfw-pcm.c | 16 +++++++++ sound/firewire/tascam/tascam-pcm.c | 16 +++++++++ 11 files changed, 182 insertions(+), 36 deletions(-) -- 2.11.0

3 22

[alsa-devel] [PATCH v1] ASoC: rt5677: Hide platform data in the module sources
by Andy Shevchenko 18 Jun '17

18 Jun '17

There is no user of legacy platform data. Remove separate header and hide its content inside module sources. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- include/sound/rt5677.h | 45 --------------------------------------------- sound/soc/codecs/rt5677.c | 27 +++++++++++++++++---------- sound/soc/codecs/rt5677.h | 30 +++++++++++++++++++++++++++++- 3 files changed, 46 insertions(+), 56 deletions(-) delete mode 100644 include/sound/rt5677.h diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h deleted file mode 100644 index a6207043ac3c..000000000000 --- a/include/sound/rt5677.h +++ /dev/null @@ -1,45 +0,0 @@ -/* - * linux/sound/rt5677.h -- Platform data for RT5677 - * - * Copyright 2013 Realtek Semiconductor Corp. - * Author: Oder Chiou <oder_chiou(a)realtek.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __LINUX_SND_RT5677_H -#define __LINUX_SND_RT5677_H - -enum rt5677_dmic2_clk { - RT5677_DMIC_CLK1 = 0, - RT5677_DMIC_CLK2 = 1, -}; - - -struct rt5677_platform_data { - /* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */ - bool in1_diff; - bool in2_diff; - bool lout1_diff; - bool lout2_diff; - bool lout3_diff; - /* DMIC2 clock source selection */ - enum rt5677_dmic2_clk dmic2_clk_pin; - - /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ - u8 gpio_config[6]; - - /* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */ - unsigned int jd1_gpio; - /* jd2 and jd3 can select 0 ~ 3 as - OFF, GPIO4, GPIO5 and GPIO6 respectively */ - unsigned int jd2_gpio; - unsigned int jd3_gpio; - - /* Set MICBIAS1 VDD 1v8 or 3v3 */ - bool micbias1_vdd_3v3; -}; - -#endif diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 36e530a36c82..29c94d1ff56c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/firmware.h> +#include <linux/of_device.h> #include <linux/property.h> #include <sound/core.h> #include <sound/pcm.h> @@ -5019,14 +5020,12 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", RT5677 }, - { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", }, + { .compatible = "realtek,rt5677", RT5677 }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); @@ -5147,7 +5146,6 @@ static void rt5677_free_irq(struct i2c_client *i2c) static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct rt5677_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5677_priv *rt5677; int ret; unsigned int val; @@ -5159,16 +5157,25 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); - rt5677->type = id->driver_data; + if (i2c->dev.of_node) { + const struct of_device_id *match_id; + + match_id = of_match_device(rt5677_of_match, &i2c->dev); + if (match_id) + rt5677->type = (enum rt5677_type)match_id->data; - if (pdata) - rt5677->pdata = *pdata; - else if (i2c->dev.of_node) rt5677_read_device_properties(rt5677, &i2c->dev); - else if (ACPI_HANDLE(&i2c->dev)) + } else if (ACPI_HANDLE(&i2c->dev)) { + const struct acpi_device_id *acpi_id; + + acpi_id = acpi_match_device(rt5677_acpi_match, &i2c->dev); + if (acpi_id) + rt5677->type = (enum rt5677_type)acpi_id->driver_data; + rt5677_read_acpi_properties(rt5677, &i2c->dev); - else + } else { return -EINVAL; + } /* pow-ldo2 and reset are optional. The codec pins may be statically * connected on the board without gpios. If the gpio device property diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d46855a42c40..97239973edc4 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -12,7 +12,6 @@ #ifndef __RT5677_H__ #define __RT5677_H__ -#include <sound/rt5677.h> #include <linux/gpio/driver.h> #include <linux/gpio/consumer.h> @@ -1761,6 +1760,35 @@ enum { RT5677_I2S4_SOURCE = (0x1 << 18), }; +enum rt5677_dmic2_clk { + RT5677_DMIC_CLK1 = 0, + RT5677_DMIC_CLK2 = 1, +}; + +struct rt5677_platform_data { + /* IN1/IN2/LOUT1/LOUT2/LOUT3 can optionally be differential */ + bool in1_diff; + bool in2_diff; + bool lout1_diff; + bool lout2_diff; + bool lout3_diff; + /* DMIC2 clock source selection */ + enum rt5677_dmic2_clk dmic2_clk_pin; + + /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ + u8 gpio_config[6]; + + /* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */ + unsigned int jd1_gpio; + /* jd2 and jd3 can select 0 ~ 3 as + OFF, GPIO4, GPIO5 and GPIO6 respectively */ + unsigned int jd2_gpio; + unsigned int jd3_gpio; + + /* Set MICBIAS1 VDD 1v8 or 3v3 */ + bool micbias1_vdd_3v3; +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: intel: Use kvzalloc() for suspend buffers
by Takashi Iwai 16 Jun '17

16 Jun '17

Intel SST driver allocates lots of pages at suspend for saving the firmware states, and this may occasionally lead to the allocation error due to the high order, ending up with the suspend failure. Use kvzalloc() so that it can fall back to vmalloc() gracefully. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst/sst.c | 22 +++++++++++----------- sound/soc/intel/atom/sst/sst.h | 8 ++++---- 2 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index f9ba71315e33..b4aa5b6f79d9 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -499,23 +499,23 @@ static int intel_sst_suspend(struct device *dev) fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL); if (!fw_save) return -ENOMEM; - fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); + fw_save->iram = kvzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); if (!fw_save->iram) { ret = -ENOMEM; goto iram; } - fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); + fw_save->dram = kvzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); if (!fw_save->dram) { ret = -ENOMEM; goto dram; } - fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); + fw_save->sram = kvzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); if (!fw_save->sram) { ret = -ENOMEM; goto sram; } - fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); + fw_save->ddr = kvzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); if (!fw_save->ddr) { ret = -ENOMEM; goto ddr; @@ -530,11 +530,11 @@ static int intel_sst_suspend(struct device *dev) ctx->ops->reset(ctx); return 0; ddr: - kfree(fw_save->sram); + kvfree(fw_save->sram); sram: - kfree(fw_save->dram); + kvfree(fw_save->dram); dram: - kfree(fw_save->iram); + kvfree(fw_save->iram); iram: kfree(fw_save); return ret; @@ -562,10 +562,10 @@ static int intel_sst_resume(struct device *dev) memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE); memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base); - kfree(fw_save->sram); - kfree(fw_save->dram); - kfree(fw_save->iram); - kfree(fw_save->ddr); + kvfree(fw_save->sram); + kvfree(fw_save->dram); + kvfree(fw_save->iram); + kvfree(fw_save->ddr); kfree(fw_save); block = sst_create_block(ctx, 0, FW_DWNL_ID); diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 5c9a51cc77aa..2ac1aa8e13d1 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -338,10 +338,10 @@ struct sst_shim_regs64 { }; struct sst_fw_save { - void *iram; - void *dram; - void *sram; - void *ddr; + void *iram; /* allocated via kvmalloc() */ + void *dram; /* allocated via kvmalloc() */ + void *sram; /* allocated via kvmalloc() */ + void *ddr; /* allocated via kvmalloc() */ }; /** -- 2.13.1

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: rt5677: Move platform code to where it belongs
by Andy Shevchenko 16 Jun '17

16 Jun '17

There are few issues with current approach: - ACPI enumeration should be done by IDs, and not by instances - Platform code with board related stuff should reside in corresponding module Fix those issues here. Andy Shevchenko (3): ASoC: rt5677: Introduce proper table for ACPI enumeration ASoC: rt5677: Move platform code to board file ASoC: Intel: bdw-rt5677: Switch to devm_acpi_dev_add_driver_gpios() sound/soc/codecs/rt5677.c | 32 +++++++------------------------- sound/soc/intel/boards/bdw-rt5677.c | 26 ++++++++++++++++++++++++++ 2 files changed, 33 insertions(+), 25 deletions(-) -- 2.11.0

2 6

[alsa-devel] [PATCH v2] ASoC: ak4613: add hw_constraint rule for Sampling Rate
by Kuninori Morimoto 16 Jun '17

16 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ak4613 accepts all range of Sampling Rate, but it depends on inputed master clock. This patch adds hw constraint rule for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- v1 -> v2 - Merged v1's [1/2][2/2] into this patch. - Removed CKS settings (= was [1/2]). It was Master mode settings which is not yet supported - tidyup ak4613_hw_constraints() method sound/soc/codecs/ak4613.c | 61 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 61 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 27c44c7..690edeb 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -94,6 +94,8 @@ struct ak4613_interface { struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; + struct snd_pcm_hw_constraint_list constraint; + unsigned int sysclk; unsigned int fmt; u8 oc; @@ -252,6 +254,50 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, mutex_unlock(&priv->lock); } +static void ak4613_hw_constraints(struct ak4613_priv *priv, + struct snd_pcm_runtime *runtime) +{ + static const unsigned int ak4613_rates[] = { + 32000, + 44100, + 48000, + 64000, + 88200, + 96000, + 176400, + 192000, + }; + struct snd_pcm_hw_constraint_list *constraint = &priv->constraint; + unsigned int fs; + int i; + + constraint->list = ak4613_rates; + constraint->mask = 0; + constraint->count = 0; + + /* + * Slave Mode + * Normal: [32kHz, 48kHz] : 256fs,384fs or 512fs + * Double: [64kHz, 96kHz] : 256fs + * Quad : [128kHz,192kHz]: 128fs + * + * Master mode + * Normal: [32kHz, 48kHz] : 256fs or 512fs + * Double: [64kHz, 96kHz] : 256fs + * Quad : [128kHz,192kHz]: 128fs + */ + for (i = 0; i < ARRAY_SIZE(ak4613_rates); i++) { + /* minimum fs on each range */ + fs = (ak4613_rates[i] <= 96000) ? 256 : 128; + + if (priv->sysclk >= ak4613_rates[i] * fs) + constraint->count = i + 1; + } + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, constraint); +} + static int ak4613_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -260,6 +306,19 @@ static int ak4613_dai_startup(struct snd_pcm_substream *substream, priv->cnt++; + ak4613_hw_constraints(priv, substream->runtime); + + return 0; +} + +static int ak4613_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->sysclk = freq; + return 0; } @@ -411,6 +470,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, + .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, .hw_params = ak4613_dai_hw_params, }; @@ -529,6 +589,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; + priv->sysclk = 0; mutex_init(&priv->lock); -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: simple-card-utils: add asoc_simple_card_of_parse_widgets()
by Kuninori Morimoto 16 Jun '17

16 Jun '17

Hi Mark Simple/OF-Audio card are using same DT properties, but not using same functions for it. Now, simple-card is only supporting "widgets" feature. I hope it will be supported on simple-scu/of-audio/of-audio-scu card too (I'm not widgets user, thus, I can't test). Then, DT properties needs to be shared with all Simple/OF-Audio cards. This means if needs common function on simple-card-utils. Currently only 1 user exist, thus, this is a little bit aggressive, but encapsulation is one of simple card util's purpose. This patches adds common widgets function on utils, and use it from simple-card. Kuninori Morimoto (2): ASoC: simple-card-utils: add asoc_simple_card_of_parse_widgets() ASoC: simple-card: use asoc_simple_card_of_parse_widgets() include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ sound/soc/generic/simple-card.c | 10 +++------- 3 files changed, 24 insertions(+), 7 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 0/2] ASoC: stm32: sai: Add H7 support
by olivier moysan 16 Jun '17

16 Jun '17

This patch-set adds support of the Serial Audio Interface (SAI) IP on H7 STM32 platform family. olivier moysan (2): dt-bindings: stm32: add h7 support for sai ASoC: stm32: sai: add h7 support .../devicetree/bindings/sound/st,stm32-sai.txt | 18 ++--- sound/soc/stm/stm32_sai.c | 13 ++- sound/soc/stm/stm32_sai.h | 72 +++++++++++++++-- sound/soc/stm/stm32_sai_sub.c | 92 ++++++++++++++++++---- 4 files changed, 164 insertions(+), 31 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v1 1/3] ASoC: rt5677: Introduce proper table for ACPI enumeration
by Andy Shevchenko 16 Jun '17

16 Jun '17

I2C devices are enumerated by IDs, and not by instances. Make it clear by using proper module device table for ACPI case. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/codecs/rt5677.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 65ac4518ad06..64cf6cc41ad1 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5030,7 +5030,6 @@ static const struct regmap_config rt5677_regmap = { static const struct i2c_device_id rt5677_i2c_id[] = { { "rt5677", RT5677 }, { "rt5676", RT5676 }, - { "RT5677CE:00", RT5677 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -5041,6 +5040,14 @@ static const struct of_device_id rt5677_of_match[] = { }; MODULE_DEVICE_TABLE(of, rt5677_of_match); +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_match[] = { + { "RT5677CE", RT5677 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_match); +#endif + static const struct acpi_gpio_params plug_det_gpio = { RT5677_GPIO_PLUG_DET, 0, false }; static const struct acpi_gpio_params mic_present_gpio = { RT5677_GPIO_MIC_PRESENT_L, 0, false }; static const struct acpi_gpio_params headphone_enable_gpio = { RT5677_GPIO_HP_AMP_SHDN_L, 0, false }; @@ -5301,6 +5308,7 @@ static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .of_match_table = rt5677_of_match, + .acpi_match_table = ACPI_PTR(rt5677_acpi_match), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, -- 2.11.0

2 4

[alsa-devel] [PATCH v3] ASoC: dapm: Add new widget type snd_soc_dapm_dsp_component
by Liam Girdwood 16 Jun '17

16 Jun '17

Add a new widget type to represent internal DSP components and allow DSP drivers and firmware to define topologies using this widget type. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Takashi, I'll send a subsequent patch for alsa-lib if Mark is happy with this version. Changes V2. o No need for multiple types since core does not care. Changes v1. o Added some documentation. o Split codec widget into encoder and decoder to avoid confusion with HW CODECS. Documentation/sound/soc/dapm.rst | 4 ++++ include/sound/soc-dapm.h | 1 + include/uapi/sound/asoc.h | 3 ++- sound/soc/soc-topology.c | 1 + 4 files changed, 8 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index a27f42befa4d..540a86012e29 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -105,6 +105,10 @@ Pre Special PRE widget (exec before all others) Post Special POST widget (exec after all others) +DSP Component + Internal DSP component not used by the ASoC core but may be used + by DSP driver and DSP firmware. + (Widgets are defined in include/sound/soc-dapm.h) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..87133c974991 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_dsp_component, /* DSP internal component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..e805ef651538 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,8 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_DSP_COMPONENT 16 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DSP_COMPONENT /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 12e189701924..17751736957f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,7 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_DSP_COMPONENT, snd_soc_dapm_dsp_component}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- 2.11.0

3 4

[alsa-devel] [PATCH 0/4] ALSA: Follow standard EXPORT_SYMBOL() declarations
by Takashi Iwai 16 Jun '17

16 Jun '17

Hi, this is a very very trivial cleanup, to make EXPORT_SYMBOL*() declarations to follow the standard style, i.e. right after the function definition. I didn't want this kind of changes because it makes the backporting difficult in general. But recently we've changed the core stuff enough, so it's good time to stir the code for this kind of cleanups, too. thanks, Takashi === Takashi Iwai (4): ALSA: pcm: Follow standard EXPORT_SYMBOL() declarations ALSA: seq: Follow standard EXPORT_SYMBOL() declarations ALSA: timer: Follow standard EXPORT_SYMBOL() declarations ALSA: core: Follow standard EXPORT_SYMBOL() declarations sound/core/info.c | 6 ------ sound/core/info_oss.c | 1 - sound/core/init.c | 8 +------- sound/core/isadma.c | 3 --- sound/core/memalloc.c | 13 ++++--------- sound/core/memory.c | 2 -- sound/core/misc.c | 1 - sound/core/oss/mixer_oss.c | 3 +-- sound/core/pcm_lib.c | 21 --------------------- sound/core/pcm_memory.c | 6 ------ sound/core/pcm_misc.c | 11 ----------- sound/core/pcm_native.c | 7 ------- sound/core/seq/seq_clientmgr.c | 8 -------- sound/core/seq/seq_lock.c | 1 - sound/core/seq/seq_memory.c | 2 -- sound/core/seq/seq_midi_emul.c | 9 ++++----- sound/core/seq/seq_midi_event.c | 21 ++++++++------------- sound/core/seq/seq_ports.c | 2 -- sound/core/seq/seq_virmidi.c | 3 +-- sound/core/sound.c | 2 -- sound/core/sound_oss.c | 3 --- sound/core/timer.c | 27 +++++++++++++-------------- 22 files changed, 32 insertions(+), 128 deletions(-) -- 2.13.1

1 4

[alsa-devel] [PATCH 0/2] Add support for TI TAS6424 digital amplifier
by Michael Stecklein 16 Jun '17

16 Jun '17

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

1 2

Re: [alsa-devel] Add ALC256 HP depop function
by Takashi Iwai 16 Jun '17

16 Jun '17

On Fri, 16 Jun 2017 11:14:01 +0200, Kailang wrote: > > Hi Takashi, > > This patch is support ALC256 for HP depop function. > It also could solve some ALC256 machine for plug headset cause power off issue. Thanks, applied now. Takashi

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Apply quirks to Broxton-T, too
by Takashi Iwai 16 Jun '17

16 Jun '17

Broxton-T was a forgotten child and we didn't apply the quirks for Skylake+ properly. Meanwhile, a quirk for reducing the DMA latency seems specific to the early Broxton model, so we leave as is. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e3c696c46a21..01eb1dc7b5b3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -370,11 +370,12 @@ enum { #define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71) #define IS_KBL_H(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa2f0) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_BXT_T(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x1a98) #define IS_GLK(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x3198) #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) -#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \ - IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci) || \ - IS_GLK(pci) || IS_CFL(pci) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci) || \ + IS_BXT_T(pci) || IS_KBL(pci) || IS_KBL_LP(pci) || \ + IS_KBL_H(pci) || IS_GLK(pci) || IS_CFL(pci)) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", -- 2.13.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Add Coffelake PCI ID
by Subhransu S. Prusty 16 Jun '17

16 Jun '17

From: Megha Dey <megha.dey(a)intel.com> Coffelake is another Intel part, so need to add PCI ID for it. Signed-off-by: Megha Dey <megha.dey(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/pci/hda/hda_intel.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1770f085c2a6..e3c696c46a21 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -371,9 +371,10 @@ enum { #define IS_KBL_H(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa2f0) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_GLK(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x3198) +#define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) #define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \ IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci) || \ - IS_GLK(pci) + IS_GLK(pci) || IS_CFL(pci) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2378,6 +2379,9 @@ static void azx_shutdown(struct pci_dev *pci) /* Kabylake-H */ { PCI_DEVICE(0x8086, 0xa2f0), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Coffelake */ + { PCI_DEVICE(0x8086, 0xa348), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: ak4613: add PCM rule
by Kuninori Morimoto 16 Jun '17

16 Jun '17

Hi Mark These patches adds hw_constraint rule to ak4613. Otherwise, Renesas platform might use un-supported sampling rate Kuninori Morimoto (2): ASoC: ak4613: add use CTRL2::CKS = 00 ASoC: ak4613: add hw_constraint rule for Sampling Rate sound/soc/codecs/ak4613.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 1 deletion(-) -- 1.9.1

2 4

[alsa-devel] [asoc:topic/rcar 36/37] sound/soc/sh/rcar/ssi.c:285:5-14: WARNING: Unsigned expression compared with zero: main_rate < 0 (fwd)
by Julia Lawall 16 Jun '17

16 Jun '17

Please check on line 285. julia ---------- Forwarded message ---------- Date: Fri, 16 Jun 2017 04:21:36 +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/rcar 36/37] sound/soc/sh/rcar/ssi.c:285:5-14: WARNING: Unsigned expression compared with zero: main_rate < 0 tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rcar head: 8cc03722403a5153054993388857e4e565dcdfb1 commit: ef4cf5d6a143e04e149ad81fc491fe10855544fe [36/37] ASoC: rsnd: add rsnd_ssi_clk_query() :::::: branch date: 3 hours ago :::::: commit date: 3 hours ago >> sound/soc/sh/rcar/ssi.c:285:5-14: WARNING: Unsigned expression compared with zero: main_rate < 0 git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout ef4cf5d6a143e04e149ad81fc491fe10855544fe vim +285 sound/soc/sh/rcar/ssi.c e7d850dd1 Kuninori Morimoto 2015-10-26 269 if (ssi_parent_mod && !rsnd_ssi_is_parent(mod, io)) e7d850dd1 Kuninori Morimoto 2015-10-26 270 return 0; e7d850dd1 Kuninori Morimoto 2015-10-26 271 b4c83b171 Kuninori Morimoto 2015-12-17 272 if (rsnd_ssi_is_multi_slave(mod, io)) b4c83b171 Kuninori Morimoto 2015-12-17 273 return 0; b4c83b171 Kuninori Morimoto 2015-12-17 274 e7d850dd1 Kuninori Morimoto 2015-10-26 275 if (ssi->usrcnt > 1) { e7d850dd1 Kuninori Morimoto 2015-10-26 276 if (ssi->rate != rate) { e7d850dd1 Kuninori Morimoto 2015-10-26 277 dev_err(dev, "SSI parent/child should use same rate\n"); e7d850dd1 Kuninori Morimoto 2015-10-26 278 return -EINVAL; e7d850dd1 Kuninori Morimoto 2015-10-26 279 } e7d850dd1 Kuninori Morimoto 2015-10-26 280 e7d850dd1 Kuninori Morimoto 2015-10-26 281 return 0; e7d850dd1 Kuninori Morimoto 2015-10-26 282 } e7d850dd1 Kuninori Morimoto 2015-10-26 283 ef4cf5d6a Kuninori Morimoto 2017-06-15 284 main_rate = rsnd_ssi_clk_query(priv, rate, chan, &idx); ef4cf5d6a Kuninori Morimoto 2017-06-15 @285 if (main_rate < 0) { ef4cf5d6a Kuninori Morimoto 2017-06-15 286 dev_err(dev, "unsupported clock rate\n"); ef4cf5d6a Kuninori Morimoto 2017-06-15 287 return -EIO; ef4cf5d6a Kuninori Morimoto 2017-06-15 288 } ae5c32230 Kuninori Morimoto 2013-07-21 289 b76e218ae Kuninori Morimoto 2015-09-10 290 ret = rsnd_adg_ssi_clk_try_start(mod, main_rate); ef4cf5d6a Kuninori Morimoto 2017-06-15 291 if (ret < 0) ef4cf5d6a Kuninori Morimoto 2017-06-15 292 return ret; ae5c32230 Kuninori Morimoto 2013-07-21 293 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] Applied "ASoC: rockchip: Fix build" to the asoc tree
by Mark Brown 15 Jun '17

15 Jun '17

The patch ASoC: rockchip: Fix build 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 b2e0a42e32d4eb9d5ee99264616dbba8fc288944 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Thu, 15 Jun 2017 20:13:33 +0100 Subject: [PATCH] ASoC: rockchip: Fix build Reported-by: Christophe Jaillet <christophe.jaillet(a)wanadoo.fr> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index ce09dee2202e..80db9b8f6605 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -641,7 +641,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } - soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai + soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai, sizeof(*soc_dai), GFP_KERNEL); if (!soc_dai) { err = -ENOMEM; -- 2.11.0

1 0

[alsa-devel] [PATCH 0/7] ASoC: simple-card: add asoc_simple_card_convert_fixup/parse_routing
by Kuninori Morimoto 15 Jun '17

15 Jun '17

Hi Mark Simple/OF-Audio card are using same DT properties, but not using same functions for it. These patches adds such function on simple-card-utils and uses it from simple/simple-scu/audio/audio-scu cards. These are for convert_rate/convert_channels/routing Kuninori Morimoto (7): ASoC: simple-card-utils: add asoc_simple_card_convert_fixup() ASoC: simple-scu-card: use asoc_simple_card_convert_fixup() ASoC: audio-graph-scu-card: use asoc_simple_card_convert_fixup() ASoC: simple-card-utils: add asoc_simple_card_of_parse_routing() ASoC: simple-card: use asoc_simple_card_of_parse_routing() ASoC: simple-scu-card: use asoc_simple_card_of_parse_routing() ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_routing() include/sound/simple_card_utils.h | 14 ++++++++ sound/soc/generic/audio-graph-scu-card.c | 28 +++------------ sound/soc/generic/simple-card-utils.c | 62 ++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 10 ++---- sound/soc/generic/simple-scu-card.c | 26 +++----------- 5 files changed, 88 insertions(+), 52 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

2 14

[alsa-devel] [PATCH 0/4] ASoC: rsnd: add PCM rule
by Kuninori Morimoto 15 Jun '17

15 Jun '17

Hi Mark These patches adds hw_constraints rule to Renesas sound driver. It supports TDM and Multi SSI features, possible Channel / Sampling Rate are depends on its situation. Kuninori Morimoto (4): ASoC: rsnd: rename "slots" related variable ASoC: rsnd: add rsnd_adg_clk_query() ASoC: rsnd: add rsnd_ssi_clk_query() ASoC: rsnd: add rsnd_soc_hw_rule/constraint() sound/soc/sh/rcar/adg.c | 53 ++++++------ sound/soc/sh/rcar/core.c | 208 ++++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/dvc.c | 7 +- sound/soc/sh/rcar/rsnd.h | 27 ++++-- sound/soc/sh/rcar/ssi.c | 98 +++++++++++++--------- 5 files changed, 289 insertions(+), 104 deletions(-) -- 1.9.1

2 8

[alsa-devel] Multiplexable codecs?
by Ricard Wanderlof 15 Jun '17

15 Jun '17

I have a usecase where there is a requirement to have two codecs which are multiplexed on the same I2S channel. (In fact one of the 'codecs' is actually a digital microphone, but it generates an I2S stream just the same). In the ALSA SoC framework I know there is support for having separate codecs on the right and left channels, and also for having different codecs for playback and capture. But is there any way in the framework that two codecs can be toggled between at runtime? Normally a machine driver would configure a single codec which essentially is statically allocated. Of course the whole driver could be removed and another one loaded, but the thing here is that it is only for capture that the codecs are multiplexed, for playback it's always the same codec, and I don't want to disrupt playback just because a change is required on the capture side. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [GIT PULL] KBL firmware update
by Vinod Koul 15 Jun '17

15 Jun '17

Hi, Please consider this pull request to get updated audio dsp fw for KBl platform. This version updates to 9.21.00.2042. The following changes since commit 37857004a430e96dc837db7f967b6d0279053de8: linux-firmware: add firmware image for Redpine 9113 chipset (2017-06-01 10:22:17 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git kbl for you to fetch changes up to cb8cdaaff39c1a453350bcd8c9fa4635661cc135: linux-firmware: intel: Update Kabylake audio firmware (2017-06-13 14:54:03 +0530) ---------------------------------------------------------------- Subhransu S. Prusty (1): linux-firmware: intel: Update Kabylake audio firmware WHENCE | 4 +++- intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v2042.bin | Bin 0 -> 238920 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_kbl_v2042.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] ASoC: rockchip: Fix an error handling in 'rockchip_i2s_probe'
by Christophe JAILLET 15 Jun '17

15 Jun '17

If this memory allocation fail, we must disable what has been enabled. Do not return immediately but go thrue the error handling path instead. Also use 'devm_kmemdup' instead of 'devm_kzalloc+memcpy' to simplify code. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/rockchip/rockchip_i2s.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af5e0da144f7..f3aa3dfe914b 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -643,12 +643,13 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } - soc_dai = devm_kzalloc(&pdev->dev, + soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai sizeof(*soc_dai), GFP_KERNEL); - if (!soc_dai) - return -ENOMEM; + if (!soc_dai) { + err = -ENOMEM; + goto err_pm_disable; + } - memcpy(soc_dai, &rockchip_i2s_dai, sizeof(*soc_dai)); if (!of_property_read_u32(node, "rockchip,playback-channels", &val)) { if (val >= 2 && val <= 8) soc_dai->playback.channels_max = val; -- 2.11.0

3 6

[alsa-devel] [PATCH v2] ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.
by Liam Girdwood 15 Jun '17

15 Jun '17

Add some DAPM widget types to better support the construction of DAPM graphs within DSPs Changes v1. o Added some documentation. o Split codec widget into encoder and decoder to avoid confusion with HW CODECS. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/soc/dapm.rst | 18 ++++++++++++++++++ include/sound/soc-dapm.h | 7 +++++++ include/uapi/sound/asoc.h | 10 +++++++++- sound/soc/soc-topology.c | 8 ++++++++ 4 files changed, 42 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index a27f42befa4d..78d63f6a35fa 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -105,6 +105,24 @@ Pre Special PRE widget (exec before all others) Post Special POST widget (exec after all others) +Buffer + Inter widget audio data buffer within a DSP. +Pipeline + Collection of widgets within a DSP, can represent a sequence of DSP + widgets and their collective configuration. +Effect + Widget that performs an audio processing effect. +SRC + Sample Rate Converter within DSP or CODEC +ASRC + Asynchronous Sample Rate Converter within DSP or CODEC +Encoder + Widget that encodes audio data from one format (usually PCM) to another + usually more compressed format. +Decoder + Widget that decodes audio data from a compressed format to an + uncompressed format like PCM. + (Widgets are defined in include/sound/soc-dapm.h) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..68fea6d97199 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,13 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_buffer, /* DSP/CODEC internal buffer */ + snd_soc_dapm_pipeline, /* DSP/CODEC internal pipeline */ + snd_soc_dapm_effect, /* DSP/CODEC effect component */ + snd_soc_dapm_src, /* DSP/CODEC SRC component */ + snd_soc_dapm_asrc, /* DSP/CODEC ASRC component */ + snd_soc_dapm_encoder, /* FW/SW audio encoder component */ + snd_soc_dapm_decoder, /* FW/SW audio decoder component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..8a4fdf3e7e2e 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,15 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_PIPELINE 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_ENCODER 22 +#define SND_SOC_TPLG_DAPM_DECODER 23 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DECODER /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 12e189701924..b82d2755c2ac 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,14 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_BUFFER, snd_soc_dapm_buffer}, + {SND_SOC_TPLG_DAPM_PIPELINE, snd_soc_dapm_pipeline}, + {SND_SOC_TPLG_DAPM_EFFECT, snd_soc_dapm_effect}, + {SND_SOC_TPLG_DAPM_SIGGEN, snd_soc_dapm_siggen}, + {SND_SOC_TPLG_DAPM_SRC, snd_soc_dapm_src}, + {SND_SOC_TPLG_DAPM_ASRC, snd_soc_dapm_asrc}, + {SND_SOC_TPLG_DAPM_ENCODER, snd_soc_dapm_encoder}, + {SND_SOC_TPLG_DAPM_DECODER, snd_soc_dapm_decoder}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- 2.11.0

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[alsa-devel] [PATCH v4 0/3] ASoC: Intel: Add Kabylake machine driver for RT5663+RT5514+MAX98927
by Harsha Priya 14 Jun '17

14 Jun '17

This patch set adds a machine driver for Kabylake platform which has RT5663, RT5514 and MAX9827 codecs. It also adds the quirk data for the platform driver to load the correct machine driver. For this particular machine, * the RT5663 is a headset codec connected on SSP0 * the RT5514 is a dmic codec connected on SSP1 * the MAX98987 is a speaker codec connected on SSP1 Harsha Priya (3): ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927 ASoC: Intel: Convert all sst_codecs data definition to c99 style ASoC: Intel: Add Kabylake RT5663+RT5514+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 640 +++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 37 +- 4 files changed, 690 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c -- 1.9.1

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[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Add Kabylake machine driver for RT5663+RT5514+MAX98927
by Harsha Priya 14 Jun '17

14 Jun '17

This patch set adds a machine driver for Kabylake platform which has RT5663, RT5514 and MAX9827 codecs. It also adds the quirk data for the platform driver to load the correct machine driver. For this particular machine, * the RT5663 is a headset codec connected on SSP0 * the RT5514 is a dmic codec connected on SSP1 * the MAX98987 is a speaker codec connected on SSP1 Harsha Priya (3): ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927 ASoC: Intel: Convert all sst_codecs data defintion to c99 style ASoC: Intel: Add Kabylake RT5663+RT5514+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 640 +++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 37 +- 4 files changed, 690 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c -- 1.9.1

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[alsa-devel] [PATCH] ALSA: pcm: Don't treat NULL chmap as a fatal error
by Takashi Iwai 14 Jun '17

14 Jun '17

The standard PCM chmap helper callbacks treat the NULL info->chmap as a fatal error and spews the kernel warning with stack trace when CONFIG_SND_DEBUG is on. This was OK, originally it was supposed to be always static and non-NULL. But, as the recent addition of Intel LPE audio driver shows, the chmap content may vary dynamically, and it can be even NULL when disconnected. The user still sees the kernel warning unnecessarily. For clearing such a confusion, this patch simply removes the snd_BUG_ON() in each place, just returns an error without warning. Cc: <stable(a)vger.kernel.org> # v4.11+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5088d4b8db22..009e6c98754e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2492,7 +2492,7 @@ static int pcm_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_substream *substream; const struct snd_pcm_chmap_elem *map; - if (snd_BUG_ON(!info->chmap)) + if (!info->chmap) return -EINVAL; substream = snd_pcm_chmap_substream(info, idx); if (!substream) @@ -2524,7 +2524,7 @@ static int pcm_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int __user *dst; int c, count = 0; - if (snd_BUG_ON(!info->chmap)) + if (!info->chmap) return -EINVAL; if (size < 8) return -ENOMEM; -- 2.13.1

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[alsa-devel] [PATCH] topology: Add support for missing fields parser.
by Liam Girdwood 14 Jun '17

14 Jun '17

The alsa-lib topology parser is missing some fields for certain objects that are part of the ABI. This patch adds the missing fields to the parser. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/pcm.c | 183 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 183 insertions(+) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index daef20e..0f4deb4 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -383,6 +383,49 @@ int tplg_parse_stream_caps(snd_tplg_t *tplg, tplg_dbg("\t\t%s: %d\n", id, sc->channels_max); continue; } + + if (strcmp(id, "periods_min") == 0) { + sc->periods_min = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->periods_min); + continue; + } + + if (strcmp(id, "periods_max") == 0) { + sc->periods_max = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->periods_max); + continue; + } + + if (strcmp(id, "period_size_min") == 0) { + sc->period_size_min = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->period_size_min); + continue; + } + + if (strcmp(id, "period_size_max") == 0) { + sc->period_size_max = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->period_size_max); + continue; + } + + if (strcmp(id, "buffer_size_min") == 0) { + sc->buffer_size_min = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->buffer_size_min); + continue; + } + + if (strcmp(id, "buffer_size_max") == 0) { + sc->buffer_size_max = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->buffer_size_max); + continue; + } + + if (strcmp(id, "sig_bits") == 0) { + sc->sig_bits = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->sig_bits); + continue; + } + } return 0; @@ -572,6 +615,17 @@ int tplg_parse_pcm(snd_tplg_t *tplg, continue; } + if (strcmp(id, "compress") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (strcmp(val, "true") == 0) + pcm->compress = 1; + + tplg_dbg("\t%s: %s\n", id, val); + continue; + } + if (strcmp(id, "dai") == 0) { err = tplg_parse_compound(tplg, n, tplg_parse_fe_dai, elem); @@ -655,6 +709,26 @@ int tplg_parse_dai(snd_tplg_t *tplg, continue; } + if (strcmp(id, "playback") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + dai->playback = atoi(val); + tplg_dbg("\t%s: %d\n", id, dai->playback); + continue; + } + + + if (strcmp(id, "capture") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + dai->capture = atoi(val); + tplg_dbg("\t%s: %d\n", id, dai->capture); + continue; + } + + /* stream capabilities */ if (strcmp(id, "pcm") == 0) { err = tplg_parse_compound(tplg, n, @@ -997,6 +1071,23 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, continue; } + if (strcmp(id, "bclk_freq") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->bclk_rate = atoi(val); + continue; + } + + if (strcmp(id, "bclk_invert") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "true")) + hw_cfg->invert_bclk = true; + continue; + } + if (strcmp(id, "fsync") == 0) { if (snd_config_get_string(n, &val) < 0) return -EINVAL; @@ -1005,6 +1096,98 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, hw_cfg->fsync_master = true; continue; } + + if (strcmp(id, "fsync_invert") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "true")) + hw_cfg->invert_fsync = true; + continue; + } + + if (strcmp(id, "fsync_freq") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->fsync_rate = atoi(val); + continue; + } + + if (strcmp(id, "mclk_freq") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->mclk_rate = atoi(val); + continue; + } + + if (strcmp(id, "mclk") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "master")) + hw_cfg->mclk_direction = true; + continue; + } + + if (strcmp(id, "pm_gate_clocks") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "true")) + hw_cfg->clock_gated = true; + continue; + } + + if (strcmp(id, "tdm_slots") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tdm_slots = atoi(val); + continue; + } + + if (strcmp(id, "tdm_slot_width") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tdm_slot_width = atoi(val); + continue; + } + + if (strcmp(id, "tx_slots") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tx_slots = atoi(val); + continue; + } + + if (strcmp(id, "rx_slots") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->rx_slots = atoi(val); + continue; + } + + if (strcmp(id, "tx_channels") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tx_channels = atoi(val); + continue; + } + + if (strcmp(id, "rx_channels") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->rx_channels = atoi(val); + continue; + } + } return 0; -- 1.9.1

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[alsa-devel] [PATCH] alsatplg: fix topology compiler long option parsing
by Liam Girdwood 14 Jun '17

14 Jun '17

verbose, compile and output options all have a parameter. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- topology/topology.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/topology/topology.c b/topology/topology.c index ffe2f05..097c255 100644 --- a/topology/topology.c +++ b/topology/topology.c @@ -54,10 +54,10 @@ int main(int argc, char *argv[]) snd_tplg_t *snd_tplg; static const char short_options[] = "hc:v:o:"; static const struct option long_options[] = { - {"help", 0, 0, 'h'}, - {"verbose", 0, 0, 'v'}, - {"compile", 0, 0, 'c'}, - {"output", 0, 0, 'o'}, + {"help", 0, NULL, 'h'}, + {"verbose", 1, NULL, 'v'}, + {"compile", 1, NULL, 'c'}, + {"output", 1, NULL, 'o'}, {0, 0, 0, 0}, }; char *source_file = NULL, *output_file = NULL; -- 1.9.1

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[alsa-devel] [PATCH 0/2] ALSA: pcm: remove unused local commands
by Takashi Sakamoto 14 Jun '17

14 Jun '17

Hi, A callback of 'struct snd_pcm_ops.ioctl' is internally used for ALSA pcm core to communicate to each drivers for some purposes; e.g. ask the size of FIFO in handled devices. However, in 'include/sound/pcm.h', there're some commands for the callback, which has never been used. This commit removes the unused commands to prevent developers from undocumented and unreferenced features. Takashi Sakamoto (2): ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO internal command ALSA: pcm: remove SNDRV_PCM_IOCTL1_GSTATE internal command include/sound/pcm.h | 4 ++-- sound/core/pcm_lib.c | 2 -- sound/core/pcm_native.c | 6 +----- 3 files changed, 3 insertions(+), 9 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: simple_card_utils: add EXPORT_SYMBOL_GPL() for asoc_simple_card_clk_xxx()
by Kuninori Morimoto 14 Jun '17

14 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 891caea41746 ("ASoC: simple_card_utils: add asoc_simple_card_clk_xxx()") added new asoc_simple_card_clk_xxx(), but, it didn't have EXPORT_SYMBOL_GPL(). This patch adds it. Otherwise, we will get below error ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-audio-graph-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-audio-graph-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-audio-graph-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-audio-graph-card.ko] undefined! Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b3fc8ce..26d64fa 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -160,11 +160,13 @@ int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) { return clk_prepare_enable(dai->clk); } +EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { clk_disable_unprepare(dai->clk); } +EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable); int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, -- 1.9.1

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