February 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/6] ASoC: Fix of-node refcount unbalance
by Takashi Iwai 20 Feb '19

20 Feb '19

Hi, this is a patchset covering the OF-node refcount problem on a few ASoC drivers. thanks, Takashi === Takashi Iwai (6): ASoC: fsl: Fix of-node refcount unbalance in fsl_ssi_probe_from_dt() ASoC: simple-card: Fix missing of_node_put() at simple_dai_link_of() ASoC: simple-card: Fix of-node refcount unbalance in DAI-link parser ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of() ASoC: qcom: Fix of-node refcount unbalance in qcom_snd_parse_of() ASoC: samsung: Fix of-node refcount unbalance in odroid_audio_probe() sound/soc/fsl/fsl_ssi.c | 6 ++++-- sound/soc/generic/simple-card.c | 19 +++++++++++++------ sound/soc/qcom/apq8016_sbc.c | 21 ++++++++++++++++----- sound/soc/qcom/common.c | 10 ++++++++-- sound/soc/samsung/odroid.c | 5 +++-- 5 files changed, 44 insertions(+), 17 deletions(-) -- 2.16.4

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[alsa-devel] [PATCH] ALSA: hda/realtek - Reduce click noise on Dell Precision 5820 headphone
by Takashi Iwai 20 Feb '19

20 Feb '19

Dell Precision 5820 with ALC3234 codec (which is equivalent with ALC255) shows click noises at (runtime) PM resume on the headphone. The biggest source of the noise comes from the cleared headphone pin control at resume, which is done via the standard shutup procedure. Although we have an override of the standard shutup callback to replace with NOP, this would skip other needed stuff (e.g. the pull down of headset power). So, instead, this "fixes" the behavior of alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag. When this flag is set, Realtek codec won't call the standard snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this flag instead of overriding spec->shutup callback itself. This allows us to apply the similar fix for other entries easily if needed in future. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 57 ++++++++++++++++++++++++++----------------- 1 file changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1212955f7ab5..5855648b500c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -118,6 +118,7 @@ struct alc_spec { unsigned int has_alc5505_dsp:1; unsigned int no_depop_delay:1; unsigned int done_hp_init:1; + unsigned int no_shutup_pins:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -476,6 +477,14 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) set_eapd(codec, *p, on); } +static void alc_shutup_pins(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->no_shutup_pins) + snd_hda_shutup_pins(codec); +} + /* generic shutup callback; * just turning off EAPD and a little pause for avoiding pop-noise */ @@ -486,7 +495,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) alc_auto_setup_eapd(codec, false); if (!spec->no_depop_delay) msleep(200); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } /* generic EAPD initialization */ @@ -814,7 +823,7 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); else - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc_reboot_notify(struct hda_codec *codec) @@ -2922,7 +2931,7 @@ static void alc269_shutup(struct hda_codec *codec) (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static struct coef_fw alc282_coefs[] = { @@ -3025,14 +3034,15 @@ static void alc282_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(85); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); alc_write_coef_idx(codec, 0x78, coef78); } @@ -3138,15 +3148,16 @@ static void alc283_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(100); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); alc_write_coef_idx(codec, 0x43, 0x9614); } @@ -3212,14 +3223,15 @@ static void alc256_shutup(struct hda_codec *codec) /* NOTE: call this before clearing the pin, otherwise codec stalls */ alc_update_coef_idx(codec, 0x46, 0, 3 << 12); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc225_init(struct hda_codec *codec) @@ -3306,7 +3318,7 @@ static void alc225_shutup(struct hda_codec *codec) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc_default_init(struct hda_codec *codec) @@ -3360,14 +3372,15 @@ static void alc_default_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(85); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc294_hp_init(struct hda_codec *codec) @@ -3384,8 +3397,9 @@ static void alc294_hp_init(struct hda_codec *codec) msleep(100); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ @@ -4981,16 +4995,12 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } -static void alc_no_shutup(struct hda_codec *codec) -{ -} - static void alc_fixup_no_shutup(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_PRE_PROBE) { struct alc_spec *spec = codec->spec; - spec->shutup = alc_no_shutup; + spec->no_shutup_pins = 1; } } @@ -6639,6 +6649,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), + SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.16.4

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[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Allow compressed buffers in any memory region
by Charles Keepax 20 Feb '19

20 Feb '19

From: Andrew Ford <aford(a)opensource.cirrus.com> Currently, compressed buffers can only be specified in the XM memory region. There is no reason to have such a restriction with the newer meta-data based way of specifying the buffers, so remove it. Signed-off-by: Andrew Ford <aford(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1dd291cebe67..12ef85e85c29 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -344,6 +344,7 @@ struct wm_adsp_compr_buf { u32 irq_count; int read_index; int avail; + int host_buf_mem_type; }; struct wm_adsp_compr { @@ -3219,14 +3220,14 @@ static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, static inline int wm_adsp_buffer_read(struct wm_adsp_compr_buf *buf, unsigned int field_offset, u32 *data) { - return wm_adsp_read_data_word(buf->dsp, WMFW_ADSP2_XM, + return wm_adsp_read_data_word(buf->dsp, buf->host_buf_mem_type, buf->host_buf_ptr + field_offset, data); } static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, unsigned int field_offset, u32 data) { - return wm_adsp_write_data_word(buf->dsp, WMFW_ADSP2_XM, + return wm_adsp_write_data_word(buf->dsp, buf->host_buf_mem_type, buf->host_buf_ptr + field_offset, data); } @@ -3264,6 +3265,8 @@ static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) if (!buf->host_buf_ptr) return -EIO; + buf->host_buf_mem_type = WMFW_ADSP2_XM; + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); return 0; @@ -3282,6 +3285,7 @@ wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf) if (!ctl->enabled) continue; + buf->host_buf_mem_type = ctl->alg_region.type; return ctl; } -- 2.11.0

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[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Fix a typo in model documentation
by Takashi Iwai 20 Feb '19

20 Feb '19

Some garbage was taken via copy-and-paste error. Clean up. Fixes: a26d96c7802e ("ALSA: hda/realtek - Comprehensive model list for ALC259 & co") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/hd-audio/models.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 368a07a165f5..ef6101a78136 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -254,7 +254,7 @@ alc274-dell-aio ALC274 fixups on Dell AIO machines alc255-dummy-lineout Dell Precision 3930 fixups -alc255-dell-headset"}, +alc255-dell-headset Dell Precision 3630 fixups alc295-hp-x360 HP Spectre X360 fixups -- 2.16.4

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[alsa-devel] [PATCH 0/3] ALSA: Fix of-node refcount unbalance
by Takashi Iwai 19 Feb '19

19 Feb '19

Hi, here is a few patches to address the unbalanced OF-node refcount on powerpc and ac97 driver code. Another patchset for ASoC will follow later. Takashi === Takashi Iwai (3): ALSA: ac97: Fix of-node refcount unbalance ALSA: aoa: Fix of-node refcount unbalance ALSA: ppc: Fix of-node refcount unbalance sound/ac97/bus.c | 2 +- sound/aoa/core/gpio-feature.c | 1 + sound/ppc/tumbler.c | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) -- 2.16.4

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[alsa-devel] [PATCH] ASoC: fsl_spdif: fix sysclk_df type
by Viorel Suman 19 Feb '19

19 Feb '19

According to RM SPDIF STC SYSCLK_DF field is 9-bit wide, values being in 0..511 range. Use a proper type to handle sysclk_df. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a26686e..4842e6d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -96,7 +96,7 @@ struct fsl_spdif_priv { bool dpll_locked; u32 txrate[SPDIF_TXRATE_MAX]; u8 txclk_df[SPDIF_TXRATE_MAX]; - u8 sysclk_df[SPDIF_TXRATE_MAX]; + u16 sysclk_df[SPDIF_TXRATE_MAX]; u8 txclk_src[SPDIF_TXRATE_MAX]; u8 rxclk_src; struct clk *txclk[SPDIF_TXRATE_MAX]; @@ -376,7 +376,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, struct platform_device *pdev = spdif_priv->pdev; unsigned long csfs = 0; u32 stc, mask, rate; - u8 clk, txclk_df, sysclk_df; + u16 sysclk_df; + u8 clk, txclk_df; int ret; switch (sample_rate) { @@ -1109,8 +1110,9 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; - u32 sysclk_dfmin, sysclk_dfmax; - u32 txclk_df, sysclk_df, arate; + u32 arate; + u16 sysclk_dfmin, sysclk_dfmax, sysclk_df; + u8 txclk_df; /* The sysclk has an extra divisor [2, 512] */ sysclk_dfmin = is_sysclk ? 2 : 1; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsl_spdif: fix TXCLK_DF mask
by Viorel Suman 19 Feb '19

19 Feb '19

According to RM SPDIF TXCLK_DF mask is 7-bit wide. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_spdif.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 7666dab..e6c61e0 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -152,7 +152,7 @@ enum spdif_gainsel { #define STC_TXCLK_ALL_EN_MASK (1 << STC_TXCLK_ALL_EN_OFFSET) #define STC_TXCLK_ALL_EN (1 << STC_TXCLK_ALL_EN_OFFSET) #define STC_TXCLK_DF_OFFSET 0 -#define STC_TXCLK_DF_MASK (0x7ff << STC_TXCLK_DF_OFFSET) +#define STC_TXCLK_DF_MASK (0x7f << STC_TXCLK_DF_OFFSET) #define STC_TXCLK_DF(x) ((((x) - 1) << STC_TXCLK_DF_OFFSET) & STC_TXCLK_DF_MASK) #define STC_TXCLK_SRC_MAX 8 -- 2.7.4

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[alsa-devel] Applied "ASoC: codecs: pcm186x: Fix energysense SLEEP bit" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: codecs: pcm186x: Fix energysense SLEEP bit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 05bd7fcdd06b19a10f069af1bea3ad9abac038d7 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Date: Tue, 19 Feb 2019 16:29:28 +0000 Subject: [PATCH] ASoC: codecs: pcm186x: Fix energysense SLEEP bit The ADCs are sleeping when the SLEEP bit is set and running when it's cleared, so the bit should be inverted. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/pcm186x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index c36a391fec8a..c5fcc632f670 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -158,7 +158,7 @@ static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { * Put the codec into SLEEP mode when not in use, allowing the * Energysense mechanism to operate. */ - SND_SOC_DAPM_ADC("ADC", "HiFi Capture", PCM186X_POWER_CTRL, 1, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", PCM186X_POWER_CTRL, 1, 1), }; static const struct snd_soc_dapm_widget pcm1865_dapm_widgets[] = { @@ -184,8 +184,8 @@ static const struct snd_soc_dapm_widget pcm1865_dapm_widgets[] = { * Put the codec into SLEEP mode when not in use, allowing the * Energysense mechanism to operate. */ - SND_SOC_DAPM_ADC("ADC1", "HiFi Capture 1", PCM186X_POWER_CTRL, 1, 0), - SND_SOC_DAPM_ADC("ADC2", "HiFi Capture 2", PCM186X_POWER_CTRL, 1, 0), + SND_SOC_DAPM_ADC("ADC1", "HiFi Capture 1", PCM186X_POWER_CTRL, 1, 1), + SND_SOC_DAPM_ADC("ADC2", "HiFi Capture 2", PCM186X_POWER_CTRL, 1, 1), }; static const struct snd_soc_dapm_route pcm1863_dapm_routes[] = { -- 2.20.1

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[alsa-devel] [PATCH v2 1/2] ASoC: codecs: pcm186x: fix wrong usage of DECLARE_TLV_DB_SCALE()
by Codrin.Ciubotariu＠microchip.com 19 Feb '19

19 Feb '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v2: - fixed title - added correct macro; - revert the removal of a new line; sound/soc/codecs/pcm186x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 809b7e9f03ca..c36a391fec8a 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -42,7 +42,7 @@ struct pcm186x_priv { bool is_master_mode; }; -static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 4000, 50); +static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 50, 0); static const struct snd_kcontrol_new pcm1863_snd_controls[] = { SOC_DOUBLE_R_S_TLV("ADC Capture Volume", PCM186X_PGA_VAL_CH1_L, -- 2.17.1

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[alsa-devel] [PATCH 1/2] ASoC: codecs: pcm186x: fix wrong usage of DECLARE_TLV_DB_LINEAR()
by Codrin.Ciubotariu＠microchip.com 19 Feb '19

19 Feb '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/pcm186x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 809b7e9f03ca..7947279243f9 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -6,7 +6,6 @@ * Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ - #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -42,7 +41,7 @@ struct pcm186x_priv { bool is_master_mode; }; -static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 4000, 50); +static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 50, 0); static const struct snd_kcontrol_new pcm1863_snd_controls[] = { SOC_DOUBLE_R_S_TLV("ADC Capture Volume", PCM186X_PGA_VAL_CH1_L, -- 2.17.1

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