July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5663: Correct the mixer switch setting and remove redundant routing path
by Oder Chiou 19 Jul '17

19 Jul '17

The patch corrects the mixer siwtch setting that was the mistake in the previous commitment. And remove the redundant routing paths. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index d324817..38c4027 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -59,7 +59,7 @@ struct rt5663_priv { }; static const struct reg_sequence rt5663_patch_list[] = { - { 0x002a, 0x8080 }, + { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, }; @@ -482,7 +482,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0023, 0x0039 }, { 0x0026, 0xc0c0 }, { 0x0029, 0x8080 }, - { 0x002a, 0x8080 }, + { 0x002a, 0x8020 }, { 0x002c, 0x000c }, { 0x002d, 0x0000 }, { 0x0040, 0x0808 }, @@ -1967,8 +1967,8 @@ static const struct snd_kcontrol_new rt5663_sto1_dac_l_mix[] = { }; 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 */ @@ -2358,11 +2358,9 @@ static const struct snd_soc_dapm_route rt5663_dapm_routes[] = { { "DAC R1", NULL, "ADDA MIXR" }, { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, - { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, - { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt274: add rt274 codec driver
by Mark Brown 19 Jul '17

19 Jul '17

On Wed, Jul 19, 2017 at 02:08:58AM +0000, Bard Liao wrote: > I have sent a patch for it. But I was wondering why checkpatch.pl can't > find the style issue? No idea, sorry.

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[alsa-devel] Patch for Digidesign Mbox 2 (set sample rate)
by Viktor Radnai 19 Jul '17

19 Jul '17

Hi there, I have recently bought a Digidesign Mbox 2 Mini and found the lack of support for 44100Hz sample rate in the driver limiting (the sample rate was hard-set to 48000Hz by the original author even though the device supports it, because he couldn't get the frame rate switching to work reliably). I had a go at hacking the driver and came up with the following patch which appears to work on the MBox 2 Mini: https://pastebin.com/kzzTDj9T I have built and tested this with Linux 4.8.15 without any realtime patches. I am planning to pick up a "full size" Mbox 2 as well, in the meantime if anyone on the list owns one I would appreciate if you could test this and let me know the result. There is a possibility that it will not work without the "magic" init sequence that I have removed because it does not look like it's needed on the Mini. Although I would be a bit surprised because the two devices are apparently identical to the point that the Mini allows the use of it's physically nonexistent features. Once someone could confirm that this works, I would like to get it merged. This is my first attempt at submitting a patch for any Linux device driver so please let me know if I need to do anything else :) Cheers, Vik -- My other sig is hilarious

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[alsa-devel] Alsa library plugin license
by Jean-Francois Dockes 19 Jul '17

19 Jul '17

Hello, I was requested to study the possibility to implement an alsa library pcm plugin doing specific audio processing. I had a look around, but I could not find an explicit answer to this question: is a closed source alsa library plugin possible ? My guess would be that it has to be GPL'd, but I have to ask. Cheers, jf

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[alsa-devel] [PATCH] sound: fsl_dma: remove dma_object path member
by Rob Herring 19 Jul '17

19 Jul '17

dma_object.path is unused, so rather than fix it to work with DT full_name changes, just remove it. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org --- sound/soc/fsl/fsl_dma.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index ccadefceeff2..b38dd328c783 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -897,20 +896,19 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) ret = of_address_to_resource(ssi_np, 0, &res); if (ret) { - dev_err(&pdev->dev, "could not determine resources for %s\n", - ssi_np->full_name); + dev_err(&pdev->dev, "could not determine resources for %pOF\n", + ssi_np); of_node_put(ssi_np); return ret; } - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + dma = kzalloc(sizeof(*dma), GFP_KERNEL); if (!dma) { dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 2.11.0

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[alsa-devel] [PATCH] sound: Convert to using %pOF instead of full_name
by Rob Herring 18 Jul '17

18 Jul '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- sound/aoa/codecs/tas.c | 4 ++-- sound/soc/fsl/imx-audmux.c | 16 ++++++++-------- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 733b6365dad6..15c05755d270 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -905,8 +905,8 @@ static int tas_i2c_probe(struct i2c_client *client, goto fail; } printk(KERN_DEBUG - "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - (unsigned int)client->addr, node->full_name); + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); return 0; fail: mutex_destroy(&tas->mtx); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..392d5eef356d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -268,13 +268,13 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, ret = of_property_read_u32(child, "fsl,audmux-port", &port); if (ret) { - dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%s\"\n", - child->full_name); + dev_warn(&pdev->dev, "Failed to get fsl,audmux-port of child node \"%pOF\"\n", + child); continue; } if (!of_property_read_bool(child, "fsl,port-config")) { - dev_warn(&pdev->dev, "child node \"%s\" does not have property fsl,port-config\n", - child->full_name); + dev_warn(&pdev->dev, "child node \"%pOF\" does not have property fsl,port-config\n", + child); continue; } @@ -292,15 +292,15 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, } if (ret != -EOVERFLOW) { - dev_err(&pdev->dev, "Failed to read u32 at index %d of child %s\n", - i, child->full_name); + dev_err(&pdev->dev, "Failed to read u32 at index %d of child %pOF\n", + i, child); continue; } if (audmux_type == IMX31_AUDMUX) { if (i % 2) { - dev_err(&pdev->dev, "One pdcr value is missing in child node %s\n", - child->full_name); + dev_err(&pdev->dev, "One pdcr value is missing in child node %pOF\n", + child); continue; } imx_audmux_v2_configure_port(port, ptcr, pdcr); -- 2.11.0

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[alsa-devel] [PATCH] ASoC: sgtl5000: add AVC configuration controls
by Richard Leitner 18 Jul '17

18 Jul '17

For the configuration of the automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening, following controls are added: + AVC Threshold Volume: threshold where audio is compressed when the measured level is above or expanded when below + AVC Max Gain Volume: maximum gain which can be applied when the measured audio level is below threshold + AVC Hard Limiter Switch: when enabled the signal is limited to the programmed threshold. + AVC Integrator Response These controls use the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers for configuration. This patch depends on "ASoC: sgtl5000: add avc enable control" Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:144: CHECK: Alignment should match open parenthesis patch:147: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- sound/soc/codecs/sgtl5000.c | 85 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0477f2e..2bb3235 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,63 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = clamp((int)ucontrol->value.integer.value[0], 0, 96); + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +467,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -435,7 +512,15 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + /* Automatic Volume Control (DAP AVC) */ SOC_SINGLE("AVC Enable Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ -- 2.1.4

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[alsa-devel] [PATCH] ALSA: hda: constify pci_device_id.
by Arvind Yadav 18 Jul '17

18 Jul '17

pci_device_id are not supposed to change at runtime. All functions working with pci_device_id provided by <linux/pci.h> work with const pci_device_id. So mark the non-const structs as const. File size before: text data bss dec hex filename 2610 304 8 2922 b6a sound/hda/hdac_i915.o File size After adding 'const': text data bss dec hex filename 2674 240 8 2922 b6a sound/hda/hdac_i915.o Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/hda/hdac_i915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 0659bf3..038a180 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -336,7 +336,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); /* check whether intel graphics is present */ static bool i915_gfx_present(void) { - static struct pci_device_id ids[] = { + static const struct pci_device_id ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_BASE_CLASS_DISPLAY << 16, .class_mask = 0xff << 16 }, -- 2.7.4

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[alsa-devel] [PATCH v2 1/7] ALSA: als100: fix format string overflow warning
by Arnd Bergmann 18 Jul '17

18 Jul '17

The compiler sees that the format string might overflow for the longname: sound/isa/als100.c: In function 'snd_als100_pnp_detect': sound/isa/als100.c:225:27: error: ', dma ' directive writing 6 bytes into a region of size between 0 and 64 [-Werror=format-overflow=] sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/isa/als100.c:225:3: note: 'sprintf' output between 24 and 113 bytes into a destination of size 80 sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", Open-coding "shortname" here gets us below the limit, and using snprintf() is a good idea too. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/isa/als100.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/isa/als100.c b/sound/isa/als100.c index bc9ea306ee02..6a2c5b48f3ae 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -222,15 +222,16 @@ static int snd_card_als100_probe(int dev, if (pid->driver_data == SB_HW_DT019X) { strcpy(card->driver, "DT-019X"); strcpy(card->shortname, "Diamond Tech. DT-019X"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev]); + snprintf(card->longname, sizeof(card->longname), + "Diamond Tech. DT-019X, %s at 0x%lx, irq %d, dma %d", + chip->name, chip->port, irq[dev], dma8[dev]); } else { strcpy(card->driver, "ALS100"); strcpy(card->shortname, "Avance Logic ALS100"); - sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, chip->name, chip->port, - irq[dev], dma8[dev], dma16[dev]); + snprintf(card->longname, sizeof(card->longname), + "Avance Logic ALS100, %s at 0x%lx, irq %d, dma %d&%d", + chip->name, chip->port, irq[dev], dma8[dev], + dma16[dev]); } if ((error = snd_sb16dsp_pcm(chip, 0)) < 0) { -- 2.9.0

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[alsa-devel] [PATCH] ASoC: rt5663: Modify the default value for ASRC function
by Oder Chiou 18 Jul '17

18 Jul '17

The patch modifies the default value for ASRC function. It could prevent the pop noise and recording no sound with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ac6dd25..c46d47f 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -60,6 +60,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8080 }, + { 0x0086, 0x0028 }, }; static const struct reg_default rt5663_v2_reg[] = { @@ -509,7 +510,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0082, 0x0000 }, { 0x0083, 0x0000 }, { 0x0084, 0x0000 }, - { 0x0086, 0x0008 }, + { 0x0086, 0x0028 }, { 0x0087, 0x0000 }, { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, -- 2.7.4

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