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[alsa-devel] [PATCH] ALSA: fireface: Off by one in latter_handle_midi_msg()
by Dan Carpenter 25 Jan '19

25 Jan '19

The > should be >= or otherwise we potentially read one element beyond the end of the ff->tx_midi_substreams[] array. Fixes: 73f5537fb209 ("ALSA: fireface: support tx MIDI functionality of Fireface UCX") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/fireface/ff-protocol-latter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c index 817af4447349..438aeed9a1ab 100644 --- a/sound/firewire/fireface/ff-protocol-latter.c +++ b/sound/firewire/fireface/ff-protocol-latter.c @@ -275,7 +275,7 @@ static void latter_handle_midi_msg(struct snd_ff *ff, unsigned int offset, struct snd_rawmidi_substream *substream; unsigned int len; - if (index > ff->spec->midi_in_ports) + if (index >= ff->spec->midi_in_ports) return; switch (data & 0x0000000f) { -- 2.17.1

3 2

[alsa-devel] [PATCH v2] ASoC: intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabled
by Hans de Goede 25 Jan '19

25 Jan '19

Users have been seeing sound stability issues with max98090 codecs since: commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") At first that commit broke sound for Chromebook Swanky and Clapper models, the problem was that the machine-driver has been controlling the wrong clock on those models since support for them was added. This was hidden by clk-pmc-atom.c keeping the actual clk on unconditionally. With the machine-driver controlling the proper clock, sound works again but we are seeing bug reports describing it as: low volume, "sounds like played at 10x speed" and instable. When these issues are hit the following message is seen in dmesg: "max98090 i2c-193C9890:00: PLL unlocked". Attempts have been made to fix this by inserting a delay between enabling the clk and enabling and checking the pll, but this has not helped. It seems that at least on boards which use pmc_plt_clk_0 as clock, if we ever disable the clk, the pll looses its lock and after that we get various issues. This commit fixes this by enabling the clock once at probe time on these boards. In essence this restores the old behavior of clk-pmc-atom.c always keeping the clk on on these boards. Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-by: Mogens Jensen <mogens-jensen(a)protonmail.com> Reported-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in v2: -Only enable the codec clock once and keep it enabled on boards which use pmc_plt_clk_0 as clock and leave the behavior unchanged on other boards --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 47 +++++++++++++++++--- 1 file changed, 41 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 08a5152e635a..e7620017e725 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -42,6 +42,7 @@ struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; bool ts3a227e_present; + int quirks; }; static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -53,6 +54,10 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); int ret; + /* See the comment in snd_cht_mc_probe() */ + if (ctx->quirks & QUIRK_PMC_PLT_CLK_0) + return 0; + codec_dai = snd_soc_card_get_codec_dai(card, CHT_CODEC_DAI); if (!codec_dai) { dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); @@ -222,6 +227,10 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) "jack detection gpios not added, error %d\n", ret); } + /* See the comment in snd_cht_mc_probe() */ + if (ctx->quirks & QUIRK_PMC_PLT_CLK_0) + return 0; + /* * The firmware might enable the clock at * boot (this information may or may not @@ -420,16 +429,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int ret_val = 0; struct cht_mc_private *drv; const char *mclk_name; - int quirks = 0; - - dmi_id = dmi_first_match(cht_max98090_quirk_table); - if (dmi_id) - quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + drv->quirks = (unsigned long)dmi_id->driver_data; + drv->ts3a227e_present = acpi_dev_found("104C227E"); if (!drv->ts3a227e_present) { /* no need probe TI jack detection chip */ @@ -446,7 +454,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - if (quirks & QUIRK_PMC_PLT_CLK_0) + if (drv->quirks & QUIRK_PMC_PLT_CLK_0) mclk_name = "pmc_plt_clk_0"; else mclk_name = "pmc_plt_clk_3"; @@ -459,6 +467,21 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return PTR_ERR(drv->mclk); } + /* + * Boards which have the MAX98090's clk connected to clk_0 do not seem + * to like it if we muck with the clock. If we disable the clock when + * it is unused we get "max98090 i2c-193C9890:00: PLL unlocked" errors + * and the PLL never seems to lock again. + * So for these boards we enable it here once and leave it at that. + */ + if (drv->quirks & QUIRK_PMC_PLT_CLK_0) { + ret_val = clk_prepare_enable(drv->mclk); + if (ret_val < 0) { + dev_err(&pdev->dev, "MCLK enable error: %d\n", ret_val); + return ret_val; + } + } + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); if (ret_val) { dev_err(&pdev->dev, @@ -469,11 +492,23 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return ret_val; } +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->quirks & QUIRK_PMC_PLT_CLK_0) + clk_disable_unprepare(ctx->mclk); + + return 0; +} + static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: soc-core: Fix null pointer dereference in soc_find_component
by Rohit kumar 25 Jan '19

25 Jan '19

From: Ajit Pandey <ajitp(a)codeaurora.org> soc_find_component() may lead to null pointer exception if both arguments i.e of_node and name is NULL. Add NULL check before calling soc_find_component(). Also fix some typos. Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-core.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0934b36..df05fb8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1131,11 +1131,13 @@ static int soc_init_dai_link(struct snd_soc_card *card, } /* - * Defer card registartion if platform dai component is not added to + * Defer card registration if platform dai component is not added to * component list. */ - if (!soc_find_component(link->platform->of_node, link->platform->name)) - return -EPROBE_DEFER; + if (link->platform->of_node || link->platform->name) + if (!soc_find_component(link->platform->of_node, + link->platform->name)) + return -EPROBE_DEFER; /* * CPU device may be specified by either name or OF node, but @@ -1150,11 +1152,12 @@ static int soc_init_dai_link(struct snd_soc_card *card, } /* - * Defer card registartion if cpu dai component is not added to + * Defer card registration if cpu dai component is not added to * component list. */ - if (!soc_find_component(link->cpu_of_node, link->cpu_name)) - return -EPROBE_DEFER; + if (link->cpu_of_node || link->cpu_name) + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; /* * At least one of CPU DAI name or CPU device name/node must be -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

6 25

[alsa-devel] [PATCH][V2] ASoC: Intel: make const arrays static, reduces object code size
by Colin King 24 Jan '19

24 Jan '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const arrays on the stack but instead make it static. Makes the object code smaller, for example: Before: text data bss dec hex filename 14107 8832 224 23163 5a7b bytcht_es8316.o After: text data bss dec hex filename 14015 8896 224 23135 5a5f bytcht_es8316.o (gcc version 8.2.0 x86_64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: also fix up bytcr_rt5640.c and bytcr_rt5651.c, thanks to Pierre-Louis Bossart for noticing these two. --- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index fa9c4cf97686..1364e4e601d8 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -437,7 +437,7 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { - const char * const mic_name[] = { "in1", "in2" }; + static const char * const mic_name[] = { "in1", "in2" }; struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ca8b4d5ff70f..a79466c8fb29 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1149,7 +1149,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) { - const char * const map_name[] = { "dmic1", "dmic2", "in1", "in3" }; + static const char * const map_name[] = { "dmic1", "dmic2", "in1", "in3" }; const struct dmi_system_id *dmi_id; struct byt_rt5640_private *priv; struct snd_soc_acpi_mach *mach; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b618d984e2d5..e6945d11c8ab 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -919,7 +919,7 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { - const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; + static const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; struct byt_rt5651_private *priv; struct snd_soc_acpi_mach *mach; struct device *codec_dev; -- 2.19.1

3 2

[alsa-devel] [PATCH] ALSA: hda: runtime PM is always active
by Sameer Pujar 24 Jan '19

24 Jan '19

The runtime PM count is incremented and set to active during hda codec device init, but it is decremented and set to suspend during exit only. Hence the runtime PM status is always active and hda device cannot be put to runtime suspend. Keeping device usage active for entire period, though nothing really happening on the device, seems unnecessary. This patch avoides incrementing runtime PM usage count of the device. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> Reviewed-by: Mohan Kumar D <mkumard(a)nvidia.com> --- sound/hda/hdac_device.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 95b073e..72aa342 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -31,9 +31,6 @@ static void default_release(struct device *dev) * * Returns zero for success or a negative error code. * - * This function increments the runtime PM counter and marks it active. - * The caller needs to turn it off appropriately later. - * * The caller needs to set the device's release op properly by itself. */ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, @@ -55,8 +52,6 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->bus = bus; codec->addr = addr; codec->type = HDA_DEV_CORE; - pm_runtime_set_active(&codec->dev); - pm_runtime_get_noresume(&codec->dev); atomic_set(&codec->in_pm, 0); err = snd_hdac_bus_add_device(bus, codec); @@ -123,7 +118,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); */ void snd_hdac_device_exit(struct hdac_device *codec) { - pm_runtime_put_noidle(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); -- 2.7.4

2 5

[alsa-devel] [RFC PATCH] ASoC: pcm512x: Implement the set_bclk_ratio interface
by Dimitris Papavasiliou 24 Jan '19

24 Jan '19

>From 5e6ad135fd063d4b22cd962de43499a161bbc7db Mon Sep 17 00:00:00 2001 From: Dimitris Papavasiliou <dpapavas(a)gmail.com> Date: Fri, 11 Jan 2019 22:13:27 +0200 Subject: [RFC PATCH] ASoC: pcm512x: Implement the set_bclk_ratio interface Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external oscillators, to generate 44.1 or 48kHz multiples and are forced to resort to hacks[1] in order to support 24-bit data without ending up with fractional dividers. This patch allows the machine driver to use 32-bit frames for 24-bit data to avoid such issues. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.h… Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: The patch also includes a related change in the calculation of the bclk divider. The current calculation is not very clear to me, as I can't find much concrete information on what this sample rate fraction is, how it's calculated and how, therefore, its denominator would fit into the calculation. It also yields slightly wrong dividers in certain cases, which the machine driver for my HiFiBerry DAC+ Pro board seemingly tries to circumvent, by updating the rate fraction so as to suit this calculation. The updated calculation seems to work fine in my tests and, as far as I can see, should correctly yield the smallest bit clock rate that would fit the frame. Please comment if anything looks off. sound/soc/codecs/pcm512x.c | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4cc24a5d5c31..8cd728f9a1eb 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -55,6 +55,7 @@ struct pcm512x_priv { unsigned long overclock_dsp; int mute; struct mutex mutex; + unsigned int bclk_ratio; }; /* @@ -915,16 +916,21 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, int fssp; int gpio; - lrclk_div = snd_soc_params_to_frame_size(params); - if (lrclk_div == 0) { - dev_err(dev, "No LRCLK?\n"); - return -EINVAL; + if (pcm512x->bclk_ratio > 0) { + lrclk_div = pcm512x->bclk_ratio; + } else { + lrclk_div = snd_soc_params_to_frame_size(params); + + if (lrclk_div == 0) { + dev_err(dev, "No LRCLK?\n"); + return -EINVAL; + } } if (!pcm512x->pll_out) { sck_rate = clk_get_rate(pcm512x->sclk); - bclk_div = params->rate_den * 64 / lrclk_div; - bclk_rate = DIV_ROUND_CLOSEST(sck_rate, bclk_div); + bclk_rate = params_rate(params) * lrclk_div; + bclk_div = DIV_ROUND_CLOSEST(sck_rate, bclk_rate); mck_rate = sck_rate; } else { @@ -1383,6 +1389,16 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + pcm512x->bclk_ratio = ratio; + + return 0; +} + static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; @@ -1435,6 +1451,7 @@ static const struct snd_soc_dai_ops pcm512x_dai_ops = { .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, .digital_mute = pcm512x_digital_mute, + .set_bclk_ratio = pcm512x_set_bclk_ratio, }; static struct snd_soc_dai_driver pcm512x_dai = { -- 2.11.0

4 10

[alsa-devel] [PATCH][next] ASoC: Intel: bytcht_es8316: make mic_name static, reduces object code size
by Colin King 24 Jan '19

24 Jan '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const array mic_name on the stack but instead make it static. Makes the object code smaller by 28 bytes: Before: text data bss dec hex filename 14107 8832 224 23163 5a7b bytcht_es8316.o After: text data bss dec hex filename 14015 8896 224 23135 5a5f bytcht_es8316.o (gcc version 8.2.0 x86_64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bytcht_es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index fa9c4cf97686..1364e4e601d8 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -437,7 +437,7 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { - const char * const mic_name[] = { "in1", "in2" }; + static const char * const mic_name[] = { "in1", "in2" }; struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabled
by Hans de Goede 24 Jan '19

24 Jan '19

Users have been seeing sound stability issues with max98090 codecs since: commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") At first that commit broke sound for Chromebook Swanky and Clapper models, the problem was that the machine-driver has been controlling the wrong clock on those models since support for them was added. This was hidden by clk-pmc-atom.c keeping the actual clk on unconditionally. With the machine-driver controlling the proper clock, sound works again but we are seeing bug reports describing it as: low volume, "sounds like played at 10x speed" and instable. When these issues are hit the following message is seen in dmesg: "max98090 i2c-193C9890:00: PLL unlocked". Attempts have been made to fix this by inserting a delay between enabling the clk and enabling and checking the pll, but this has not helped. It seems that if we ever disable the clk, the pll looses its lock and then we get various issues. This commit fixes this by enabling the clock once at probe time, in essence restoring the old behavior of clk-pmc-atom.c always keeping the clk on, but then only on machines with a max98090 codec. Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Mogens Jensen <mogens-jensen(a)protonmail.com> Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 75 ++++++-------------- 1 file changed, 20 insertions(+), 55 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 08a5152e635a..549d6ce12ec4 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -44,43 +44,11 @@ struct cht_mc_private { bool ts3a227e_present; }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, CHT_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = clk_prepare_enable(ctx->mclk); - if (ret < 0) { - dev_err(card->dev, - "could not configure MCLK state"); - return ret; - } - } else { - clk_disable_unprepare(ctx->mclk); - } - - return 0; -} - static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cht_audio_map[] = { @@ -97,10 +65,6 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"codec_in0", NULL, "ssp2 Rx" }, {"codec_in1", NULL, "ssp2 Rx" }, {"ssp2 Rx", NULL, "HiFi Capture"}, - {"Headphone", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "Platform Clock"}, - {"Int Mic", NULL, "Platform Clock"}, - {"Ext Spk", NULL, "Platform Clock"}, }; static const struct snd_kcontrol_new cht_mc_controls[] = { @@ -222,25 +186,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) "jack detection gpios not added, error %d\n", ret); } - /* - * The firmware might enable the clock at - * boot (this information may or may not - * be reflected in the enable clock register). - * To change the rate we must disable the clock - * first to cover these cases. Due to common - * clock framework restrictions that do not allow - * to disable a clock that has not been enabled, - * we need to enable the clock first. - */ - ret = clk_prepare_enable(ctx->mclk); - if (!ret) - clk_disable_unprepare(ctx->mclk); - - ret = clk_set_rate(ctx->mclk, CHT_PLAT_CLK_3_HZ); - - if (ret) - dev_err(runtime->dev, "unable to set MCLK rate\n"); - return ret; } @@ -459,6 +404,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return PTR_ERR(drv->mclk); } + /* + * The MAX98090 does not seem to like it if we muck with its clock, + * so we enable it here once and leave it at that. + */ + ret_val = clk_prepare_enable(drv->mclk); + if (ret_val < 0) { + dev_err(&pdev->dev, "Failed to enable MCLK: %d\n", ret_val); + return ret_val; + } + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); if (ret_val) { dev_err(&pdev->dev, @@ -469,11 +424,21 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return ret_val; } +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + + clk_disable_unprepare(ctx->mclk); + return 0; +} + static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) -- 2.20.1

2 3

[alsa-devel] [PATCH] ALSA: fireface: comment cleanup about destination address of async transactions for MIDI messages
by Takashi Sakamoto 24 Jan '19

24 Jan '19

In Fireface series, registration of higher 4 bytes of destination address for asynchronous transaction of MIDI messages is done by a write transaction to model-specific register. On the other hand, registration of lower 4 bytes of the address is selectable from 4 options. A register for this registration includes the other purpose options such as input attenuation. Thus this driver expects userspace applications to configure the register. Actual behaviour for the asynchronous transaction is different depending on protocols. In former protocol, destination offset of each transaction is the same as the registered address even if it is block request. In latter models, destination offset of each transaction is the offset of previous transaction plus 4 byte and the transaction is quadlet request. This commit cleanups comments about the above mechanism. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-former.c | 23 +++++++++++++ sound/firewire/fireface/ff-protocol-latter.c | 18 ++++++++++ sound/firewire/fireface/ff-transaction.c | 36 ++++---------------- 3 files changed, 48 insertions(+), 29 deletions(-) diff --git a/sound/firewire/fireface/ff-protocol-former.c b/sound/firewire/fireface/ff-protocol-former.c index e0acf40a02ee..8d1c2c6e907b 100644 --- a/sound/firewire/fireface/ff-protocol-former.c +++ b/sound/firewire/fireface/ff-protocol-former.c @@ -396,6 +396,10 @@ static void ff800_finish_session(struct snd_ff *ff) FF800_ISOC_COMM_STOP, &reg, sizeof(reg), 0); } +// Fireface 800 doesn't allow drivers to register lower 4 bytes of destination +// address. +// A write transaction to clear registered higher 4 bytes of destination address +// has an effect to suppress asynchronous transaction from device. static void ff800_handle_midi_msg(struct snd_ff *ff, unsigned int offset, __le32 *buf, size_t length) { @@ -525,6 +529,25 @@ static void ff400_finish_session(struct snd_ff *ff) FF400_ISOC_COMM_STOP, &reg, sizeof(reg), 0); } +// For Fireface 400, lower 4 bytes of destination address is configured by bit +// flag in quadlet register (little endian) at 0x'0000'801'0051c. Drivers can +// select one of 4 options: +// +// bit flags: offset of destination address +// - 0x04000000: 0x'....'....'0000'0000 +// - 0x08000000: 0x'....'....'0000'0080 +// - 0x10000000: 0x'....'....'0000'0100 +// - 0x20000000: 0x'....'....'0000'0180 +// +// Drivers can suppress the device to transfer asynchronous transactions by +// using below 2 bits. +// - 0x01000000: suppress transmission +// - 0x02000000: suppress transmission +// +// Actually, the register is write-only and includes the other options such as +// input attenuation. This driver allocates destination address with '0000'0000 +// in its lower offset and expects userspace application to configure the +// register for it. static void ff400_handle_midi_msg(struct snd_ff *ff, unsigned int offset, __le32 *buf, size_t length) { diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c index 817af4447349..0fbc1950327f 100644 --- a/sound/firewire/fireface/ff-protocol-latter.c +++ b/sound/firewire/fireface/ff-protocol-latter.c @@ -266,6 +266,24 @@ static void latter_dump_status(struct snd_ff *ff, struct snd_info_buffer *buffer // NOTE: transactions are transferred within 0x00-0x7f in allocated range of // address. This seems to be for check of discontinuity in receiver side. +// +// Like Fireface 400, drivers can select one of 4 options for lower 4 bytes of +// destination address by bit flags in quadlet register (little endian) at +// 0x'ffff'0000'0014: +// +// bit flags: offset of destination address +// - 0x00002000: 0x'....'....'0000'0000 +// - 0x00004000: 0x'....'....'0000'0080 +// - 0x00008000: 0x'....'....'0000'0100 +// - 0x00010000: 0x'....'....'0000'0180 +// +// Drivers can suppress the device to transfer asynchronous transactions by +// clear these bit flags. +// +// Actually, the register is write-only and includes the other settings such as +// input attenuation. This driver allocates for the first option +// (0x'....'....'0000'0000) and expects userspace application to configure the +// register for it. static void latter_handle_midi_msg(struct snd_ff *ff, unsigned int offset, __le32 *buf, size_t length) { diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index d8a8b01b39a1..0d6ad19363b8 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -165,35 +165,13 @@ static int allocate_own_address(struct snd_ff *ff, int i) return err; } -/* - * Controllers are allowed to register higher 4 bytes of address to receive - * the transactions. Different models have different registers for this purpose; - * e.g. 0x'0000'8010'03f4 for Fireface 400. - * The controllers are not allowed to register lower 4 bytes of the address. - * They are forced to select one of 4 options for the part of address by writing - * corresponding bits to 0x'0000'8010'051f. - * - * The 3rd-6th bits of this register are flags to indicate lower 4 bytes of - * address to which the device transferrs the transactions. In short: - * - 0x20: 0x'....'....'0000'0180 - * - 0x10: 0x'....'....'0000'0100 - * - 0x08: 0x'....'....'0000'0080 - * - 0x04: 0x'....'....'0000'0000 - * - * This driver configure 0x'....'....'0000'0000 to receive MIDI messages from - * units. The 3rd bit of the register should be configured, however this driver - * deligates this task to userspace applications due to a restriction that this - * register is write-only and the other bits have own effects. - * - * Unlike Fireface 800, Fireface 400 cancels transferring asynchronous - * transactions when the 1st and 2nd of the register stand. These two bits have - * the same effect. - * - 0x02, 0x01: cancel transferring - * - * On the other hand, the bits have no effect on Fireface 800. This model - * cancels asynchronous transactions when the higher 4 bytes of address is - * overwritten with zero. - */ +// Controllers are allowed to register higher 4 bytes of destination address to +// receive asynchronous transactions for MIDI messages, while the way to +// register lower 4 bytes of address is different depending on protocols. For +// details, please refer to comments in protocol implementations. +// +// This driver expects userspace applications to configure registers for the +// lower address because in most cases such registers has the other settings. int snd_ff_transaction_reregister(struct snd_ff *ff) { struct fw_card *fw_card = fw_parent_device(ff->unit)->card; -- 2.19.1

2 1

[alsa-devel] [PATCH v9 0/4] Media Device Allocator API
by shuah＠kernel.org 24 Jan '19

24 Jan '19

From: Shuah Khan <shuah(a)kernel.org> Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - This patch series is tested on 4.20-rc6 - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 321 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 705 insertions(+), 22 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

4 16

[alsa-devel] Possible ALSA loopback bug?
by Timothy Pearson 24 Jan '19

24 Jan '19

I've been running a dedicated machine with ALSA loopback, feeding darkice on the capture side and (currently) using mplayer to provide audio on the playback side. I've seen sporadic hangs for a couple of years on this setup, but recently with the switch to mplayer I have additional information that points more to ALSA than the other applications. When the audio hangs, mplayer prints a continuous stream of: "Audio device got stuck!" The only way to "unstick" the audio stream is to terminate and restart mplayer. Restarting darkice does not fix the stuck stream. There is no pulseaudio on this sytem. What would be the next steps to debug this setup? It's shown up on two entirely different installs at this point (Ubuntu 14.04 and Debian 9). Thanks! -- Timothy Pearson Raptor Engineering +1 (415) 727-8645 (direct line) +1 (512) 690-0200 (switchboard) https://www.raptorengineering.com

2 5

[alsa-devel] [PATCH v3 0/3] ASoC: Intel: Update broxton machine driver
by Yong Zhi 23 Jan '19

23 Jan '19

Hello, This patchset is an update to the following v2 posted by Naveen last year: https://patchwork.kernel.org/patch/10335295/ https://patchwork.kernel.org/patch/10335293/ https://patchwork.kernel.org/patch/10335297/ 1/5 and 5/5 of original series were merged. The changes have been tested for both bxt_da7219_max98357a and glk_da7219_max98357a. Yong Zhi (3): ASoC: Intel: Headset button support in broxton machine driver ASoC: Intel: Add Geminilake Dialog Maxim machine driver ASoC: Intel: glk: Add DAI links for Multi-Playback sound/soc/intel/boards/bxt_da7219_max98357a.c | 101 +++++++++++++++++++++++--- 1 file changed, 89 insertions(+), 12 deletions(-) -- 2.7.4

1 3

[alsa-devel] [PATCH] ALSA: proc: Avoid possible leaks of snd_info_entry objects
by Takashi Iwai 23 Jan '19

23 Jan '19

This patch changes the parent pointer assignment of snd_info_entry object to be always non-NULL. More specifically,check the parent argument in snd_info_create_module_entry() & co, and assign snd_proc_root if NULL is passed there. This assures that the proc object is always freed when the root is freed, so avoid possible memory leaks. For example, some error paths (e.g. snd_info_register() error at snd_minor_info_init()) may leave snd_info_entry object although the proc file itself is freed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index fe502bc5e6d2..2dfb6389c084 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -741,7 +741,11 @@ struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry *parent) { - struct snd_info_entry *entry = snd_info_create_entry(name, parent); + struct snd_info_entry *entry; + + if (!parent) + parent = snd_proc_root; + entry = snd_info_create_entry(name, parent); if (entry) entry->module = module; return entry; @@ -762,7 +766,11 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry * parent) { - struct snd_info_entry *entry = snd_info_create_entry(name, parent); + struct snd_info_entry *entry; + + if (!parent) + parent = card->proc_root; + entry = snd_info_create_entry(name, parent); if (entry) { entry->module = card->module; entry->card = card; -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: pcm: Simplify proc file destruction
by Takashi Iwai 23 Jan '19

23 Jan '19

The proc files are recursively freed by calling with the root snd_info_entry object, so we don't have to keep each object for releasing one by one. Move the release of the PCM stream proc root at the beginning, so that we can remove the redundant code and resource. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 11 --------- sound/core/pcm.c | 66 +++++++++---------------------------------------- sound/core/pcm_memory.c | 16 ++---------- 3 files changed, 13 insertions(+), 80 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a20d3a48df00..eae6d2b82d7a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -481,15 +481,6 @@ struct snd_pcm_substream { #endif #ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; - struct snd_info_entry *proc_info_entry; - struct snd_info_entry *proc_hw_params_entry; - struct snd_info_entry *proc_sw_params_entry; - struct snd_info_entry *proc_status_entry; - struct snd_info_entry *proc_prealloc_entry; - struct snd_info_entry *proc_prealloc_max_entry; -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_info_entry *proc_xrun_injection_entry; -#endif #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* misc flags */ unsigned int hw_opened: 1; @@ -511,10 +502,8 @@ struct snd_pcm_str { #endif #ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; - struct snd_info_entry *proc_info_entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG unsigned int xrun_debug; /* 0 = disabled, 1 = verbose, 2 = stacktrace */ - struct snd_info_entry *proc_xrun_debug_entry; #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index ca1ea3cf9350..bca0bdf3e33c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -536,12 +536,9 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) entry = snd_info_create_card_entry(pcm->card, "info", pstr->proc_root); if (entry) { snd_info_set_text_ops(entry, pstr, snd_pcm_stream_proc_info_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - pstr->proc_info_entry = entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG entry = snd_info_create_card_entry(pcm->card, "xrun_debug", @@ -551,24 +548,15 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) entry->c.text.write = snd_pcm_xrun_debug_write; entry->mode |= 0200; entry->private_data = pstr; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - pstr->proc_xrun_debug_entry = entry; #endif return 0; } static int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - snd_info_free_entry(pstr->proc_xrun_debug_entry); - pstr->proc_xrun_debug_entry = NULL; -#endif - snd_info_free_entry(pstr->proc_info_entry); - pstr->proc_info_entry = NULL; snd_info_free_entry(pstr->proc_root); pstr->proc_root = NULL; return 0; @@ -597,45 +585,33 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_info_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_info_entry = entry; entry = snd_info_create_card_entry(card, "hw_params", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_hw_params_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_hw_params_entry = entry; entry = snd_info_create_card_entry(card, "sw_params", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_sw_params_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_sw_params_entry = entry; entry = snd_info_create_card_entry(card, "status", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_status_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_status_entry = entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG entry = snd_info_create_card_entry(card, "xrun_injection", @@ -645,40 +621,18 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) entry->c.text.read = NULL; entry->c.text.write = snd_pcm_xrun_injection_write; entry->mode = S_IFREG | 0200; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_xrun_injection_entry = entry; #endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } -static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) -{ - snd_info_free_entry(substream->proc_info_entry); - substream->proc_info_entry = NULL; - snd_info_free_entry(substream->proc_hw_params_entry); - substream->proc_hw_params_entry = NULL; - snd_info_free_entry(substream->proc_sw_params_entry); - substream->proc_sw_params_entry = NULL; - snd_info_free_entry(substream->proc_status_entry); - substream->proc_status_entry = NULL; -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - snd_info_free_entry(substream->proc_xrun_injection_entry); - substream->proc_xrun_injection_entry = NULL; -#endif - snd_info_free_entry(substream->proc_root); - substream->proc_root = NULL; - return 0; -} #else /* !CONFIG_SND_VERBOSE_PROCFS */ static inline int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) { return 0; } -static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) { return 0; } #endif /* CONFIG_SND_VERBOSE_PROCFS */ static const struct attribute_group *pcm_dev_attr_groups[]; @@ -911,15 +865,17 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) #if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif + + /* free all proc files under the stream */ + snd_pcm_stream_proc_done(pstr); + substream = pstr->substream; while (substream) { substream_next = substream->next; snd_pcm_timer_done(substream); - snd_pcm_substream_proc_done(substream); kfree(substream); substream = substream_next; } - snd_pcm_stream_proc_done(pstr); #if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4b5356a10315..9a98bc61461f 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -93,12 +93,6 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { snd_pcm_lib_preallocate_dma_free(substream); -#ifdef CONFIG_SND_VERBOSE_PROCFS - snd_info_free_entry(substream->proc_prealloc_max_entry); - substream->proc_prealloc_max_entry = NULL; - snd_info_free_entry(substream->proc_prealloc_entry); - substream->proc_prealloc_entry = NULL; -#endif return 0; } @@ -203,21 +197,15 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) entry->c.text.write = snd_pcm_lib_preallocate_proc_write; entry->mode |= 0200; entry->private_data = substream; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_prealloc_entry = entry; if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc_max", substream->proc_root)) != NULL) { entry->c.text.read = snd_pcm_lib_preallocate_max_proc_read; entry->private_data = substream; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_prealloc_max_entry = entry; } #else /* !CONFIG_SND_VERBOSE_PROCFS */ -- 2.16.4

1 0

[alsa-devel] [PATCH v5 0/8] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 23 Jan '19

23 Jan '19

Thankyou for reviewing v4 patchset, here is v5 addressing comments from v4. Major change is moving the driver out of MFD as there are no immediate plans to add support to other drivers on this codec. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support without much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is our integration branch incase someone want to try this out: https://git.linaro.org/landing-teams/working/qualcomm/kernel.git/log/?h=int… Thanks, Srini Changes since v4 (https://lkml.org/lkml/2018/9/17/21/) - Move the driver out of MFD as discussed with Lee - defer support to MBHC for more testing! Srinivas Kandagatla (8): ASoC: dt-bindings: update wcd9335 bindings. ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: add slim support .../bindings/sound/qcom,wcd9335.txt | 10 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 576 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5275 +++++++++++++++++ sound/soc/codecs/wcd9335.h | 640 ++ sound/soc/qcom/apq8096.c | 71 +- 8 files changed, 6623 insertions(+), 6 deletions(-) create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c create mode 100644 sound/soc/codecs/wcd9335.h -- 2.20.1

2 12

[alsa-devel] How ALSA handle PCM playing when the ALSA modules are removing
by Zhu, Yingjiang 23 Jan '19

23 Jan '19

Hi, Takashi I am doing an ALSA module test with insert module/remove module loop, when playing a wave during this loop, there will be kernel oops. Does ALSA need handle this case? Or just forbid wave playing during the module load/unload? If this need to be handled, module unload will wait the wave play ending, or just stop the wave play at the first of the load/unload? BR Yingjiang

2 2

[alsa-devel] [PATCH] ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter
by shumingf＠realtek.com 22 Jan '19

22 Jan '19

From: Shuming Fan <shumingf(a)realtek.com> AD/DA ASRC function control two ASRC clock sources separately. Whether AD/DA filter select which clock source, we enable AD/DA ASRC function for all cases. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index bd45d03..9d5acd2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1784,7 +1784,9 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, {"DAC STO1 ASRC", NULL, "DA ASRC"}, {"DAC STO1 ASRC", NULL, "CLKDET"}, -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: core: Fix multi-CODEC setups" to the asoc tree
by Mark Brown 22 Jan '19

22 Jan '19

The patch ASoC: core: Fix multi-CODEC setups 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 3f6a125230d8bfcbfe0c06ff0b8eaccbc727acd7 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 22 Jan 2019 17:36:11 +0000 Subject: [PATCH] ASoC: core: Fix multi-CODEC setups Revert 10dff9b0d (ASoC: soc-core: use for_each_link_codecs() for dai_link codecs) for now as Sylwester Nawrocki reports that it causes oopses on at least Odroid boards. Reported-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c92b4aff57b..8a58fa86675a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -870,7 +870,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_component *component; struct snd_soc_dai **codec_dais; @@ -905,8 +905,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->num_codecs = dai_link->num_codecs; /* Find CODEC from registered CODECs */ + /* we can use for_each_rtd_codec_dai() after this */ codec_dais = rtd->codec_dais; - for_each_link_codecs(dai_link, i, codecs) { + for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n", -- 2.20.1

1 0

[alsa-devel] ALC255 Fixup (help needed)
by Stephan Lachnit 22 Jan '19

22 Jan '19

I have a Xiaomi Mi Air 2018 Laptop with one combined 3.5mm jack at the left side of the keyboard, connected via an ALC255. While audio output with jack detection works fine, alsa doesn't get any mic input. Internal mic and speakers work fine. I tried a bit around with hdajackretask and found that Pin ID 0x19 is wired to the mic input. I uploaded pictures of my config to my stack overflow question: https://stackoverflow.com/questions/54277467/how-to-contribute-to-alsa-inte… I know there should be a patch written for sound/pci/hda/patch_realtek.c which should look somewhat like this: [ALC255_FIXUP_XIAOMI_AIR18] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x01a1913c }, /* combinded audio jack */ }, but I really have no idea what the third and fourth line means and just copied from the Dell fixup. Would be nice if someone could help me to write a proper patch (I have some basic C/C++ knowledge, but no idea about alsa internals). System Info: Debian Unstable libasound2 1.1.7-2 i5 8250U w/ Realtek ALC255 This is my first attempt to contribute to open source project, so I hope I didn't miss anything important. Best regards, Stephan Lachnit

1 0

[alsa-devel] [PATCH 00/21] Sound Open Firmware (SOF) - Intel support
by Pierre-Louis Bossart 22 Jan '19

22 Jan '19

This patch series introduces the support for Intel devices (Baytrail, CherryTrail, Broadwell/Haswell, ApolloLake, GeminiLake, CannonLake, IceLake). Support for Skylake and KabyLake is incomplete at the moment but the basic building blocks required are already present. The series starts with the 'legacy' devices, then all the more recent ones and concludes with ACPI/PCI hooks. SKL+ platforms can be compiled without HDaudio link/codec support, in which case the hdac library is not used. When support for HDMI or HDaudio codecs is selected, parts of this library are used. With Takashi's latest patches, it is understood some fixes will be needed before merge, they will be provided as an update when merged by Mark. Patch 19/21 ('ALSA-HDA-export-process_unsol_events') was submitted as a dependency last week but not yet merged. It is provided for completeness should anyone try to compile the code. The machine drivers supported by SOF are not provided as part of this series, mainly because of likely conflicts with other patches in flight to support HDaudio fallback and additional work required for upstream. The build will only work in 'nocodec' mode for now. Support for SPI-based devices, compressed streams is also not provided for now since it's not ready for upstream just yet. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Pierre Keyon Jie (3): ASoC: SOF: Intel: Add hda-bus support and initialization ASoC: SOF: Intel: SKL, CNL, APL platform DAIs ALSA: HDA: export process_unsol_events() Liam Girdwood (15): ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. ASoC: SOF: Intel: Add HSW HW DSP support ASoC: SOF: Intel: Add support for BDW HW DSP support ASoC: SOF: Intel: Add APL/CNL HW DSP support ASoC: SOF: Intel: Add HDA controller for Intel DSP ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. ASoC: SOF: Intel: Add Intel specific HDA firmware loader ASoC: SOF: Intel: Add Intel specific HDA PCM operations ASoC: SOF: Intel: Add Intel specific HDA stream operations ASoC: SOF: Intel: Add Intel specific HDA trace operations ASoC: SOF: Intel: Add platform differentiation for SKL, APL and CNL ASoC: SOF: Add ACPI device support ASoC: SOF: Add PCI device support ASoC: SOF: Add Build support for SOF core and Intel drivers Pierre-Louis Bossart (3): ASoC: SOF: Intel: Add support for HDAudio codecs ASoC: SOF: Intel: Add SKL-specific code loader ASoC: Intel: Kconfig: expose common option between SST and SOF drivers include/sound/hdaudio.h | 1 + sound/hda/hdac_bus.c | 7 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/intel/Kconfig | 7 +- sound/soc/sof/Kconfig | 82 +++ sound/soc/sof/Makefile | 17 + sound/soc/sof/intel/Kconfig | 136 +++++ sound/soc/sof/intel/Makefile | 17 + sound/soc/sof/intel/apl.c | 99 ++++ sound/soc/sof/intel/bdw.c | 700 +++++++++++++++++++++++ sound/soc/sof/intel/byt.c | 805 +++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 242 ++++++++ sound/soc/sof/intel/hda-bus.c | 107 ++++ sound/soc/sof/intel/hda-codec.c | 185 ++++++ sound/soc/sof/intel/hda-ctrl.c | 166 ++++++ sound/soc/sof/intel/hda-dai.c | 368 ++++++++++++ sound/soc/sof/intel/hda-dsp.c | 417 ++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 387 +++++++++++++ sound/soc/sof/intel/hda-loader-skl.c | 531 ++++++++++++++++++ sound/soc/sof/intel/hda-loader.c | 369 ++++++++++++ sound/soc/sof/intel/hda-pcm.c | 232 ++++++++ sound/soc/sof/intel/hda-stream.c | 661 ++++++++++++++++++++++ sound/soc/sof/intel/hda-trace.c | 79 +++ sound/soc/sof/intel/hda.c | 704 +++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 537 ++++++++++++++++++ sound/soc/sof/intel/hsw.c | 701 +++++++++++++++++++++++ sound/soc/sof/intel/shim.h | 159 ++++++ sound/soc/sof/intel/skl.c | 92 +++ sound/soc/sof/sof-acpi-dev.c | 269 +++++++++ sound/soc/sof/sof-pci-dev.c | 361 ++++++++++++ 31 files changed, 8435 insertions(+), 5 deletions(-) create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/bdw.c create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/cnl.c create mode 100644 sound/soc/sof/intel/hda-bus.c create mode 100644 sound/soc/sof/intel/hda-codec.c create mode 100644 sound/soc/sof/intel/hda-ctrl.c create mode 100644 sound/soc/sof/intel/hda-dai.c create mode 100644 sound/soc/sof/intel/hda-dsp.c create mode 100644 sound/soc/sof/intel/hda-ipc.c create mode 100644 sound/soc/sof/intel/hda-loader-skl.c create mode 100644 sound/soc/sof/intel/hda-loader.c create mode 100644 sound/soc/sof/intel/hda-pcm.c create mode 100644 sound/soc/sof/intel/hda-stream.c create mode 100644 sound/soc/sof/intel/hda-trace.c create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h create mode 100644 sound/soc/sof/intel/hsw.c create mode 100644 sound/soc/sof/intel/shim.h create mode 100644 sound/soc/sof/intel/skl.c create mode 100644 sound/soc/sof/sof-acpi-dev.c create mode 100644 sound/soc/sof/sof-pci-dev.c -- 2.17.1

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[alsa-devel] [PATCH 0/6] ALSA: fireface: support MIDI functionality of Fireface UCX
by Takashi Sakamoto 22 Jan '19

22 Jan '19

Hi, This patchset is a supplement of my previous one and to complete support for Fireface UCX: [alsa-devel] [PATCH 0/9] ALSA: fireface: add support for Fireface UCX http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144465.ht… The mechanism to use MIDi functionality of Fireface UCX is different from the other supported devices; Fireface 800 and Fireface 400. This patchset extends abstraction layer of hardware specification and communication protocol. Unfortunately, a bitflag to decide lower part of address to transfer asynchronous transaction for MIDI message is in option register which has the other options. This driver expects userspace applications to configure the option as expected. In detail please refer to third patch. For testers, a remote branch is available to backport a series of patches to Linux kernel v4.17 or later: https://github.com/takaswie/snd-firewire-improve/tree/topic/ff-ucx-midi A userspace tool to configure addressed register is available in another remote branch for hinawa-utils: https://github.com/takaswie/hinawa-utils/tree/topic/ff-ucx As I noted in previous patchset, this work is based on my packet analysis from rent device for a short term (2 weeks). I'm happy to get your response till next weekend. Regards Takashi Sakamoto (6): ALSA: fireface: change prototype of handler for async transaction with MIDI messages ALSA: fireface: add model-dependent parameter for address range to receive async transaction ALSA: fireface: support tx MIDI functionality of Fireface UCX ALSA: fireface: add model-dependent parameter for address to receive async transaction for MIDI messages ALSA: fireface: add protocol-specific operation to fill transaction buffer with MIDI messages ALSA: fireface: support rx MIDI functionality for Fireface UCX sound/firewire/fireface/ff-midi.c | 2 +- sound/firewire/fireface/ff-protocol-former.c | 29 +++- sound/firewire/fireface/ff-protocol-latter.c | 139 +++++++++++++++++++ sound/firewire/fireface/ff-transaction.c | 44 +++--- sound/firewire/fireface/ff.c | 31 +++-- sound/firewire/fireface/ff.h | 10 +- 6 files changed, 210 insertions(+), 45 deletions(-) -- 2.19.1

2 7

[alsa-devel] [PATCH 0/7] ALSA: Refactoring PCM locking code
by Takashi Iwai 22 Jan '19

22 Jan '19

Hi, this is a patch set to refactor the PCM locking code and to reduce the global lock. At least, this gets rid of the global rwlock, while the global rwsem is still needed for some non-atomic ops and for avoiding the deadlock at linking. Takashi === Takashi Iwai (7): ALSA: pcm: Call snd_card_unref() inside in_pcm_file() ALSA: pcm: Unify snd_pcm_group initialization ALSA: pcm: Make PCM linked list consistent while re-grouping ALSA: pcm: Avoid confusing loop in snd_pcm_unlink() ALSA: pcm: More fine-grained PCM link locking ALSA: pcm: Remove down_write() hack for snd_pcm_link_rwsem ALSA: pcm: Cleanup snd_pcm_stream_lock() & co include/sound/pcm.h | 3 +- sound/core/pcm.c | 4 +- sound/core/pcm_local.h | 1 + sound/core/pcm_native.c | 290 ++++++++++++++++++++++++++++-------------------- 4 files changed, 171 insertions(+), 127 deletions(-) -- 2.16.4

1 7

[alsa-devel] [PATCH v2 0/6] Runtime PM support (hda/tegra)
by Sameer Pujar 22 Jan '19

22 Jan '19

Background ========== The device power management operations can be controlled with the help of runtime power management (runtime PM) framework. In this case hda driver can register runtime PM callbacks with the power management core (PM core). Idea is to put the device in low power mode, when it is not getting used. The clocks and power domains can be turned off when no use case is running on the device. Current patch series adds necessary support to manage hda device at runtime. Change log ========== v1: ------ Patch-1: "ALSA: hda/tegra: runtime power management support" * runtime PM enable/disable added in device probe/remove * runtime PM calls are forbidden if AZX_DCAPS_PM_RUNTIME is not enabled in driver flags * worker thread uses pm_runtime_get_sync to invoke ->runtime_resume and pm_runtime_put to invoke ->runtime_suspend. The callbacks are added in subsequent patches Patch-2: "ALSA: hda/tegra: get clock handles early in probe" * Clock handles required for hda are acquired early in the probe. This is done to make use of runtime PM framework to enable/disable clocks in the callbacks. Patch-3: "ALSA: hda/tegra: add runtime PM callbacks" * adds runtime PM callbacks for ->runtime_suspend(), ->runtime_resume() * only skeleton for the callbacks is added, implementation is done in the subsequent patches Patch-4: "ALSA: hda/tegra: remove redundant clock enable API" * As worked thread is calling pm_runtime_get_sync() now, no need to enable the clock explicitly. This can be managed from callback. * Idea is to have clock enable and disable functionalities in runtime PM callbacks and the driver can make use of the above. Any device access should be preceded by ->runtime_resume() call. Patch-5: "ALSA: hda/tegra: implement runtime suspend/resume" * clock enable and controller initialization is moved to ->runtime_reesume() and the reverse is done during ->runtime_suspend() * Also during system wide power transitions, pm_runtime_force_resume() is invoked in system resume() and similarly for suspend case. Patch-6: "ALSA: hda/tegra: fix kernel panic" * kernel panic is happening befoe controller init is done * added a flag "probed" to indicate the completion of device init * this flag is checked before doing any device access. Patch-7: "ALSA: hda/tegra: add driver flag for runtime PM" * driver flag AZX_DCAPS_PM_RUNTIME is added if CONFIG_PM is enabled. * otherwise runtime PM calls will be forbidden and act as dummy calls v1-->v2 ------- Patch-1: "ALSA: hda/tegra: runtime power management support" * no change Patch-2: "ALSA: hda/tegra: get clock handles early in probe" * no change Patch-3: "ALSA: hda/tegra: add runtime PM callbacks" * no change Patch-4: "ALSA: hda/tegra: remove redundant clock enable API" * no change Patch-5: "ALSA: hda/tegra: implement runtime suspend/resume" * Patch-6(v1) commit is melded into this commit * instead of new flag "probed", chip->running status is used to know the completion of probe Patch-6: "ALSA: hda/tegra: add driver flag for runtime PM" * This commit is similar to Patch-7(v1) * AZX_DCAPS_PM_RUNTIME is unconditionally added and CONFIG_PM under which above flag was added is removed ========= Sameer Pujar (6): ALSA: hda/tegra: runtime power management support ALSA: hda/tegra: get clock handles early in probe ALSA: hda/tegra: add runtime PM callbacks ALSA: hda/tegra: remove redundant clock enable API ALSA: hda/tegra: implement runtime suspend/resume ALSA: hda/tegra: add driver flag for runtime PM sound/pci/hda/hda_tegra.c | 120 +++++++++++++++++++++++++++++++++------------- 1 file changed, 88 insertions(+), 32 deletions(-) -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 7

[alsa-devel] [PATCH 0/7] Runtime PM support (hda/tegra)
by Sameer Pujar 22 Jan '19

22 Jan '19

Background ========== The device power management operations can be controlled with the help of runtime power management (runtime PM) framework. In this case hda driver can register runtime PM callbacks with the power management core (PM core). Idea is to put the device in low power mode, when it is not getting used. The clocks and power domains can be turned off when no use case is running on the device. Current patch series adds necessary support to manage hda device at runtime. Change log ========== v1: ---- Patch-1: "ALSA: hda/tegra: runtime power management support" * runtime PM enable/disable added in device probe/remove * runtime PM calls are forbidden if AZX_DCAPS_PM_RUNTIME is not enabled in driver flags * worker thread uses pm_runtime_get_sync to invoke ->runtime_resume and pm_runtime_put to invoke ->runtime_suspend. The callbacks are added in subsequent patches Patch-2: "ALSA: hda/tegra: get clock handles early in probe" * Clock handles required for hda are acquired early in the probe. This is done to make use of runtime PM framework to enable/disable clocks in the callbacks. Patch-3: "ALSA: hda/tegra: add runtime PM callbacks" * adds runtime PM callbacks for ->runtime_suspend(), ->runtime_resume() * only skeleton for the callbacks is added, implementation is done in the subsequent patches Patch-4: "ALSA: hda/tegra: remove redundant clock enable API" * As worked thread is calling pm_runtime_get_sync() now, no need to enable the clock explicitly. This can be managed from callback. * Idea is to have clock enable and disable functionalities in runtime PM callbacks and the driver can make use of the above. Any device access should be preceded by ->runtime_resume() call. Patch-5: "ALSA: hda/tegra: implement runtime suspend/resume" * clock enable and controller initialization is moved to ->runtime_reesume() and the reverse is done during ->runtime_suspend() * Also during system wide power transitions, pm_runtime_force_resume() is invoked in system resume() and similarly for suspend case. Patch-6: "ALSA: hda/tegra: fix kernel panic" * kernel panic is happening befoe controller init is done * added a flag "probed" to indicate the completion of device init * this flag is checked before doing any device access. Patch-7: "ALSA: hda/tegra: add driver flag for runtime PM" * driver flag AZX_DCAPS_PM_RUNTIME is added if CONFIG_PM is enabled. * otherwise runtime PM calls will be forbidden and act as dummy calls ========= Sameer Pujar (7): ALSA: hda/tegra: runtime power management support ALSA: hda/tegra: get clock handles early in probe ALSA: hda/tegra: add runtime PM callbacks ALSA: hda/tegra: remove redundant clock enable API ALSA: hda/tegra: implement runtime suspend/resume ALSA: hda/tegra: fix kernel panic ALSA: hda/tegra: add driver flag for runtime PM sound/pci/hda/hda_tegra.c | 120 +++++++++++++++++++++++++++++++++------------- 1 file changed, 88 insertions(+), 32 deletions(-) -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 13

[alsa-devel] Noise in begnning of alsa application
by Chakravarthi Pradeep 21 Jan '19

21 Jan '19

I have ALSA driver for capture device which works fine with VLC. when I try to capture and render using own application having some issues. When I start play, it has some noise in beginning and audio is discontinuous in between. Discontinuous means, some audio frame are getting dropped in between. cut cut sound for sine wave. I have tried snd_pcm_drop and snd_pcm_drain but no use. Kindly suggest how to remove the starting noise in ALSA. -- Thanks and Regards Chakravarthi Pradeep.K

3 5

[alsa-devel] [PATCH] ASoC: MAINTAINERS: fsl: Change Fabio's email address
by Fabio Estevam 21 Jan '19

21 Jan '19

I prefer to use my personal email address for kernel related work. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a3d1222b81ad..f3de979b8e58 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6178,7 +6178,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi <timur(a)kernel.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> -R: Fabio Estevam <fabio.estevam(a)nxp.com> +R: Fabio Estevam <festevam(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained @@ -10954,7 +10954,7 @@ F: include/linux/nvmem-consumer.h F: include/linux/nvmem-provider.h NXP SGTL5000 DRIVER -M: Fabio Estevam <fabio.estevam(a)nxp.com> +M: Fabio Estevam <festevam(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: Documentation/devicetree/bindings/sound/sgtl5000.txt -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: soc-core: add new snd_soc_flush_all_delayed_work()" to the asoc tree
by Mark Brown 21 Jan '19

21 Jan '19

The patch ASoC: soc-core: add new snd_soc_flush_all_delayed_work() 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 65462e445f78cb2f9378443be7ba8b7e07300694 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 21 Jan 2019 09:32:37 +0900 Subject: [PATCH] ASoC: soc-core: add new snd_soc_flush_all_delayed_work() soc-core is calling flush_delayed_work() many times for same purpose. Same code in many places makes code un-understandable. This patch adds new snd_soc_flush_all_delayed_work() for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c63921675d5..eeb794d8a262 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -425,6 +425,14 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); +static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + for_each_card_rtds(card, rtd) + flush_delayed_work(&rtd->delayed_work); +} + static void codec2codec_close_delayed_work(struct work_struct *work) { /* @@ -494,8 +502,7 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { @@ -2228,11 +2235,8 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd; - /* make sure any delayed work runs */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); /* free the ALSA card at first; this syncs with pending operations */ snd_card_free(card->snd_card); @@ -2275,8 +2279,7 @@ int snd_soc_poweroff(struct device *dev) * Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); snd_soc_dapm_shutdown(card); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: soc-core: merge card resources cleanup method" to the asoc tree
by Mark Brown 21 Jan '19

21 Jan '19

The patch ASoC: soc-core: merge card resources cleanup method 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 53e947a0e1f770b9707febb7054b856878945d50 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 21 Jan 2019 09:32:42 +0900 Subject: [PATCH] ASoC: soc-core: merge card resources cleanup method We need to cleanup card resources when snd_soc_instantiate_card() was failed, or when snd_soc_unbind_card() was called. But they are cleanuping card resources on each way. Same code in many places makes code un-understandable. This patch reuses soc_cleanup_card_resources() for cleanuping code resource. Then, it makes avoiding cleanup order. It will be called from snd_soc_instantiate_card() and snd_soc_unbind_card(). Then, original soc_cleanup_card_resources() included snd_soc_flush_all_delayed_work(), but it is now separated. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 103 ++++++++++++++++++------------------------- 1 file changed, 43 insertions(+), 60 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eeb794d8a262..d59b5ea9fa25 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2008,6 +2008,29 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) } } +static int soc_cleanup_card_resources(struct snd_soc_card *card) +{ + /* free the ALSA card at first; this syncs with pending operations */ + if (card->snd_card) + snd_card_free(card->snd_card); + + /* remove and free each DAI */ + soc_remove_dai_links(card); + soc_remove_pcm_runtimes(card); + + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + + snd_soc_dapm_free(&card->dapm); + soc_cleanup_card_debugfs(card); + + /* remove the card */ + if (card->remove) + card->remove(card); + + return 0; +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -2017,6 +2040,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + card->dapm.bias_level = SND_SOC_BIAS_OFF; + card->dapm.dev = card->dev; + card->dapm.card = card; + list_add(&card->dapm.list, &card->dapm_list); + /* check whether any platform is ignore machine FE and using topology */ soc_check_tplg_fes(card); @@ -2024,14 +2052,14 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) for_each_card_prelinks(card, i, dai_link) { ret = soc_bind_dai_link(card, dai_link); if (ret != 0) - goto base_error; + goto probe_end; } /* bind aux_devs too */ for (i = 0; i < card->num_aux_devs; i++) { ret = soc_bind_aux_dev(card, i); if (ret != 0) - goto base_error; + goto probe_end; } /* add predefined DAI links to the list */ @@ -2045,16 +2073,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: can't create sound card for card %s: %d\n", card->name, ret); - goto base_error; + goto probe_end; } soc_init_card_debugfs(card); - card->dapm.bias_level = SND_SOC_BIAS_OFF; - card->dapm.dev = card->dev; - card->dapm.card = card; - list_add(&card->dapm.list, &card->dapm_list); - #ifdef CONFIG_DEBUG_FS snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); #endif @@ -2076,7 +2099,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (card->probe) { ret = card->probe(card); if (ret < 0) - goto card_probe_error; + goto probe_end; } /* probe all components used by DAI links on this card */ @@ -2087,7 +2110,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2095,7 +2118,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe auxiliary components */ ret = soc_probe_aux_devices(card); if (ret < 0) - goto probe_dai_err; + goto probe_end; /* * Find new DAI links added during probing components and bind them. @@ -2107,10 +2130,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; ret = soc_bind_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; } /* probe all DAI links on this card */ @@ -2121,7 +2144,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2168,7 +2191,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: %s late_probe() failed: %d\n", card->name, ret); - goto probe_aux_dev_err; + goto probe_end; } } @@ -2178,33 +2201,17 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: failed to register soundcard %d\n", ret); - goto probe_aux_dev_err; + goto probe_end; } card->instantiated = 1; dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); - mutex_unlock(&card->mutex); - mutex_unlock(&client_mutex); - - return 0; -probe_aux_dev_err: - soc_remove_aux_devices(card); - -probe_dai_err: - soc_remove_dai_links(card); - -card_probe_error: - if (card->remove) - card->remove(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - snd_card_free(card->snd_card); +probe_end: + if (ret < 0) + soc_cleanup_card_resources(card); -base_error: - soc_remove_pcm_runtimes(card); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2233,31 +2240,6 @@ static int soc_probe(struct platform_device *pdev) return snd_soc_register_card(card); } -static int soc_cleanup_card_resources(struct snd_soc_card *card) -{ - /* make sure any delayed work runs */ - snd_soc_flush_all_delayed_work(card); - - /* free the ALSA card at first; this syncs with pending operations */ - snd_card_free(card->snd_card); - - /* remove and free each DAI */ - soc_remove_dai_links(card); - soc_remove_pcm_runtimes(card); - - /* remove auxiliary devices */ - soc_remove_aux_devices(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - - /* remove the card */ - if (card->remove) - card->remove(card); - - return 0; -} - /* removes a socdev */ static int soc_remove(struct platform_device *pdev) { @@ -2823,6 +2805,7 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) if (card->instantiated) { card->instantiated = false; snd_soc_dapm_shutdown(card); + snd_soc_flush_all_delayed_work(card); soc_cleanup_card_resources(card); if (!unregister) list_add(&card->list, &unbind_card_list); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: soc-core: reduce if/else nest on soc_probe_link_dais" to the asoc tree
by Mark Brown 21 Jan '19

21 Jan '19

The patch ASoC: soc-core: reduce if/else nest on soc_probe_link_dais 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 52293596f5afff10d14e033aa3edfc801a31b3a1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 21 Jan 2019 09:32:50 +0900 Subject: [PATCH] ASoC: soc-core: reduce if/else nest on soc_probe_link_dais Deep nested codec is not readable. Let's reduce if/else nest. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 37 +++++++++++++++++-------------------- 1 file changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d59b5ea9fa25..7fc10a41e0e9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1612,27 +1612,24 @@ static int soc_probe_link_dais(struct snd_soc_card *card, dai_link->stream_name); return ret; } - } else { - - if (!dai_link->params) { - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; - } - ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); - if (ret < 0) - return ret; - ret = soc_link_dai_pcm_new(rtd->codec_dais, - rtd->num_codecs, rtd); - if (ret < 0) - return ret; - } else { - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + } else if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; } + ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); + if (ret < 0) + return ret; + ret = soc_link_dai_pcm_new(rtd->codec_dais, + rtd->num_codecs, rtd); + if (ret < 0) + return ret; + } else { + INIT_DELAYED_WORK(&rtd->delayed_work, + codec2codec_close_delayed_work); } return 0; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: simple-card: rename to asoc_simple_card_canonicalize_platform()
by Kuninori Morimoto 21 Jan '19

21 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current simple-card is using asoc_simple_card_canonicalize_dailink(). Its naming is "dailink", but is for "platform". We already have asoc_simple_card_canonicalize_cpu() for "cpu", let's follow same naming rule. It never return error, so, void function is better idea. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 11 +++-------- sound/soc/generic/simple-card-utils.c | 7 ++----- sound/soc/generic/simple-card.c | 11 +++-------- 4 files changed, 9 insertions(+), 22 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index ab5a2ba..7afe453 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -108,7 +108,7 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai); -int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link); +void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 42b077c..bb12351 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -307,14 +307,12 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, "prefix"); } + asoc_simple_card_canonicalize_platform(dai_link); + ret = asoc_simple_card_of_parse_tdm(ep, dai); if (ret) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); if (ret < 0) @@ -405,10 +403,6 @@ static int graph_dai_link_of(struct graph_priv *priv, if (ret < 0) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, @@ -419,6 +413,7 @@ static int graph_dai_link_of(struct graph_priv *priv, dai_link->ops = &graph_ops; dai_link->init = graph_dai_init; + asoc_simple_card_canonicalize_platform(dai_link); asoc_simple_card_canonicalize_cpu(dai_link, of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 3c0901d..5c1424f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -394,16 +394,13 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); -int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) +void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ if (!dai_link->platforms->of_node) dai_link->platforms->of_node = dai_link->cpu_of_node; - - return 0; - } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_dailink); +EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_platform); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d8a0d1e..08df261 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -297,14 +297,12 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, simple_get_conversion(dev, np, &dai_props->adata); + asoc_simple_card_canonicalize_platform(dai_link); + ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) return ret; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - snprintf(prop, sizeof(prop), "%smclk-fs", prefix); of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(node, prop, &dai_props->mclk_fs); @@ -409,10 +407,6 @@ static int simple_dai_link_of(struct simple_priv *priv, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - goto dai_link_of_err; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, @@ -424,6 +418,7 @@ static int simple_dai_link_of(struct simple_priv *priv, dai_link->init = simple_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_card_canonicalize_platform(dai_link); dai_link_of_err: of_node_put(node); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: dapm: Skip suspending widgets with ignore flag
by Zhiwei Jiang 21 Jan '19

21 Jan '19

For endpoint wigdets which have set the suspend ignore flag, asoc shouldn't mark them as dirty when ASoC suspend function is called. Signed-off-by: Zhiwei Jiang <qq282012236(a)gmail.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..68c49090f561 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -278,7 +278,7 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { - if (w->is_ep) { + if (w->is_ep && !w->ignore_suspend) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) dapm_widget_invalidate_output_paths(w); -- 2.19.0

2 1

[alsa-devel] testing hardware with alsaloop.
by tom＠hitechsys.com 21 Jan '19

21 Jan '19

Hello Im running alsaloop test to test the audio on our hardware. Im specifically running this command - alsaloop -C hw:0,0 -P hw:1,0 -S 1 -l 480 -c 6 -v When I measure the delay through my hardware I get 11mSec - which is fine, and the audio quality is good as well. However after several hours of running this loop test the delay has increased to 90mSec. Can someone advice what is going wrong here please. Best regards Tom Favell | Hi Tech Systems Ltd http://www.hitechsys.com For support go to https://support.hitechsys.com/ T: +44 (0) 1256 780880 M: +44 (0) 79797 55359 http://www.hitechsys.com © 2019 Hi Tech Systems Ltd. Holbrook House, Oakley Lane, Basingstoke, RG23 7JY, UK Registered in England No. 2130346 | VAT No. GB 450390370 The information contained in this e-mail is confidential and is intended solely for the use of the named addressee. Access, copying or re-use of the information in it by any other person is not authorized. The Internet is not a secure transmission medium and therefore the content of this e-mail may have been changed without the consent of the originator. If you are not the intended recipient, please destroy this email and contact the sender.http://www.hitechsys.com

1 0

[alsa-devel] [PATCH] ALSA: hda - Add mute LED support for HP ProBook 470 G5
by Anthony Wong 21 Jan '19

21 Jan '19

From: Anthony Wong <anthony.wong(a)canonical.com> Support speaker and mic mute LEDs on HP ProBook 470 G5. BugLink: https://bugs.launchpad.net/bugs/1811254 Signed-off-by: Anthony Wong <anthony.wong(a)canonical.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 88ce2f1..8b5071b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -966,6 +966,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.7.4

2 1

[alsa-devel] [PATCH 0/9] ALSA: fireface: add support for Fireface UCX
by Takashi Sakamoto 21 Jan '19

21 Jan '19

Hi, In 2012, RME GmbH shipped Fireface UFX. This model is one of latter models of Fireface series and support both of IEEE 1394 bus and USB. Furthermore, it supports two types of remote control unit (Basic Remote and Advanced Remote Control) with by 9pin mini-din connector. This patchset adds support for this model, as a part of devices by ALSA firewire stack (therefore it's not a device of USB). At present, maintain of isochronous communication and synchronization status are cleared but asynchronous communication for MIDI messages are not yet, therefore this commit adds support for isochronous communication for PCM frames. This work is based on my packet analysis from rent device for a short term (2 weeks). I'm happy to get your responses till next weekend. Driver code in a remote branch is available for backport till Linux kernel v4.17 or later for your test. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Regards Takashi Sakamoto (9): ALSA: fireface: rename protocol layer for former models ALSA: fireface: unify protocol layer for FF400/FF800 ALSA: fireface: obsolete proc node to leave one node ALSA: fireface: add protocol-dependent operation to dump status ALSA: fireface: add protocol-dependent operation to switch mode to fetch PCM frame ALSA: fireface: add protocol-dependent operation to get clock status ALSA: fireface: code refactoring for dump of sync status ALSA: fireface: code refactoring to parse of clock configuration ALSA: fireface: add support for Fireface UCX sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 4 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 217 +------- sound/firewire/fireface/ff-protocol-ff400.c | 161 ------ sound/firewire/fireface/ff-protocol-ff800.c | 143 ----- sound/firewire/fireface/ff-protocol-former.c | 549 +++++++++++++++++++ sound/firewire/fireface/ff-protocol-latter.c | 273 +++++++++ sound/firewire/fireface/ff-stream.c | 40 +- sound/firewire/fireface/ff-transaction.c | 63 --- sound/firewire/fireface/ff.c | 41 +- sound/firewire/fireface/ff.h | 13 +- 12 files changed, 886 insertions(+), 621 deletions(-) delete mode 100644 sound/firewire/fireface/ff-protocol-ff400.c delete mode 100644 sound/firewire/fireface/ff-protocol-ff800.c create mode 100644 sound/firewire/fireface/ff-protocol-former.c create mode 100644 sound/firewire/fireface/ff-protocol-latter.c -- 2.19.1

2 10

[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Switch to modern UUID API
by Andy Shevchenko 21 Jan '19

21 Jan '19

Switch the driver to use modern UUID API, i.e. guid_t type and accompanying functions, such as guid_equal(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 12 ++++++------ sound/soc/intel/skylake/skl-sst-dsp.h | 6 +++--- sound/soc/intel/skylake/skl-sst-utils.c | 23 +++++++---------------- sound/soc/intel/skylake/skl-sst.c | 4 ++-- sound/soc/intel/skylake/skl-topology.c | 24 ++++++++++++------------ sound/soc/intel/skylake/skl-topology.h | 6 +++--- 6 files changed, 33 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..86f8ebc77a32 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1268,12 +1268,12 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) { struct skl_sst *ctx = skl->skl_sst; struct skl_module_inst_id *pin_id; - uuid_le *uuid_mod, *uuid_tplg; + guid_t *uuid_mod, *uuid_tplg; struct skl_module *skl_module; struct uuid_module *module; int i, ret = -EIO; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (list_empty(&ctx->uuid_list)) { dev_err(ctx->dev, "Module list is empty\n"); @@ -1281,7 +1281,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) } list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { mconfig->id.module_id = module->id; if (mconfig->module) mconfig->module->loadable = module->is_loadable; @@ -1298,7 +1298,7 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; - if (!uuid_le_cmp(*uuid_mod, *uuid_tplg)) { + if (guid_equal(uuid_mod, uuid_tplg)) { mconfig->module = skl_module; ret = 0; break; @@ -1310,13 +1310,13 @@ static int skl_get_module_info(struct skl *skl, struct skl_module_cfg *mconfig) list_for_each_entry(module, &ctx->uuid_list, list) { for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } for (i = 0; i < MAX_OUT_QUEUE; i++) { pin_id = &mconfig->m_out_pin[i].id; - if (!uuid_le_cmp(pin_id->mod_uuid, module->uuid)) + if (guid_equal(&pin_id->mod_uuid, &module->uuid)) pin_id->module_id = module->id; } } diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index e1d6f6719f7e..cbc7a93d56c2 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -177,7 +177,7 @@ struct skl_dsp_loader_ops { #define MAX_INSTANCE_BUFF 2 struct uuid_module { - uuid_le uuid; + guid_t uuid; int id; int is_loadable; int max_instance; @@ -241,8 +241,8 @@ void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, unsigned int offset, int index); -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id); +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id); +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id); int skl_get_pvt_instance_id_map(struct skl_sst *ctx, int module_id, int instance_id); void skl_freeup_uuid_list(struct skl_sst *ctx); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 2ae405617876..85551321c35b 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -21,17 +21,11 @@ #include "../common/sst-dsp-priv.h" #include "skl-sst-ipc.h" - -#define UUID_STR_SIZE 37 #define DEFAULT_HASH_SHA256_LEN 32 /* FW Extended Manifest Header id = $AE1 */ #define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 -struct UUID { - u8 id[16]; -}; - union seg_flags { u32 ul; struct { @@ -65,7 +59,7 @@ struct module_type { struct adsp_module_entry { u32 struct_id; u8 name[8]; - struct UUID uuid; + u8 uuid[16]; struct module_type type; u8 hash1[DEFAULT_HASH_SHA256_LEN]; u32 entry_point; @@ -184,13 +178,13 @@ static inline int skl_pvtid_128(struct uuid_module *module) * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique */ -int skl_get_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int instance_id) +int skl_get_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int instance_id) { struct uuid_module *module; int pvt_id; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { pvt_id = skl_pvtid_128(module); if (pvt_id >= 0) { @@ -214,13 +208,13 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * * This frees a 128 bit private unique id previously generated */ -int skl_put_pvt_id(struct skl_sst *ctx, uuid_le *uuid_mod, int *pvt_id) +int skl_put_pvt_id(struct skl_sst *ctx, guid_t *uuid_mod, int *pvt_id) { int i; struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { + if (guid_equal(uuid_mod, &module->uuid)) { if (*pvt_id != 0) i = (*pvt_id) / 64; @@ -247,7 +241,6 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, struct adsp_fw_hdr *adsp_hdr; struct adsp_module_entry *mod_entry; int i, num_entry, size; - uuid_le *uuid_bin; const char *buf; struct skl_sst *skl = ctx->thread_context; struct uuid_module *module; @@ -279,8 +272,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, return -EINVAL; } - mod_entry = (struct adsp_module_entry *) - (buf + offset + adsp_hdr->len); + mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); num_entry = adsp_hdr->num_modules; @@ -307,8 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, goto free_uuid_list; } - uuid_bin = (uuid_le *)mod_entry->uuid.id; - memcpy(&module->uuid, uuid_bin, sizeof(module->uuid)); + guid_copy(&module->uuid, (guid_t *)&mod_entry->uuid); module->id = (i | (index << 12)); module->is_loadable = mod_entry->type.load_type; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..92806119a755 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -420,9 +420,9 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - uuid_le *uuid_mod; + guid_t *uuid_mod; - uuid_mod = (uuid_le *)guid; + uuid_mod = (guid_t *)guid; snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", "intel/dsp_fw_", uuid_mod, ".bin"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 76dde12cc2bb..160a5c864b3f 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -577,7 +577,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) int ret = 0; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; w = w_module->w; mconfig = w->priv; @@ -585,7 +585,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (mconfig->id.module_id < 0) { dev_err(skl->skl_sst->dev, "module %pUL id not populated\n", - (uuid_le *)mconfig->guid); + (guid_t *)mconfig->guid); return -EIO; } @@ -619,7 +619,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) * FE/BE params */ skl_tplg_update_module_params(w, ctx); - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; mconfig->id.pvt_id = skl_get_pvt_id(ctx, uuid_mod, mconfig->id.instance_id); if (mconfig->id.pvt_id < 0) @@ -658,9 +658,9 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, struct skl_module_cfg *mconfig = NULL; list_for_each_entry(w_module, &pipe->w_list, node) { - uuid_le *uuid_mod; + guid_t *uuid_mod; mconfig = w_module->w->priv; - uuid_mod = (uuid_le *)mconfig->guid; + uuid_mod = (guid_t *)mconfig->guid; if (mconfig->module->loadable && ctx->dsp->fw_ops.unload_mod && mconfig->m_state > SKL_MODULE_UNINIT) { @@ -916,12 +916,12 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, return 0; } -static int skl_get_module_id(struct skl_sst *ctx, uuid_le *uuid) +static int skl_get_module_id(struct skl_sst *ctx, guid_t *uuid) { struct uuid_module *module; list_for_each_entry(module, &ctx->uuid_list, list) { - if (uuid_le_cmp(*uuid, module->uuid) == 0) + if (guid_equal(uuid, &module->uuid)) return module->id; } @@ -2121,11 +2121,11 @@ static int skl_tplg_add_pipe(struct device *dev, return 0; } -static int skl_tplg_get_uuid(struct device *dev, u8 *guid, +static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) { if (uuid_tkn->token == SKL_TKN_UUID) { - memcpy(guid, &uuid_tkn->uuid, 16); + guid_copy(guid, (guid_t *)&uuid_tkn->uuid); return 0; } @@ -2151,7 +2151,7 @@ static int skl_tplg_fill_pin(struct device *dev, break; case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, m_pin[pin_index].id.mod_uuid.b, + ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); if (ret < 0) return ret; @@ -2666,7 +2666,7 @@ static int skl_tplg_get_tokens(struct device *dev, case SND_SOC_TPLG_TUPLE_TYPE_UUID: if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, mconfig->guid, + ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, array->uuid); is_module_guid = false; } else { @@ -3484,7 +3484,7 @@ static int skl_tplg_get_manifest_uuid(struct device *dev, if (uuid_tkn->token == SKL_TKN_UUID) { mod = skl->modules[ref_count]; - memcpy(&mod->uuid, &uuid_tkn->uuid, sizeof(uuid_tkn->uuid)); + guid_copy(&mod->uuid, (guid_t *)&uuid_tkn->uuid); ref_count++; } else { dev_err(dev, "Not an UUID token tkn %d\n", uuid_tkn->token); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 82282cac9751..9311e248b6f9 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -224,7 +224,7 @@ struct skl_mod_inst_map { struct skl_uuid_inst_map { u16 inst_id; u16 reserved; - uuid_le mod_uuid; + guid_t mod_uuid; } __packed; struct skl_kpb_params { @@ -236,7 +236,7 @@ struct skl_kpb_params { }; struct skl_module_inst_id { - uuid_le mod_uuid; + guid_t mod_uuid; int module_id; u32 instance_id; int pvt_id; @@ -369,7 +369,7 @@ struct skl_module_res { }; struct skl_module { - uuid_le uuid; + guid_t uuid; u8 loadable; u8 input_pin_type; u8 output_pin_type; -- 2.18.0

3 4

[alsa-devel] [PATCH 0/2] ASoC: soc-core: use snd_soc_dai_link_component for cpu
by Kuninori Morimoto 21 Jan '19

21 Jan '19

Hi Mark These add snd_soc_dai_link_component support for CPU. I know multi-CPU support patch was posted to ML, but I think it is not yet accepted (or rejected ?). If these patches were accepted, we can finally switch to modern style dai_link, and remove legacy style dai_link. Kuninori Morimoto (2): ASoC: soc-core: add .num_platform for dai_link ASoC: soc-core: use snd_soc_dai_link_component for cpu For switching to modern, I added "use modern style dai_link" and "remove legacy style dai_link" patches as sample. If these have no objection, I will post full patch to ML. Kuninori Morimoto (3): ASoC: audio-graph/simple-card: use modern style dai_link for CPU ASoC: atmel: atmel-classd: use modern style dai_link ASoC: soc-core: remove regacy dai_link binding method --- Kuninori Morimoto

3 13

[alsa-devel] [PATCH v2] ASoC: soc-core: add .num_platforms for dai_link
by Kuninori Morimoto 21 Jan '19

21 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current snd_soc_dai_link is starting to use snd_soc_dai_link_component style for Platform, but it is assuming single Platform so far. But We will need to have multi Platform support in the not far future. Currently only simple card is using it as sound card driver, and other drivers are converted to it from legacy style by snd_soc_init_platform(). To avoid future problem of multi Platform support, let's add num_platforms before it is too late. In the same time, to make it same naming mothed, "platform" should be "platforms". This patch fixup it too. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - pickup this patch from posted patch-set include/sound/simple_card_utils.h | 2 +- include/sound/soc.h | 3 ++- sound/soc/generic/audio-graph-card.c | 5 +++-- sound/soc/generic/simple-card-utils.c | 4 ++-- sound/soc/generic/simple-card.c | 7 ++++--- sound/soc/soc-core.c | 21 +++++++++++++++------ 6 files changed, 27 insertions(+), 15 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 6d69ed2..ab5a2ba 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -75,7 +75,7 @@ void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); &dai_link->codec_dai_name, \ list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, dai_link->platform, \ + asoc_simple_card_parse_dai(node, dai_link->platforms, \ &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, diff --git a/include/sound/soc.h b/include/sound/soc.h index c31b6d1..3089257 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -961,7 +961,8 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; - struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link_component *platforms; + unsigned int num_platforms; int id; /* optional ID for machine driver link identification */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3ec96cd..42b077c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -26,7 +26,7 @@ struct graph_priv { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ - struct snd_soc_dai_link_component platform; + struct snd_soc_dai_link_component platforms; struct asoc_simple_card_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; @@ -687,7 +687,8 @@ static int graph_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platforms = &dai_props[i].platforms; + dai_link[i].num_platforms = 1; } priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 336895f..3c0901d 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -397,8 +397,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (!dai_link->platform->of_node) - dai_link->platform->of_node = dai_link->cpu_of_node; + if (!dai_link->platforms->of_node) + dai_link->platforms->of_node = dai_link->cpu_of_node; return 0; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 479de23..d8a0d1e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -21,7 +21,7 @@ struct simple_priv { struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ - struct snd_soc_dai_link_component platform; + struct snd_soc_dai_link_component platforms; struct asoc_simple_card_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; @@ -732,7 +732,8 @@ static int simple_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platforms = &dai_props[i].platforms; + dai_link[i].num_platforms = 1; } priv->dai_props = dai_props; @@ -782,7 +783,7 @@ static int simple_probe(struct platform_device *pdev) codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; - platform = dai_link->platform; + platform = dai_link->platforms; platform->name = cinfo->platform; card->name = (cinfo->card) ? cinfo->card : cinfo->name; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bf405d6..58218d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -921,7 +921,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* find one from the set of registered platforms */ for_each_component(component) { - if (!snd_soc_is_matching_component(dai_link->platform, + if (!snd_soc_is_matching_component(dai_link->platforms, component)) continue; @@ -1035,7 +1035,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link_component *platform = dai_link->platform; + struct snd_soc_dai_link_component *platform = dai_link->platforms; /* * REMOVE ME @@ -1055,7 +1055,8 @@ static int snd_soc_init_platform(struct snd_soc_card *card, if (!platform) return -ENOMEM; - dai_link->platform = platform; + dai_link->platforms = platform; + dai_link->num_platforms = 1; dai_link->legacy_platform = 1; platform->name = dai_link->platform_name; platform->of_node = dai_link->platform_of_node; @@ -1145,11 +1146,19 @@ static int soc_init_dai_link(struct snd_soc_card *card, } } + /* FIXME */ + if (link->num_platforms > 1) { + dev_err(card->dev, + "ASoC: multi platform is not yet supported %s\n", + link->name); + return -EINVAL; + } + /* * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform->name && link->platform->of_node) { + if (link->platforms->name && link->platforms->of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); @@ -1160,7 +1169,7 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Defer card registartion if platform dai component is not added to * component list. */ - if (!soc_find_component(link->platform->of_node, link->platform->name)) + if (!soc_find_component(link->platforms->of_node, link->platforms->name)) return -EPROBE_DEFER; /* @@ -1959,7 +1968,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platform->name = component->name; + dai_link->platforms->name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; -- 2.7.4

2 2

[alsa-devel] simple-audio-card and headphone jack detect
by Paul Cercueil 19 Jan '19

19 Jan '19

Hello, I am using the simple-card and simple-amplifier drivers with the following devicetree: amp: audio-amp { compatible = "simple-audio-amplifier"; enable-gpios = <&gpc 15 GPIO_ACTIVE_HIGH>; }; sound-card { compatible = "simple-audio-card"; simple-audio-card,name = "rs90-audio"; simple-audio-card,format = "i2s"; simple-audio-card,widgets = "Speaker", "Speaker", "Headphone", "Headphones"; simple-audio-card,routing = "INL", "LHPOUT", "INR", "RHPOUT", "Headphones", "LHPOUT", "Headphones", "RHPOUT", "Speaker", "OUTL", "Speaker", "OUTR"; simple-audio-card,hp-det-gpio = <&gpd 16 GPIO_ACTIVE_LOW>; simple-audio-card,aux-devs = <&amp>; simple-audio-card,cpu { sound-dai = <&aic>; }; simple-audio-card,codec { sound-dai = <&codec>; bitclock-master; frame-master; }; }; My codec outputs on the LHPOUT/RHPOUT lines. INL/INR and OUTL/OUTR are the inputs/outputs of the simple-amplifier. So far so good, when no audio is playing the audio amp is disabled (GPIO is low), and it gets enabled as soon as I start to play audio. What is not working, and what I am trying to achieve, is to have the external audio amplifier disabled as soon as headphones are plugged. Is that something the core handles for me? Any suggestions to make it work? Thanks, -Paul

1 0

[alsa-devel] [PATCH] ASoC: fsl_sai: Remove expensive print in irq handler
by Daniel Baluta 18 Jan '19

18 Jan '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> When stopping audio, ASoC will first stop DMA then CPU DAI. Sometimes there is a delay between DMA stop and CPU DAI stop, which triggers an underrun error. Now, because of the delay introduced by dev_err another underrun error will occur causing a vicious circle making impossible to stop CPU DAI. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4163f2cfc06f..db9e0872f73d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -62,10 +62,10 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) dev_dbg(dev, "isr: Start of Tx word detected\n"); if (flags & FSL_SAI_CSR_SEF) - dev_warn(dev, "isr: Tx Frame sync error detected\n"); + dev_dbg(dev, "isr: Tx Frame sync error detected\n"); if (flags & FSL_SAI_CSR_FEF) { - dev_warn(dev, "isr: Transmit underrun detected\n"); + dev_dbg(dev, "isr: Transmit underrun detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } @@ -96,10 +96,10 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) dev_dbg(dev, "isr: Start of Rx word detected\n"); if (flags & FSL_SAI_CSR_SEF) - dev_warn(dev, "isr: Rx Frame sync error detected\n"); + dev_dbg(dev, "isr: Rx Frame sync error detected\n"); if (flags & FSL_SAI_CSR_FEF) { - dev_warn(dev, "isr: Receive overflow detected\n"); + dev_dbg(dev, "isr: Receive overflow detected\n"); /* FIFO reset for safety */ xcsr |= FSL_SAI_CSR_FR; } -- 2.17.1

3 2

[alsa-devel] [PATCH v5 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 18 Jan '19

18 Jan '19

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- include/sound/cs35l36.h | 44 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36.c | 1959 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 446 ++++++++++ 5 files changed, 2456 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..1ee7527 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,44 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +struct cs35l36_vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_pcm_inv; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + int irq_drv_sel; + int irq_gpio_sel; + struct cs35l36_vpbr_cfg vpbr_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f18..77cb861 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -465,6 +466,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c..7d8a372 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -310,6 +311,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..955b90e --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1959 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +struct reg_default cs35l36_reg[] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_TST_FS_MON0: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_TST_FS_MON0: + return true; + default: + return false; + } +} + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->pdata.ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->pdata.ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK, + fs1 | (fs2 << CS35L36_FS2_WINDOW_SHIFT)); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + if (cs35l36->pdata.irq_drv_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + cs35l36->pdata.irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (cs35l36->pdata.irq_gpio_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + cs35l36->pdata.irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .precious_reg = cs35l36_precious_reg, + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "cirrus,irq-drive-select", &val) >= 0) + pdata->irq_drv_sel = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,irq-gpio-select", &val) >= 0) + pdata->irq_gpio_sel = val | CS35L36_VALID_PDATA; + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static void cs35l36_apply_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + struct irq_data *irq_d; + int ret, irq_pol, chip_irq_pol, i; + u32 reg_id, reg_revid, l37_id_reg; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_apply_vpbr_config(cs35l36); + + irq_d = irq_get_irq_data(i2c_client->irq); + if (IS_ERR(irq_d)) { + dev_err(&i2c_client->dev, "Invalid IRQ: %d\n", i2c_client->irq); + ret = PTR_ERR(irq_d); + goto err; + } + + irq_pol = irqd_get_trigger_type(irq_d); + + switch (irq_pol) { + case IRQF_TRIGGER_FALLING: + case IRQF_TRIGGER_LOW: + chip_irq_pol = 0; + break; + case IRQF_TRIGGER_RISING: + case IRQF_TRIGGER_HIGH: + chip_irq_pol = 1; + break; + default: + dev_err(cs35l36->dev, "Invalid IRQ polarity: %d\n", irq_pol); + ret = -EINVAL; + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + chip_irq_pol << CS35L36_INT_POL_SEL_SHIFT); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, 1); + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, + ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..f6e38c6 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,446 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

4 7

[alsa-devel] Applied "ASoC: soc.h: add explanation of legacy/modern style of dai_link" to the asoc tree
by Mark Brown 18 Jan '19

18 Jan '19

The patch ASoC: soc.h: add explanation of legacy/modern style of dai_link 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 62bc79d35ebb55451112979babea864975cfd16d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 18 Jan 2019 11:20:41 +0900 Subject: [PATCH] ASoC: soc.h: add explanation of legacy/modern style of dai_link Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system. Because of this background, it is directly using xxx_name / xxx_of_node / xxx_dai_name on dai_link. Let's call it as legacy style here. More complex style system like multi CPU multi Platform (= DMA) will coming. To supporting it, we can use snd_soc_dai_link_component on dai_link. Let's call it as modern style here. But current ALSA SoC can't support it so far. Thus, we need to have multi CPU / multi Codec / multi Platform style in the future on ALSA SoC. Currently we already have multi Codec support. Platform is starting to use modern style on dai_link, but still style only. Multi Platform is not yet implemented. And we still don't have multi CPU support on ALSA SoC, and not have modern style either. Currently, if driver is using legacy style Codec/Platform, it will be converted to modern style on soc-core. This means, we are using glue code for legacy vs modern style so far on ALSA SoC. We can fully switch to modern style on all drivers if ALSA SoC supported modern style for CPU, and then, legacy style code will be removed from ALSA SoC. Untile then, we need to keep both legacy/modern style and its glue code. This patch adds such future plan and background on soc.h Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 20 ++++++++++++++++++-- 2 files changed, 54 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f111b0d2..c31b6d122ff6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -891,6 +891,18 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ + + /* + * cpu_name + * cpu_of_node + * cpu_dai_name + * + * These are legacy style, and will be replaced to + * modern style (= snd_soc_dai_link_component) in the future, + * but, not yet supported so far. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MAY specify the link's CPU-side device, either by device name, * or by DT/OF node, but not both. If this information is omitted, @@ -906,6 +918,19 @@ struct snd_soc_dai_link { * only, which only works well when that device exposes a single DAI. */ const char *cpu_dai_name; + + /* + * codec_name + * codec_of_node + * codec_dai_name + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Driver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MUST specify the link's codec, either by device name, or by * DT/OF node, but not both. @@ -918,6 +943,17 @@ struct snd_soc_dai_link { struct snd_soc_dai_link_component *codecs; unsigned int num_codecs; + /* + * platform_name + * platform_of_node + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Driver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 890d6c9c2752..de2851f1b3df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1029,9 +1029,14 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link_component *platform = dai_link->platform; /* - * FIXME + * REMOVE ME * - * this function should be removed in the future + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Driver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ if (!platform || dai_link->legacy_platform) { @@ -1059,6 +1064,17 @@ static int snd_soc_init_platform(struct snd_soc_card *card, static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* + * REMOVE ME + * + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Driver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link + */ + /* Legacy codec/codec_dai link is a single entry in multicodec */ if (dai_link->codec_name || dai_link->codec_of_node || dai_link->codec_dai_name) { -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix oops on re-probe
by Russell King 18 Jan '19

18 Jan '19

hdmi-codec oopses the kernel when it is unbound from a successfully bound audio subsystem, and is then rebound: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = ee3f0000 [0000001c] *pgd=3cc59831 Internal error: Oops: 817 [#1] PREEMPT ARM Modules linked in: ext2 snd_soc_spdif_tx vmeta dove_thermal snd_soc_kirkwood ofpart marvell_cesa m25p80 orion_wdt mtd spi_nor des_generic gpio_ir_recv snd_soc_kirkwood_spdif bmm_dmabuf auth_rpcgss nfsd autofs4 etnaviv thermal_sys hwmon gpu_sched tda9950 CPU: 0 PID: 1005 Comm: bash Not tainted 4.20.0+ #1762 Hardware name: Marvell Dove (Cubox) PC is at hdmi_dai_probe+0x68/0x80 LR is at find_held_lock+0x20/0x94 pc : [<c04c7de0>] lr : [<c0063bf4>] psr: 600f0013 sp : ee15bd28 ip : eebd8b1c fp : c093b488 r10: ee048000 r9 : eebdab18 r8 : ee048600 r7 : 00000001 r6 : 00000000 r5 : 00000000 r4 : ee82c100 r3 : 00000006 r2 : 00000001 r1 : c067e38c r0 : ee82c100 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none[ 297.318599] Control: 10c5387d Table: 2e3f0019 DAC: 00000051 Process bash (pid: 1005, stack limit = 0xee15a248) ... [<c04c7de0>] (hdmi_dai_probe) from [<c04b7060>] (soc_probe_dai.part.9+0x34/0x70) [<c04b7060>] (soc_probe_dai.part.9) from [<c04b81a8>] (snd_soc_instantiate_card+0x734/0xc9c) [<c04b81a8>] (snd_soc_instantiate_card) from [<c04b8b6c>] (snd_soc_add_component+0x29c/0x378) [<c04b8b6c>] (snd_soc_add_component) from [<c04b8c8c>] (snd_soc_register_component+0x44/0x54) [<c04b8c8c>] (snd_soc_register_component) from [<c04c64b4>] (devm_snd_soc_register_component+0x48/0x84) [<c04c64b4>] (devm_snd_soc_register_component) from [<c04c7be8>] (hdmi_codec_probe+0x150/0x260) [<c04c7be8>] (hdmi_codec_probe) from [<c0373124>] (platform_drv_probe+0x48/0x98) This happens because hdmi_dai_probe() attempts to access the HDMI codec private data, but this has not been assigned by hdmi_dai_probe() before it calls devm_snd_soc_register_component(). Move the call to dev_set_drvdata() before devm_snd_soc_register_component() to avoid this oops. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- Discovered and patched on 4.20. sound/soc/codecs/hdmi-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 1c628af373eb..6bec7f36eb4c 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -799,6 +799,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) if (hcd->spdif) hcp->daidrv[i] = hdmi_spdif_dai; + dev_set_drvdata(dev, hcp); + ret = devm_snd_soc_register_component(dev, &hdmi_driver, hcp->daidrv, dai_count); if (ret) { @@ -806,8 +808,6 @@ static int hdmi_codec_probe(struct platform_device *pdev) __func__, ret); return ret; } - - dev_set_drvdata(dev, hcp); return 0; } -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: imx-sgtl5000: put of nodes if finding codec fails" to the asoc tree
by Mark Brown 18 Jan '19

18 Jan '19

The patch ASoC: imx-sgtl5000: put of nodes if finding codec fails 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 d9866572486802bc598a3e8576a5231378d190de Mon Sep 17 00:00:00 2001 From: Stefan Agner <stefan(a)agner.ch> Date: Fri, 18 Jan 2019 10:06:52 +0100 Subject: [PATCH] ASoC: imx-sgtl5000: put of nodes if finding codec fails Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..594bde3b0ded 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -111,7 +111,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: imx-sgtl5000: lower log level for potential probe deferral cases" to the asoc tree
by Mark Brown 18 Jan '19

18 Jan '19

The patch ASoC: imx-sgtl5000: lower log level for potential probe deferral cases 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 691beb02d9ff364000a8de4e74898600b9caee89 Mon Sep 17 00:00:00 2001 From: Stefan Agner <stefan(a)agner.ch> Date: Fri, 18 Jan 2019 10:06:53 +0100 Subject: [PATCH] ASoC: imx-sgtl5000: lower log level for potential probe deferral cases Not finding the codec/SSI instance can be due to probe deferral. Do not print error messages in those cases. Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 594bde3b0ded..9790a2a8ec2d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -104,13 +104,13 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); + dev_dbg(&pdev->dev, "failed to find SSI platform device\n"); ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); + dev_dbg(&pdev->dev, "failed to find codec platform device\n"); ret = -EPROBE_DEFER; goto fail; } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: imx-sgtl5000: don't print EPROBE_DEFER as error" to the asoc tree
by Mark Brown 18 Jan '19

18 Jan '19

The patch ASoC: imx-sgtl5000: don't print EPROBE_DEFER as error 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 e379ee969ecbc807c5b233460c2a88b6ee06bde3 Mon Sep 17 00:00:00 2001 From: Stefan Agner <stefan(a)agner.ch> Date: Fri, 18 Jan 2019 10:06:54 +0100 Subject: [PATCH] ASoC: imx-sgtl5000: don't print EPROBE_DEFER as error Probe deferral is to be expected during normal operation, so avoid printing an error when it is encountered. Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9790a2a8ec2d..b6cb80480b60 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -157,7 +157,9 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); goto fail; } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: fsl_spdif: don't print EPROBE_DEFER as error" to the asoc tree
by Mark Brown 18 Jan '19

18 Jan '19

The patch ASoC: fsl_spdif: don't print EPROBE_DEFER as error 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 1aabff2508cbd184b0141f2d1b8712bacbb78cef Mon Sep 17 00:00:00 2001 From: Stefan Agner <stefan(a)agner.ch> Date: Fri, 18 Jan 2019 10:06:55 +0100 Subject: [PATCH] ASoC: fsl_spdif: don't print EPROBE_DEFER as error Probe deferral is to be expected during normal operation, so avoid printing an error when it is encountered. Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 740b90df44bb..a26686e7281c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1320,7 +1320,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); return ret; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: imx-spdif: don't print EPROBE_DEFER as error" to the asoc tree
by Mark Brown 18 Jan '19

18 Jan '19

The patch ASoC: imx-spdif: don't print EPROBE_DEFER as error 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 2363d85f4e04f7218efc66b6167a405cccbc8ff7 Mon Sep 17 00:00:00 2001 From: Stefan Agner <stefan(a)agner.ch> Date: Fri, 18 Jan 2019 10:06:56 +0100 Subject: [PATCH] ASoC: imx-spdif: don't print EPROBE_DEFER as error Probe deferral is to be expected during normal operation, so avoid printing an error when it is encountered. Removing the goto would not be strictly necessary. However, if code gets added later, the cleanup in the EPROBE_DEFER case likely would get missed. Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/imx-spdif.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index fb896b2c9ba3..797d66e43d49 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -67,10 +67,8 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) goto end; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); - goto end; - } end: of_node_put(spdif_np); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: soc-core: remove error due to probe deferral
by Stefan Agner 18 Jan '19

18 Jan '19

Deferred probes shouldn't cause error messages in the boot log, so change the dev_err() to the more harmless dev_info(). Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b29d0f65611e..ba9b09517c5f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -903,8 +903,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", - codecs[i].dai_name); + dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n", + codecs[i].dai_name); goto _err_defer; } snd_soc_rtdcom_add(rtd, codec_dais[i]->component); -- 2.20.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v5.0
by Mark Brown 18 Jan '19

18 Jan '19

The following changes since commit a7a850dba82498a1e050d8d153cae67ce0edb3b2: Merge branch 'asoc-4.21' into asoc-next (2018-12-18 12:23:59 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.0-rc2 for you to fetch changes up to 4cb79ef9c6c4413427cd70afbb1f3bc01e9b7abf: ASoC: amd: Fix potential NULL pointer dereference (2019-01-15 19:06:23 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.0 Quite a big batch of fixes here. There's a couple of things going on, the main one is that we found some issues with not deferring probe when we should, causing us to skip some driver initialization. The fixes for this then in turn exposed some issues with how we were searching for components which had previously gone unnoticed due to the original issue. There's also been the normal driver specific stuff and there's been what looks like several batches of automated scanning for issues which have generated quite a large set of smaller fixes for potential crashes and missed error handling. ---------------------------------------------------------------- Should probably have pointed out that the DaVinci runtime PM fix here is a big diff but mostly code motion, it fixes problems where the device would stop working after runtime PM. Ajit Pandey (1): ASoC: soc-core: defer card probe until all component is added to list Curtis Malainey (1): ASoC: soc-core: fix init platform memory handling Dan Carpenter (6): ASoC: dma-sh7760: cleanup a debug printk ASoC: pcm512x: Fix a double unlock in pcm512x_digital_mute() ASoC: qdsp6: q6asm-dai: Off by one in of_q6asm_parse_dai_data() ASoC: qdsp6: q6asm-dai: Fix a NULL vs IS_ERR() bug ASoC: qdsp6: q6asm-dai: Fix a small memory leak ALSA: compress: prevent potential divide by zero bugs Geert Uytterhoeven (1): ASoC: xlnx: Grammar s/the the/the/ Gustavo A. R. Silva (2): ASoC: rt5514-spi: Fix potential NULL pointer dereference ASoC: amd: Fix potential NULL pointer dereference Kangjie Lu (1): ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages Mac Chiang (1): ASoC: Intel: Boards: move the codec PLL configuration to _init Mark Brown (1): ASoC: core: Make snd_soc_find_component() more robust Maruthi Srinivas Bayyavarapu (1): ASoC: xlnx: change license header format style Matthias Reichl (1): ASoC: core: Don't defer probe on optional, NULL components Peter Ujfalusi (1): ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks Rander Wang (1): ASoC: Intel: Haswell/Broadwell: fix setting for .dynamic field Rohit kumar (2): ASoC: sdm845: set jack only for a specific backend ASoC: soc-core: Hold client_mutex around soc_init_dai_link() Shuming Fan (2): ASoC: rt5682: Fix recording no sound issue ASoC: rt5682: Fix PLL source register definitions Silvio Cesare (2): ASoC: dapm: change snprintf to scnprintf for possible overflow ASoC: imx-audmux: change snprintf to scnprintf for possible overflow Sinan Kaya (1): ASoC: Intel: atom: Make PCI dependency explicit Yizhuo (1): ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized b-ak (1): ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode include/sound/soc.h | 6 ++ sound/core/compress_offload.c | 3 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/codecs/pcm512x.c | 11 +-- sound/soc/codecs/rt274.c | 5 +- sound/soc/codecs/rt5514-spi.c | 2 + sound/soc/codecs/rt5682.c | 1 + sound/soc/codecs/rt5682.h | 24 ++--- sound/soc/codecs/tlv320aic32x4.c | 4 + sound/soc/fsl/imx-audmux.c | 24 ++--- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 45 ++------- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 17 +++- sound/soc/qcom/sdm845.c | 31 ++++-- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/soc-core.c | 34 +++++-- sound/soc/soc-dapm.c | 10 +- sound/soc/ti/davinci-mcasp.c | 136 ++++++++++++-------------- sound/soc/xilinx/Kconfig | 2 +- sound/soc/xilinx/xlnx_i2s.c | 15 ++- 23 files changed, 208 insertions(+), 184 deletions(-)

2 1

[alsa-devel] [PATCH v2 1/5] ASoC: imx-sgtl5000: put of nodes if finding codec fails
by Stefan Agner 18 Jan '19

18 Jan '19

Make sure to properly put the of node in case finding the codec fails. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Stefan Agner <stefan(a)agner.ch> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v2: - Reordered patches, make sure this is the first in the patchset -- Stefan sound/soc/fsl/imx-sgtl5000.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..594bde3b0ded 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -111,7 +111,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EPROBE_DEFER; + ret = -EPROBE_DEFER; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 2.20.1

2 9

[alsa-devel] [PATCH] alsamixer: define _GNU_SOURCE to get exp10
by makepost＠firemail.cc 18 Jan '19

18 Jan '19

Fixes Master, Headphones and Speaker stuck at -8 percentage after building with Clang 7.0.1 and getting warned about implicit declaration of exp10, which is behind _GNU_SOURCE as a non-standard feature. Thanks Takashi Iwai for the CFLAGS suggestion on the mailing list. GCC build is not affected, so not adding a compiler check to the configure script. uClibc-ng has got exp10 since 1.0.12, so the fallback macro is no longer needed. However, alsa-utils relies on gettext so might need further patches to actually work on uClibc systems. Signed-off-by: makepost <makepost(a)firemail.cc> diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 94bd0fe..48cfbe2 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -36,11 +36,6 @@ #include <stdbool.h> #include "volume_mapping.h" -#ifdef __UCLIBC__ -/* 10^x = 10^(log e^x) = (e^x)^log10 = e^(x * log 10) */ -#define exp10(x) (exp((x) * log(10))) -#endif /* __UCLIBC__ */ - #define MAX_LINEAR_DB_SCALE 24 static inline bool use_linear_dB_scale(long dBmin, long dBmax) diff --git a/configure.ac b/configure.ac index 7938996..0d54942 100644 --- a/configure.ac +++ b/configure.ac @@ -350,6 +350,8 @@ if test x$alsamixer = xtrue; then AC_MSG_RESULT([$CURSESLIB]) + + CFLAGS="$CFLAGS -D_GNU_SOURCE" AC_SUBST(CURSESINC) -- 2.20.1

2 3

[alsa-devel] Timing issues between ALSA and i915 drivers
by Pierre-Louis Bossart 18 Jan '19

18 Jan '19

Hi, I could use some feedback on HDMI audio issues exposed during the 4.21 merge window. By accident (misleading documentation) we ended up enabling the Skylake driver instead of the HDaudio legacy, and broke audio on a number of Skylake and ApolloLake devices where the HDMI/iDISP codec was not detected (bit 2 not set in the codec_mask). Linus' Dell XPS13 9350 was the first to be impacted of course... After debugging a bit, this issue can be resolved by either a) compiling both SOUND and DRM as built-ins (y instead of m) b) moving the calls snd_hdac_i915_init() to the probe function instead of the worker queue (code at https://github.com/plbossart/sound/commits/fix/skl-hdmi) Both solutions point to timing issues. During internal reviews I was alerted to the fact that the suggested fix essentially reverts patch ab1b732d53c18 ('ASoC: Intel: Skylake: Move i915 registration to worker thread') which was introduced to solve DRM lockup issues. In other words, we are at a point where we either have DRM lockups or can't detect the HDMI audio codec, none of which are too good. I don't have the background for the DRM lockup stuff, nor do I understand why snd_hdac_i915_init has to be called from a thread context. Is this really a requirement? I also don't get what sort of timing issues would come from an initialization. What happens on the i915 side and is there some sort of mandatory delay between the completion of the i915_init and issuing a snd_hdac_chip_readw(bus, STATESTS) to get the codec masks on the HDaudio links? Thanks for any pointers or comments. -Pierre

2 7

[alsa-devel] [PATCH v2] ASoC: soc.h: add explanation of legacy/modern style of dai_link
by Kuninori Morimoto 18 Jan '19

18 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system. Because of this background, it is directly using xxx_name / xxx_of_node / xxx_dai_name on dai_link. Let's call it as legacy style here. More complex style system like multi CPU multi Platform (= DMA) will coming. To supporting it, we can use snd_soc_dai_link_component on dai_link. Let's call it as modern style here. But current ALSA SoC can't support it so far. Thus, we need to have multi CPU / multi Codec / multi Platform style in the future on ALSA SoC. Currently we already have multi Codec support. Platform is starting to use modern style on dai_link, but still style only. Multi Platform is not yet implemented. And we still don't have multi CPU support on ALSA SoC, and not have modern style either. Currently, if driver is using legacy style Codec/Platform, it will be converted to modern style on soc-core. This means, we are using glue code for legacy vs modern style so far on ALSA SoC. We can fully switch to modern style on all drivers if ALSA SoC supported modern style for CPU, and then, legacy style code will be removed from ALSA SoC. Untile then, we need to keep both legacy/modern style and its glue code. This patch adds such future plan and background on soc.h Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fixup "Driver" typo include/sound/soc.h | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 20 ++++++++++++++++++-- 2 files changed, 54 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f11..c31b6d1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -891,6 +891,18 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ + + /* + * cpu_name + * cpu_of_node + * cpu_dai_name + * + * These are legacy style, and will be replaced to + * modern style (= snd_soc_dai_link_component) in the future, + * but, not yet supported so far. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MAY specify the link's CPU-side device, either by device name, * or by DT/OF node, but not both. If this information is omitted, @@ -906,6 +918,19 @@ struct snd_soc_dai_link { * only, which only works well when that device exposes a single DAI. */ const char *cpu_dai_name; + + /* + * codec_name + * codec_of_node + * codec_dai_name + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Driver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MUST specify the link's codec, either by device name, or by * DT/OF node, but not both. @@ -919,6 +944,17 @@ struct snd_soc_dai_link { unsigned int num_codecs; /* + * platform_name + * platform_of_node + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Driver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link * do not need a platform. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c0aaff..bf405d6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1038,9 +1038,14 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link_component *platform = dai_link->platform; /* - * FIXME + * REMOVE ME * - * this function should be removed in the future + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Driver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ if (!platform || dai_link->legacy_platform) { @@ -1068,6 +1073,17 @@ static int snd_soc_init_platform(struct snd_soc_card *card, static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* + * REMOVE ME + * + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Driver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link + */ + /* Legacy codec/codec_dai link is a single entry in multicodec */ if (dai_link->codec_name || dai_link->codec_of_node || dai_link->codec_dai_name) { -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc.h: add explanation of legacy/modern style of dai_link
by Kuninori Morimoto 18 Jan '19

18 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system. Because of this background, it is directly using xxx_name / xxx_of_node / xxx_dai_name on dai_link. Let's call it as legacy style here. More complex style system like multi CPU multi Platform (= DMA) will coming. To supporting it, we can use snd_soc_dai_link_component on dai_link. Let's call it as modern style here. But current ALSA SoC can't support it so far. Thus, we need to have multi CPU / multi Codec / multi Platform style in the future on ALSA SoC. Currently we already have multi Codec support. Platform is starting to use modern style on dai_link, but still style only. Multi Platform is not yet implemented. And we still don't have multi CPU support on ALSA SoC, and not have modern style either. Currently, if driver is using legacy style Codec/Platform, it will be converted to modern style on soc-core. This means, we are using glue code for legacy vs modern style so far on ALSA SoC. We can fully switch to modern style on all drivers if ALSA SoC supported modern style for CPU, and then, legacy style code will be removed from ALSA SoC. Untile then, we need to keep both legacy/modern style and its glue code. This patch adds such future plan and background on soc.h Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 20 ++++++++++++++++++-- 2 files changed, 54 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f11..d17e368 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -891,6 +891,18 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ + + /* + * cpu_name + * cpu_of_node + * cpu_dai_name + * + * These are legacy style, and will be replaced to + * modern style (= snd_soc_dai_link_component) in the future, + * but, not yet supported so far. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MAY specify the link's CPU-side device, either by device name, * or by DT/OF node, but not both. If this information is omitted, @@ -906,6 +918,19 @@ struct snd_soc_dai_link { * only, which only works well when that device exposes a single DAI. */ const char *cpu_dai_name; + + /* + * codec_name + * codec_of_node + * codec_dai_name + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Dirver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ /* * You MUST specify the link's codec, either by device name, or by * DT/OF node, but not both. @@ -919,6 +944,17 @@ struct snd_soc_dai_link { unsigned int num_codecs; /* + * platform_name + * platform_of_node + * + * These are legacy style, it will be converted to modern style + * (= snd_soc_dai_link_component) automatically in soc-core + * if driver is using legacy style. + * Dirver shouldn't use both legacy and modern style in the same time. + * If modern style was supported for CPU, all driver will switch + * to use it, and, legacy style code will be removed from ALSA SoC. + */ + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link * do not need a platform. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c0aaff..f01a223 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1038,9 +1038,14 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link_component *platform = dai_link->platform; /* - * FIXME + * REMOVE ME * - * this function should be removed in the future + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Dirver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ if (!platform || dai_link->legacy_platform) { @@ -1068,6 +1073,17 @@ static int snd_soc_init_platform(struct snd_soc_card *card, static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* + * REMOVE ME + * + * This is glue code for Legacy vs Modern dai_link. + * This function will be removed if all derivers are switched to + * modern style dai_link. + * Dirver shouldn't use both legacy and modern style in the same time. + * see + * soc.h :: struct snd_soc_dai_link + */ + /* Legacy codec/codec_dai link is a single entry in multicodec */ if (dai_link->codec_name || dai_link->codec_of_node || dai_link->codec_dai_name) { -- 2.7.4

2 2

[alsa-devel] [PATCH 1/5] ASoC: fsl_spdif: don't print EPROBE_DEFER as error
by Stefan Agner 18 Jan '19

18 Jan '19

Probe deferral is to be expected during normal operation, so avoid printing an error when it is encountered. Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 740b90df44bb..a26686e7281c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1320,7 +1320,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); return ret; -- 2.20.1

4 15

[alsa-devel] [PATCH] ASoC: rt5682: add default pdata for i2s mode
by Bard liao 17 Jan '19

17 Jan '19

Add a default pdata which can fit most HW design. So we don't need to add a lot of DMI checking in this driver. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/rt5682.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 89c43b26c379..bd45d0343913 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,6 +43,12 @@ static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { "VBAT", }; +static const struct rt5682_platform_data i2s_default_platform_data = { + .dmic1_data_pin = RT5682_DMIC1_DATA_GPIO2, + .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, + .jd_src = RT5682_JD1, +}; + struct rt5682_priv { struct snd_soc_component *component; struct rt5682_platform_data pdata; @@ -2534,6 +2540,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5682); + rt5682->pdata = i2s_default_platform_data; + if (pdata) rt5682->pdata = *pdata; else -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: qcom: Kconfig: select max98927 for sdm845
by Cheng-Yi Chiang 16 Jan '19

16 Jan '19

Select SND_SOC_MAX98927 for SND_SOC_SDM845. Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- This is part of the patch https://patchwork.kernel.org/patch/10696507/ RT5663 part was added in the patch e9dc919043c5 ("ASoC: sdm845: add rt5663 codec select") This patch adds max98927. Thanks! sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 804ae0d93058..7948e993adba 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -102,6 +102,7 @@ config SND_SOC_SDM845 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 + select SND_SOC_MAX98927 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.20.1.97.g81188d93c3-goog

2 1

[alsa-devel] [PATCH] ASoC: sdm845: Set DAI format for dmic codec
by Cheng-Yi Chiang 16 Jan '19

16 Jan '19

Set codec DAI format for dmic codec in startup. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 6f66a58e23ca..882f52ed8231 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -248,12 +248,14 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) break; case SECONDARY_MI2S_TX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; if (++(data->sec_mi2s_clk_count) == 1) { snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT, MI2S_BCLK_RATE, SNDRV_PCM_STREAM_CAPTURE); } snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); break; case QUATERNARY_TDM_RX_0: -- 2.20.1.97.g81188d93c3-goog

2 1

[alsa-devel] [PATCH v2 0/4] Add NXP AUDMIX device and machine drivers
by Viorel Suman 16 Jan '19

16 Jan '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V1: 1. Original patch split into distinct patches for the device driver and DT binding documentation. 2. Replaced AMIX with AUDMIX in both code and file names as it looks more RM-compliant. 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. 4. Added machine driver and related DT binding documentation. Viorel Suman (4): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver ASoC: add imx-audmix DT binding documentation .../devicetree/bindings/sound/fsl,audmix.txt | 44 ++ .../devicetree/bindings/sound/imx-audmix.txt | 24 + sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 551 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 101 ++++ sound/soc/fsl/imx-audmix.c | 328 ++++++++++++ 7 files changed, 1069 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audmix.txt create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c -- 2.7.4

4 9

[alsa-devel] [PATCH] ASoC: core: Don't defer probe on optional, NULL components
by Matthias Reichl 15 Jan '19

15 Jan '19

cpu and platform are optional components in DAI links. For example codec-codec links usually have no platform set. Call snd_soc_find_component only if the name or of_node of a cpu or platform is set. Otherwise it will return NULL and soc_init_dai_link bails out immediately with -EPROBE_DEFER, meaning registering a card with NULL cpu or platform in DAI links can never succeed. Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/soc/soc-core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b680c673c553..aae450ba4f08 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1135,7 +1135,8 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Defer card registartion if platform dai component is not added to * component list. */ - if (!soc_find_component(link->platform->of_node, link->platform->name)) + if ((link->platform->of_node || link->platform->name) && + !soc_find_component(link->platform->of_node, link->platform->name)) return -EPROBE_DEFER; /* @@ -1154,7 +1155,8 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Defer card registartion if cpu dai component is not added to * component list. */ - if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + if ((link->cpu_of_node || link->cpu_name) && + !soc_find_component(link->cpu_of_node, link->cpu_name)) return -EPROBE_DEFER; /* -- 2.19.2

2 1

[alsa-devel] [PATCH] ASoC: rt5682: Fix PLL source register definitions
by shumingf＠realtek.com 15 Jan '19

15 Jan '19

From: Shuming Fan <shumingf(a)realtek.com> Fix typo which causes headphone no sound while using BCLK as PLL source. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.h | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d82a830..96944cf 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -849,18 +849,18 @@ #define RT5682_SCLK_SRC_PLL2 (0x2 << 13) #define RT5682_SCLK_SRC_SDW (0x3 << 13) #define RT5682_SCLK_SRC_RCCLK (0x4 << 13) -#define RT5682_PLL1_SRC_MASK (0x3 << 10) -#define RT5682_PLL1_SRC_SFT 10 -#define RT5682_PLL1_SRC_MCLK (0x0 << 10) -#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) -#define RT5682_PLL1_SRC_SDW (0x2 << 10) -#define RT5682_PLL1_SRC_RC (0x3 << 10) -#define RT5682_PLL2_SRC_MASK (0x3 << 8) -#define RT5682_PLL2_SRC_SFT 8 -#define RT5682_PLL2_SRC_MCLK (0x0 << 8) -#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) -#define RT5682_PLL2_SRC_SDW (0x2 << 8) -#define RT5682_PLL2_SRC_RC (0x3 << 8) +#define RT5682_PLL2_SRC_MASK (0x3 << 10) +#define RT5682_PLL2_SRC_SFT 10 +#define RT5682_PLL2_SRC_MCLK (0x0 << 10) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL2_SRC_SDW (0x2 << 10) +#define RT5682_PLL2_SRC_RC (0x3 << 10) +#define RT5682_PLL1_SRC_MASK (0x3 << 8) +#define RT5682_PLL1_SRC_SFT 8 +#define RT5682_PLL1_SRC_MCLK (0x0 << 8) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL1_SRC_SDW (0x2 << 8) +#define RT5682_PLL1_SRC_RC (0x3 << 8) -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: dapm: change snprintf to scnprintf for possible overflow" to the asoc tree
by Mark Brown 15 Jan '19

15 Jan '19

The patch ASoC: dapm: change snprintf to scnprintf for possible overflow 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 e581e151e965bf1f2815dd94620b638fec4d0a7e Mon Sep 17 00:00:00 2001 From: Silvio Cesare <silvio.cesare(a)gmail.com> Date: Sat, 12 Jan 2019 16:28:43 +0100 Subject: [PATCH] ASoC: dapm: change snprintf to scnprintf for possible overflow Change snprintf to scnprintf. There are generally two cases where using snprintf causes problems. 1) Uses of size += snprintf(buf, SIZE - size, fmt, ...) In this case, if snprintf would have written more characters than what the buffer size (SIZE) is, then size will end up larger than SIZE. In later uses of snprintf, SIZE - size will result in a negative number, leading to problems. Note that size might already be too large by using size = snprintf before the code reaches a case of size += snprintf. 2) If size is ultimately used as a length parameter for a copy back to user space, then it will potentially allow for a buffer overflow and information disclosure when size is greater than SIZE. When the size is used to index the buffer directly, we can have memory corruption. This also means when size = snprintf... is used, it may also cause problems since size may become large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel configuration. The solution to these issues is to use scnprintf which returns the number of characters actually written to the buffer, so the size variable will never exceed SIZE. Signed-off-by: Silvio Cesare <silvio.cesare(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: Kees Cook <keescook(a)chromium.org> Cc: Will Deacon <will.deacon(a)arm.com> Cc: Greg KH <greg(a)kroah.com> Signed-off-by: Willy Tarreau <w(a)1wt.eu> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..2c4c13419539 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2019,19 +2019,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -2044,7 +2044,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: rt5514-spi: Fix potential NULL pointer dereference
by Gustavo A. R. Silva 15 Jan '19

15 Jan '19

There is a potential NULL pointer dereference in case devm_kzalloc() fails and returns NULL. Fix this by adding a NULL check on rt5514_dsp. This issue was detected with the help of Coccinelle. Fixes: 6eebf35b0e4a ("ASoC: rt5514: add rt5514 SPI driver") Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/rt5514-spi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 4d46f4567c3a..bec2eefa8b0f 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -280,6 +280,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp = devm_kzalloc(component->dev, sizeof(*rt5514_dsp), GFP_KERNEL); + if (!rt5514_dsp) + return -ENOMEM; rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); -- 2.20.1

2 1

[alsa-devel] [PATCH v2 4/8] ASoC: imx-audmux: change snprintf to scnprintf for possible overflow
by Willy Tarreau 15 Jan '19

15 Jan '19

From: Silvio Cesare <silvio.cesare(a)gmail.com> Change snprintf to scnprintf. There are generally two cases where using snprintf causes problems. 1) Uses of size += snprintf(buf, SIZE - size, fmt, ...) In this case, if snprintf would have written more characters than what the buffer size (SIZE) is, then size will end up larger than SIZE. In later uses of snprintf, SIZE - size will result in a negative number, leading to problems. Note that size might already be too large by using size = snprintf before the code reaches a case of size += snprintf. 2) If size is ultimately used as a length parameter for a copy back to user space, then it will potentially allow for a buffer overflow and information disclosure when size is greater than SIZE. When the size is used to index the buffer directly, we can have memory corruption. This also means when size = snprintf... is used, it may also cause problems since size may become large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel configuration. The solution to these issues is to use scnprintf which returns the number of characters actually written to the buffer, so the size variable will never exceed SIZE. Signed-off-by: Silvio Cesare <silvio.cesare(a)gmail.com> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: Kees Cook <keescook(a)chromium.org> Cc: Will Deacon <will.deacon(a)arm.com> Cc: Greg KH <greg(a)kroah.com> Signed-off-by: Willy Tarreau <w(a)1wt.eu> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Reviewed-by: Kees Cook <keescook(a)chromium.org> -- v2: - adjust subject line - added alsa-devel & Mark - added acked-by - added reviewed-by The patches in this series were sent by Silvio to the security list to fix a minor memory disclosure issue after this article was published: http://blog.infosectcbr.com.au/2018/11/memory-bugs-in-multiple-linux-kernel… sound/soc/fsl/imx-audmux.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 392d5eef356d..99e07b01a2ce 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); -- 2.19.2

3 2

[alsa-devel] [PATCH] ASoC: amd: Fix potential NULL pointer dereference
by Gustavo A. R. Silva 15 Jan '19

15 Jan '19

Check return value from call to devm_kzalloc() in order to prevent a potential NULL pointer dereference. Also, notice that it makes no sense to allocate any resources if res = platform_get_resource(pdev, IORESOURCE_MEM, 0); fails, so move the call to devm_kzalloc() below the mentioned code. Lastly, improve the use of sizeof in the call to devm_kzalloc() by changing it from sizeof(struct i2s_dev_data) to sizeof(*adata) This issue was detected with the help of Coccinelle. Fixes: ac289c7ec0bc ("ASoC: amd: add ACP3x PCM platform driver") Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 022a8912c8a2..3d58338fa3cf 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -611,14 +611,16 @@ static int acp3x_audio_probe(struct platform_device *pdev) } irqflags = *((unsigned int *)(pdev->dev.platform_data)); - adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), - GFP_KERNEL); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); return -ENODEV; } + adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); + if (!adata) + return -ENOMEM; + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); -- 2.20.1

2 1

[alsa-devel] [PATCH v1 0/3] ASoC: last minute fixes
by Marcel Ziswiler 15 Jan '19

15 Jan '19

This series comes with some last minutes fixes and clean-up. Changes in v1: - Split from the Tegra series as suggested by Mark. - Fix issue in soc-core rather than working around it in tegra_sgtl5000. Marcel Ziswiler (3): ASoC: tegra_sgtl5000: fix device_node refcounting ASoC: soc-core: fix trivial checkpatch issues ASoC: soc-core: fix platform name vs. of_node assignement sound/soc/soc-core.c | 162 ++++++++++++++++++++++++--------------- sound/soc/tegra/tegra_sgtl5000.c | 17 +++- 2 files changed, 116 insertions(+), 63 deletions(-) -- 2.14.4

5 30

[alsa-devel] next/pending-fixes boot bisection: v5.0-rc2-361-g08b435ee366b on bcm2837-rpi-3-b-32
by kernelci.org bot 15 Jan '19

15 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc2-361-g08b435ee366b on bcm2837-rpi-3-b-32 Summary: Start: 08b435ee366b Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3d7a2759b51458916edd4d Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc2-361-g08b435ee366b… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc2-361-g08b435ee366b… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: bcm2837-rpi-3-b-32 CPU arch: arm Lab: lab-baylibre Config: bcm2835_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [3719876809e745b9db5293d418600c194bbf5c23] sbitmap: Protect swap_lock from softirqs git bisect good 3719876809e745b9db5293d418600c194bbf5c23 # bad: [08b435ee366bab63250c375ffea3c94cb02fcd93] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 08b435ee366bab63250c375ffea3c94cb02fcd93 # good: [f80eb2d45c4e34750337d1210a78bfe2ebf43b73] Merge remote-tracking branch 'ipsec/master' git bisect good f80eb2d45c4e34750337d1210a78bfe2ebf43b73 # bad: [e242412b491b4e34643f82c6f422e964beb81704] Merge remote-tracking branch 'input-current/for-linus' git bisect bad e242412b491b4e34643f82c6f422e964beb81704 # bad: [eb1c2383c8b929fcc5a56f5698ff3fe3d0ec54e6] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad eb1c2383c8b929fcc5a56f5698ff3fe3d0ec54e6 # bad: [04eb1efcd614d6f067b76a355b3a3599667959dc] ASoC: soc-core: Hold client_mutex around soc_init_dai_link() git bisect bad 04eb1efcd614d6f067b76a355b3a3599667959dc # good: [a41d9dbf5dac5b6a1283ee8001f22807d18352ea] ASoC: qdsp6: q6asm-dai: Fix a small memory leak git bisect good a41d9dbf5dac5b6a1283ee8001f22807d18352ea # good: [6175471755075d256c1c654151fc1cad183c1e33] ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks git bisect good 6175471755075d256c1c654151fc1cad183c1e33 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/pending-fixes boot bisection: v5.0-rc2-361-g08b435ee366b on sun8i-h2-plus-libretech-all-h3-cc
by kernelci.org bot 15 Jan '19

15 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc2-361-g08b435ee366b on sun8i-h2-plus-libretech-all-h3-cc Summary: Start: 08b435ee366b Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3d795459b514563e6edd38 Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc2-361-g08b435ee366b… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc2-361-g08b435ee366b… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: sun8i-h2-plus-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [3719876809e745b9db5293d418600c194bbf5c23] sbitmap: Protect swap_lock from softirqs git bisect good 3719876809e745b9db5293d418600c194bbf5c23 # bad: [08b435ee366bab63250c375ffea3c94cb02fcd93] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 08b435ee366bab63250c375ffea3c94cb02fcd93 # good: [f80eb2d45c4e34750337d1210a78bfe2ebf43b73] Merge remote-tracking branch 'ipsec/master' git bisect good f80eb2d45c4e34750337d1210a78bfe2ebf43b73 # bad: [e242412b491b4e34643f82c6f422e964beb81704] Merge remote-tracking branch 'input-current/for-linus' git bisect bad e242412b491b4e34643f82c6f422e964beb81704 # bad: [eb1c2383c8b929fcc5a56f5698ff3fe3d0ec54e6] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad eb1c2383c8b929fcc5a56f5698ff3fe3d0ec54e6 # bad: [04eb1efcd614d6f067b76a355b3a3599667959dc] ASoC: soc-core: Hold client_mutex around soc_init_dai_link() git bisect bad 04eb1efcd614d6f067b76a355b3a3599667959dc # good: [a41d9dbf5dac5b6a1283ee8001f22807d18352ea] ASoC: qdsp6: q6asm-dai: Fix a small memory leak git bisect good a41d9dbf5dac5b6a1283ee8001f22807d18352ea # good: [6175471755075d256c1c654151fc1cad183c1e33] ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks git bisect good 6175471755075d256c1c654151fc1cad183c1e33 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/master boot bisection: next-20190115 on sun8i-h3-libretech-all-h3-cc
by kernelci.org bot 15 Jan '19

15 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/master boot bisection: next-20190115 on sun8i-h3-libretech-all-h3-cc Summary: Start: 5aa6589e17bd Add linux-next specific files for 20190115 Details: https://kernelci.org/boot/id/5c3d77f659b51450786edd44 Plain log: https://storage.kernelci.org//next/master/next-20190115/arm/sunxi_defconfig… HTML log: https://storage.kernelci.org//next/master/next-20190115/arm/sunxi_defconfig… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: master Target: sun8i-h3-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [3719876809e745b9db5293d418600c194bbf5c23] sbitmap: Protect swap_lock from softirqs git bisect good 3719876809e745b9db5293d418600c194bbf5c23 # bad: [5aa6589e17bd4dbc1856e37281f00a290d733ab0] Add linux-next specific files for 20190115 git bisect bad 5aa6589e17bd4dbc1856e37281f00a290d733ab0 # bad: [f2dd4e3054741e22c4c97c8a0fa05faa7239d60a] Merge remote-tracking branch 'crypto/master' git bisect bad f2dd4e3054741e22c4c97c8a0fa05faa7239d60a # bad: [3ba858cbc004b32f8767d295b9b566eb0381da0c] Merge remote-tracking branch 'qcom/for-next' git bisect bad 3ba858cbc004b32f8767d295b9b566eb0381da0c # bad: [4cc2339ea8900debf88333934fa9b1dbc96ee2aa] Merge remote-tracking branch 'btrfs-fixes/next-fixes' git bisect bad 4cc2339ea8900debf88333934fa9b1dbc96ee2aa # good: [259764fd4e5b47aedd6317d2edf5d813333909c5] Merge remote-tracking branch 'net/master' git bisect good 259764fd4e5b47aedd6317d2edf5d813333909c5 # bad: [29568b4049ff4ef23ce0a2dd46cbaed94b589c78] Merge remote-tracking branch 'usb-serial-fixes/usb-linus' git bisect bad 29568b4049ff4ef23ce0a2dd46cbaed94b589c78 # good: [798729dc036d5bb384e24bb48af5c9ea6554d945] Merge remote-tracking branch 'sound-current/for-linus' git bisect good 798729dc036d5bb384e24bb48af5c9ea6554d945 # bad: [d9c51542207a29b7b06c11f1085a1e3b8eb24f07] Merge branch 'asoc-5.0' into asoc-linus git bisect bad d9c51542207a29b7b06c11f1085a1e3b8eb24f07 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/pending-fixes boot bisection: v5.0-rc2-361-g08b435ee366b on sun8i-h3-libretech-all-h3-cc
by kernelci.org bot 15 Jan '19

15 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc2-361-g08b435ee366b on sun8i-h3-libretech-all-h3-cc Summary: Start: 08b435ee366b Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3d795559b51455d36edd69 Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc2-361-g08b435ee366b… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc2-361-g08b435ee366b… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: sun8i-h3-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [3719876809e745b9db5293d418600c194bbf5c23] sbitmap: Protect swap_lock from softirqs git bisect good 3719876809e745b9db5293d418600c194bbf5c23 # bad: [08b435ee366bab63250c375ffea3c94cb02fcd93] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 08b435ee366bab63250c375ffea3c94cb02fcd93 # good: [f80eb2d45c4e34750337d1210a78bfe2ebf43b73] Merge remote-tracking branch 'ipsec/master' git bisect good f80eb2d45c4e34750337d1210a78bfe2ebf43b73 # bad: [e242412b491b4e34643f82c6f422e964beb81704] Merge remote-tracking branch 'input-current/for-linus' git bisect bad e242412b491b4e34643f82c6f422e964beb81704 # bad: [eb1c2383c8b929fcc5a56f5698ff3fe3d0ec54e6] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad eb1c2383c8b929fcc5a56f5698ff3fe3d0ec54e6 # bad: [04eb1efcd614d6f067b76a355b3a3599667959dc] ASoC: soc-core: Hold client_mutex around soc_init_dai_link() git bisect bad 04eb1efcd614d6f067b76a355b3a3599667959dc # good: [a41d9dbf5dac5b6a1283ee8001f22807d18352ea] ASoC: qdsp6: q6asm-dai: Fix a small memory leak git bisect good a41d9dbf5dac5b6a1283ee8001f22807d18352ea # good: [6175471755075d256c1c654151fc1cad183c1e33] ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks git bisect good 6175471755075d256c1c654151fc1cad183c1e33 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/master boot bisection: next-20190115 on sun8i-h2-plus-libretech-all-h3-cc
by kernelci.org bot 15 Jan '19

15 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/master boot bisection: next-20190115 on sun8i-h2-plus-libretech-all-h3-cc Summary: Start: 5aa6589e17bd Add linux-next specific files for 20190115 Details: https://kernelci.org/boot/id/5c3d783659b51451106edd6c Plain log: https://storage.kernelci.org//next/master/next-20190115/arm/sunxi_defconfig… HTML log: https://storage.kernelci.org//next/master/next-20190115/arm/sunxi_defconfig… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: master Target: sun8i-h2-plus-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [3719876809e745b9db5293d418600c194bbf5c23] sbitmap: Protect swap_lock from softirqs git bisect good 3719876809e745b9db5293d418600c194bbf5c23 # bad: [5aa6589e17bd4dbc1856e37281f00a290d733ab0] Add linux-next specific files for 20190115 git bisect bad 5aa6589e17bd4dbc1856e37281f00a290d733ab0 # bad: [f2dd4e3054741e22c4c97c8a0fa05faa7239d60a] Merge remote-tracking branch 'crypto/master' git bisect bad f2dd4e3054741e22c4c97c8a0fa05faa7239d60a # bad: [3ba858cbc004b32f8767d295b9b566eb0381da0c] Merge remote-tracking branch 'qcom/for-next' git bisect bad 3ba858cbc004b32f8767d295b9b566eb0381da0c # bad: [4cc2339ea8900debf88333934fa9b1dbc96ee2aa] Merge remote-tracking branch 'btrfs-fixes/next-fixes' git bisect bad 4cc2339ea8900debf88333934fa9b1dbc96ee2aa # good: [259764fd4e5b47aedd6317d2edf5d813333909c5] Merge remote-tracking branch 'net/master' git bisect good 259764fd4e5b47aedd6317d2edf5d813333909c5 # bad: [29568b4049ff4ef23ce0a2dd46cbaed94b589c78] Merge remote-tracking branch 'usb-serial-fixes/usb-linus' git bisect bad 29568b4049ff4ef23ce0a2dd46cbaed94b589c78 # good: [798729dc036d5bb384e24bb48af5c9ea6554d945] Merge remote-tracking branch 'sound-current/for-linus' git bisect good 798729dc036d5bb384e24bb48af5c9ea6554d945 # bad: [d9c51542207a29b7b06c11f1085a1e3b8eb24f07] Merge branch 'asoc-5.0' into asoc-linus git bisect bad d9c51542207a29b7b06c11f1085a1e3b8eb24f07 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] [PATCH] ASoC: core: Make snd_soc_find_component() more robust
by Mark Brown 15 Jan '19

15 Jan '19

There are some use cases where you're checking for a lot of things on a card and it makes sense that you might end up trying to call snd_soc_find_component() without either a name or an of_node. Currently in that case we try to dereference the name and crash but it's more useful to allow the caller to just treat that as a case where we don't find anything, that error handling will already exist. Inspired by a patch from Ajit Pandey fixing some callers. Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cdcc417c94ca..b680c673c553 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -742,7 +742,7 @@ static struct snd_soc_component *soc_find_component( if (of_node) { if (component->dev->of_node == of_node) return component; - } else if (strcmp(component->name, name) == 0) { + } else if (name && strcmp(component->name, name) == 0) { return component; } } -- 2.20.1

1 1

[alsa-devel] [PATCH 1/3] ASoC: cs4341: fix waring unused-function
by Anders Roxell 15 Jan '19

15 Jan '19

The driver cs4341 can be built with SPI and/or I2C, but it has to be one of them at least. When I2C is set as a module we see the warning below: sound/soc/codecs/cs4341.c:213:12: warning: ‘cs4341_probe’ defined but not used [-Wunused-function] static int cs4341_probe(struct device *dev) ^~~~~~~~~~~~ Rework so that we use IS_ENABLED instead of defined. Also change so SND_SOC_CS4341 depends on SND_SOC_I2C_AND_SPI to we dont' get a link error when SND_SOC_CS4341=y, I2C=m and REGMAP_I2C=m is set. Suggested-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/cs4341.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 87cb9c51e6f5..87ecb38e57f4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -546,7 +546,7 @@ config SND_SOC_CS43130 config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" - depends on I2C || SPI_MASTER + depends on SND_SOC_I2C_AND_SPI select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index d2e616a89fd4..ade7477d04f1 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -223,7 +223,7 @@ static int cs4341_probe(struct device *dev) &cs4341_dai, 1); } -#if defined(CONFIG_I2C) +#if IS_ENABLED(CONFIG_I2C) static int cs4341_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -317,7 +317,7 @@ static int __init cs4341_init(void) { int ret = 0; -#if defined(CONFIG_I2C) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&cs4341_i2c_driver); if (ret) return ret; @@ -332,7 +332,7 @@ module_init(cs4341_init); static void __exit cs4341_exit(void) { -#if defined(CONFIG_I2C) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&cs4341_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- 2.19.2

2 4

[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: remove few unused variables
by Srinivas Kandagatla 15 Jan '19

15 Jan '19

This patch removes unused variables which also fixes below warnings: msm8916-wcd-digital.c:245:30: warning: 'rx2_mix2_inp1_chain_enum' defined but not used [-Wunused-const-variable=] static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( ^~~~~~~~~~~~~~~~~~~~~~~~ msm8916-wcd-digital.c:234:30: warning: 'rx_mix2_inp1_chain_enum' defined but not used [-Wunused-const-variable=] static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( ^~~~~~~~~~~~~~~~~~~~~~~ msm8916-wcd-digital.c:224:26: warning: 'adc2_mux_text' defined but not used [-Wunused-const-variable=] static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; ^~~~~~~~~~~~~ msm8916-wcd-digital.c:223:26: warning: 'rx_mix2_text' defined but not used [-Wunused-const-variable=] Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-digital.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 3063dedd21cf..423bfebabed4 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -220,8 +220,6 @@ static const char *const dec_mux_text[] = { }; static const char *const cic_mux_text[] = { "AMIC", "DMIC" }; -static const char *const rx_mix2_text[] = { "ZERO", "IIR1", "IIR2" }; -static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; /* RX1 MIX1 */ static const struct soc_enum rx_mix1_inp_enum[] = { @@ -230,10 +228,6 @@ static const struct soc_enum rx_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B2_CTL, 0, 6, rx_mix1_text), }; -/* RX1 MIX2 */ -static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( - LPASS_CDC_CONN_RX1_B3_CTL, 0, 3, rx_mix2_text); - /* RX2 MIX1 */ static const struct soc_enum rx2_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B1_CTL, 0, 6, rx_mix1_text), @@ -241,10 +235,6 @@ static const struct soc_enum rx2_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B2_CTL, 0, 6, rx_mix1_text), }; -/* RX2 MIX2 */ -static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE( - LPASS_CDC_CONN_RX2_B3_CTL, 0, 3, rx_mix2_text); - /* RX3 MIX1 */ static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B1_CTL, 0, 6, rx_mix1_text), -- 2.20.1

2 1

[alsa-devel] [PATCH v2] ASoC: msm8916-wcd-digital: Select REGMAP_MMIO to fix build error
by Axel Lin 15 Jan '19

15 Jan '19

Fix below build error: ERROR: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-msm8916-digital.ko] undefined! Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: select REGMAP_MMIO is for SND_SOC_MSM8916_WCD_DIGITAL, not SND_SOC_MSM8916_WCD_ANALOG. sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 87cb9c51e6f5..17049b9bc178 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -706,6 +706,7 @@ config SND_SOC_MSM8916_WCD_ANALOG config SND_SOC_MSM8916_WCD_DIGITAL tristate "Qualcomm MSM8916 WCD DIGITAL Codec" + select REGMAP_MMIO config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: xlnx: parse AES audio parameters
by Maruthi Srinivas Bayyavarapu 15 Jan '19

15 Jan '19

AES channel status carries various audio parameters. If channel status is detected, current patch extracts sample rate and bit depth parameters of the incoming stream during capture. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 143 ++++++++++++++++++++++++++++++++++ 1 file changed, 143 insertions(+) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index d2194da..97177d3 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -13,6 +13,7 @@ #include <linux/of_irq.h> #include <linux/sizes.h> +#include <sound/asoundef.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -63,6 +64,7 @@ #define PERIODS_MAX 6 #define PERIOD_BYTES_MIN 192 #define PERIOD_BYTES_MAX (50 * 1024) +#define XLNX_PARAM_UNKNOWN 0 enum bit_depth { BIT_DEPTH_8, @@ -117,6 +119,129 @@ static const struct snd_pcm_hardware xlnx_pcm_hardware = { .periods_max = PERIODS_MAX, }; +enum { + AES_TO_AES, + AES_TO_PCM, + PCM_TO_PCM, + PCM_TO_AES +}; + +static void xlnx_parse_aes_params(u32 chsts_reg1_val, u32 chsts_reg2_val, + struct device *dev) +{ + u32 padded, srate, bit_depth, status[2]; + + if (chsts_reg1_val & IEC958_AES0_PROFESSIONAL) { + status[0] = chsts_reg1_val & 0xff; + status[1] = (chsts_reg1_val >> 16) & 0xff; + + switch (status[0] & IEC958_AES0_PRO_FS) { + case IEC958_AES0_PRO_FS_44100: + srate = 44100; + break; + case IEC958_AES0_PRO_FS_48000: + srate = 48000; + break; + case IEC958_AES0_PRO_FS_32000: + srate = 32000; + break; + case IEC958_AES0_PRO_FS_NOTID: + default: + srate = XLNX_PARAM_UNKNOWN; + break; + } + + switch (status[1] & IEC958_AES2_PRO_SBITS) { + case IEC958_AES2_PRO_WORDLEN_NOTID: + case IEC958_AES2_PRO_SBITS_20: + padded = 0; + break; + case IEC958_AES2_PRO_SBITS_24: + padded = 4; + break; + default: + bit_depth = XLNX_PARAM_UNKNOWN; + goto log_params; + } + + switch (status[1] & IEC958_AES2_PRO_WORDLEN) { + case IEC958_AES2_PRO_WORDLEN_20_16: + bit_depth = 16 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_22_18: + bit_depth = 18 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_23_19: + bit_depth = 19 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_24_20: + bit_depth = 20 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_NOTID: + default: + bit_depth = XLNX_PARAM_UNKNOWN; + break; + } + + } else { + status[0] = (chsts_reg1_val >> 24) & 0xff; + status[1] = chsts_reg2_val & 0xff; + + switch (status[0] & IEC958_AES3_CON_FS) { + case IEC958_AES3_CON_FS_44100: + srate = 44100; + break; + case IEC958_AES3_CON_FS_48000: + srate = 48000; + break; + case IEC958_AES3_CON_FS_32000: + srate = 32000; + break; + default: + srate = XLNX_PARAM_UNKNOWN; + break; + } + + if (status[1] & IEC958_AES4_CON_MAX_WORDLEN_24) + padded = 4; + else + padded = 0; + + switch (status[1] & IEC958_AES4_CON_WORDLEN) { + case IEC958_AES4_CON_WORDLEN_20_16: + bit_depth = 16 + padded; + break; + case IEC958_AES4_CON_WORDLEN_22_18: + bit_depth = 18 + padded; + break; + case IEC958_AES4_CON_WORDLEN_23_19: + bit_depth = 19 + padded; + break; + case IEC958_AES4_CON_WORDLEN_24_20: + bit_depth = 20 + padded; + break; + case IEC958_AES4_CON_WORDLEN_21_17: + bit_depth = 17 + padded; + break; + case IEC958_AES4_CON_WORDLEN_NOTID: + default: + bit_depth = XLNX_PARAM_UNKNOWN; + break; + } + } + +log_params: + if (srate != XLNX_PARAM_UNKNOWN) + dev_info(dev, "sample rate = %d\n", srate); + else + dev_info(dev, "sample rate = unknown\n"); + + if (bit_depth != XLNX_PARAM_UNKNOWN) + dev_info(dev, "bit_depth = %d\n", bit_depth); + else + dev_info(dev, "bit_depth = unknown\n"); +} + static int xlnx_formatter_pcm_reset(void __iomem *mmio_base) { u32 val, retries = 0; @@ -302,8 +427,12 @@ static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { u32 low, high, active_ch, val, bytes_per_ch, bits_per_sample; + u32 aes_reg1_val, aes_reg2_val; int status; u64 size; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); struct snd_pcm_runtime *runtime = substream->runtime; struct xlnx_pcm_stream_param *stream_data = runtime->private_data; @@ -311,6 +440,20 @@ static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, if (active_ch > stream_data->ch_limit) return -EINVAL; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + stream_data->xfer_mode == AES_TO_PCM) { + val = readl(stream_data->mmio + XLNX_AUD_STS); + if (val & AUD_STS_CH_STS_MASK) { + aes_reg1_val = readl(stream_data->mmio + + XLNX_AUD_CH_STS_START); + aes_reg2_val = readl(stream_data->mmio + + XLNX_AUD_CH_STS_START + 0x4); + + xlnx_parse_aes_params(aes_reg1_val, aes_reg2_val, + component->dev); + } + } + size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: soc-core: fix init platform memory handling" to the asoc tree
by Mark Brown 15 Jan '19

15 Jan '19

The patch ASoC: soc-core: fix init platform memory handling 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 09ac6a817bd687e7f5dac00470262efdd72f9319 Mon Sep 17 00:00:00 2001 From: Curtis Malainey <cujomalainey(a)chromium.org> Date: Thu, 10 Jan 2019 16:21:04 -0800 Subject: [PATCH] ASoC: soc-core: fix init platform memory handling snd_soc_init_platform initializes pointers to snd_soc_dai_link which is statically allocated and it does this by devm_kzalloc. In the event of an EPROBE_DEFER the memory will be freed and the pointers are left dangling. snd_soc_init_platform sees the dangling pointers and assumes they are pointing to initialized memory and does not reallocate them on the second probe attempt which results in a use after free bug since devm has freed the memory from the first probe attempt. Since the intention for snd_soc_dai_link->platform is that it can be set statically by the machine driver we need to respect the pointer in the event we did not set it but still catch dangling pointers. The solution is to add a flag to track whether the pointer was dynamically allocated or not. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 11 ++++++----- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7..e665f111b0d2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -985,6 +985,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* + * This driver uses legacy platform naming. Set by the core, machine + * drivers should not modify this value. + */ + unsigned int legacy_platform:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0934b36645b3..cdcc417c94ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { + if (!platform || dai_link->legacy_platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); if (!platform) return -ENOMEM; - dai_link->platform = platform; - platform->name = dai_link->platform_name; - platform->of_node = dai_link->platform_of_node; - platform->dai_name = NULL; + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; } /* if there's no platform we match on the empty platform */ -- 2.20.1

1 0

[alsa-devel] [PATCH] make huawei matebook X right speaker work
by Tomas Espeleta 14 Jan '19

14 Jan '19

Hard coded coefficients to make Huawuei Matebook X right speaker work. The Matebook X has a ALC298, please refer to bug 197801 on how these numbers were reverse engineered from the Windows driver The reversed engineered sequence represents a repeating pattern of verbs, and the only values that are changing periodically are written on indexes 0x23 and 0x25: 0x500, 0x23 0x400, VALUE1 0x500, 0x25 0x400, VALUE2 * fixes bug 197801 * skipped reading sequences (0x500 - 0xc00 sequences are ignored) * static values from reverse engineering are used Signed-off-by: Tomas Espeleta <tomas.espeleta(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 70 +++++++++++++++++++++++++++++++++-- 1 file changed, 67 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b3e7a18ca78..ec3972d9e30d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3628,6 +3628,64 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec, vref); } +struct hda_alc298_mbxinit { + unsigned char value_0x23; + unsigned char value_0x25; +}; + +void alc298_stereo_seq(struct hda_codec *codec, + struct hda_alc298_mbxinit *initval, int first) +{ + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x0); + alc_write_coef_idx(codec, 0x26, 0xb000); + + if (first) + snd_hda_codec_write(codec, 0x21, 0, 0xf09, 0x0); + + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x23, initval->value_0x23); + + if (initval->value_0x23 != 0x1e) + alc_write_coef_idx(codec, 0x25, initval->value_0x25); + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, 0x4b0, 0x10); +} + +void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + // Initialization magic + static struct hda_alc298_mbxinit dac_init[] = { + {0xc, 0x0}, {0xd, 0x0}, {0xe, 0x0}, {0xf, 0x0}, + {0x10, 0x0}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x0}, + {0x1d, 0x0}, {0x1e, 0x0}, {0x1f, 0x0}, + {0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24}, + {0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f}, + {0x2b, 0x2}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x0}, + {0x2f, 0x0}, + {0x30, 0x0}, {0x31, 0x0}, {0x32, 0x0}, {0x33, 0x0}, + {0x34, 0x0}, {0x35, 0x1}, {0x36, 0x93}, {0x37, 0xc}, + {0x38, 0x0}, {0x39, 0x0}, {0x3a, 0xf8}, {0x38, 0x80}, + {} + }; + struct hda_alc298_mbxinit *seq = dac_init; + + // Start + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x0); + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x22, 0x31); + alc_write_coef_idx(codec, 0x23, 0xb); + alc_write_coef_idx(codec, 0x25, 0x0); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, 0x4b0, 0x10); + + for (; seq->value_0x23; seq++) + alc298_stereo_seq(codec, seq, seq == dac_init); +} + static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3638,7 +3696,6 @@ static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, } } - /* update mute-LED according to the speaker mute state via mic VREF pin */ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) { @@ -5582,6 +5639,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC298_FIXUP_HUAWEI_MBX_STEREO, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -5880,6 +5938,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, + [ALC298_FIXUP_HUAWEI_MBX_STEREO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_huawei_mbx_stereo, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6778,8 +6842,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), - SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), - SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC298_FIXUP_HUAWEI_MBX_STEREO), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC298_FIXUP_HUAWEI_MBX_STEREO), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.20.1

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[alsa-devel] Fwd: [Alsa-user] AlsaMixer v1.1.8 Intel HDA -8 master volume
by makepost＠firemail.cc 14 Jan '19

14 Jan '19

Hello alsa-devel, forwarding from the silent alsa-user, with clarification. Found out this is specific for alsa-utils' alsamixer/volume_mapping.c, in which Clang 7.0.1 warns about exp10 implicit declaration. Broke after removal of _GNU_SOURCE in v1.1.7 apparently.[1] Uclibc-ng has exp10 since 1.0.12 according to its release notes, and exp10 is likely behind _GNU_SOURCE like other non-standard features. Should the exp10 fallback macro lower in the file have been removed instead?[2] [1] http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=6e90e4fe8e0052… [2] http://git.alsa-project.org/?p=alsa-utils.git;a=blob;f=alsamixer/volume_map… -------- Original Message -------- Subject: [Alsa-user] AlsaMixer v1.1.8 Intel HDA -8 master volume Date: 2019-01-11 21:48 From: makepost(a)firemail.cc To: alsa-user(a)lists.sourceforge.net Hi, after upgrading alsa-lib and alsa-utils to 1.1.8, Master, Headphones and Speaker in AlsaMixer are stuck at -8 percentage. Adjusting attempts immediately set max volume, still showing -8. Muting and unmuting work fine. PCM control is not affected. Realtek ALC283, vanilla 4.20.0 kernel. Attaching codec info and amixer output. What other info should I post for debug? _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

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[alsa-devel] [PATCH v3 0/5] Stripe control functionality
by Sameer Pujar 14 Jan '19

14 Jan '19

Background ========== Azalia Controller fetches command and audio data via DMA and send them to codec through SDO(Serial Data Out) lines. SDO is for outboung and it broadcasts to all codecs. There is atleast one SDO line present, but there can be multiple SDO lines supported for extended bandwidth. This is essential when a platform supports multiple hdmi/dp sinks and there is a requirement for higher resolution audio playback. In such cases simultaneous audio playback data can be striped across multiple SDOs. Global Capabilities(GCAP) Register indicates the capabilities of the controller. Bits 2:1 of GCAP can be read to know the number of supported SDO lines (below is from HD audio spec) 00: 1 SDO 01: 2 SDO 10: 4 SDO 11: Reserved Stripe control verb reports and controls the stripe capability of an Audio Output Converter. This verb needs to be implemented only for an audio output converter and only if the stripe bit of the Audio Widget Capabilities parameter is 1. Stripe control: get code(0xf24) and set code(0x724) Change log ========== v1-->v2: -------- Patch 1: "ALSA: hda: add verbs for stripe control" * no change Patch 2: "ALSA: hda: Add api to program stripe control bits" * no change Patch 3: "ALSA: hda: add register offset for stripe control" * added mask for stripe control programming Patch 4: "ALSA: hda: program stripe bits for controller" * used stripe control mask instead of hard coded value Patch 5: "ALSA: hda: program stripe control for codec" * added conditional check to know if striping is supported. If supported, then only stripe verb is implemented. v2-->v3: -------- Patch 1: "ALSA: hda: add verbs for stripe control" * no change Patch 2: "ALSA: hda: Add api to program stripe control bits" * added braces around stripe value calucation as per spec Patch 3: "ALSA: hda: add register offset for stripe control" * no change Patch 4: "ALSA: hda: program stripe bits for controller" * no change Patch 5: "ALSA: hda: program stripe control for codec" * no change =========== Sameer Pujar (5): ALSA: hda: add verbs for stripe control ALSA: hda: Add api to program stripe control bits ALSA: hda: add register offset for stripe control ALSA: hda: program stripe bits for controller ALSA: hda: program stripe control for codec include/sound/hda_register.h | 2 ++ include/sound/hda_verbs.h | 2 ++ include/sound/hdaudio.h | 3 +++ sound/hda/hdac_stream.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 10 +++++++++- 5 files changed, 56 insertions(+), 1 deletion(-) -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[alsa-devel] [PATCH 1/3] drm/i915/audio: Track temporary rpm wakerefs
by Chris Wilson 14 Jan '19

14 Jan '19

Track the temporary rpm wakerefs used within audio so that they may be marked as complete and the tracking cancelled upon release. Signed-off-by: Chris Wilson <chris(a)chris-wilson.co.uk> --- drivers/gpu/drm/i915/intel_audio.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index de26cd0a5497..92e27359c2e3 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -756,12 +756,13 @@ static void i915_audio_component_codec_wake_override(struct device *kdev, bool enable) { struct drm_i915_private *dev_priv = kdev_to_i915(kdev); + intel_wakeref_t wakeref; u32 tmp; if (!IS_GEN(dev_priv, 9)) return; - i915_audio_component_get_power(kdev); + wakeref = intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); /* * Enable/disable generating the codec wake signal, overriding the @@ -779,7 +780,7 @@ static void i915_audio_component_codec_wake_override(struct device *kdev, usleep_range(1000, 1500); } - i915_audio_component_put_power(kdev); + intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO, wakeref); } /* Get CDCLK in kHz */ @@ -850,12 +851,13 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, int port, struct i915_audio_component *acomp = dev_priv->audio_component; struct intel_encoder *encoder; struct intel_crtc *crtc; + intel_wakeref_t wakeref; int err = 0; if (!HAS_DDI(dev_priv)) return 0; - i915_audio_component_get_power(kdev); + wakeref = intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); mutex_lock(&dev_priv->av_mutex); /* 1. get the pipe */ @@ -875,7 +877,7 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, int port, unlock: mutex_unlock(&dev_priv->av_mutex); - i915_audio_component_put_power(kdev); + intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO, wakeref); return err; } -- 2.20.1

3 10

[alsa-devel] [PATCH v2 0/5] Stripe control functionality
by Sameer Pujar 14 Jan '19

14 Jan '19

Background ========== Azalia Controller fetches command and audio data via DMA and send them to codec through SDO(Serial Data Out) lines. SDO is for outboung and it broadcasts to all codecs. There is atleast one SDO line present, but there can be multiple SDO lines supported for extended bandwidth. This is essential when a platform supports multiple hdmi/dp sinks and there is a requirement for higher resolution audio playback. In such cases simultaneous audio playback data can be striped across multiple SDOs. Global Capabilities(GCAP) Register indicates the capabilities of the controller. Bits 2:1 of GCAP can be read to know the number of supported SDO lines (below is from HD audio spec) 00: 1 SDO 01: 2 SDO 10: 4 SDO 11: Reserved Stripe control verb reports and controls the stripe capability of an Audio Output Converter. This verb needs to be implemented only for an audio output converter and only if the stripe bit of the Audio Widget Capabilities parameter is 1. Stripe control: get code(0xf24) and set code(0x724) Change log ========== v1-->v2: -------- Patch 1: "ALSA: hda: add verbs for stripe control" * no change Patch 2: "ALSA: hda: Add api to program stripe control bits" * no change Patch 3: "ALSA: hda: add register offset for stripe control" * added mask for stripe control programming Patch 4: "ALSA: hda: program stripe bits for controller" * used stripe control mask instead of hard coded value Patch 5: "ALSA: hda: program stripe control for codec" * added conditional check to know if striping is supported. If supported, then only stripe verb is implemented. =========== Sameer Pujar (5): ALSA: hda: add verbs for stripe control ALSA: hda: Add api to program stripe control bits ALSA: hda: add register offset for stripe control ALSA: hda: program stripe bits for controller ALSA: hda: program stripe control for codec include/sound/hda_register.h | 2 ++ include/sound/hda_verbs.h | 2 ++ include/sound/hdaudio.h | 3 +++ sound/hda/hdac_stream.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 10 +++++++++- 5 files changed, 56 insertions(+), 1 deletion(-) -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 11

[alsa-devel] [PATCH 0/4] slimbus: minor fixes.
by Srinivas Kandagatla 14 Jan '19

14 Jan '19

Here are some minor fixes to slimbus core for issues found while testing wcd9335. First one is make the slimbus core match full device id, second one is to avoid using big lock in status callback function, third one is to use status function rather than directly invoking the status ops. finally last one is to fix missing spinlock init. Thanks, srini Srinivas Kandagatla (4): slimbus: core: match full device id slimbus: core: do not call device_status under a lock slimbus: core: use slim_device_update_status() to update status slimbus: core: add missing spin_lock_init on txn_lock drivers/slimbus/core.c | 45 +++++++++++++++++++++++------------------- 1 file changed, 25 insertions(+), 20 deletions(-) -- 2.20.1

2 6

[alsa-devel] next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on bcm2837-rpi-3-b-32
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on bcm2837-rpi-3-b-32 Summary: Start: 727b373740d1 Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3c42c459b5145f216edd41 Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: bcm2837-rpi-3-b-32 CPU arch: arm Lab: lab-baylibre Config: bcm2835_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [727b373740d1d3c6d3273cfc6e81d5e9a229b12d] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 727b373740d1d3c6d3273cfc6e81d5e9a229b12d # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [52bb35a9520ac1746fb34687c5582695813aa00a] Merge remote-tracking branch 'char-misc.current/char-misc-linus' git bisect bad 52bb35a9520ac1746fb34687c5582695813aa00a # bad: [4baf5101d553aae944fa7dc7bd13772eb559d78b] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad 4baf5101d553aae944fa7dc7bd13772eb559d78b # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on sun8i-h3-libretech-all-h3-cc
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on sun8i-h3-libretech-all-h3-cc Summary: Start: 727b373740d1 Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3c423359b5145dde6edd37 Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: sun8i-h3-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [727b373740d1d3c6d3273cfc6e81d5e9a229b12d] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 727b373740d1d3c6d3273cfc6e81d5e9a229b12d # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [52bb35a9520ac1746fb34687c5582695813aa00a] Merge remote-tracking branch 'char-misc.current/char-misc-linus' git bisect bad 52bb35a9520ac1746fb34687c5582695813aa00a # bad: [4baf5101d553aae944fa7dc7bd13772eb559d78b] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad 4baf5101d553aae944fa7dc7bd13772eb559d78b # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on sun8i-h2-plus-libretech-all-h3-cc
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on sun8i-h2-plus-libretech-all-h3-cc Summary: Start: 727b373740d1 Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3c41f159b5145bec6edd6d Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: sun8i-h2-plus-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [727b373740d1d3c6d3273cfc6e81d5e9a229b12d] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 727b373740d1d3c6d3273cfc6e81d5e9a229b12d # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [52bb35a9520ac1746fb34687c5582695813aa00a] Merge remote-tracking branch 'char-misc.current/char-misc-linus' git bisect bad 52bb35a9520ac1746fb34687c5582695813aa00a # bad: [4baf5101d553aae944fa7dc7bd13772eb559d78b] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad 4baf5101d553aae944fa7dc7bd13772eb559d78b # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] next/master boot bisection: next-20190114 on sun8i-h2-plus-libretech-all-h3-cc
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/master boot bisection: next-20190114 on sun8i-h2-plus-libretech-all-h3-cc Summary: Start: 8ce4d582f564 Add linux-next specific files for 20190114 Details: https://kernelci.org/boot/id/5c3c411659b51457a76edd34 Plain log: https://storage.kernelci.org//next/master/next-20190114/arm/sunxi_defconfig… HTML log: https://storage.kernelci.org//next/master/next-20190114/arm/sunxi_defconfig… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: master Target: sun8i-h2-plus-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [8ce4d582f5647052b86f364a070aa26268810d7e] Add linux-next specific files for 20190114 git bisect bad 8ce4d582f5647052b86f364a070aa26268810d7e # bad: [bb937d2e6697b49f6e3fa55ec09951ebbc8717ac] Merge remote-tracking branch 'crypto/master' git bisect bad bb937d2e6697b49f6e3fa55ec09951ebbc8717ac # bad: [04bab95e799a00ffe1b2980e1c786ac9a94f8026] Merge remote-tracking branch 'renesas/next' git bisect bad 04bab95e799a00ffe1b2980e1c786ac9a94f8026 # bad: [5726796d7b631fe9af2202c1ce66232a79984a19] Merge remote-tracking branch 'devicetree-fixes/dt/linus' git bisect bad 5726796d7b631fe9af2202c1ce66232a79984a19 # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [e1faf5c97ee3194c3f3e0b5dc86b3e1490c323d9] Merge remote-tracking branch 'usb-serial-fixes/usb-linus' git bisect bad e1faf5c97ee3194c3f3e0b5dc86b3e1490c323d9 # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # bad: [04eb1efcd614d6f067b76a355b3a3599667959dc] ASoC: soc-core: Hold client_mutex around soc_init_dai_link() git bisect bad 04eb1efcd614d6f067b76a355b3a3599667959dc # good: [a41d9dbf5dac5b6a1283ee8001f22807d18352ea] ASoC: qdsp6: q6asm-dai: Fix a small memory leak git bisect good a41d9dbf5dac5b6a1283ee8001f22807d18352ea # good: [6175471755075d256c1c654151fc1cad183c1e33] ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks git bisect good 6175471755075d256c1c654151fc1cad183c1e33 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

2 1

[alsa-devel] next/master boot bisection: next-20190114 on sun8i-h3-libretech-all-h3-cc
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/master boot bisection: next-20190114 on sun8i-h3-libretech-all-h3-cc Summary: Start: 8ce4d582f564 Add linux-next specific files for 20190114 Details: https://kernelci.org/boot/id/5c3c41ac59b5145b496edd2a Plain log: https://storage.kernelci.org//next/master/next-20190114/arm/sunxi_defconfig… HTML log: https://storage.kernelci.org//next/master/next-20190114/arm/sunxi_defconfig… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: master Target: sun8i-h3-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [8ce4d582f5647052b86f364a070aa26268810d7e] Add linux-next specific files for 20190114 git bisect bad 8ce4d582f5647052b86f364a070aa26268810d7e # bad: [bb937d2e6697b49f6e3fa55ec09951ebbc8717ac] Merge remote-tracking branch 'crypto/master' git bisect bad bb937d2e6697b49f6e3fa55ec09951ebbc8717ac # bad: [04bab95e799a00ffe1b2980e1c786ac9a94f8026] Merge remote-tracking branch 'renesas/next' git bisect bad 04bab95e799a00ffe1b2980e1c786ac9a94f8026 # bad: [5726796d7b631fe9af2202c1ce66232a79984a19] Merge remote-tracking branch 'devicetree-fixes/dt/linus' git bisect bad 5726796d7b631fe9af2202c1ce66232a79984a19 # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [e1faf5c97ee3194c3f3e0b5dc86b3e1490c323d9] Merge remote-tracking branch 'usb-serial-fixes/usb-linus' git bisect bad e1faf5c97ee3194c3f3e0b5dc86b3e1490c323d9 # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # bad: [04eb1efcd614d6f067b76a355b3a3599667959dc] ASoC: soc-core: Hold client_mutex around soc_init_dai_link() git bisect bad 04eb1efcd614d6f067b76a355b3a3599667959dc # good: [a41d9dbf5dac5b6a1283ee8001f22807d18352ea] ASoC: qdsp6: q6asm-dai: Fix a small memory leak git bisect good a41d9dbf5dac5b6a1283ee8001f22807d18352ea # good: [6175471755075d256c1c654151fc1cad183c1e33] ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks git bisect good 6175471755075d256c1c654151fc1cad183c1e33 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

1 0

[alsa-devel] Fix model typo
by Kailang 14 Jan '19

14 Jan '19

Hi Takashi, Attach patch was fix typo for alc225 model name. BR, Kailang

2 1

[alsa-devel] [RFC PATCH v2 1/2] ASoC: refine ASoC hdmi audio suspend/resume
by libin.yang＠intel.com 14 Jan '19

14 Jan '19

From: Libin Yang <libin.yang(a)intel.com> hdmi_codec_prepare() will trigger hdmi runtime resume, which will set the bitmask of hdev->addr. And skl_suspend() will clear the bitmask of HDA_CODEC_IDX_CONTROLLER. HDMI codec idx is not the same as HDA_CODEC_IDX_CONTROLLER, which means i915 power will not be released when suspend. On the other hand, hdmi_codec_prepare() don't need to call pm_runtime_get_sync() to wake up the audio subsystem (HDMI auido) for setting the codec registers. Turning display power on with snd_hdac_display_power() is enough. Let's use S3 without playback as an example: hdmi_codec_prepare() invokes the runtime resume of codec => snd_hdac_display_power(bus, hdev->addr, true) skl runtime resume skl_suspend() => snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); THis means hdev->addr will never release the display power when suspend. The new sequence will be: hdmi_codec_prepare() => snd_hdac_display_power(bus, hdev->addr, true) snd_hdac_display_power(bus, hdev->addr, false) skl runtime resume skl suspned Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 6 ++++-- sound/soc/intel/skylake/skl.c | 7 ------- 2 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3ab2949..782b323 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1895,7 +1895,7 @@ static int hdmi_codec_prepare(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); - pm_runtime_get_sync(&hdev->dev); + snd_hdac_display_power(hdev->bus, hdev->addr, true); /* * Power down afg. @@ -1906,6 +1906,7 @@ static int hdmi_codec_prepare(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hdac_display_power(hdev->bus, hdev->addr, false); return 0; } @@ -1915,6 +1916,7 @@ static void hdmi_codec_complete(struct device *dev) struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + snd_hdac_display_power(hdev->bus, hdev->addr, true); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); @@ -1930,7 +1932,7 @@ static void hdmi_codec_complete(struct device *dev) */ hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - pm_runtime_put_sync(&hdev->dev); + snd_hdac_display_power(hdev->bus, hdev->addr, false); } #else #define hdmi_codec_prepare NULL diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 60c9483..89f4d66 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -336,9 +336,6 @@ static int skl_suspend(struct device *dev) skl->skl_sst->fw_loaded = false; } - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - return 0; } @@ -350,10 +347,6 @@ static int skl_resume(struct device *dev) struct hdac_ext_link *hlink = NULL; int ret; - /* Turned OFF in HDMI codec driver after codec reconfiguration */ - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); - /* * resume only when we are not in suspend active, otherwise need to * restore the device -- 2.7.4

5 18

[alsa-devel] [PATCH 0/4] ASoC: wm8904: prepare for use from audio-graph-card
by Michał Mirosław 13 Jan '19

13 Jan '19

First two patches make wm8904 usable from audio-graph-card. Third is a simple micro-optimization of driver-data usage. Last fixes register access with disabled SYSCLK. Tested on custom SAMA5D2 board. Michał Mirosław (4): ASoC: wm8904: make the driver visible in Kconfig ASoC: wm8904: Automatically enable FLL when selected ASoC: wm8904: save model id directly in of_device_id.data ASoC: wm8904: enable MCLK in STANDBY sound/soc/codecs/Kconfig | 3 ++- sound/soc/codecs/wm8904.c | 42 ++++++++++++++++++++++++++++----------- sound/soc/codecs/wm8904.h | 2 +- 3 files changed, 33 insertions(+), 14 deletions(-) -- 2.19.2

3 10

[alsa-devel] [RFT][PATCH v1] gpiolib: acpi: Introduce ACPI_GPIO_QUIRK_ONLY_GPIOIO
by Andy Shevchenko 13 Jan '19

13 Jan '19

New quirk enforces search for GPIO based on its type. Note, supplied index in the mapping table must be 0. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- - it was sent few weeks ago to Hans for testing, but better to re-test - it's supposed to go via ASoC subsystem due to recent changes made for sound driver drivers/gpio/gpiolib-acpi.c | 15 ++++-- include/linux/acpi.h | 2 + sound/soc/intel/boards/bytcr_rt5651.c | 74 ++++----------------------- 3 files changed, 22 insertions(+), 69 deletions(-) diff --git a/drivers/gpio/gpiolib-acpi.c b/drivers/gpio/gpiolib-acpi.c index 259cf6ab969b..4d291b75cb9f 100644 --- a/drivers/gpio/gpiolib-acpi.c +++ b/drivers/gpio/gpiolib-acpi.c @@ -530,17 +530,24 @@ static int acpi_populate_gpio_lookup(struct acpi_resource *ares, void *data) if (ares->type != ACPI_RESOURCE_TYPE_GPIO) return 1; - if (lookup->n++ == lookup->index && !lookup->desc) { + if (!lookup->desc) { const struct acpi_resource_gpio *agpio = &ares->data.gpio; - int pin_index = lookup->pin_index; + bool gpioint = agpio->connection_type == ACPI_RESOURCE_GPIO_TYPE_INT; + int pin_index; + if (lookup->info.quirks & ACPI_GPIO_QUIRK_ONLY_GPIOIO && gpioint) + lookup->index++; + + if (lookup->n++ != lookup->index) + return 1; + + pin_index = lookup->pin_index; if (pin_index >= agpio->pin_table_length) return 1; lookup->desc = acpi_get_gpiod(agpio->resource_source.string_ptr, agpio->pin_table[pin_index]); - lookup->info.gpioint = - agpio->connection_type == ACPI_RESOURCE_GPIO_TYPE_INT; + lookup->info.gpioint = gpioint; /* * Polarity and triggering are only specified for GpioInt diff --git a/include/linux/acpi.h b/include/linux/acpi.h index 1e89c688139f..7af06794e6fd 100644 --- a/include/linux/acpi.h +++ b/include/linux/acpi.h @@ -1020,6 +1020,8 @@ struct acpi_gpio_mapping { /* Ignore IoRestriction field */ #define ACPI_GPIO_QUIRK_NO_IO_RESTRICTION BIT(0) +/* Choose one of GpioIo() type */ +#define ACPI_GPIO_QUIRK_ONLY_GPIOIO BIT(1) unsigned int quirks; }; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b618d984e2d5..5bfbbdc2a62a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -844,74 +844,18 @@ static const struct x86_cpu_id cherrytrail_cpu_ids[] = { {} }; -static const struct acpi_gpio_params first_gpio = { 0, 0, false }; -static const struct acpi_gpio_params second_gpio = { 1, 0, false }; +static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; -static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { - { "ext-amp-enable-gpios", &first_gpio, 1 }, - { }, -}; - -static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { - { "ext-amp-enable-gpios", &second_gpio, 1 }, +static const struct acpi_gpio_mapping cht_rt5651_gpios[] = { + /* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other + * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the + * GpioIo one for the ext-amp-enable-gpio. That's why quirk is enabled. + */ + { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { }, }; -/* - * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other - * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the - * GpioIo one for the ext-amp-enable-gpio and both count for the index in - * acpi_gpio_params index. So we have 2 different mappings and the code - * below figures out which one to use. - */ -struct byt_rt5651_acpi_resource_data { - int gpio_count; - int gpio_int_idx; -}; - -static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) -{ - struct byt_rt5651_acpi_resource_data *data = arg; - - if (ares->type != ACPI_RESOURCE_TYPE_GPIO) - return 0; - - if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) - data->gpio_int_idx = data->gpio_count; - - data->gpio_count++; - return 0; -} - -static void snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(struct device *codec) -{ - struct byt_rt5651_acpi_resource_data data = { 0, -1 }; - LIST_HEAD(resources); - int ret; - - ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, - snd_byt_rt5651_acpi_resource, &data); - if (ret < 0) { - dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); - return; - } - - /* All info we need is gathered during the walk */ - acpi_dev_free_resource_list(&resources); - - switch (data.gpio_int_idx) { - case 0: - byt_rt5651_gpios = byt_rt5651_amp_en_second; - break; - case 1: - byt_rt5651_gpios = byt_rt5651_amp_en_first; - break; - default: - dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", - data.gpio_int_idx); - } -} - struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -1037,7 +981,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Cherry Trail devices use an external amplifier enable gpio */ if (x86_match_cpu(cherrytrail_cpu_ids) && !byt_rt5651_gpios) - snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(codec_dev); + byt_rt5651_gpios = cht_rt5651_gpios; if (byt_rt5651_gpios) { devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); -- 2.19.2

4 5

[alsa-devel] Audio popping/clicking at boot but NOT after resume/suspend
by Kadir Çolakoğlu 12 Jan '19

12 Jan '19

Hi, I have a Dell Latitude 7480 and a stranger than usual audio popping/clicking/crackling issue. I run Fedora 29, fully updated. The thing is when I boot the laptop, the audio clicks at certain low sound volumes or events. For example when testing the speakers in the Gnome sound settings panel causes multiple clicks/pops. Listening to a audio/video file with certain silent parts causes pops/clicks. I have tried (almost) everything. Changing Pulseaudio setting does not work. Running a pure Jack system does not work. So the issue is not with Pulseaudio or Jack. I have tried disabling Intel HDA powersaving, that also does not work. But all problems disappear when I suspend and then resume the laptop. After resuming, everything works great. So at boot there are sound issues, but after a suspend/resume cycle everything is fine! See https://bugzilla.redhat.com/show_bug.cgi?id=1525104#c108 for more details and testing. I have also attached alsa-info.sh. Kadir

2 3

[alsa-devel] Fwd: Audio popping/clicking at boot but NOT after resume/suspend
by Kadir Çolakoğlu 11 Jan '19

11 Jan '19

---------- Forwarded message --------- From: Kadir Çolakoğlu <kadir(a)colakoglu.nl> Date: vr 11 jan. 2019 om 18:52 Subject: Re: [alsa-devel] Audio popping/clicking at boot but NOT after resume/suspend To: Takashi Iwai <tiwai(a)suse.de> Hi Takashi, Thanks for taking the time to look into this. I am glad you have the same laptop as a test machine. I have been really trying to solve this for a couple of days now, I have no clue so far. This is what I have tried so far: - turning snd_hda_intel powersaving (including the controller) OFF - using Jack only with Pulseaudio masked - tweaking some Pulseaudio settings - tried the latest Fedora rawhide kernel (5.0 git) - tried it with a Fedora 29 liveusb - I even reset all the BIOS settings Nothing of the above helped :( The speaker popping ONLY occurs with the speakers, not with the headphones. And it completely disappears after a suspend/resume cycle and it stays gone until the next reboot. It is good that you have the same machine, assuming all the hardware is the same, you can reproduce this quite easily I hope. If you boot a Fedora 29 liveusb (Gnome) and go to the settings -> sound -> test speakers. Let the volume stay at 50%. If you test FRONT_LEFT and FRONT_RIGHT, you will hear multiple cliks/pops. This is just a small example to easily reproduce it. The pops/clicks occur all the time when you listen to audio with short silences in between, like speech. With the same Fedora Liveusb try for example https://www.youtube.com/watch?v=7YTk_AzJ6fg (this a short tutorial video). This behaviour happens both on ac and battery, that doesn't matter. If I can help with something, let me know. I want to solve this because, while a minor issue, it is irritating. For you information, I don't mind the clicks/pops when the audiocontroller is turning on and off, I consider that normal behaviour. Kind regards, Kadir Op vr 11 jan. 2019 om 14:39 schreef Takashi Iwai <tiwai(a)suse.de>: > On Wed, 09 Jan 2019 07:26:47 +0100, > Kadir Çolakoğlu wrote: > > > > Hi, > > > > I have a Dell Latitude 7480 and a stranger than usual audio > > popping/clicking/crackling issue. I run Fedora 29, fully updated. The > thing > > is when I boot the laptop, the audio clicks at certain low sound volumes > or > > events. For example when testing the speakers in the Gnome sound settings > > panel causes multiple clicks/pops. Listening to a audio/video file with > > certain silent parts causes pops/clicks. > > > > I have tried (almost) everything. Changing Pulseaudio setting does not > > work. Running a pure Jack system does not work. So the issue is not with > > Pulseaudio or Jack. I have tried disabling Intel HDA powersaving, that > also > > does not work. > > > > But all problems disappear when I suspend and then resume the laptop. > After > > resuming, everything works great. So at boot there are sound issues, but > > after a suspend/resume cycle everything is fine! > > > > See https://bugzilla.redhat.com/show_bug.cgi?id=1525104#c108 for more > > details and testing. I have also attached alsa-info.sh. > > Does the issue happen only with the speaker, or also with the > headphone? I have a Latitude 7480 test machine in my office, and > checked 4.20 kernel now, and no problem was seen there. DE is KDE, > but it should be irrelevant... > > > thanks, > > Takashi > > > > > > Kadir > > upload=true&script=true&cardinfo= > > !!################################ > > !!ALSA Information Script v 0.4.64 > > !!################################ > > > > !!Script ran on: Mon Jan 7 20:32:15 UTC 2019 > > > > > > !!Linux Distribution > > !!------------------ > > > > Fedora release 29 (Twenty Nine) NAME=Fedora ID=fedora > PRETTY_NAME="Fedora 29 (Twenty Nine)" LOGO=fedora-logo-icon > CPE_NAME="cpe:/o:fedoraproject:fedora:29" HOME_URL=" > https://fedoraproject.org/" DOCUMENTATION_URL=" > https://docs.fedoraproject.org/en-US/fedora/f29/system-administrators-guide/" > SUPPORT_URL="https://fedoraproject.org/wiki/Communicating_and_getting_help" > BUG_REPORT_URL="https://bugzilla.redhat.com/" > REDHAT_BUGZILLA_PRODUCT="Fedora" REDHAT_BUGZILLA_PRODUCT_VERSION=29 > REDHAT_SUPPORT_PRODUCT="Fedora" REDHAT_SUPPORT_PRODUCT_VERSION=29 > PRIVACY_POLICY_URL="https://fedoraproject.org/wiki/Legal:PrivacyPolicy" > Fedora release 29 (Twenty Nine) Fedora release 29 (Twenty Nine) > > > > > > !!DMI Information > > !!--------------- > > > > Manufacturer: Dell Inc. > > Product Name: Latitude 7480 > > Product Version: > > Firmware Version: 1.13.0 > > Board Vendor: Dell Inc. > > Board Name: 00F6D3 > > > > > > !!ACPI Device Status Information > > !!--------------- > > > > /sys/bus/acpi/devices/ACPI0003:00/status 15 > > /sys/bus/acpi/devices/ACPI000C:00/status 15 > > /sys/bus/acpi/devices/DLL07A0:01/status 15 > > /sys/bus/acpi/devices/DLLK07A0:00/status 15 > > /sys/bus/acpi/devices/INT33A1:00/status 15 > > /sys/bus/acpi/devices/INT33D5:00/status 15 > > /sys/bus/acpi/devices/INT3400:00/status 15 > > /sys/bus/acpi/devices/INT3403:00/status 15 > > /sys/bus/acpi/devices/INT3403:01/status 15 > > /sys/bus/acpi/devices/INT3403:03/status 15 > > /sys/bus/acpi/devices/INT340E:00/status 15 > > /sys/bus/acpi/devices/INT3446:00/status 15 > > /sys/bus/acpi/devices/INT3F0D:00/status 15 > > /sys/bus/acpi/devices/LNXPOWER:00/status 1 > > /sys/bus/acpi/devices/LNXPOWER:01/status 1 > > /sys/bus/acpi/devices/LNXPOWER:02/status 1 > > /sys/bus/acpi/devices/LNXPOWER:03/status 1 > > /sys/bus/acpi/devices/LNXPOWER:04/status 1 > > /sys/bus/acpi/devices/LNXPOWER:05/status 1 > > /sys/bus/acpi/devices/LNXPOWER:06/status 1 > > /sys/bus/acpi/devices/LNXPOWER:07/status 1 > > /sys/bus/acpi/devices/LNXPOWER:08/status 1 > > /sys/bus/acpi/devices/LNXPOWER:09/status 1 > > /sys/bus/acpi/devices/LNXPOWER:0a/status 1 > > /sys/bus/acpi/devices/LNXPOWER:0b/status 1 > > /sys/bus/acpi/devices/LNXPOWER:0c/status 1 > > /sys/bus/acpi/devices/LNXPOWER:0d/status 1 > > /sys/bus/acpi/devices/LNXPOWER:0e/status 1 > > /sys/bus/acpi/devices/LNXPOWER:0f/status 1 > > /sys/bus/acpi/devices/LNXPOWER:10/status 1 > > /sys/bus/acpi/devices/LNXPOWER:11/status 1 > > /sys/bus/acpi/devices/LNXPOWER:12/status 1 > > /sys/bus/acpi/devices/LNXPOWER:13/status 1 > > /sys/bus/acpi/devices/PNP0103:00/status 15 > > /sys/bus/acpi/devices/PNP0C02:03/status 3 > > /sys/bus/acpi/devices/PNP0C02:05/status 3 > > /sys/bus/acpi/devices/PNP0C09:00/status 15 > > /sys/bus/acpi/devices/PNP0C0A:00/status 31 > > /sys/bus/acpi/devices/PNP0C0C:00/status 15 > > /sys/bus/acpi/devices/PNP0C0F:00/status 9 > > /sys/bus/acpi/devices/PNP0C0F:01/status 9 > > /sys/bus/acpi/devices/PNP0C0F:02/status 9 > > /sys/bus/acpi/devices/PNP0C0F:03/status 9 > > /sys/bus/acpi/devices/PNP0C0F:04/status 9 > > /sys/bus/acpi/devices/PNP0C0F:05/status 9 > > /sys/bus/acpi/devices/PNP0C0F:06/status 9 > > /sys/bus/acpi/devices/PNP0C0F:07/status 9 > > /sys/bus/acpi/devices/device:09/status 15 > > /sys/bus/acpi/devices/device:63/status 15 > > /sys/bus/acpi/devices/device:75/status 11 > > > > > > !!Kernel Information > > !!------------------ > > > > Kernel release: 4.19.13-300.fc29.x86_64 > > Operating System: GNU/Linux > > Architecture: x86_64 > > Processor: x86_64 > > SMP Enabled: Yes > > > > > > !!ALSA Version > > !!------------ > > > > Driver version: k4.19.13-300.fc29.x86_64 > > Library version: > > Utilities version: 1.1.7 > > > > > > !!Loaded ALSA modules > > !!------------------- > > > > snd_hda_intel > > > > > > !!Sound Servers on this system > > !!---------------------------- > > > > Pulseaudio: > > Installed - Yes (/usr/bin/pulseaudio) > > Running - Yes > > > > Jack: > > Installed - Yes (/usr/bin/jackd) > > Running - No > > > > > > !!Soundcards recognised by ALSA > > !!----------------------------- > > > > 0 [PCH ]: HDA-Intel - HDA Intel PCH > > HDA Intel PCH at 0xec248000 irq 131 > > > > > > !!PCI Soundcards installed in the system > > !!-------------------------------------- > > > > 00:1f.3 Audio device: Intel Corporation Sunrise Point-LP HD Audio (rev > 21) > > > > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > > !!------------------------------------------------------- > > > > 00:1f.3 0403: 8086:9d71 (rev 21) > > Subsystem: 1028:07a0 > > > > > > !!Loaded sound module options > > !!--------------------------- > > > > !!Module: snd_hda_intel > > align_buffer_size : -1 > > bdl_pos_adj : > -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > > beep_mode : > N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N > > enable : > Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > > enable_msi : -1 > > id : > (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > > index : > -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > > jackpoll_ms : > 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > > model : > (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > > patch : > (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > > pm_blacklist : Y > > position_fix : > -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > > power_save : 1 > > power_save_controller : Y > > probe_mask : > -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > > probe_only : > 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > > single_cmd : -1 > > snoop : -1 > > > > > > !!HDA-Intel Codec information > > !!--------------------------- > > --startcollapse-- > > > > Codec: Realtek ALC3246 > > Address: 0 > > AFG Function Id: 0x1 (unsol 1) > > Vendor Id: 0x10ec0256 > > Subsystem Id: 0x102807a0 > > Revision Id: 0x100002 > > No Modem Function Group found > > Default PCM: > > rates [0x560]: 44100 48000 96000 192000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Default Amp-In caps: N/A > > Default Amp-Out caps: N/A > > State of AFG node 0x01: > > Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS > > Power: setting=D0, actual=D0 > > GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 > > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > > Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out > > Control: name="Speaker Playback Volume", index=0, device=0 > > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > > Amp-Out vals: [0x3b 0x3b] > > Converter: stream=0, channel=0 > > PCM: > > rates [0x60]: 44100 48000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out > > Control: name="Headphone Playback Volume", index=0, device=0 > > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > > Device: name="ALC3246 Analog", type="Audio", device=0 > > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > > Amp-Out vals: [0x00 0x00] > > Converter: stream=0, channel=0 > > PCM: > > rates [0x60]: 44100 48000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > > Converter: stream=0, channel=0 > > Digital: > > Digital category: 0x0 > > IEC Coding Type: 0x0 > > PCM: > > rates [0x5e0]: 44100 48000 88200 96000 192000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In > > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > > Amp-In vals: [0x97 0x97] > > Converter: stream=0, channel=0 > > SDI-Select: 0 > > PCM: > > rates [0x560]: 44100 48000 96000 192000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 1 > > 0x24 > > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > > Control: name="Capture Volume", index=0, device=0 > > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > > Control: name="Capture Switch", index=0, device=0 > > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > > Device: name="ALC3246 Analog", type="Audio", device=0 > > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > > Amp-In vals: [0x80 0x80] > > Converter: stream=0, channel=0 > > SDI-Select: 0 > > PCM: > > rates [0x560]: 44100 48000 96000 192000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 1 > > 0x23 > > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > > Amp-In vals: [0x97 0x97] > > Converter: stream=0, channel=0 > > SDI-Select: 0 > > PCM: > > rates [0x560]: 44100 48000 96000 192000 > > bits [0xe]: 16 20 24 > > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 1 > > 0x22 > > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In > > Control: name="Internal Mic Boost Volume", index=0, device=0 > > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > > Amp-In vals: [0x00 0x00] > > Pincap 0x00000020: IN > > Pin Default 0x90a60140: [Fixed] Mic at Int N/A > > Conn = Digital, Color = Unknown > > DefAssociation = 0x4, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x20: IN > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In > > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > > Amp-In vals: [0x00 0x00] > > Pincap 0x00000020: IN > > Pin Default 0x40000000: [N/A] Line Out at Ext N/A > > Conn = Unknown, Color = Unknown > > DefAssociation = 0x0, Sequence = 0x0 > > Pin-ctls: 0x00: > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > > Control: name="Speaker Playback Switch", index=0, device=0 > > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-Out vals: [0x00 0x00] > > Pincap 0x00010014: OUT EAPD Detect > > EAPD 0x2: EAPD > > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > > Conn = Analog, Color = Unknown > > DefAssociation = 0x1, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x40: OUT > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 1 > > 0x02 > > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In > > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > > Amp-In vals: [0x00 0x00] > > Pincap 0x00003724: IN Detect > > Vref caps: HIZ 50 GRD 80 100 > > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > > Conn = 1/8, Color = Black > > DefAssociation = 0xf, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x20: IN VREF_HIZ > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In > > Control: name="Headset Mic Boost Volume", index=0, device=0 > > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > > Amp-In vals: [0x00 0x00] > > Pincap 0x00003724: IN Detect > > Vref caps: HIZ 50 GRD 80 100 > > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > > Conn = 1/8, Color = Black > > DefAssociation = 0xf, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x24: IN VREF_80 > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In > > Control: name="Headphone Mic Boost Volume", index=0, device=0 > > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > > Amp-In vals: [0x00 0x00] > > Pincap 0x00003724: IN Detect > > Vref caps: HIZ 50 GRD 80 100 > > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > > Conn = 1/8, Color = Black > > DefAssociation = 0xf, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x20: IN VREF_HIZ > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > > Amp-In vals: [0x00 0x00] > > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-Out vals: [0x80 0x80] > > Pincap 0x00013734: IN OUT EAPD Detect > > Vref caps: HIZ 50 GRD 80 100 > > EAPD 0x2: EAPD > > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > > Conn = 1/8, Color = Black > > DefAssociation = 0xf, Sequence = 0x0 > > Misc = NO_PRESENCE > > Pin-ctls: 0x20: IN VREF_HIZ > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 2 > > 0x02* 0x03 > > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > > Pincap 0x00000020: IN > > Pin Default 0x40700001: [N/A] Modem Hand at Ext N/A > > Conn = Unknown, Color = Unknown > > DefAssociation = 0x0, Sequence = 0x1 > > Pin-ctls: 0x20: IN > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital > > Pincap 0x00000014: OUT Detect > > Pin Default 0x421212f2: [N/A] Speaker at Ext Front > > Conn = 1/4, Color = Black > > DefAssociation = 0xf, Sequence = 0x2 > > Pin-ctls: 0x40: OUT > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 1 > > 0x06 > > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > > Processing caps: benign=0, ncoeff=91 > > Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > > Control: name="Headphone Playback Switch", index=0, device=0 > > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-Out vals: [0x80 0x80] > > Pincap 0x0001001c: OUT HP EAPD Detect > > EAPD 0x2: EAPD > > Pin Default 0x02211020: [Jack] HP Out at Ext Front > > Conn = 1/8, Color = Black > > DefAssociation = 0x2, Sequence = 0x0 > > Pin-ctls: 0xc0: OUT HP > > Unsolicited: tag=01, enabled=1 > > Power states: D0 D1 D2 D3 EPSS > > Power: setting=D3, actual=D3 > > Connection: 2 > > 0x02 0x03* > > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 > 0x80] > > Connection: 5 > > 0x18 0x19 0x1a 0x1b 0x1d > > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 > 0x80] [0x80 0x80] > > Connection: 6 > > 0x18 0x19 0x1a 0x1b 0x1d 0x12 > > Node 0x24 [Audio Selector] wcaps 0x300101: Stereo > > Connection: 2 > > 0x12* 0x13 > > Codec: Intel Kabylake HDMI > > Address: 2 > > AFG Function Id: 0x1 (unsol 0) > > Vendor Id: 0x8086280b > > Subsystem Id: 0x80860101 > > Revision Id: 0x100000 > > No Modem Function Group found > > Default PCM: > > rates [0x0]: > > bits [0x0]: > > formats [0x0]: > > Default Amp-In caps: N/A > > Default Amp-Out caps: N/A > > State of AFG node 0x01: > > Power states: D0 D3 CLKSTOP EPSS > > Power: setting=D0, actual=D0, Clock-stop-OK > > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > > Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital > > Converter: stream=0, channel=0 > > Digital: Enabled KAE > > Digital category: 0x0 > > IEC Coding Type: 0x0 > > PCM: > > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > > bits [0x1a]: 16 24 32 > > formats [0x5]: PCM AC3 > > Power states: D0 D3 EPSS > > Power: setting=D0, actual=D0 > > Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital > > Converter: stream=0, channel=0 > > Digital: Enabled KAE > > Digital category: 0x0 > > IEC Coding Type: 0x0 > > PCM: > > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > > bits [0x1a]: 16 24 32 > > formats [0x5]: PCM AC3 > > Power states: D0 D3 EPSS > > Power: setting=D0, actual=D0 > > Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital > > Converter: stream=0, channel=0 > > Digital: Enabled KAE > > Digital category: 0x0 > > IEC Coding Type: 0x0 > > PCM: > > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > > bits [0x1a]: 16 24 32 > > formats [0x5]: PCM AC3 > > Power states: D0 D3 EPSS > > Power: setting=D0, actual=D0 > > Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-Out vals: [0x00 0x00] > > Pincap 0x0b000094: OUT Detect HBR HDMI DP > > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > > Conn = Digital, Color = Unknown > > DefAssociation = 0x1, Sequence = 0x0 > > Pin-ctls: 0x00: > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D3 EPSS > > Power: setting=D0, actual=D0 > > Devices: 0 > > Connection: 0 > > In-driver Connection: 3 > > 0x02 0x03 0x04 > > Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-Out vals: [0x00 0x00] > > Pincap 0x0b000094: OUT Detect HBR HDMI DP > > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > > Conn = Digital, Color = Unknown > > DefAssociation = 0x1, Sequence = 0x0 > > Pin-ctls: 0x00: > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D3 EPSS > > Power: setting=D0, actual=D0 > > Devices: 0 > > Connection: 0 > > In-driver Connection: 3 > > 0x02 0x03 0x04 > > Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > > Amp-Out vals: [0x00 0x00] > > Pincap 0x0b000094: OUT Detect HBR HDMI DP > > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > > Conn = Digital, Color = Unknown > > DefAssociation = 0x1, Sequence = 0x0 > > Pin-ctls: 0x00: > > Unsolicited: tag=00, enabled=0 > > Power states: D0 D3 EPSS > > Power: setting=D0, actual=D0 > > Devices: 0 > > Connection: 0 > > In-driver Connection: 3 > > 0x02 0x03 0x04 > > Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono > > --endcollapse-- > > > > > > !!ALSA Device nodes > > !!----------------- > > > > crw-rw----+ 1 root audio 116, 11 Jan 7 21:15 /dev/snd/controlC0 > > crw-rw----+ 1 root audio 116, 9 Jan 7 21:15 /dev/snd/hwC0D0 > > crw-rw----+ 1 root audio 116, 10 Jan 7 21:15 /dev/snd/hwC0D2 > > crw-rw----+ 1 root audio 116, 3 Jan 7 21:23 /dev/snd/pcmC0D0c > > crw-rw----+ 1 root audio 116, 2 Jan 7 21:28 /dev/snd/pcmC0D0p > > crw-rw----+ 1 root audio 116, 8 Jan 7 21:15 /dev/snd/pcmC0D10p > > crw-rw----+ 1 root audio 116, 4 Jan 7 21:15 /dev/snd/pcmC0D3p > > crw-rw----+ 1 root audio 116, 5 Jan 7 21:15 /dev/snd/pcmC0D7p > > crw-rw----+ 1 root audio 116, 6 Jan 7 21:15 /dev/snd/pcmC0D8p > > crw-rw----+ 1 root audio 116, 7 Jan 7 21:15 /dev/snd/pcmC0D9p > > crw-rw----+ 1 root audio 116, 1 Jan 7 21:15 /dev/snd/seq > > crw-rw----+ 1 root audio 116, 33 Jan 7 21:15 /dev/snd/timer > > > > /dev/snd/by-path: > > total 0 > > drwxr-xr-x. 2 root root 60 Jan 7 21:15 . > > drwxr-xr-x. 3 root root 300 Jan 7 21:15 .. > > lrwxrwxrwx. 1 root root 12 Jan 7 21:15 pci-0000:00:1f.3 -> ../controlC0 > > > > > > !!ALSA configuration files > > !!------------------------ > > > > !!System wide config file (/etc/asound.conf) > > > > # > > # Place your global alsa-lib configuration here... > > # > > > > > > !!Aplay/Arecord output > > !!-------------------- > > > > APLAY > > > > **** List of PLAYBACK Hardware Devices **** > > card 0: PCH [HDA Intel PCH], device 0: ALC3246 Analog [ALC3246 Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > > > ARECORD > > > > **** List of CAPTURE Hardware Devices **** > > card 0: PCH [HDA Intel PCH], device 0: ALC3246 Analog [ALC3246 Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > > > !!Amixer output > > !!------------- > > > > !!-------Mixer controls for card 0 [PCH] > > > > Card hw:0 'PCH'/'HDA Intel PCH at 0xec248000 irq 131' > > Mixer name : 'Realtek ALC3246' > > Components : 'HDA:10ec0256,102807a0,00100002 > HDA:8086280b,80860101,00100000' > > Controls : 56 > > Simple ctrls : 18 > > Simple mixer control 'Master',0 > > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > > Playback channels: Mono > > Limits: Playback 0 - 87 > > Mono: Playback 59 [68%] [-21.00dB] [on] > > Simple mixer control 'Headphone',0 > > Capabilities: pvolume pswitch > > Playback channels: Front Left - Front Right > > Limits: Playback 0 - 87 > > Mono: > > Front Left: Playback 0 [0%] [-65.25dB] [off] > > Front Right: Playback 0 [0%] [-65.25dB] [off] > > Simple mixer control 'Headphone Mic',0 > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > Capture exclusive group: 0 > > Capture channels: Mono > > Mono: Capture [off] > > Simple mixer control 'Headphone Mic Boost',0 > > Capabilities: volume > > Playback channels: Front Left - Front Right > > Capture channels: Front Left - Front Right > > Limits: 0 - 3 > > Front Left: 0 [0%] [0.00dB] > > Front Right: 0 [0%] [0.00dB] > > Simple mixer control 'Speaker',0 > > Capabilities: pvolume pswitch > > Playback channels: Front Left - Front Right > > Limits: Playback 0 - 87 > > Mono: > > Front Left: Playback 87 [100%] [0.00dB] [on] > > Front Right: Playback 87 [100%] [0.00dB] [on] > > Simple mixer control 'PCM',0 > > Capabilities: pvolume > > Playback channels: Front Left - Front Right > > Limits: Playback 0 - 255 > > Mono: > > Front Left: Playback 255 [100%] [0.00dB] > > Front Right: Playback 255 [100%] [0.00dB] > > Simple mixer control 'Mic Mute-LED Mode',0 > > Capabilities: enum > > Items: 'On' 'Off' 'Follow Capture' 'Follow Mute' > > Item0: 'Follow Capture' > > Simple mixer control 'IEC958',0 > > Capabilities: pswitch pswitch-joined > > Playback channels: Mono > > Mono: Playback [off] > > Simple mixer control 'IEC958',1 > > Capabilities: pswitch pswitch-joined > > Playback channels: Mono > > Mono: Playback [off] > > Simple mixer control 'IEC958',2 > > Capabilities: pswitch pswitch-joined > > Playback channels: Mono > > Mono: Playback [off] > > Simple mixer control 'IEC958',3 > > Capabilities: pswitch pswitch-joined > > Playback channels: Mono > > Mono: Playback [off] > > Simple mixer control 'IEC958',4 > > Capabilities: pswitch pswitch-joined > > Playback channels: Mono > > Mono: Playback [off] > > Simple mixer control 'Capture',0 > > Capabilities: cvolume cswitch > > Capture channels: Front Left - Front Right > > Limits: Capture 0 - 63 > > Front Left: Capture 0 [0%] [-17.25dB] [off] > > Front Right: Capture 0 [0%] [-17.25dB] [off] > > Simple mixer control 'Auto-Mute Mode',0 > > Capabilities: enum > > Items: 'Disabled' 'Enabled' > > Item0: 'Enabled' > > Simple mixer control 'Headset Mic',0 > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > Capture exclusive group: 0 > > Capture channels: Mono > > Mono: Capture [on] > > Simple mixer control 'Headset Mic Boost',0 > > Capabilities: volume > > Playback channels: Front Left - Front Right > > Capture channels: Front Left - Front Right > > Limits: 0 - 3 > > Front Left: 0 [0%] [0.00dB] > > Front Right: 0 [0%] [0.00dB] > > Simple mixer control 'Internal Mic',0 > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > Capture exclusive group: 0 > > Capture channels: Mono > > Mono: Capture [off] > > Simple mixer control 'Internal Mic Boost',0 > > Capabilities: volume > > Playback channels: Front Left - Front Right > > Capture channels: Front Left - Front Right > > Limits: 0 - 3 > > Front Left: 0 [0%] [0.00dB] > > Front Right: 0 [0%] [0.00dB] > > > > > > !!Alsactl output > > !!-------------- > > > > --startcollapse-- > > state.PCH { > > control.1 { > > iface MIXER > > name 'Headphone Playback Volume' > > value.0 0 > > value.1 0 > > comment { > > access 'read write' > > type INTEGER > > count 2 > > range '0 - 87' > > dbmin -6525 > > dbmax 0 > > dbvalue.0 -6525 > > dbvalue.1 -6525 > > } > > } > > control.2 { > > iface MIXER > > name 'Headphone Playback Switch' > > value.0 false > > value.1 false > > comment { > > access 'read write' > > type BOOLEAN > > count 2 > > } > > } > > control.3 { > > iface MIXER > > name 'Speaker Playback Volume' > > value.0 87 > > value.1 87 > > comment { > > access 'read write' > > type INTEGER > > count 2 > > range '0 - 87' > > dbmin -6525 > > dbmax 0 > > dbvalue.0 0 > > dbvalue.1 0 > > } > > } > > control.4 { > > iface MIXER > > name 'Speaker Playback Switch' > > value.0 true > > value.1 true > > comment { > > access 'read write' > > type BOOLEAN > > count 2 > > } > > } > > control.5 { > > iface MIXER > > name 'Auto-Mute Mode' > > value Enabled > > comment { > > access 'read write' > > type ENUMERATED > > count 1 > > item.0 Disabled > > item.1 Enabled > > } > > } > > control.6 { > > iface MIXER > > name 'Capture Source' > > value 'Headset Mic' > > comment { > > access 'read write' > > type ENUMERATED > > count 1 > > item.0 'Headset Mic' > > item.1 'Headphone Mic' > > item.2 'Internal Mic' > > } > > } > > control.7 { > > iface MIXER > > name 'Capture Volume' > > value.0 0 > > value.1 0 > > comment { > > access 'read write' > > type INTEGER > > count 2 > > range '0 - 63' > > dbmin -1725 > > dbmax 3000 > > dbvalue.0 -1725 > > dbvalue.1 -1725 > > } > > } > > control.8 { > > iface MIXER > > name 'Capture Switch' > > value.0 false > > value.1 false > > comment { > > access 'read write' > > type BOOLEAN > > count 2 > > } > > } > > control.9 { > > iface MIXER > > name 'Headset Mic Boost Volume' > > value.0 0 > > value.1 0 > > comment { > > access 'read write' > > type INTEGER > > count 2 > > range '0 - 3' > > dbmin 0 > > dbmax 3000 > > dbvalue.0 0 > > dbvalue.1 0 > > } > > } > > control.10 { > > iface MIXER > > name 'Headphone Mic Boost Volume' > > value.0 0 > > value.1 0 > > comment { > > access 'read write' > > type INTEGER > > count 2 > > range '0 - 3' > > dbmin 0 > > dbmax 3000 > > dbvalue.0 0 > > dbvalue.1 0 > > } > > } > > control.11 { > > iface MIXER > > name 'Internal Mic Boost Volume' > > value.0 0 > > value.1 0 > > comment { > > access 'read write' > > type INTEGER > > count 2 > > range '0 - 3' > > dbmin 0 > > dbmax 3000 > > dbvalue.0 0 > > dbvalue.1 0 > > } > > } > > control.12 { > > iface MIXER > > name 'Mic Mute-LED Mode' > > value 'Follow Capture' > > comment { > > access 'read write' > > type ENUMERATED > > count 1 > > item.0 On > > item.1 Off > > item.2 'Follow Capture' > > item.3 'Follow Mute' > > } > > } > > control.13 { > > iface MIXER > > name 'Master Playback Volume' > > value 59 > > comment { > > access 'read write' > > type INTEGER > > count 1 > > range '0 - 87' > > dbmin -6525 > > dbmax 0 > > dbvalue.0 -2100 > > } > > } > > control.14 { > > iface MIXER > > name 'Master Playback Switch' > > value true > > comment { > > access 'read write' > > type BOOLEAN > > count 1 > > } > > } > > control.15 { > > iface CARD > > name 'Headset Mic Phantom Jack' > > value true > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.16 { > > iface CARD > > name 'Headphone Mic Jack' > > value false > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.17 { > > iface CARD > > name 'Internal Mic Phantom Jack' > > value true > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.18 { > > iface CARD > > name 'Speaker Phantom Jack' > > value true > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.19 { > > iface PCM > > name 'Playback Channel Map' > > value.0 0 > > value.1 0 > > comment { > > access read > > type INTEGER > > count 2 > > range '0 - 36' > > } > > } > > control.20 { > > iface PCM > > name 'Capture Channel Map' > > value.0 0 > > value.1 0 > > comment { > > access read > > type INTEGER > > count 2 > > range '0 - 36' > > } > > } > > control.21 { > > iface CARD > > name 'HDMI/DP,pcm=3 Jack' > > value false > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.22 { > > iface MIXER > > name 'IEC958 Playback Con Mask' > > value > '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.23 { > > iface MIXER > > name 'IEC958 Playback Pro Mask' > > value > '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.24 { > > iface MIXER > > name 'IEC958 Playback Default' > > value > '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access 'read write' > > type IEC958 > > count 1 > > } > > } > > control.25 { > > iface MIXER > > name 'IEC958 Playback Switch' > > value false > > comment { > > access 'read write' > > type BOOLEAN > > count 1 > > } > > } > > control.26 { > > iface PCM > > device 3 > > name ELD > > value '' > > comment { > > access 'read volatile' > > type BYTES > > count 0 > > } > > } > > control.27 { > > iface CARD > > name 'HDMI/DP,pcm=7 Jack' > > value false > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.28 { > > iface MIXER > > name 'IEC958 Playback Con Mask' > > index 1 > > value > '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.29 { > > iface MIXER > > name 'IEC958 Playback Pro Mask' > > index 1 > > value > '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.30 { > > iface MIXER > > name 'IEC958 Playback Default' > > index 1 > > value > '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access 'read write' > > type IEC958 > > count 1 > > } > > } > > control.31 { > > iface MIXER > > name 'IEC958 Playback Switch' > > index 1 > > value false > > comment { > > access 'read write' > > type BOOLEAN > > count 1 > > } > > } > > control.32 { > > iface PCM > > device 7 > > name ELD > > value '' > > comment { > > access 'read volatile' > > type BYTES > > count 0 > > } > > } > > control.33 { > > iface CARD > > name 'HDMI/DP,pcm=8 Jack' > > value false > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.34 { > > iface MIXER > > name 'IEC958 Playback Con Mask' > > index 2 > > value > '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.35 { > > iface MIXER > > name 'IEC958 Playback Pro Mask' > > index 2 > > value > '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.36 { > > iface MIXER > > name 'IEC958 Playback Default' > > index 2 > > value > '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access 'read write' > > type IEC958 > > count 1 > > } > > } > > control.37 { > > iface MIXER > > name 'IEC958 Playback Switch' > > index 2 > > value false > > comment { > > access 'read write' > > type BOOLEAN > > count 1 > > } > > } > > control.38 { > > iface PCM > > device 8 > > name ELD > > value '' > > comment { > > access 'read volatile' > > type BYTES > > count 0 > > } > > } > > control.39 { > > iface CARD > > name 'HDMI/DP,pcm=9 Jack' > > value false > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.40 { > > iface MIXER > > name 'IEC958 Playback Con Mask' > > index 3 > > value > '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.41 { > > iface MIXER > > name 'IEC958 Playback Pro Mask' > > index 3 > > value > '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.42 { > > iface MIXER > > name 'IEC958 Playback Default' > > index 3 > > value > '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access 'read write' > > type IEC958 > > count 1 > > } > > } > > control.43 { > > iface MIXER > > name 'IEC958 Playback Switch' > > index 3 > > value false > > comment { > > access 'read write' > > type BOOLEAN > > count 1 > > } > > } > > control.44 { > > iface PCM > > device 9 > > name ELD > > value '' > > comment { > > access 'read volatile' > > type BYTES > > count 0 > > } > > } > > control.45 { > > iface CARD > > name 'HDMI/DP,pcm=10 Jack' > > value false > > comment { > > access read > > type BOOLEAN > > count 1 > > } > > } > > control.46 { > > iface MIXER > > name 'IEC958 Playback Con Mask' > > index 4 > > value > '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.47 { > > iface MIXER > > name 'IEC958 Playback Pro Mask' > > index 4 > > value > '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access read > > type IEC958 > > count 1 > > } > > } > > control.48 { > > iface MIXER > > name 'IEC958 Playback Default' > > index 4 > > value > '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > > comment { > > access 'read write' > > type IEC958 > > count 1 > > } > > } > > control.49 { > > iface MIXER > > name 'IEC958 Playback Switch' > > index 4 > > value false > > comment { > > access 'read write' > > type BOOLEAN > > count 1 > > } > > } > > control.50 { > > iface PCM > > device 10 > > name ELD > > value '' > > comment { > > access 'read volatile' > > type BYTES > > count 0 > > } > > } > > control.51 { > > iface PCM > > device 3 > > name 'Playback Channel Map' > > value.0 0 > > value.1 0 > > value.2 0 > > value.3 0 > > value.4 0 > > value.5 0 > > value.6 0 > > value.7 0 > > comment { > > access 'read write' > > type INTEGER > > count 8 > > range '0 - 36' > > } > > } > > control.52 { > > iface PCM > > device 7 > > name 'Playback Channel Map' > > value.0 0 > > value.1 0 > > value.2 0 > > value.3 0 > > value.4 0 > > value.5 0 > > value.6 0 > > value.7 0 > > comment { > > access 'read write' > > type INTEGER > > count 8 > > range '0 - 36' > > } > > } > > control.53 { > > iface PCM > > device 8 > > name 'Playback Channel Map' > > value.0 0 > > value.1 0 > > value.2 0 > > value.3 0 > > value.4 0 > > value.5 0 > > value.6 0 > > value.7 0 > > comment { > > access 'read write' > > type INTEGER > > count 8 > > range '0 - 36' > > } > > } > > control.54 { > > iface PCM > > device 9 > > name 'Playback Channel Map' > > value.0 0 > > value.1 0 > > value.2 0 > > value.3 0 > > value.4 0 > > value.5 0 > > value.6 0 > > value.7 0 > > comment { > > access 'read write' > > type INTEGER > > count 8 > > range '0 - 36' > > } > > } > > control.55 { > > iface PCM > > device 10 > > name 'Playback Channel Map' > > value.0 0 > > value.1 0 > > value.2 0 > > value.3 0 > > value.4 0 > > value.5 0 > > value.6 0 > > value.7 0 > > comment { > > access 'read write' > > type INTEGER > > count 8 > > range '0 - 36' > > } > > } > > control.56 { > > iface MIXER > > name 'PCM Playback Volume' > > value.0 255 > > value.1 255 > > comment { > > access 'read write user' > > type INTEGER > > count 2 > > range '0 - 255' > > tlv '0000000100000008ffffec1400000014' > > dbmin -5100 > > dbmax 0 > > dbvalue.0 0 > > dbvalue.1 0 > > } > > } > > } > > --endcollapse-- > > > > > > !!All Loaded Modules > > !!------------------ > > > > Module > > fuse > > bluetooth > > ecdh_generic > > ccm > > vfat > > fat > > arc4 > > iwlmvm > > snd_soc_skl > > snd_soc_skl_ipc > > snd_soc_sst_ipc > > snd_soc_sst_dsp > > snd_hda_ext_core > > snd_soc_acpi_intel_match > > mac80211 > > snd_soc_acpi > > snd_soc_core > > snd_hda_codec_hdmi > > iwlwifi > > joydev > > snd_compress > > ac97_bus > > snd_pcm_dmaengine > > snd_hda_codec_realtek > > snd_hda_codec_generic > > mei_wdt > > hid_alps > > wmi_bmof > > dell_wmi > > intel_wmi_thunderbolt > > uvcvideo > > cfg80211 > > snd_hda_intel > > dell_rbtn > > videobuf2_vmalloc > > videobuf2_memops > > snd_hda_codec > > dell_laptop > > videobuf2_v4l2 > > intel_rapl > > videobuf2_common > > dell_smbios > > dell_wmi_descriptor > > x86_pkg_temp_thermal > > dcdbas > > intel_powerclamp > > snd_hda_core > > coretemp > > dell_smm_hwmon > > kvm_intel > > snd_hwdep > > videodev > > snd_seq > > intel_cstate > > intel_uncore > > snd_seq_device > > e1000e > > snd_pcm > > intel_rapl_perf > > media > > snd_timer > > snd > > i2c_i801 > > soundcore > > mei_me > > mei > > rtsx_pci_ms > > rfkill > > memstick > > processor_thermal_device > > intel_soc_dts_iosf > > idma64 > > intel_pch_thermal > > intel_lpss_pci > > wmi > > int3400_thermal > > acpi_thermal_rel > > intel_lpss_acpi > > intel_lpss > > int3403_thermal > > int340x_thermal_zone > > acpi_pad > > intel_hid > > pcc_cpufreq > > sparse_keymap > > dm_crypt > > i915 > > kvmgt > > mdev > > vfio > > kvm > > rtsx_pci_sdmmc > > mmc_core > > irqbypass > > i2c_algo_bit > > drm_kms_helper > > crct10dif_pclmul > > crc32_pclmul > > crc32c_intel > > drm > > ghash_clmulni_intel > > serio_raw > > rtsx_pci > > i2c_hid > > video > > > > > > !!Sysfs Files > > !!----------- > > > > /sys/class/sound/hwC0D0/init_pin_configs: > > 0x12 0x90a60140 > > 0x13 0x40000000 > > 0x14 0x90170110 > > 0x18 0x411111f0 > > 0x19 0x411111f0 > > 0x1a 0x411111f0 > > 0x1b 0x411111f0 > > 0x1d 0x40700001 > > 0x1e 0x421212f2 > > 0x21 0x02211020 > > > > /sys/class/sound/hwC0D0/driver_pin_configs: > > 0x19 0x01a1913c > > 0x1a 0x01a1913d > > > > /sys/class/sound/hwC0D0/user_pin_configs: > > > > /sys/class/sound/hwC0D0/init_verbs: > > > > /sys/class/sound/hwC0D0/hints: > > > > /sys/class/sound/hwC0D2/init_pin_configs: > > 0x05 0x18560010 > > 0x06 0x18560010 > > 0x07 0x18560010 > > > > /sys/class/sound/hwC0D2/driver_pin_configs: > > > > /sys/class/sound/hwC0D2/user_pin_configs: > > > > /sys/class/sound/hwC0D2/init_verbs: > > > > /sys/class/sound/hwC0D2/hints: > > > > > > !!ALSA/HDA dmesg > > !!-------------- > > > > [ 0.268744] ACPI: Added _OSI(Linux-Dell-Video) > > [ 0.268744] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) > > [ 0.310469] ACPI: 10 ACPI AML tables successfully acquired and loaded > > -- > > [ 12.041224] e1000e 0000:00:1f.6 0000:00:1f.6 (uninitialized): > registered PHC clock > > [ 12.071264] snd_hda_intel 0000:00:1f.3: enabling device (0000 -> 0002) > > [ 12.072382] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops > i915_audio_component_bind_ops [i915]) > > [ 12.077227] input: Dell WMI hotkeys as > /devices/platform/PNP0C14:01/wmi_bus/wmi_bus-PNP0C14:01/9DBB5994-A997-11DA-B012-B622A1EF5492/input/input8 > > -- > > [ 12.153053] cfg80211: Loaded X.509 cert 'sforshee: 00b28ddf47aef9cea7' > > [ 12.178202] snd_hda_codec_realtek hdaudioC0D0: autoconfig for > ALC3246: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker > > [ 12.178205] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 > (0x0/0x0/0x0/0x0/0x0) > > [ 12.178207] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 > (0x21/0x0/0x0/0x0/0x0) > > [ 12.178208] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 > > [ 12.178209] snd_hda_codec_realtek hdaudioC0D0: inputs: > > [ 12.178211] snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 > > [ 12.178212] snd_hda_codec_realtek hdaudioC0D0: Headphone Mic=0x1a > > [ 12.178214] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 > > [ 12.198133] Intel(R) Wireless WiFi driver for Linux > > -- > > [ 12.498714] iwlwifi 0000:02:00.0 wlp2s0: renamed from wlan0 > > [ 13.248848] input: HDA Intel PCH Headphone Mic as > /devices/pci0000:00/0000:00:1f.3/sound/card0/input12 > > [ 13.248908] input: HDA Intel PCH HDMI/DP,pcm=3 as > /devices/pci0000:00/0000:00:1f.3/sound/card0/input13 > > [ 13.248952] input: HDA Intel PCH HDMI/DP,pcm=7 as > /devices/pci0000:00/0000:00:1f.3/sound/card0/input14 > > [ 13.248995] input: HDA Intel PCH HDMI/DP,pcm=8 as > /devices/pci0000:00/0000:00:1f.3/sound/card0/input15 > > [ 13.249039] input: HDA Intel PCH HDMI/DP,pcm=9 as > /devices/pci0000:00/0000:00:1f.3/sound/card0/input16 > > [ 13.249084] input: HDA Intel PCH HDMI/DP,pcm=10 as > /devices/pci0000:00/0000:00:1f.3/sound/card0/input17 > > [ 13.494851] EXT4-fs (sda2): mounted filesystem with ordered data > mode. Opts: (null) > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

1 0

Re: [alsa-devel] [RFC] ASOC: Intel: Skylake: Broken HDaudio controller/link initialization sequences
by Takashi Iwai 11 Jan '19

11 Jan '19

[ Corrected the ML address ] On Wed, 09 Jan 2019 00:28:30 +0100, Pierre-Louis Bossart wrote: > > Hi, > > this is not a patch but a request for comments/feedback on the HDAudio > controller/link initialization sequences. After the v4.20 merge window > snafu w/ the HDaudio detection, I spent quite a bit of time to figure > out why all our HDaudio development devices worked well with the > Skylake driver, and why Linus' laptop and others didn't due to an HDMI > initialization issue. Hans de Goede was kind enough to give me SSH > remote access to his device and while I don't have a formal fix for > now I found a couple of issues that really need to be fixed so that > the Skylake driver works by design and not by accident. > > See below the programming sequences for the legacy and Intel/Skylake > drivers. There are obvious discrepancies or broken sequences such as > :- > > 1. the i915 component and display power being set in a work > queue. Forcing the init to be done upfront fixes the HDMI detection > issue on Hans' laptop with the correct codec_mask detected. This is hairy, and likely some timing issue... > 2. a call to snd_hdac_bus_reset_link() that doesn't seem to be useful > (done twice) Better to avoid the unneeded call of this, yeah. > 3. an initialization of ML (multi-link) registers *before* reading ML > capabilities on Skylake (only meaningful in the second init) and not > done in the legacy. > > 4. Missing setups for multi-link (done in legacy, not done by Skylake > drivers) I have little idea about the ML stuff, sorry. > 5. set_codec_wakeup() only initialized in legacy This might be the cause of the missing HDMI codec. The wakeup handling was added years ago for assuring that the i915 power-well turned on during probe and resume. Looking at the git log, the initial commit was 0a6735215350 ALSA: hda - reset display codec when power on > 6. init/stop/init pattern in Skylake driver. I can't think of any > reason why this is required both in the probe and the probe workqueue. Maybe some workaround for messy timing issue? No much idea, either. thanks, Takashi > I don't have any background for most of this, I only took over support > for HDaudio codecs after the initial contributor moved on, so sharing > all my findings in the hope that others have memories of these > different points. I'll get the same laptop as Linus later this week > and will be able to work more on this, but in the meantime comments > are welcome. > > Thanks > > -Pierre > > azx_create > -- azx_bus_init > ---- snd_hdac_bus_init > -- azx_probe_work > ---- azx_probe_continue > ------ snd_hdac_i915_init(bus) > ------ snd_hdac_display_power(bus, true); > ------ azx_first_init(chip); > -------- snd_hdac_bus_parse_capabilities(bus) > -------- azx_init_streams > ---------- snd_hdac_stream_init > -------- azx_alloc_stream_pages > -------- azx_init_pci(chip) > ---------- update_pci_byte(chip->pci, AZX_PCIREG_TCSEL, 0x07, 0); > -------- hda_intel_init_chip > ---------- snd_hdac_set_codec_wakeup(bus, true); !!! <<< is this needed? > ---------- pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); > ---------- azx_init_chip(chip, full_reset) > ------------ snd_hdac_bus_init_chip(chip, full_reset) > -------------- snd_hdac_bus_reset_link(bus, full_reset); > -------------- azx_int_clear > -------------- snd_hdac_bus_init_cmd_io > -------------- azx_int_enable > <<<< codec_mask 0x5 is correct > ---------- pci_write_config_dword(pci, INTEL_HDA_CGCTL, val) > ---------- snd_hdac_set_codec_wakeup(bus, false); !!! <<< is this needed? > ---------- bxt_reduce_dma_latency !!! <<< is this needed? > ---------- intel_init_lctl !!! <<< is this needed? > ------------ intel_ml_lctl_set_power(chip, 0) > ------------ intel_get_lctl_scf > ------------ intel_get_lctl_scf(chip, 1) > ------ azx_probe_codecs > ------ azx_codec_configure > ------ snd_hdac_display_power(bus, false); > > > skl_probe > -- skl_create > ---- snd_hdac_ext_bus_init > ------ snd_hdac_bus_init > -- skl_first_init > > <<< this sequence makes no sense. the i915/display is not initialized > before reading the codec_masks. enabling the usual sequence of > snd_hdac_i915_init > and snd_hdac_display_power gives the right codec_mask > > ---- snd_hdac_bus_reset_link !!! <<< is this needed, it's done later > as part of skl_init_chip? > ------ snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); > ------ snd_hdac_bus_enter_link_reset > ------ usleep_range(500, 1000); > ------ snd_hdac_bus_exit_link_reset > ------ usleep_range(1000, 1200); > ------ snd_hdac_chip_updatel(bus, GCTL, 0, AZX_GCTL_UNSOL); > ------ snd_hdac_chip_readw(bus, STATESTS); > <<<< codec_mask 0x1 is wrong, HDMI not detected > ---- snd_hdac_bus_parse_capabilities > ---- snd_hdac_ext_stream_init_all > ---- skl_init_pci > ------ skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0); > ---- skl_init_chip > ------ skl_enable_miscbdcge(bus->dev, false); > ------ snd_hdac_bus_init_chip(bus, full_reset); > -------- snd_hdac_bus_reset_link(bus, full_reset); > -------- azx_int_clear > -------- snd_hdac_bus_init_cmd_io > -------- azx_int_enable > <<< next line makes no sense, it's done before the ML capabilities are read > ------ bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); > !!! <<< is this needed? > ------ skl_enable_miscbdcge(bus->dev, true); > <<< codec_mask 0x1 is wrong > -- skl_nhlt_init > -- skl_init_dsp > ---- snd_hdac_ext_bus_ppcap_enable(bus, true); > ---- snd_hdac_ext_bus_ppcap_int_enable(bus, true); > -- snd_hdac_ext_bus_get_ml_capabilities > -- snd_hdac_bus_stop_chip > ---- azx_int_disable(bus); > ---- azx_int_clear(bus); > ---- snd_hdac_bus_stop_cmd_io(bus); > -- skl_probe_work > ---- skl_i915_init <<< this needs to be done earlier > ------ snd_hdac_i915_init > ------ snd_hdac_display_power > ---- skl_init_chip <<< why do we need to re-initialize again? > ------ skl_enable_miscbdcge(bus->dev, false); > ------ snd_hdac_bus_init_chip(bus, full_reset); > -------- snd_hdac_bus_reset_link(bus, full_reset); > -------- azx_int_clear > -------- snd_hdac_bus_init_cmd_io > -------- azx_int_enable > ------ bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); > ------ skl_enable_miscbdcge(bus->dev, true); > ---- skl_codec_create > ------- snd_hdac_bus_stop_chip > ------- skl_init_chip (on error) > ---- skl_platform_register > ---- skl_machine_device_register > ---- snd_hdac_ext_bus_link_put > ---- snd_hdac_display_power(bus, false); >

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix CM6206 register definitions
by Amadeusz Sławiński 11 Jan '19

11 Jan '19

fix typo after a recent commit causing headphones to have no sound Fixes: ad43d528a7ac (ALSA: usb-audio: Define registers for CM6206) Signed-off-by: Amadeusz Sławiński <amade(a)asmblr.net> --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 96340f23f86d..ebbadb3a7094 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -768,7 +768,7 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) * REG1: PLL binary search enable, soft mute enable. */ CM6206_REG1_PLLBIN_EN | - CM6206_REG1_SOFT_MUTE_EN | + CM6206_REG1_SOFT_MUTE_EN, /* * REG2: enable output drivers, * select front channels to the headphone output, -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: core: Fix deferral of machine drivers
by Jon Hunter 11 Jan '19

11 Jan '19

Commit daecf46ee0e5 ("ASoC: soc-core: use snd_soc_dai_link_component for platform") added a new member to the snd_soc_dai_link structure for storing a pointer for a platform link component. The pointer for this platform link component was allocated, if not already populated by the machine driver, using devm_kzalloc() such that the memory would be automatically freed on error or removal of the soundcard. However, this introduced a new problem, because if the probing of the soundcard is deferred, then although the memory allocated for the platform link component is freed, if the snd_soc_dai_link structure is declared statically by the machine driver, then the pointer in the DAI link structure will never be clearer. This means that when the soundcard is probed again, memory for the platform link component will not be allocated again because the address of the pointer was not cleared and this causes sound core to access memory that is no longer valid. In most cases this causes the following error condition to be triggered and causes probing the soundcard to fail. tegra-snd-sgtl5000 sound: ASoC: Both platform name/of_node are set for sgtl5000 Unfortunately, because this platform link component is allocated before the DAI links are added to the soundcard, there is no easy way to clear this pointer on teardown if an error occurs. The pointer for this platform link component was added for future proofing and communalising the structures for storing various data. Although a machine driver maybe used by more than one platform and so this platform data may vary from platform to platform, there is only ever a single instance for a given platform. Therefore, rather than dynamically allocate the platform link component structure, make it a static member of the snd_soc_dai_link to fix the problem. It should be noted that if the platform_name of platform_of_node members of the snd_soc_dai_link structure are populated, these will always be used regardless of if the new platform.name or platform.of_node members are populated. Fixes: daecf46ee0e5 ("ASoC: soc-core: use snd_soc_dai_link_component for platform") Reported-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- include/sound/simple_card_utils.h | 2 +- include/sound/soc.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +++- sound/soc/generic/simple-card-utils.c | 4 ++-- sound/soc/generic/simple-card.c | 6 ++++-- sound/soc/soc-core.c | 18 +++++++----------- 6 files changed, 18 insertions(+), 18 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 6d69ed2bd7b1..6d5842c3c09f 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -75,7 +75,7 @@ void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); &dai_link->codec_dai_name, \ list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, dai_link->platform, \ + asoc_simple_card_parse_dai(node, &dai_link->platform, \ &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7..8b7ffc60006a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -925,7 +925,7 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; - struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link_component platform; int id; /* optional ID for machine driver link identification */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3ec96cdc683b..e961d45ce141 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -687,7 +687,9 @@ static int graph_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platform.name = dai_props[i].platform.name; + dai_link[i].platform.of_node = dai_props[i].platform.of_node; + dai_link[i].platform.dai_name = dai_props[i].platform.dai_name; } priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 336895f7fd1e..74910c7841ec 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -397,8 +397,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (!dai_link->platform->of_node) - dai_link->platform->of_node = dai_link->cpu_of_node; + if (!dai_link->platform.of_node) + dai_link->platform.of_node = dai_link->cpu_of_node; return 0; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 479de236e694..b6402e09bba2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -732,7 +732,9 @@ static int simple_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platform.name = dai_props[i].platform.name; + dai_link[i].platform.of_node = dai_props[i].platform.of_node; + dai_link[i].platform.dai_name = dai_props[i].platform.dai_name; } priv->dai_props = dai_props; @@ -782,7 +784,7 @@ static int simple_probe(struct platform_device *pdev) codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; - platform = dai_link->platform; + platform = &dai_link->platform; platform->name = cinfo->platform; card->name = (cinfo->card) ? cinfo->card : cinfo->name; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..466099995e44 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -915,7 +915,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* find one from the set of registered platforms */ for_each_component(component) { - if (!snd_soc_is_matching_component(dai_link->platform, + if (!snd_soc_is_matching_component(&dai_link->platform, component)) continue; @@ -1026,7 +1026,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link_component *platform = dai_link->platform; + struct snd_soc_dai_link_component *platform = &dai_link->platform; /* * FIXME @@ -1034,14 +1034,10 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { - platform = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai_link_component), - GFP_KERNEL); - if (!platform) - return -ENOMEM; + if (dai_link->platform_name || dai_link->platform_of_node) { + dev_dbg(card->dev, + "ASoC: Defaulting to legacy platform data!\n"); - dai_link->platform = platform; platform->name = dai_link->platform_name; platform->of_node = dai_link->platform_of_node; platform->dai_name = NULL; @@ -1123,7 +1119,7 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform->name && link->platform->of_node) { + if (link->platform.name && link->platform.of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); @@ -1921,7 +1917,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platform->name = component->name; + dai_link->platform.name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; -- 2.7.4

3 7

[alsa-devel] [PATCH 1/5] ALSA: hda: add verbs for stripe control
by Sameer Pujar 11 Jan '19

11 Jan '19

Controllers can support multiple Serial Data Out(SDO) lines, for extended outbound bandwidth, to pump data to all codecs on the link. Codecs can sample data present on SDO. Add verbs AC_VERB_GET_STRIPE_CONTROL and AC_VERB_SET_STRIPE_CONTROL These can be used to program usage of SDO lines for codec. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Mohan Kumar D <mkumard(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- include/sound/hda_verbs.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index 2a8573a..e36b775 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -66,6 +66,7 @@ enum { #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 +#define AC_VERB_GET_STRIPE_CONTROL 0x0f24 #define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d #define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e #define AC_VERB_GET_HDMI_ELDD 0x0f2f @@ -110,6 +111,7 @@ enum { #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f #define AC_VERB_SET_EAPD 0x788 #define AC_VERB_SET_CODEC_RESET 0x7ff +#define AC_VERB_SET_STRIPE_CONTROL 0x724 #define AC_VERB_SET_CVT_CHAN_COUNT 0x72d #define AC_VERB_SET_HDMI_DIP_INDEX 0x730 #define AC_VERB_SET_HDMI_DIP_DATA 0x731 -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 9

[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: Select REGMAP_MMIO to fix build error
by Axel Lin 11 Jan '19

11 Jan '19

Fix below build error: ERROR: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-msm8916-digital.ko] undefined! Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 87cb9c51e6f5..45884859229f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -703,6 +703,7 @@ config SND_SOC_MAX9860 config SND_SOC_MSM8916_WCD_ANALOG tristate "Qualcomm MSM8916 WCD Analog Codec" depends on SPMI || COMPILE_TEST + select REGMAP_MMIO config SND_SOC_MSM8916_WCD_DIGITAL tristate "Qualcomm MSM8916 WCD DIGITAL Codec" -- 2.17.1

1 0

[alsa-devel] [PATCH v2] ASoC: soc-core: defer card probe until all component is added to list
by Rohit kumar 11 Jan '19

11 Jan '19

From: Ajit Pandey <ajitp(a)codeaurora.org> DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Changes since v1: Moved component find logic to soc-core instead of machine driver as suggested by Mark. sound/soc/soc-core.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3e..eec92f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 6

[alsa-devel] [PATCH - ALSA JACK plugin 1/2] jack: Replacing jack->channels with jack->num_ports
by laxmi.devi＠in.bosch.com 11 Jan '19

11 Jan '19

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> As jack->num_ports and jack->channels hold the same values, jack->channels is redundant and hence removed. Sanity check is added in prepare, to check if io->Channels is same as jack->num_ports. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index af2136e..d723b17 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -48,7 +48,6 @@ typedef struct { unsigned int sample_bits; snd_pcm_uframes_t min_avail; - unsigned int channels; snd_pcm_channel_area_t *areas; jack_port_t **ports; @@ -263,6 +262,12 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) snd_pcm_sw_params_t *swparams; int err; + if (io->channels != jack->num_ports) { + SNDERR("Channel count %d not equal to no. of ports %d in JACK", + io->channels, jack->num_ports); + return -EINVAL; + } + jack->hw_ptr = 0; jack->xrun_detected = false; @@ -381,7 +386,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) unsigned int psize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * - jack->channels); + jack->num_ports); unsigned int i; int err; @@ -398,7 +403,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_FORMAT, 1, &format)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_CHANNELS, - jack->channels, jack->channels)) < 0 || + jack->num_ports, jack->num_ports)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_RATE, rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, @@ -485,8 +490,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return err; } - jack->channels = jack->num_ports; - if (jack->channels == 0) { + if (jack->num_ports == 0) { SNDERR("define the %s_ports section", stream == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture"); snd_pcm_jack_free(jack); @@ -514,7 +518,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -ENOENT; } - jack->areas = calloc(jack->channels, sizeof(snd_pcm_channel_area_t)); + jack->areas = calloc(jack->num_ports, sizeof(snd_pcm_channel_area_t)); if (! jack->areas) { snd_pcm_jack_free(jack); return -ENOMEM; -- 2.7.4

2 2

[alsa-devel] [PATCH - ALSA JACK plugin 1/2] jack: Replacing jack->channels with jack->num_ports
by laxmi.devi＠in.bosch.com 11 Jan '19

11 Jan '19

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> As jack->num_ports and jack->channels hold the same values, jack->channels is redundant and hence removed. Sanity check is added in prepare, to check if io->Channels is same as jack->num_ports. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index af2136e..d723b17 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -48,7 +48,6 @@ typedef struct { unsigned int sample_bits; snd_pcm_uframes_t min_avail; - unsigned int channels; snd_pcm_channel_area_t *areas; jack_port_t **ports; @@ -263,6 +262,12 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) snd_pcm_sw_params_t *swparams; int err; + if (io->channels != jack->num_ports) { + SNDERR("Channel count %d not equal to no. of ports %d in JACK", + io->channels, jack->num_ports); + return -EINVAL; + } + jack->hw_ptr = 0; jack->xrun_detected = false; @@ -381,7 +386,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) unsigned int psize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * - jack->channels); + jack->num_ports); unsigned int i; int err; @@ -398,7 +403,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_FORMAT, 1, &format)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_CHANNELS, - jack->channels, jack->channels)) < 0 || + jack->num_ports, jack->num_ports)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_RATE, rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, @@ -485,8 +490,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return err; } - jack->channels = jack->num_ports; - if (jack->channels == 0) { + if (jack->num_ports == 0) { SNDERR("define the %s_ports section", stream == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture"); snd_pcm_jack_free(jack); @@ -514,7 +518,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -ENOENT; } - jack->areas = calloc(jack->channels, sizeof(snd_pcm_channel_area_t)); + jack->areas = calloc(jack->num_ports, sizeof(snd_pcm_channel_area_t)); if (! jack->areas) { snd_pcm_jack_free(jack); return -ENOMEM; -- 2.7.4

1 1

[alsa-devel] [PATCH] ASoC: soc-core: add soc_cleanup_component()
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to cleanup component when soc_probe_component() was failed, or when soc_remove_component() was called. But they are cleanuping component on each way. Same code in many places makes code un-understandable. This patch adds new soc_cleanup_component() and call it from snd_probe_component() and snd_remove_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2aa9f95..e0a326f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -939,19 +939,23 @@ static int soc_bind_dai_link(struct snd_soc_card *card, return -EPROBE_DEFER; } +static void soc_cleanup_component(struct snd_soc_component *component) +{ + list_del(&component->card_list); + soc_cleanup_component_debugfs(component); + component->card = NULL; + module_put(component->dev->driver->owner); +} + static void soc_remove_component(struct snd_soc_component *component) { if (!component->card) return; - list_del(&component->card_list); - if (component->driver->remove) component->driver->remove(component); - soc_cleanup_component_debugfs(component); - component->card = NULL; - module_put(component->dev->driver->owner); + soc_cleanup_component(component); } static void soc_remove_dai(struct snd_soc_dai *dai, int order) @@ -1333,6 +1337,7 @@ static int soc_probe_component(struct snd_soc_card *card, component->card = card; dapm->card = card; INIT_LIST_HEAD(&dapm->list); + INIT_LIST_HEAD(&component->card_list); soc_set_name_prefix(card, component); soc_init_component_debugfs(component); @@ -1395,12 +1400,9 @@ static int soc_probe_component(struct snd_soc_card *card, /* see for_each_card_components */ list_add(&component->card_list, &card->component_dev_list); - return 0; - err_probe: - soc_cleanup_component_debugfs(component); - component->card = NULL; - module_put(component->dev->driver->owner); + if (ret < 0) + soc_cleanup_component(component); return ret; } -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc-core: reduce if/else nest on soc_probe_link_dais
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Deep nested code is not readable. Let's reduce if/else nest. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 37 +++++++++++++++++-------------------- 1 file changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f3fe476..d20509e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1584,27 +1584,24 @@ static int soc_probe_link_dais(struct snd_soc_card *card, dai_link->stream_name); return ret; } - } else { - - if (!dai_link->params) { - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; - } - ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); - if (ret < 0) - return ret; - ret = soc_link_dai_pcm_new(rtd->codec_dais, - rtd->num_codecs, rtd); - if (ret < 0) - return ret; - } else { - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + } else if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; } + ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); + if (ret < 0) + return ret; + ret = soc_link_dai_pcm_new(rtd->codec_dais, + rtd->num_codecs, rtd); + if (ret < 0) + return ret; + } else { + INIT_DELAYED_WORK(&rtd->delayed_work, + codec2codec_close_delayed_work); } return 0; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc-core: merge card resources cleanup method
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to cleanup card resources when snd_soc_instantiate_card() was failed, or when snd_soc_unbind_card() was called. But they are cleanuping card resources on each way. Same code in many places makes code un-understandable. This patch reuses soc_cleanup_card_resources() for cleanuping code resource. Then, it makes avoiding cleanup order. It will be called from snd_soc_instantiate_card() and snd_soc_unbind_card(). Then, original soc_cleanup_card_resources() included snd_soc_flush_all_delayed_work(), but it is now separated. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 103 +++++++++++++++++++++------------------------------ 1 file changed, 43 insertions(+), 60 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8b661eb..f3fe476 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1980,6 +1980,29 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) } } +static int soc_cleanup_card_resources(struct snd_soc_card *card) +{ + /* free the ALSA card at first; this syncs with pending operations */ + if (card->snd_card) + snd_card_free(card->snd_card); + + /* remove and free each DAI */ + soc_remove_dai_links(card); + soc_remove_pcm_runtimes(card); + + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + + snd_soc_dapm_free(&card->dapm); + soc_cleanup_card_debugfs(card); + + /* remove the card */ + if (card->remove) + card->remove(card); + + return 0; +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1989,6 +2012,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + card->dapm.bias_level = SND_SOC_BIAS_OFF; + card->dapm.dev = card->dev; + card->dapm.card = card; + list_add(&card->dapm.list, &card->dapm_list); + /* check whether any platform is ignore machine FE and using topology */ soc_check_tplg_fes(card); @@ -1996,14 +2024,14 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) for_each_card_prelinks(card, i, dai_link) { ret = soc_bind_dai_link(card, dai_link); if (ret != 0) - goto base_error; + goto probe_end; } /* bind aux_devs too */ for (i = 0; i < card->num_aux_devs; i++) { ret = soc_bind_aux_dev(card, i); if (ret != 0) - goto base_error; + goto probe_end; } /* add predefined DAI links to the list */ @@ -2017,16 +2045,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: can't create sound card for card %s: %d\n", card->name, ret); - goto base_error; + goto probe_end; } soc_init_card_debugfs(card); - card->dapm.bias_level = SND_SOC_BIAS_OFF; - card->dapm.dev = card->dev; - card->dapm.card = card; - list_add(&card->dapm.list, &card->dapm_list); - #ifdef CONFIG_DEBUG_FS snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); #endif @@ -2048,7 +2071,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (card->probe) { ret = card->probe(card); if (ret < 0) - goto card_probe_error; + goto probe_end; } /* probe all components used by DAI links on this card */ @@ -2059,7 +2082,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2067,7 +2090,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe auxiliary components */ ret = soc_probe_aux_devices(card); if (ret < 0) - goto probe_dai_err; + goto probe_end; /* * Find new DAI links added during probing components and bind them. @@ -2079,10 +2102,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; ret = soc_bind_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; } /* probe all DAI links on this card */ @@ -2093,7 +2116,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2140,7 +2163,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: %s late_probe() failed: %d\n", card->name, ret); - goto probe_aux_dev_err; + goto probe_end; } } @@ -2150,33 +2173,17 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: failed to register soundcard %d\n", ret); - goto probe_aux_dev_err; + goto probe_end; } card->instantiated = 1; dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); - mutex_unlock(&card->mutex); - mutex_unlock(&client_mutex); - - return 0; -probe_aux_dev_err: - soc_remove_aux_devices(card); - -probe_dai_err: - soc_remove_dai_links(card); - -card_probe_error: - if (card->remove) - card->remove(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - snd_card_free(card->snd_card); +probe_end: + if (ret < 0) + soc_cleanup_card_resources(card); -base_error: - soc_remove_pcm_runtimes(card); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2205,31 +2212,6 @@ static int soc_probe(struct platform_device *pdev) return snd_soc_register_card(card); } -static int soc_cleanup_card_resources(struct snd_soc_card *card) -{ - /* make sure any delayed work runs */ - snd_soc_flush_all_delayed_work(card); - - /* free the ALSA card at first; this syncs with pending operations */ - snd_card_free(card->snd_card); - - /* remove and free each DAI */ - soc_remove_dai_links(card); - soc_remove_pcm_runtimes(card); - - /* remove auxiliary devices */ - soc_remove_aux_devices(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - - /* remove the card */ - if (card->remove) - card->remove(card); - - return 0; -} - /* removes a socdev */ static int soc_remove(struct platform_device *pdev) { @@ -2795,6 +2777,7 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) if (card->instantiated) { card->instantiated = false; snd_soc_dapm_shutdown(card); + snd_soc_flush_all_delayed_work(card); soc_cleanup_card_resources(card); if (!unregister) list_add(&card->list, &unbind_card_list); -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc-core: add new snd_soc_flush_all_delayed_work()
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-core is calling flush_delayed_work() many times for same purpose. Same code in many places makes code un-understandable. This patch adds new snd_soc_flush_all_delayed_work() for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0934b36..8b661eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -425,6 +425,14 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); +static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + for_each_card_rtds(card, rtd) + flush_delayed_work(&rtd->delayed_work); +} + static void codec2codec_close_delayed_work(struct work_struct *work) { /* @@ -494,8 +502,7 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { @@ -2200,11 +2207,8 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd; - /* make sure any delayed work runs */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); /* free the ALSA card at first; this syncs with pending operations */ snd_card_free(card->snd_card); @@ -2247,8 +2251,7 @@ int snd_soc_poweroff(struct device *dev) * Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - for_each_card_rtds(card, rtd) - flush_delayed_work(&rtd->delayed_work); + snd_soc_flush_all_delayed_work(card); snd_soc_dapm_shutdown(card); -- 2.7.4

1 0

[alsa-devel] ALSA 1.1.8 Release - POLLRDNORM undefined
by Jay Foster 11 Jan '19

11 Jan '19

I am attempting to build the 1.1.8 release of alsa-utils and getting an error about POLLRDNORM (and others) undefined. This error comes from axfer/waiter-select.c. axfer/waiter-select.c includes "waiter.h", which includes "poll.h". This build is for a linux target. poll.h includes sys/poll.h which includes bits/poll.h, which only defines POLLRDNORM (and others) when __USE_XOPEN is defined (which it is not). However, linux/poll.h includes asm/poll.h which includes asm-generic/poll.h which does define POLLRDNORM (and others). Should I change my build to define__USE_XOPEN or should I patch waiter.h to include linux/poll.h instead of just poll.h? Jay

4 10

[alsa-devel] axfer: FTBFS of axfer with glibc-2.11 or before due to missing definition of POLLRDNORM
by GitHub issues - edited 11 Jan '19

11 Jan '19

alsa-project/alsa-utils issue #9 was edited from takaswie: Reported by Jay Foster in alsa-devel list. http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144024.ht… The v1.1.8 release brings FTBFS of axfer in his legacy toolchain (glibc v2.9) for arm platform, due to missing definition `POLLRDNORM` and the others, used in `axfer/waiter-select.c`. An option of `-D_GNU_SOURCE` for CFLAGS can suppress the issue. With glibc v2.22, no issues. During development period of axfer, this issue has never been got. More recent version of glibc (v2.28) was used and the macro is available without any explicit options. Issue URL : https://github.com/alsa-project/alsa-utils/issues/9 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] axfer: FTBFS of axfer with glibc-2.11 or before due to missing definition of POLLRDNORM
by GitHub issues - opened 11 Jan '19

11 Jan '19

alsa-project/alsa-utils issue #9 was opened from takaswie: Reported by Jay Foster in alsa-devel list. http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144024.ht… The v1.1.8 release brings FTBFS of axfer in his legacy toolchain (glibc v2.9) for arm platform, due to missing definition `POLLRDNORM`, used in `axfer/waiter-select.c`. An option of `-D_GNU_SOURCE` for CFLAGS can suppress the issue. With glibc v2.22, no issues. During development period of axfer, this issue has never been got. More recent version of glibc (v2.28) was used and the macro is available without any explicit options. Issue URL : https://github.com/alsa-project/alsa-utils/issues/9 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: soc-core: fix init platform memory handling
by Curtis Malainey 11 Jan '19

11 Jan '19

snd_soc_init_platform initializes pointers to snd_soc_dai_link which is statically allocated and it does this by devm_kzalloc. In the event of an EPROBE_DEFER the memory will be freed and the pointers are left dangling. snd_soc_init_platform sees the dangling pointers and assumes they are pointing to initialized memory and does not reallocate them on the second probe attempt which results in a use after free bug since devm has freed the memory from the first probe attempt. Since the intention for snd_soc_dai_link->platform is that it can be set statically by the machine driver we need to respect the pointer in the event we did not set it but still catch dangling pointers. The solution is to add a flag to track whether the pointer was dynamically allocated or not. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 11 ++++++----- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7e..e665f111b0d27 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -985,6 +985,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* + * This driver uses legacy platform naming. Set by the core, machine + * drivers should not modify this value. + */ + unsigned int legacy_platform:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0934b36645b3e..cdcc417c94ca1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { + if (!platform || dai_link->legacy_platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); if (!platform) return -ENOMEM; - dai_link->platform = platform; - platform->name = dai_link->platform_name; - platform->of_node = dai_link->platform_of_node; - platform->dai_name = NULL; + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; } /* if there's no platform we match on the empty platform */ -- 2.20.1.97.g81188d93c3-goog

1 1

[alsa-devel] [PATCH] ASoC: Intel: sst: Delay machine device creation until after initialization
by Stephan Gerhold 10 Jan '19

10 Jan '19

Right now, the machine devices are created early, before the SST context is initialized. This means that SST might not be fully initialized if sst_acpi_probe() fails later on (e.g. after sst_platform_get_resources() if the IRQ does not exist). However, at least sst-mfld-platform assumes that sst_register_dsp() was called to set the (global) "sst" device pointer, which happens only later in sst_acpi_probe() when sst_context_init() is called. This may cause a NULL pointer dereference later when the ALSA device is first opened: BUG: unable to handle kernel NULL pointer dereference at 0000000000000008 PGD 0 P4D 0 Oops: 0000 [#1] PREEMPT SMP PTI CPU: 0 PID: 790 Comm: pulseaudio Not tainted 4.20.0-rc6-mainline-00161-g6531e115b7ab #1 Hardware name: ASUSTeK COMPUTER INC. ME176C/ME176C, BIOS 5.6.5 09/16/2015 RIP: 0010:sst_handle_vb_timer+0x61/0x1b0 [snd_soc_sst_atom_hifi2_platform] Code: 44 24 04 e9 84 00 00 00 31 c9 c7 04 24 ff ff ff ff 66 89 4c 24 06 84 db 0f 84 90 00 00 00 48 8b 05 c4 23 01 00 be 01 00 00 00 <48> 8b 78 08 48 8b 40 10 48 8b 40 48 e8 2e 5e d7 f0 89 c3 85 c0 78 RSP: 0018:ffff9d968099fa30 EFLAGS: 00010202 RAX: 0000000000000000 RBX: 0000000000000001 RCX: 0000000000000000 RDX: 0000000080000001 RSI: 0000000000000001 RDI: 00000000ffffffff RBP: ffff968d33384618 R08: 0000000000000001 R09: 00000000000002e3 R10: ffff968d333a0800 R11: 0000000000000000 R12: ffff968d34bc7c00 R13: ffff968d333a3eb0 R14: 0000000000000001 R15: ffff968d333a08c0 FS: 00007f63a7e7b200(0000) GS:ffff968d37600000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000008 CR3: 00000000297b8000 CR4: 00000000001006f0 Call Trace: sst_enable_ssp+0x24/0x40 [snd_soc_sst_atom_hifi2_platform] soc_pcm_open+0xeb/0x960 [snd_soc_core] ? __debugfs_create_file+0xcd/0x120 dpcm_be_dai_startup+0x183/0x3c0 [snd_soc_core] dpcm_fe_dai_open+0x10c/0xab0 [snd_soc_core] snd_pcm_open_substream+0x7f/0x140 [snd_pcm] snd_pcm_open+0xe6/0x220 [snd_pcm] ? wake_up_q+0x70/0x70 snd_pcm_playback_open+0x3d/0x70 [snd_pcm] chrdev_open+0xa3/0x1b0 ? cdev_put.part.0+0x20/0x20 do_dentry_open+0x12f/0x350 path_openat+0x2d1/0x14e0 ? inotify_handle_event+0x17b/0x1e0 do_filp_open+0x93/0x100 ? snd_card_file_remove+0x14b/0x170 [snd] ? __check_object_size+0x102/0x189 ? _raw_spin_unlock+0x12/0x30 do_sys_open+0x186/0x210 do_syscall_64+0x55/0x160 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x7f63a992f4c2 Code: 25 00 00 41 00 3d 00 00 41 00 74 4c 48 8d 05 85 70 0d 00 8b 00 85 c0 75 6d 89 f2 b8 01 01 00 00 48 89 fe bf 9c ff ff ff 0f 05 <48> 3d 00 f0 ff ff 0f 87 a2 00 00 00 48 8b 4c 24 28 64 48 33 0c 25 RSP: 002b:00007ffe71196b70 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 0000000000080802 RCX: 00007f63a992f4c2 RDX: 0000000000080802 RSI: 00007ffe71196d20 RDI: 00000000ffffff9c RBP: 00000000ffffffff R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007ffe71196c00 R13: 0000000000000004 R14: 00007ffe71196d20 R15: 0000558466055a80 CR2: 0000000000000008 ---[ end trace 34534a02650ee26c ]--- This can be avoided if the machine device creation is delayed in sst_acpi_probe() until after sst_context_init(), when sst_register_dsp() is guaranteed to have already been called. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- An other option to fix this would be to add proper NULL checks in the probe method of sst-mfld-platform and/or sst_enable_ssp(). Maybe this should be done additionally, but at least in my opinion there is not much point in registering the machine devices if they end up being broken anyway. sound/soc/intel/atom/sst/sst_acpi.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index ac542535b9d5..493d32923815 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -345,6 +345,18 @@ static int sst_acpi_probe(struct platform_device *pdev) mach->mach_params.acpi_ipc_irq_index = pdata->res_info->acpi_ipc_irq_index; + /* Fill sst platform data */ + ctx->pdata = pdata; + strcpy(ctx->firmware_name, mach->fw_filename); + + ret = sst_platform_get_resources(ctx); + if (ret) + return ret; + + ret = sst_context_init(ctx); + if (ret < 0) + return ret; + plat_dev = platform_device_register_data(dev, pdata->platform, -1, NULL, 0); if (IS_ERR(plat_dev)) { @@ -365,18 +377,6 @@ static int sst_acpi_probe(struct platform_device *pdev) return PTR_ERR(mdev); } - /* Fill sst platform data */ - ctx->pdata = pdata; - strcpy(ctx->firmware_name, mach->fw_filename); - - ret = sst_platform_get_resources(ctx); - if (ret) - return ret; - - ret = sst_context_init(ctx); - if (ret < 0) - return ret; - sst_configure_runtime_pm(ctx); platform_set_drvdata(pdev, ctx); return ret; -- 2.20.0

2 4

[alsa-devel] Pioneer DDJ-SX3 usb audio quirks
by ard 10 Jan '19

10 Jan '19

Hi Guys, I am trying to get my pioneer to work. Everything works except for the PCM, which seems to be nothing more than an isosynchronous stream of 12 channels S32_LE output and 10 channels of S32_LE input. (Determined by usbpcap data patterns and mixxx channel selection). I tried to make a quirk out of it, but I got puzzled by the ifnum. The PCM channels are both endpoints of a single interface, so I wonder how I should write that down. This is the lsusb -v: Bus 001 Device 005: ID 2b73:0023 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x2b73 idProduct 0x0023 bcdDevice 1.02 iManufacturer 1 Pioneer DJ Corporation iProduct 2 PIONEER DJ DDJ-SX3 iSerial 3 201706 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 137 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 65 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 3 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 3 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 2 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 2 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x5f70 (Bus Powered) Debug Mode And here is my preliminary quirk (yes, I need to remove the octo capture ;-) ) commit 55a009f7e77acf6d75a297e0a01839fdeb580e65 Author: Ard van Breemen <ard(a)kwaak.net> Date: Wed Jan 9 15:36:46 2019 +0100 try 1 SX3 diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 15cbe2565703..8493dbe536bd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3324,5 +3324,60 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * PIONEER DJ DDJ-SX3 + * PCM is 12 channels out, 10 channels in @ 44.1 fixed + * interface 0, vendor class alt setting 1 for endpoints 5 and 0x86 + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0023), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "", */ + /* .product_name = "OCTO-CAPTURE", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 10, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC ---- Since this is a copy paste of the roland octo capture with fixes for the numbers I know, I am still wondering about this: Should I use .ifnum=1 for every snd_usb_audio_quirk? And in eacht audioformat should I use .iface = 0? I also wonder what the .altset_idx means. Regards, Ard van Breemen -- .signature not found

2 2

[alsa-devel] [PATCH] ASoC: soc-core: Hold client_mutex around soc_init_dai_link()
by Rohit kumar 10 Jan '19

10 Jan '19

soc_init_dai_link() calls soc_find_component() which needs to be within client_mutex lock. Add client_mutex lock around soc_init_dai_link() in snd_soc_register_card() to avoid lockdep warning. Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eec92f1..0934b36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,6 +1027,7 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; + /* * FIXME * @@ -2754,15 +2755,18 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + mutex_lock(&client_mutex); for_each_card_prelinks(card, i, link) { ret = soc_init_dai_link(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + mutex_unlock(&client_mutex); return ret; } } + mutex_unlock(&client_mutex); dev_set_drvdata(card->dev, card); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[alsa-devel] [PATCH -next] ASoC: Intel: bytcht_es8316: Set correct platform drvdata in snd_byt_cht_es8316_mc_probe()
by Wei Yongjun 10 Jan '19

10 Jan '19

The snd_byt_cht_es8316_mc_remove() use the platform drvdata as a type of 'struct byt_cht_es8316_private', but snd_byt_cht_es8316_mc_probe() set it to 'struct snd_soc_card', fix to pass correct platform drvdata to platform_set_drvdata(). Fixes: 0d3e91da0750 ("ASoC: Intel: bytcht_es8316: Add external speaker mux support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/intel/boards/bytcht_es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index cdf2061..42111d8 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -538,7 +538,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) dev_err(dev, "snd_soc_register_card failed: %d\n", ret); return ret; } - platform_set_drvdata(pdev, &byt_cht_es8316_card); + platform_set_drvdata(pdev, priv); return 0; }

5 7

[alsa-devel] only 1.00 dB of playback volume change
by rm.riches＠jacob21819.net 10 Jan '19

10 Jan '19

(I sent this first to alsa-user and was instructed to send it to alsa-devel, as well.) With a new OS installation on a pre-existing machine, some of the USB audio devices now have only 1.00 dB of playback volume adjustment, which of course is not enough to be terribly useful. The affected USB devices show up as "Audio Advantage MicroII" in the output of "aplay -L". From LSB, they show up as these: Bus 001 Device 004: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ Audio Controller Bus 001 Device 003: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ Audio Controller I just installed Devuan Ascii on the machine. It's running kernel 4.9.0-8-amd64 and alsa-utils version 1.1.3-1. Doing "amixer -c 1 contents" or "amixer -c 2 contents" shows this for playback volume: numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=-1.00dB,max=0.00dB and numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=77,77 | dBminmax-min=-1.00dB,max=0.00dB It looks like I'm running kernel 4.9.0-8-amd64 and alsa-utils version 1.1.3-1. Using alsamixer to run the controls from end to end, I cannot perceive any difference in volume level. In contrast, the other device I use (either HDA Intel or HDA NVidia) has around a 60dB range according to amixer, and running the control in alsamixer from end to end has a huge range of gain. Previously, the exact same hardware was running Debian 7/Wheezy and then Slackware 14.2. With both those operating systems, the USB sound devices had a very adequate volume control range, certainly a whole lot more than 1dB. Being as the USB sound devices are the ones my wife uses, I'd be extremely grateful if somebody could point me a to way (if one exists) to a usefully greater range of volume control. Any suggestions? Assuming this qualifies as a bug, what channel should I use to report it so it can be queued to be fixed? Thanks, Robert

2 1

[alsa-devel] [PATCH v2 0/2] ALSA: HDAudio: enable dynamic selection between legacy and Skylake drivers
by Pierre-Louis Bossart 09 Jan '19

09 Jan '19

Use detection logic based on the PCI class/subclass/prog-if in the HDaudio legacy probe, and add a module parameter to bypass or force the automatic detection should it ever fail (or in case of missing firmware or topology files) This patchset was tested on 1. an SKL HP device with the DSP disabled 2. a WHL Acer Swift3 laptop 3. APL NUC6 4. KBL NUC7 5 reboot tests were done for each device with the DSP presence detected reliably. The "pci_binding" parameter was also tested on the WHL device. The dynamic selection adds about 5ms worst case. [ 2.813741] snd_hda_intel 0000:00:1f.3: The DSP is enabled on this platform, aborting probe [ 2.850588] snd_soc_skl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 The SOF driver will use exactly the same mechanism, and add a mutual exclusion with the Skylake/SST driver to avoid having 3 drivers for the same PCI ID. Two is complicated enough. Changes since v1 (Feedback from Takashi) a) added explanations and renamed Kconfigs to SND_HDA_INTEL_DSP_DETECTION_XYZ b) less verbose dev_info c) fixed pci_binding parameter Pierre-Louis Bossart (2): ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection include/sound/hdaudio.h | 6 ++++ sound/pci/hda/Kconfig | 62 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 54 +++++++++++++++++++++++++---- sound/soc/intel/Kconfig | 6 ++++ sound/soc/intel/skylake/skl.c | 48 ++++++++++++++++++-------- 6 files changed, 155 insertions(+), 23 deletions(-) base-commit: ba26f635bf7253b42b08135152b18a2c8082abb7 -- 2.17.1

3 5

[alsa-devel] ASoC: Intel: sst: Missing IRQ at index 5 on BYT-T device
by Stephan Gerhold 09 Jan '19

09 Jan '19

Hi, I have an Intel Bay Trail (BYT-T) tablet that was originally shipped with Android. With the right quirks, bytcr-rt5640 is working fine, but there is a problem in sst_acpi.c that is preventing it from working with a mainline kernel: Even though this is a BYT-T device, there is no IRQ at index 5 in the ACPI DSDT table. This means that SST will fail to probe, and actually leads to a NULL pointer dereference later when the ALSA device is first opened. (I have submitted a possible solution for this as "[PATCH] ASoC: Intel: sst: Delay machine device creation until after initialization") The correct IRQ is actually located on index 0, just like it is already being used for BYT-CR devices. So if I force is_byt_cr() to return TRUE, everything works as expected. Here is the relevant part from the ACPI DSDT table: Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F28" /* Intel SST Audio DSP */) // _HID: Hardware ID Name (_CID, "80860F28" /* Intel SST Audio DSP */) // _CID: Compatible ID Name (_DDN, "Intel(R) Low Power Audio Controller - 80860F28") // _DDN: DOS Device Name Name (_SUB, "80867270") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (RBUF, ResourceTemplate () { Memory32Fixed (ReadWrite, 0x12345678, // Address Base 0x00200000, // Address Length _Y08) Memory32Fixed (ReadWrite, 0xFE830000, // Address Base 0x00001000, // Address Length _Y09) Memory32Fixed (ReadWrite, 0x55AA55AA, // Address Base 0x00200000, // Address Length _Y0A) Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001D, } }) Unlike many of the other DSDT dumps I've looked at, there is only one interrupt listed. Full ACPI DSDT table is at [1]. Since there is no IRQ at index 5, platform_get_irq will return -ENXIO. Couldn't we fall back to index 0 in this case? I would say that if the seemingly "correct" IRQ at index 5 does not even exist, we still have a better chance of picking the right one if we try the one at index 0. Or we could check the number of interrupts that are actually available. The other option would be some kind of DMI-based quirk, but personally I would prefer to avoid that.. (In my opinion, there is way too much device specific code with the quirks etc already...) Or do you have any other suggestions? Thanks, Stephan [1]: https://github.com/me176c-dev/me176c-acpi/blob/f48c78c11b0819b899f988407b6e…

5 26

[alsa-devel] Applied "ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated" to the asoc tree
by Mark Brown 09 Jan '19

09 Jan '19

The patch ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated 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 04d979d7a7bac2f645cd827ea37e5ffa5b4e1f97 Mon Sep 17 00:00:00 2001 From: b-ak <anur.bhargav(a)gmail.com> Date: Wed, 9 Jan 2019 22:41:21 +0530 Subject: [PATCH] ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated SND_SOC_DAPM_MICBIAS is deprecated, replace it with SND_SOC_DAPM_SUPPLY. MICBIAS voltage wasn't supplied to the microphone with the older SND_SOC_DAPM_MICBIAS widget, hence the microphone wouldn't work. This patch fixes the problem. Signed-off-by: b-ak <anur.bhargav(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 30 +++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic32x4.h | 1 + 2 files changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e2b5a11b16d1..521663d8b585 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -79,6 +79,32 @@ struct aic32x4_priv { struct device *dev; }; +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Change Mic Bias Registor */ + snd_soc_component_update_bits(component, AIC32X4_MICBIAS, + AIC32x4_MICBIAS_MASK, + AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); + printk(KERN_DEBUG "%s: Mic Bias will be turned ON\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, AIC32X4_MICBIAS, + AIC32x4_MICBIAS_MASK, 0); + printk(KERN_DEBUG "%s: Mic Bias will be turned OFF\n", + __func__); + break; + } + + return 0; +} + + static int aic32x4_get_mfp1_gpio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -450,7 +476,9 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, in3r_to_lmixer_controls), - SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", AIC32X4_MICBIAS, 6, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index e9df49edbf19..c2d74025bf4b 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -195,6 +195,7 @@ int aic32x4_remove(struct device *dev); /* AIC32X4_MICBIAS */ #define AIC32X4_MICBIAS_LDOIN BIT(3) #define AIC32X4_MICBIAS_2075V 0x60 +#define AIC32x4_MICBIAS_MASK GENMASK(6, 3) /* AIC32X4_LMICPGANIN */ #define AIC32X4_LMICPGANIN_IN2R_10K 0x10 -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: Boards: move the codec PLL configuration to _init
by mac.chiang＠intel.com 09 Jan '19

09 Jan '19

From: Mac Chiang <mac.chiang(a)intel.com> move the codec PLL to rt5682_codec_init, because codec only need to config the clock source/PLL once. As the result, remove the platform_clock_controls since no need to control clock anymore. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/glk_rt5682_max98357a.c | 45 +++++---------------------- 1 file changed, 7 insertions(+), 38 deletions(-) diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c74c4f1..8f83b18 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -55,39 +55,6 @@ enum { GLK_DPCM_AUDIO_HDMI3_PB, }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop sysclk: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); - if (ret < 0) { - dev_err(card->dev, "can't set codec pll: %d\n", ret); - return ret; - } - } - - if (ret) - dev_err(card->dev, "failed to start internal clk: %d\n", ret); - - return ret; -} - static const struct snd_kcontrol_new geminilake_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = { { "Spk", NULL, "Speaker" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, /* digital mics */ @@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_jack *jack; int ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + /* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); -- 2.7.4

3 2

[alsa-devel] [PATCH v3 0/9] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 09 Jan '19

09 Jan '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. Since the last push there has been some activity getting surround sound working and this is included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. Anyway, I've moved the more controversial patches to the top of the stack as I'm expecting feedback. Have a great Xmas and New Year, BR, CK --- v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Marcus Cooper (9): ASoC: sun4i-i2s: Adjust regmap settings ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Fix offset mask ASoC: sun4i-i2s: Correct divider calculations ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Do not divide clocks when slave ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Add multichannel functionality sound/soc/sunxi/sun4i-i2s.c | 399 ++++++++++++++++++++++++------------ 1 file changed, 267 insertions(+), 132 deletions(-) -- 2.20.1

4 19

Re: [alsa-devel] [PATCH] ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated
by Mark Brown 09 Jan '19

09 Jan '19

On Wed, Jan 09, 2019 at 10:41:21PM +0530, b-ak wrote: > SND_SOC_DAPM_MICBIAS is deprecated, replace it with SND_SOC_DAPM_SUPPLY. > > MICBIAS voltage wasn't supplied to the microphone with the older > SND_SOC_DAPM_MICBIAS widget, hence the microphone wouldn't work. > > This patch fixes the problem. Note that the reason we've not done that already for everything is that the DAPM graph needs updating for the new routing in machine drivers as the two widgets are wired differently. However this is fine here as there aren't any in tree users I can see so I'll apply, thanks!

1 0

[alsa-devel] [PATCH 0/2] ASoC: da7219: Improvements to DAI clocks handling in driver
by Adam Thomson 09 Jan '19

09 Jan '19

This small patchset provides improvements to CCF based clock handling relating to the DAI clocks. In the first patch MCLK is made parent of DAI clocks, if MCLK has been provided to the driver, which means that MCLK will automatically be enabled as a prerequisite to DAI clocks. The second patch adds the ability to read the WCLK rate provided by the codec when DAI clocks are featured. This is more useful than the returned rate simply being the rate of the parent clock, which in this case would be MCLK. Adam Thomson (2): ASoC: da7219: MCLK should be enabled before DAI clocks ASoC: da7219: Add recalc_rate function to return DAI clock rate sound/soc/codecs/da7219.c | 77 ++++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/da7219.h | 1 + 2 files changed, 67 insertions(+), 11 deletions(-) -- 1.9.1

2 5

Re: [alsa-devel] hp_pin was NULL value
by Takashi Iwai 09 Jan '19

09 Jan '19

On Wed, 09 Jan 2019 12:19:14 +0100, Kailang wrote: > > Hi Takashi, > > But spec->init_hook will call at resume back. > alc294_hp_init() function just need to call at one time. As already mentioned, even if it has to be called only once and does something for the hardware, it'll be missing if the machine goes hibernate (S4) and resume. In S4, the machine boots from scratch and switches immediately to the saved image that calls resume callback. What's worse for us is that this switch happens usually in initrd, i.e. before the sound driver gets loaded. So, the machine goes to resume without processing alc294_hp_init(). IOW, even if it's needed only once, you'll have to call it at resume unless it has a significant drawback. (Strictly speaking, we may use thaw PM callback, but the HD-audio codec driver doesn't use it, so there is only resume callback for now.) > And it need to call it before call spec->init_hook. But ALC294 (and ALC700) have no init_hook, so far? > ALC225 also need to call similar procedure. I will post it later. OK, thanks. Takashi > > BR, > Kailang > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年1月9日下午 05:56 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: hp_pin was NULL value > > On Wed, 09 Jan 2019 10:45:25 +0100, > Kailang wrote: > > > > >>But this made me wonder whether we don't need to call this function at resume as well? Currently it's called only at probing. > > Yes, it only call at probing. Just need to call it at boot time. > > > > -----Original Message----- > > From: Takashi Iwai <tiwai(a)suse.de> > > Sent: Wednesday, January 9, 2019 5:43 PM > > To: Kailang <kailang(a)realtek.com> > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > Subject: Re: hp_pin was NULL value > > > > On Wed, 09 Jan 2019 10:31:33 +0100, > > Kailang wrote: > > > > > > Hi Takashi, > > > > > > Could I move the alc294_hp_init(codec) to below line. > > > Because hp_pin = spec->gen.autocfg.hp_pins[0] was null value when alc294_hp_init(codec) at original line. > > > Or move alc269_parse_auto_config() upward. > > > > It looks OK to me. But this made me wonder whether we don't need to call this function at resume as well? Currently it's called only at probing. > > Hrm, but it modifies many COEFs, and are these all preserved with > suspend? For example, there is also hibernation (S4), which is a > switch from the boot without the audio driver initialization. Then > we'll resume from the uninitialized state. I guess other COEF changes > in that area should be moved as the additional init hook as well. > > Actually, if we do call this at resume, the change would be easier, a > patch like below. > > > thanks, > > Takashi > > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -7404,6 +7404,20 @@ static void alc294_hp_init(struct hda_codec *codec) > msleep(50); > } > > +static void alc294_init(struct hda_codec *codec) > +{ > + /* UAJ MIC Vref control by verb */ > + alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); > + alc294_hp_init(codec); > +} > + > +static void alc700_init(struct hda_codec *codec) > +{ > + /* Combo jack auto trigger control */ > + alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); > + alc294_hp_init(codec); > +} > + > /* > */ > static int patch_alc269(struct hda_codec *codec) > @@ -7528,8 +7542,7 @@ static int patch_alc269(struct hda_codec *codec) > 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 */ > - alc294_hp_init(codec); > + spec->init_hook = alc294_init; > break; > case 0x10ec0300: > spec->codec_variant = ALC269_TYPE_ALC300; > @@ -7540,8 +7553,7 @@ static int patch_alc269(struct hda_codec *codec) > case 0x10ec0703: > spec->codec_variant = ALC269_TYPE_ALC700; > spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ > - alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ > - alc294_hp_init(codec); > + spec->init_hook = alc700_init; > break; > > } > > > ------Please consider the environment before printing this e-mail.

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[alsa-devel] pcm_hw bug when calling snd_pcm_sw_params twice
by Jamey Sharp 09 Jan '19

09 Jan '19

I would have reported this to your bug tracker, but I can't find it; the links on alsa-project.org are dead. I've attached a small test program that demonstrates that calling snd_pcm_sw_params twice changes the value of the period_event flag in the sw_params struct, at least on pcm_hw devices. In src/pcm/pcm_hw.c, in snd_pcm_hw_sw_params, there's an early `sw_set_period_event(params, 0)` call. Its effect is not undone if the function returns early. This is easiest to trigger by calling it with unchanged parameters, although there are other early-exit paths with the same problem. I don't actually care about this bug, but since I noticed it while reading the alsa-lib source code trying to figure out what a period event is for, I thought I'd go ahead and report it. Jamey

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[alsa-devel] hp_pin was NULL value
by Kailang 09 Jan '19

09 Jan '19

Hi Takashi, Could I move the alc294_hp_init(codec) to below line. Because hp_pin = spec->gen.autocfg.hp_pins[0] was null value when alc294_hp_init(codec) at original line. Or move alc269_parse_auto_config() upward. BR, Kailang --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7514,7 +7514,6 @@ static int patch_alc269(struct hda_codec *codec) 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 */ - alc294_hp_init(codec); break; case 0x10ec0300: spec->codec_variant = ALC269_TYPE_ALC300; @@ -7526,7 +7525,6 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ - alc294_hp_init(codec); break; } @@ -7553,6 +7551,16 @@ static int patch_alc269(struct hda_codec *codec) if (err < 0) goto error; + switch (codec->core.vendor_id) { + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + case 0x10ec0700: + case 0x10ec0701: + case 0x10ec0703: + alc294_hp_init(codec); + break; + } if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) { err = set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); if (err < 0)

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[alsa-devel] [PATCH 1/2] ALSA: hda: Fix mismatches for register mask and value in hdac controller
by Keyon Jie 09 Jan '19

09 Jan '19

E.g. for azx_int_enable(), we should set both mask and value to be "AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN"(the mask was 0) to enable controller CIE and GIE. We have similar issues on setting AZX_GCTL_RESET and AZX_GCTL_UNSOL, here try to correct all of them. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/hdac_controller.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 74244d8e2909..b2e9454f5816 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -376,7 +376,7 @@ void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus) { unsigned long timeout; - snd_hdac_chip_updateb(bus, GCTL, 0, AZX_GCTL_RESET); + snd_hdac_chip_updateb(bus, GCTL, AZX_GCTL_RESET, AZX_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); while (!snd_hdac_chip_readb(bus, GCTL) && time_before(jiffies, timeout)) @@ -415,7 +415,7 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) } /* Accept unsolicited responses */ - snd_hdac_chip_updatel(bus, GCTL, 0, AZX_GCTL_UNSOL); + snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, AZX_GCTL_UNSOL); /* detect codecs */ if (!bus->codec_mask) { @@ -431,7 +431,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_reset_link); static void azx_int_enable(struct hdac_bus *bus) { /* enable controller CIE and GIE */ - snd_hdac_chip_updatel(bus, INTCTL, 0, AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN); + snd_hdac_chip_updatel(bus, INTCTL, + AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN, + AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN); } /* disable interrupts */ -- 2.17.1

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Re: [alsa-devel] This will solve jack plug and unplug noise
by Kailang 09 Jan '19

09 Jan '19

Hi Takashi, Update ALC225 depop for jack plug and unplug noise. Add this two patch. It will solve jack plug and unplug noise for ALC225 of Dell headset mode. BR, Kailang

2 1

[alsa-devel] [PATCH -next] ALSA: usb-audio: Remove set but not used variable 'first_ch_bits'
by YueHaibing 09 Jan '19

09 Jan '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/usb/mixer.c: In function 'parse_audio_feature_unit': sound/usb/mixer.c:1838:28: warning: variable 'first_ch_bits' set but not used [-Wunused-but-set-variable] It never used since 2.6 Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/usb/mixer.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e7d441d..8ad1a24 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1835,7 +1835,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, { int channels, i, j; struct usb_audio_term iterm; - unsigned int master_bits, first_ch_bits; + unsigned int master_bits; int err, csize; struct uac_feature_unit_descriptor *hdr = _ftr; __u8 *bmaControls; @@ -1926,10 +1926,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, break; } - if (channels > 0) - first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize); - else - first_ch_bits = 0; if (state->mixer->protocol == UAC_VERSION_1) { /* check all control types */

2 1

[alsa-devel] [PATCH 0/2] ASoC: soc-core: add snd_soc_of_lookup_component()
by Rohit kumar 09 Jan '19

09 Jan '19

Platform DAI component probe is not called if it is not present in component list during sound card registration. Add snd_soc_of_lookup_component() to get component info from device_node. This can be used by machine driver to make sure that card components are registered before calling snd_soc_register_card(). Ajit Pandey (2): ASoC: soc-core: add snd_soc_of_lookup_component() ASoC: qcom: defer probe if platform dai is not registered include/sound/soc.h | 1 + sound/soc/qcom/common.c | 9 +++++++++ sound/soc/soc-core.c | 28 ++++++++++++++++++++++++++++ 3 files changed, 38 insertions(+) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 4

[alsa-devel] [PATCH] ASoC: ADAU7002: Add optional delay before start of capture
by Agrawal, Akshu 08 Jan '19

08 Jan '19

On capture through some of dmic we observe a glitch at the start of record. This is because we start capturing even before dmic is ready to send out data. The optional delay will be applied after enabling the mic. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/codecs/adau7002.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c index fdff868..a8deb37 100644 --- a/sound/soc/codecs/adau7002.c +++ b/sound/soc/codecs/adau7002.c @@ -8,6 +8,7 @@ */ #include <linux/acpi.h> +#include <linux/delay.h> #include <linux/init.h> #include <linux/module.h> #include <linux/of.h> @@ -15,12 +16,55 @@ #include <sound/soc.h> +struct adau7002_priv { + int wakeup_delay; +}; + +static int adau7002_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct adau7002_priv *adau7002 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (adau7002->wakeup_delay) + msleep(adau7002->wakeup_delay); + break; + } + + return 0; +} + +static int adau7002_component_probe(struct snd_soc_component *component) +{ + struct adau7002_priv *adau7002; + + adau7002 = devm_kzalloc(component->dev, sizeof(*adau7002), + GFP_KERNEL); + if (!adau7002) + return -ENOMEM; + + device_property_read_u32(component->dev, "wakeup-delay-ms", + &adau7002->wakeup_delay); + + snd_soc_component_set_drvdata(component, adau7002); + + return 0; +} + static const struct snd_soc_dapm_widget adau7002_widgets[] = { + SND_SOC_DAPM_AIF_OUT_E("ADAU AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0, adau7002_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_INPUT("PDM_DAT"), SND_SOC_DAPM_REGULATOR_SUPPLY("IOVDD", 0, 0), }; static const struct snd_soc_dapm_route adau7002_routes[] = { + { "ADAU AIF", NULL, "PDM_DAT"}, { "Capture", NULL, "PDM_DAT" }, { "Capture", NULL, "IOVDD" }, }; @@ -40,6 +84,7 @@ }; static const struct snd_soc_component_driver adau7002_component_driver = { + .probe = adau7002_component_probe, .dapm_widgets = adau7002_widgets, .num_dapm_widgets = ARRAY_SIZE(adau7002_widgets), .dapm_routes = adau7002_routes, -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: rt274: Variable "buf" in function rt274_jack_detect() could be uninitialized" to the asoc tree
by Mark Brown 08 Jan '19

08 Jan '19

The patch ASoC: rt274: Variable "buf" in function rt274_jack_detect() could be uninitialized 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 4a8191aa9e057ea38279ef9e809265ba3966be40 Mon Sep 17 00:00:00 2001 From: Yizhuo <yzhai003(a)ucr.edu> Date: Mon, 7 Jan 2019 12:12:32 -0800 Subject: [PATCH] ASoC: rt274: Variable "buf" in function rt274_jack_detect() could be uninitialized In function rt274_jack_detect(), local variable "buf" could be uninitialized if function regmap_read() returns -EINVAL. However, it will be used to calculate "hp" and "mic" and make their value unpredictable while those value are used in the caller. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt274.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 9e88f7b25d38..adf59039a3b6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -353,6 +353,7 @@ static void rt274_index_sync(struct snd_soc_component *component) static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) { unsigned int buf; + int ret; *hp = false; *mic = false; @@ -360,9 +361,15 @@ static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) if (!rt274->component) return -EINVAL; - regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + ret = regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + if (ret) + return ret; + *hp = buf & 0x80000000; - regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + ret = regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + if (ret) + return ret; + *mic = buf & 0x80000000; pr_debug("*hp = %d *mic = %d\n", *hp, *mic); -- 2.20.1

1 0

[alsa-devel] [PATCH - ALSA JACK plugin 0/2] jack: Speedup of start/stop/recovery
by twischer＠de.adit-jv.com 08 Jan '19

08 Jan '19

Hi all, the following patches minimizes the delay of calling snd_pcm_start() and snd_pcm_stop() for the ALSA JACK plugin. This also makes an Xrun recovery faster. Best regards Timo

2 4

[alsa-devel] [PATCH] ALSA: cs46xx: Potential NULL dereference in probe
by Dan Carpenter 08 Jan '19

08 Jan '19

The "chip->dsp_spos_instance" can be NULL on some of the ealier error paths in snd_cs46xx_create(). Reported-by: "Yavuz, Tuba" <tuba(a)ece.ufl.edu> Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- We don't really consider NULL derefences as security bugs unless they can be triggered remotely, but it's definitely worth fixing. sound/pci/cs46xx/dsp_spos.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 598d140bb7cb..5fc497c6d738 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -903,6 +903,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; + if (!ins) + return 0; + snd_info_free_entry(ins->proc_sym_info_entry); ins->proc_sym_info_entry = NULL; -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt274: Variable "buf" in function rt274_jack_detect() could be uninitialized
by Yizhuo 07 Jan '19

07 Jan '19

In function rt274_jack_detect(), local variable "buf" could be uninitialized if function regmap_read() returns -EINVAL. However, it will be used to calculate "hp" and "mic" and make their value unpredictable while those value are used in the caller. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/soc/codecs/rt274.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 8f92e5c4dd9d..332a3e955bdb 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -353,6 +353,7 @@ static void rt274_index_sync(struct snd_soc_codec *codec) static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) { unsigned int buf; + int ret; *hp = false; *mic = false; @@ -360,9 +361,15 @@ static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) if (!rt274->codec) return -EINVAL; - regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + ret = regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + if (ret) + return ret; + *hp = buf & 0x80000000; - regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + ret = regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + if (ret) + return ret; + *mic = buf & 0x80000000; pr_debug("*hp = %d *mic = %d\n", *hp, *mic); -- 2.17.1

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[alsa-devel] Applied "ASoC: Intel: common: Add quirk for PoV P1006W tablet" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: common: Add quirk for PoV P1006W tablet 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 d3dcc5882ca95c9207b5232395c291d34a511627 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 30 Dec 2018 00:00:19 +0100 Subject: [PATCH] ASoC: Intel: common: Add quirk for PoV P1006W tablet The Point of View TAB-P1006W-232 (v1.0) tablet uses 10EC5640 as ACPI HID, but it has a rt5651 codec add a quirk for this. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../intel/common/soc-acpi-intel-byt-match.c | 35 +++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 47a90909b956..96f9c553fe6c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -21,6 +21,7 @@ static unsigned long byt_machine_id; #define BYT_THINKPAD_10 1 +#define BYT_POV_P1006W 2 static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) { @@ -28,6 +29,11 @@ static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) return 1; } +static int byt_pov_p1006w_quirk_cb(const struct dmi_system_id *id) +{ + byt_machine_id = BYT_POV_P1006W; + return 1; +} static const struct dmi_system_id byt_table[] = { { @@ -58,6 +64,17 @@ static const struct dmi_system_id byt_table[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"), }, }, + { + /* Point of View mobii wintab p1006w (v1.0) */ + .callback = byt_pov_p1006w_quirk_cb, + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "BayTrail"), + /* Note 105b is Foxcon's USB/PCI vendor id */ + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "105B"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "0E57"), + }, + }, { } }; @@ -71,16 +88,30 @@ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .asoc_plat_name = "sst-mfld-platform", }; +static struct snd_soc_acpi_mach byt_pov_p1006w = { + .id = "10EC5640", + .drv_name = "bytcr_rt5651", + .fw_filename = "intel/fw_sst_0f28.bin", + .board = "bytcr_rt5651", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5651.tplg", + .asoc_plat_name = "sst-mfld-platform", +}; + static struct snd_soc_acpi_mach *byt_quirk(void *arg) { struct snd_soc_acpi_mach *mach = arg; dmi_check_system(byt_table); - if (byt_machine_id == BYT_THINKPAD_10) + switch (byt_machine_id) { + case BYT_THINKPAD_10: return &byt_thinkpad_10; - else + case BYT_POV_P1006W: + return &byt_pov_p1006w; + default: return mach; + } } struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[] = { -- 2.20.1

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[alsa-devel] Applied "ASoC: rt5651: Add ACPI ID 10EC5640" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: rt5651: Add ACPI ID 10EC5640 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 d306873589c5a4c13df7176cd73d66ebfa690064 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 30 Dec 2018 00:00:20 +0100 Subject: [PATCH] ASoC: rt5651: Add ACPI ID 10EC5640 Some BYT platforms have a RT5651 codec while using an ACPI node with a HID of 10EC5640 to describe the coded. Add the 10EC5640 HID to the acpi_device_id list, so that the rt5651 will bind to the codec on these devices. Like the rt5645 and rt5670 drivers which also have the 10EC5640 ACPI HID in their acpi_device_id list for similar reasons, the rt5651 driver checks the codecs device-id register so that it will only bind if the codec actually is a rt5651 and it will ignore actual rt5640 codecs. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 3882e238ff99..9a007c162631 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2138,6 +2138,7 @@ MODULE_DEVICE_TABLE(of, rt5651_of_match); #ifdef CONFIG_ACPI static const struct acpi_device_id rt5651_acpi_match[] = { { "10EC5651", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5651_acpi_match); -- 2.20.1

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[alsa-devel] Applied "ASoC: rt5651: Add support for jack detect using an external GPIO" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: rt5651: Add support for jack detect using an external GPIO 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 c2ec9d957d2bf49d69afb1b872cb2363c6cb5862 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 30 Dec 2018 00:00:21 +0100 Subject: [PATCH] ASoC: rt5651: Add support for jack detect using an external GPIO Some board designs hook the jack-detect up to an external GPIO, rather then to one of the codec pins, add support for this. Figuring out which GPIO to use is pretty much board specific so I've chosen to let the machine driver pass the gpio_desc as data argument to snd_soc_component_set_jack() rather then add support for getting the GPIO to the codec driver. This keeps the codec code nice and clean. Note that using an external GPIO for this conflicts with button-press support, so this commit disables button-press support when an external GPIO is used. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 54 +++++++++++++++++++++++++++------------ sound/soc/codecs/rt5651.h | 1 + 2 files changed, 39 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 9a007c162631..75994297c896 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/regmap.h> #include <linux/platform_device.h> @@ -1621,6 +1622,12 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component) struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); int val; + if (rt5651->gpiod_hp_det) { + val = gpiod_get_value_cansleep(rt5651->gpiod_hp_det); + dev_dbg(component->dev, "jack-detect gpio %d\n", val); + return val; + } + val = snd_soc_component_read32(component, RT5651_INT_IRQ_ST); dev_dbg(component->dev, "irq status %#04x\n", val); @@ -1761,6 +1768,13 @@ static int rt5651_detect_headset(struct snd_soc_component *component) return SND_JACK_HEADPHONE; } +static bool rt5651_support_button_press(struct rt5651_priv *rt5651) +{ + /* Button press support only works with internal jack-detection */ + return (rt5651->hp_jack->status & SND_JACK_MICROPHONE) && + rt5651->gpiod_hp_det == NULL; +} + static void rt5651_jack_detect_work(struct work_struct *work) { struct rt5651_priv *rt5651 = @@ -1785,15 +1799,15 @@ static void rt5651_jack_detect_work(struct work_struct *work) WARN_ON(rt5651->ovcd_irq_enabled); rt5651_enable_micbias1_for_ovcd(component); report = rt5651_detect_headset(component); - if (report == SND_JACK_HEADSET) { + dev_dbg(component->dev, "detect report %#02x\n", report); + snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); + if (rt5651_support_button_press(rt5651)) { /* Enable ovcd IRQ for button press detect. */ rt5651_enable_micbias1_ovcd_irq(component); } else { /* No more need for overcurrent detect. */ rt5651_disable_micbias1_for_ovcd(component); } - dev_dbg(component->dev, "detect report %#02x\n", report); - snd_soc_jack_report(rt5651->hp_jack, report, SND_JACK_HEADSET); } else if (rt5651->ovcd_irq_enabled && rt5651_micbias1_ovcd(component)) { dev_dbg(component->dev, "OVCD IRQ\n"); @@ -1837,16 +1851,20 @@ static void rt5651_cancel_work(void *data) } static void rt5651_enable_jack_detect(struct snd_soc_component *component, - struct snd_soc_jack *hp_jack) + struct snd_soc_jack *hp_jack, + struct gpio_desc *gpiod_hp_det) { struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); - - /* IRQ output on GPIO1 */ - snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1, - RT5651_GP1_PIN_MASK, RT5651_GP1_PIN_IRQ); + bool using_internal_jack_detect = true; /* Select jack detect source */ switch (rt5651->jd_src) { + case RT5651_JD_NULL: + rt5651->gpiod_hp_det = gpiod_hp_det; + if (!rt5651->gpiod_hp_det) + return; /* No jack detect */ + using_internal_jack_detect = false; + break; case RT5651_JD1_1: snd_soc_component_update_bits(component, RT5651_JD_CTRL2, RT5651_JD_TRG_SEL_MASK, RT5651_JD_TRG_SEL_JD1_1); @@ -1865,16 +1883,20 @@ static void rt5651_enable_jack_detect(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5651_IRQ_CTRL1, RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN); break; - case RT5651_JD_NULL: - return; default: dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); return; } - /* Enable jack detect power */ - snd_soc_component_update_bits(component, RT5651_PWR_ANLG2, - RT5651_PWR_JD_M, RT5651_PWR_JD_M); + if (using_internal_jack_detect) { + /* IRQ output on GPIO1 */ + snd_soc_component_update_bits(component, RT5651_GPIO_CTRL1, + RT5651_GP1_PIN_MASK, RT5651_GP1_PIN_IRQ); + + /* Enable jack detect power */ + snd_soc_component_update_bits(component, RT5651_PWR_ANLG2, + RT5651_PWR_JD_M, RT5651_PWR_JD_M); + } /* Set OVCD threshold current and scale-factor */ snd_soc_component_write(component, RT5651_PR_BASE + RT5651_BIAS_CUR4, @@ -1903,7 +1925,7 @@ static void rt5651_enable_jack_detect(struct snd_soc_component *component, RT5651_MB1_OC_STKY_MASK, RT5651_MB1_OC_STKY_EN); rt5651->hp_jack = hp_jack; - if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + if (rt5651_support_button_press(rt5651)) { rt5651_enable_micbias1_for_ovcd(component); rt5651_enable_micbias1_ovcd_irq(component); } @@ -1920,7 +1942,7 @@ static void rt5651_disable_jack_detect(struct snd_soc_component *component) disable_irq(rt5651->irq); rt5651_cancel_work(rt5651); - if (rt5651->hp_jack->status & SND_JACK_MICROPHONE) { + if (rt5651_support_button_press(rt5651)) { rt5651_disable_micbias1_ovcd_irq(component); rt5651_disable_micbias1_for_ovcd(component); snd_soc_jack_report(rt5651->hp_jack, 0, SND_JACK_BTN_0); @@ -1933,7 +1955,7 @@ static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *jack, void *data) { if (jack) - rt5651_enable_jack_detect(component, jack); + rt5651_enable_jack_detect(component, jack, data); else rt5651_disable_jack_detect(component); diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index ac6de6fb5414..41fcb8b5eb40 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2073,6 +2073,7 @@ struct rt5651_priv { struct regmap *regmap; /* Jack and button detect data */ struct snd_soc_jack *hp_jack; + struct gpio_desc *gpiod_hp_det; struct work_struct jack_detect_work; struct delayed_work bp_work; bool ovcd_irq_enabled; -- 2.20.1

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[alsa-devel] Applied "ASoC: Intel: bytcr_rt5651: Revert "Fix DMIC map headsetmic mapping"" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: bytcr_rt5651: Revert "Fix DMIC map headsetmic mapping" 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 aee48a9ffa5a128bf4e433c57c39e015ea5b0208 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 30 Dec 2018 00:00:22 +0100 Subject: [PATCH] ASoC: Intel: bytcr_rt5651: Revert "Fix DMIC map headsetmic mapping" Commit 37c7401e8c1f ("ASoC: Intel: bytcr_rt5651: Fix DMIC map headsetmic mapping"), changed the headsetmic mapping from IN3P to IN2P, this was based on the observation that all bytcr_rt5651 devices I have access to (7 devices) where all using IN3P for the headsetmic. This was an attempt to unifify / simplify the mapping, but it was wrong. None of those devices was actually using a digital internal mic. Now I've access to a Point of View TAB-P1006W-232 (v1.0) tabler, which does use a DMIC and it does have its headsetmic connected to IN2P, showing that the original mapping was correct, so this commit reverts the change changing the mapping back to IN2P. Fixes: 37c7401e8c1f ("ASoC: Intel: bytcr_rt5651: Fix DMIC map ... mapping") Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index e528995668b7..0ed844f2ad01 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -266,7 +266,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { {"DMIC L1", NULL, "Internal Mic"}, {"DMIC R1", NULL, "Internal Mic"}, - {"IN3P", NULL, "Headset Mic"}, + {"IN2P", NULL, "Headset Mic"}, }; static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_map[] = { -- 2.20.1

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[alsa-devel] Applied "ASoC: Intel: bytcr_rt5651: Add quirks module parameter" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: bytcr_rt5651: Add quirks module parameter 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 7eb187313eef4c8faa49f70c9c7d8918e1052207 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 30 Dec 2018 00:00:23 +0100 Subject: [PATCH] ASoC: Intel: bytcr_rt5651: Add quirks module parameter Add quirks module parameter to allow manually specifying quirks from the kernel commandline (or modprobe.conf). Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 0ed844f2ad01..6d8ef9dd106e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -98,6 +98,10 @@ struct byt_rt5651_private { static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP; +static unsigned int quirk_override; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + static void log_quirks(struct device *dev) { if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_DMIC_MAP) @@ -973,6 +977,12 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* check quirks before creating card */ dmi_check_system(byt_rt5651_quirk_table); + if (quirk_override) { + dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n", + (unsigned int)byt_rt5651_quirk, quirk_override); + byt_rt5651_quirk = quirk_override; + } + /* Must be called before register_card, also see declaration comment. */ ret_val = byt_rt5651_add_codec_device_props(codec_dev); if (ret_val) { -- 2.20.1

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[alsa-devel] [PATCH v2 0/7] Asoc: Various Intel rt5651 patches
by Hans de Goede 07 Jan '19

07 Jan '19

Hi All, Here is a v2 of my previous set with various Intel rt5651 patches. This version contains a small fix to: [PATCH v2 6/7] ASoC: Intel: bytcr_rt5651: Add support for jack-detect I forgot to pass the gpio to snd_soc_component_set_jack from the resume handler too (instead of only from the init function). Other then that there are no changes, so I've kept Pierre-Louis' Acked-by for the entire series. Regards, Hans

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[alsa-devel] Applied "ASoC: Intel: bytcr_rt5651: Add quirk for PoV TAB-P1006W-232 (v1.0) tablet" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: bytcr_rt5651: Add quirk for PoV TAB-P1006W-232 (v1.0) tablet 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 fee3e1cbd6cd74925286a571b567ec18728818a7 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 30 Dec 2018 00:00:25 +0100 Subject: [PATCH] ASoC: Intel: bytcr_rt5651: Add quirk for PoV TAB-P1006W-232 (v1.0) tablet Add a DMI quirk for the Point of View TAB-P1006W-232 (v1.0) tablet, this tablet is special in a number of ways: 1) It uses the 2nd GPIO resource in the ACPI tables for jack-detect rather then using the rt5651 codec's builtin jack-detect functionality 2) It uses the 3th GPIO resource in the ACPI tables to control the external amplifier rather then the usual first non GpioInt resource and the GPIO is active-low. 3) It is a BYTCR device, without a CHAN package and it uses SSP0-AIF1 rather then the default SSP0-AIF2. 4) Its internal mic is a digital mic (the first x86 rt5651 device that I'm aware of which does this), combined with having its headset-mic connected to IN2. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 48 ++++++++++++++++++++++++--- 1 file changed, 43 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 9a2ee9080897..b618d984e2d5 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -95,6 +95,8 @@ struct byt_rt5651_private { struct snd_soc_jack jack; }; +static const struct acpi_gpio_mapping *byt_rt5651_gpios; + /* Default: jack-detect on JD1_1, internal mic on in2, headsetmic on in3 */ static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP; @@ -365,6 +367,22 @@ static int byt_rt5651_aif1_hw_params(struct snd_pcm_substream *substream, return byt_rt5651_prepare_and_enable_pll1(codec_dai, rate, bclk_ratio); } +static const struct acpi_gpio_params pov_p1006w_hp_detect = { 1, 0, false }; +static const struct acpi_gpio_params pov_p1006w_ext_amp_en = { 2, 0, true }; + +static const struct acpi_gpio_mapping byt_rt5651_pov_p1006w_gpios[] = { + { "hp-detect-gpios", &pov_p1006w_hp_detect, 1, }, + { "ext-amp-enable-gpios", &pov_p1006w_ext_amp_en, 1, }, + { }, +}; + +static int byt_rt5651_pov_p1006w_quirk_cb(const struct dmi_system_id *id) +{ + byt_rt5651_quirk = (unsigned long)id->driver_data; + byt_rt5651_gpios = byt_rt5651_pov_p1006w_gpios; + return 1; +} + static int byt_rt5651_quirk_cb(const struct dmi_system_id *id) { byt_rt5651_quirk = (unsigned long)id->driver_data; @@ -440,6 +458,23 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { .driver_data = (void *)(BYT_RT5651_MCLK_EN | BYT_RT5651_IN1_MAP), }, + { + /* Point of View mobii wintab p1006w (v1.0) */ + .callback = byt_rt5651_pov_p1006w_quirk_cb, + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "BayTrail"), + /* Note 105b is Foxcon's USB/PCI vendor id */ + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "105B"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "0E57"), + }, + .driver_data = (void *)(BYT_RT5651_DMIC_MAP | + BYT_RT5651_OVCD_TH_2000UA | + BYT_RT5651_OVCD_SF_0P75 | + BYT_RT5651_DMIC_EN | + BYT_RT5651_MCLK_EN | + BYT_RT5651_SSP0_AIF1), + }, { /* VIOS LTH17 */ .callback = byt_rt5651_quirk_cb, @@ -848,7 +883,7 @@ static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) return 0; } -static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec) +static void snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(struct device *codec) { struct byt_rt5651_acpi_resource_data data = { 0, -1 }; LIST_HEAD(resources); @@ -866,10 +901,10 @@ static void snd_byt_rt5651_mc_add_amp_en_gpio_mapping(struct device *codec) switch (data.gpio_int_idx) { case 0: - devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_second); + byt_rt5651_gpios = byt_rt5651_amp_en_second; break; case 1: - devm_acpi_dev_add_driver_gpios(codec, byt_rt5651_amp_en_first); + byt_rt5651_gpios = byt_rt5651_amp_en_first; break; default: dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", @@ -1001,8 +1036,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } /* Cherry Trail devices use an external amplifier enable gpio */ - if (x86_match_cpu(cherrytrail_cpu_ids)) { - snd_byt_rt5651_mc_add_amp_en_gpio_mapping(codec_dev); + if (x86_match_cpu(cherrytrail_cpu_ids) && !byt_rt5651_gpios) + snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(codec_dev); + + if (byt_rt5651_gpios) { + devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child( &pdev->dev, "ext-amp-enable", 0, codec_dev->fwnode, -- 2.20.1

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[alsa-devel] [PATCH 0/2] Add support for AK4497 codec
by Daniel Baluta 07 Jan '19

07 Jan '19

AK4497 is almost register compatible with AK4458. Almost means that AK4497 has an extra "debug" register that we currently do not use. Daniel Baluta (2): ASoC: ak4458: Add support for AK4497 ASoC: dt-bindings: Document support for ak4497 .../devicetree/bindings/sound/ak4458.txt | 2 +- sound/soc/codecs/ak4458.c | 79 ++++++++++++++++++- 2 files changed, 77 insertions(+), 4 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH 00/21] ASoC: codecs: remove warnings with Sparse, Coccinelle and W=1
by Pierre-Louis Bossart 07 Jan '19

07 Jan '19

This patchset contains mostly trivial fixes to make tools happy. I don't get paid more by sending lots of small patches, my only goal is to make sure tools don't spit out too many warnings and drown potential problems. Thanks for Bard Liao for his help, and Adam Thompson and Oder Chiou for off-line reviews. Bard liao (4): ASoC: rt5645: remove unused mux define ASoC: rt5670: remove unused mux/mixer define ASoC: rt5645: store eq kcontrol byte in __be ASoC: rl6437a: use __be32 for a __be32 buf Pierre-Louis Bossart (17): ASoC: dmic: declare trigger function as static ASoC: max98090: remove unused constant variables ASoC: es8316: remove unused constant variables ASoC: codecs: fix kernel doc descriptions ASoC: max98383: fix boolean assignments to true/false ASoC: rt298: fix boolean tests ASoC: cs4271: fix boolean assignments ASoC: rt274: fix boolean tests ASoc: rt286: fix boolean tests ASoC: rt5640: fix boolean assignments ASoC: max98927: fix boolean assignments ASoC: rt5651: fix boolean assignments ASoC: nau8824: fix boolean assignment ASoC: tscs42xx.c: fix boolean test ASoC: mt6351: remove unneeded variable ASoC: da7219: fix endianness issues ASoC: da7219: use logical AND sound/soc/codecs/cs4271.c | 4 +-- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/da7219.c | 8 ++--- sound/soc/codecs/dmic.c | 4 +-- sound/soc/codecs/es8316.c | 4 --- sound/soc/codecs/max98090.c | 15 -------- sound/soc/codecs/max98373.c | 6 ++-- sound/soc/codecs/max98927.c | 8 ++--- sound/soc/codecs/mt6351.c | 4 +-- sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 1 + sound/soc/codecs/rl6347a.c | 4 +-- sound/soc/codecs/rt274.c | 8 ++--- sound/soc/codecs/rt286.c | 8 ++--- sound/soc/codecs/rt298.c | 12 +++---- sound/soc/codecs/rt5514.c | 1 + sound/soc/codecs/rt5640.c | 6 ++-- sound/soc/codecs/rt5645.c | 66 +++++++++-------------------------- sound/soc/codecs/rt5651.c | 6 ++-- sound/soc/codecs/rt5670.c | 54 ---------------------------- sound/soc/codecs/rt5677.c | 8 ++--- sound/soc/codecs/tscs42xx.c | 2 +- 22 files changed, 63 insertions(+), 170 deletions(-) base-commit: 14f9b94a3274b047126c8c0ff39dc495e8d1acab -- 2.17.1

3 44

[alsa-devel] [PATCH] ASoC: eliminate left-over from Raumfeld machine driver removal
by Daniel Mack 07 Jan '19

07 Jan '19

Commit f84a6273dd9107c ("ASoC: pxa: remove raumfeld machine driver") removed the Raumfeld ASoC machine driver but forgot to kill one line in the Makefile. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/Makefile | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 0ab2a9dcb720..ea4929d73318 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -30,7 +30,6 @@ snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o snd-soc-imote2-objs := imote2.o -snd-soc-raumfeld-objs := raumfeld.o snd-soc-brownstone-objs := brownstone.o snd-soc-ttc-dkb-objs := ttc-dkb.o -- 2.19.2

2 1

[alsa-devel] [PATCH v2 0/2] Add MICFIL DAI support
by Cosmin Samoila 07 Jan '19

07 Jan '19

PDM is a popular way to deliver audio from microphones to the processor in several applications, such as mobile telephones. However, current digital-audio systems use multibit audio signal (also known as multibit PCM) to represent the signal. For this purpose a set of FIR, CIC or/and Half Band filters are usually implemented on DSPs or software. This block implements the required digital interface to provide a 16-bit audio signal from a PDM microphone bitstream in a configurable output sampling rate. The implementation of this digital interface is based on the application of digital signal processing techniques in hardware. The PDM Microphone Interface architecture was designed to gate saving and minimal power consumption. It implements a bunch of filters to transform a 1-bit PDM bitstream to a 16-bit PCM signal in the audio band. To avoid aliasing frequencies in passband, the overall filter has 80 dB stopband attenuation and passband ripple less than 0.2dB. The whole module is implemented to work in a multichannel mode. All channels have the same configuration but each input channel could be turned on/off independently. Changes since v1: - removed the hwvad support from patches since it is a separate functionality and it needs some design rework. Will address HWVAD in later series of patches after analyzing comunity feedback and design guildlines. - removed clock source choice for clock driver and use the AudioPLL1 as clock source. Having only one clock source it is sufficient for basic functionality. - removed dma-names and status from dt-bindings - changed volume control names and type to S_VALUE Cosmin-Gabriel Samoila (2): ASoC: micfil: Add bindings for MICFIL DAI ASoC: Add MICFIL SoC Digital Audio Interface driver. .../devicetree/bindings/sound/fsl,micfil.txt | 32 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_micfil.c | 895 +++++++++++++++++++++ sound/soc/fsl/fsl_micfil.h | 283 +++++++ 5 files changed, 1221 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 sound/soc/fsl/fsl_micfil.c create mode 100644 sound/soc/fsl/fsl_micfil.h -- 2.7.4

2 4

[alsa-devel] Applied "ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode 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 667e9334fa64da2273e36ce131b05ac9e47c5769 Mon Sep 17 00:00:00 2001 From: b-ak <anur.bhargav(a)gmail.com> Date: Mon, 7 Jan 2019 22:30:22 +0530 Subject: [PATCH] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode During the bootup of the kernel, the DAPM bias level is in the OFF state. As soon as the DAPM framework kicks in it pushes the codec into STANDBY state. The probe function doesn't prepare the clock, and STANDBY state does a clk_disable_unprepare() without checking the previous state. This leads to an OOPS. Not transitioning from an OFF state to the STANDBY state fixes the problem. Signed-off-by: b-ak <anur.bhargav(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e2b5a11b16d1..f03195d2ab2e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -822,6 +822,10 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + /* Initial cold start */ + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + break; + /* Switch off BCLK_N Divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, AIC32X4_BCLKEN, 0); -- 2.20.1

1 0

[alsa-devel] [PATCH 0/3] add Xilinx SPDIF audio driver
by Maruthi Srinivas Bayyavarapu 07 Jan '19

07 Jan '19

SPDIF IP supports playback and capture of audio in AES format. Each IP instance can only work in either transmit or receive mode exclusively. The patchset adds support for playback and capture functionality. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,spdif: Document spdif bindings ASoC: xlnx: add SPDIF audio driver ASoC: xlnx: enable SPDIF driver build .../devicetree/bindings/sound/xlnx,spdif.txt | 28 ++ sound/soc/xilinx/Kconfig | 7 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_spdif.c | 339 +++++++++++++++++++++ 4 files changed, 376 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,spdif.txt create mode 100644 sound/soc/xilinx/xlnx_spdif.c -- 2.7.4

2 6

[alsa-devel] Applied "ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages 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 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 Mon Sep 17 00:00:00 2001 From: Kangjie Lu <kjlu(a)umn.edu> Date: Tue, 25 Dec 2018 20:29:48 -0600 Subject: [PATCH] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages snd_pcm_lib_malloc_pages() may fail, so let's check its status and return its error code upstream. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index afc559866095..91a2436ce952 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + int ret; + + ret = + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (ret) + return ret; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); return 0; } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: sgtl5000: Allow SCLK pad drive strength to be changed" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: sgtl5000: Allow SCLK pad drive strength to be changed 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 5dc4ca2996840db569e43d00420c10499140274a Mon Sep 17 00:00:00 2001 From: Alison Wang <alison.wang(a)nxp.com> Date: Wed, 26 Dec 2018 08:59:53 +0800 Subject: [PATCH] ASoC: sgtl5000: Allow SCLK pad drive strength to be changed This patch introduces "sclk-strength" property to allow SCLK pad drive strength to be changed via device tree. When running playback test on LS1028ARDB, Tx Frame sync error interrupt will occur sometimes. Some noises also exist. After changing SCLK pad drive strength to the maximum value, the issues are gone. Signed-off-by: Alison Wang <alison.wang(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/sgtl5000.txt | 9 +++++++++ sound/soc/codecs/sgtl5000.c | 19 ++++++++++++++++++- 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 9c58f724396a..9d9ff5184939 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -37,6 +37,15 @@ VDDIO 1.8V 2.5V 3.3V 2 = 3.33 mA 5.74 mA 8.03 mA 3 = 4.99 mA 8.61 mA 12.05 mA +- sclk-strength: the SCLK pad strength. Possible values are: +0, 1, 2 and 3 as per the table below: + +VDDIO 1.8V 2.5V 3.3V +0 = Disable +1 = 1.66 mA 2.87 mA 4.02 mA +2 = 3.33 mA 5.74 mA 8.03 mA +3 = 4.99 mA 8.61 mA 12.05 mA + Example: sgtl5000: codec@a { diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index add18d6d77da..a6a4748c97f9 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -116,6 +116,13 @@ enum { I2S_LRCLK_STRENGTH_HIGH, }; +enum { + I2S_SCLK_STRENGTH_DISABLE, + I2S_SCLK_STRENGTH_LOW, + I2S_SCLK_STRENGTH_MEDIUM, + I2S_SCLK_STRENGTH_HIGH, +}; + /* sgtl5000 private structure in codec */ struct sgtl5000_priv { int sysclk; /* sysclk rate */ @@ -129,6 +136,7 @@ struct sgtl5000_priv { u8 micbias_resistor; u8 micbias_voltage; u8 lrclk_strength; + u8 sclk_strength; }; /* @@ -1302,7 +1310,9 @@ static int sgtl5000_probe(struct snd_soc_component *component) SGTL5000_DAC_MUTE_RIGHT | SGTL5000_DAC_MUTE_LEFT); - reg = ((sgtl5000->lrclk_strength) << SGTL5000_PAD_I2S_LRCLK_SHIFT | 0x5f); + reg = ((sgtl5000->lrclk_strength) << SGTL5000_PAD_I2S_LRCLK_SHIFT | + (sgtl5000->sclk_strength) << SGTL5000_PAD_I2S_SCLK_SHIFT | + 0x1f); snd_soc_component_write(component, SGTL5000_CHIP_PAD_STRENGTH, reg); snd_soc_component_write(component, SGTL5000_CHIP_ANA_CTRL, @@ -1542,6 +1552,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, sgtl5000->lrclk_strength = value; } + sgtl5000->sclk_strength = I2S_SCLK_STRENGTH_LOW; + if (!of_property_read_u32(np, "sclk-strength", &value)) { + if (value > I2S_SCLK_STRENGTH_HIGH) + value = I2S_SCLK_STRENGTH_LOW; + sgtl5000->sclk_strength = value; + } + /* Ensure sgtl5000 will start with sane register values */ sgtl5000_fill_defaults(client); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: rsnd: update BSDSR/BSDISR handling
by Kuninori Morimoto 07 Jan '19

07 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current BSDSR/BSDISR are using temporary/generic settings, but it can't handle all SRCx/SoC. It needs to handle correctry. Otherwise, sampling rate converted sound channel will be broken if it was TDM. One note is that it needs to overwrite settings on E3 case. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: chaoliang qin <chaoliang.qin.jg(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 125 ++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 115 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 50348a2..db81e06 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -14,6 +14,7 @@ */ #include "rsnd.h" +#include <linux/sys_soc.h> #define SRC_NAME "src" @@ -134,20 +135,83 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } +const static u32 bsdsr_table_pattern1[] = { + 0x01800000, /* 6 - 1/6 */ + 0x01000000, /* 6 - 1/4 */ + 0x00c00000, /* 6 - 1/3 */ + 0x00800000, /* 6 - 1/2 */ + 0x00600000, /* 6 - 2/3 */ + 0x00400000, /* 6 - 1 */ +}; + +const static u32 bsdsr_table_pattern2[] = { + 0x02400000, /* 6 - 1/6 */ + 0x01800000, /* 6 - 1/4 */ + 0x01200000, /* 6 - 1/3 */ + 0x00c00000, /* 6 - 1/2 */ + 0x00900000, /* 6 - 2/3 */ + 0x00600000, /* 6 - 1 */ +}; + +const static u32 bsisr_table[] = { + 0x00100060, /* 6 - 1/6 */ + 0x00100040, /* 6 - 1/4 */ + 0x00100030, /* 6 - 1/3 */ + 0x00100020, /* 6 - 1/2 */ + 0x00100020, /* 6 - 2/3 */ + 0x00100020, /* 6 - 1 */ +}; + +const static u32 chan288888[] = { + 0x00000006, /* 1 to 2 */ + 0x000001fe, /* 1 to 8 */ + 0x000001fe, /* 1 to 8 */ + 0x000001fe, /* 1 to 8 */ + 0x000001fe, /* 1 to 8 */ + 0x000001fe, /* 1 to 8 */ +}; + +const static u32 chan244888[] = { + 0x00000006, /* 1 to 2 */ + 0x0000001e, /* 1 to 4 */ + 0x0000001e, /* 1 to 4 */ + 0x000001fe, /* 1 to 8 */ + 0x000001fe, /* 1 to 8 */ + 0x000001fe, /* 1 to 8 */ +}; + +const static u32 chan222222[] = { + 0x00000006, /* 1 to 2 */ + 0x00000006, /* 1 to 2 */ + 0x00000006, /* 1 to 2 */ + 0x00000006, /* 1 to 2 */ + 0x00000006, /* 1 to 2 */ + 0x00000006, /* 1 to 2 */ +}; + +static const struct soc_device_attribute ov_soc[] = { + { .soc_id = "r8a77990" }, /* E3 */ + { /* sentinel */ } +}; + static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + const struct soc_device_attribute *soc = soc_device_match(ov_soc); int is_play = rsnd_io_is_play(io); int use_src = 0; u32 fin, fout; u32 ifscr, fsrate, adinr; u32 cr, route; - u32 bsdsr, bsisr; u32 i_busif, o_busif, tmp; + const u32 *bsdsr_table; + const u32 *chptn; uint ratio; + int chan; + int idx; if (!runtime) return; @@ -155,6 +219,8 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, fin = rsnd_src_get_in_rate(priv, io); fout = rsnd_src_get_out_rate(priv, io); + chan = rsnd_runtime_channel_original(io); + /* 6 - 1/6 are very enough ratio for SRC_BSDSR */ if (fin == fout) ratio = 0; @@ -173,8 +239,7 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, /* * SRC_ADINR */ - adinr = rsnd_get_adinr_bit(mod, io) | - rsnd_runtime_channel_original(io); + adinr = rsnd_get_adinr_bit(mod, io) | chan; /* * SRC_IFSCR / SRC_IFSVR @@ -207,21 +272,56 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, /* * SRC_BSDSR / SRC_BSISR + * + * see + * Combination of Register Setting Related to + * FSO/FSI Ratio and Channel, Latency */ switch (rsnd_mod_id(mod)) { + case 0: + chptn = chan288888; + bsdsr_table = bsdsr_table_pattern1; + break; + case 1: + case 3: + case 4: + chptn = chan244888; + bsdsr_table = bsdsr_table_pattern1; + break; + case 2: + case 9: + chptn = chan222222; + bsdsr_table = bsdsr_table_pattern1; + break; case 5: case 6: case 7: case 8: - bsdsr = 0x02400000; /* 6 - 1/6 */ - bsisr = 0x00100060; /* 6 - 1/6 */ + chptn = chan222222; + bsdsr_table = bsdsr_table_pattern2; break; default: - bsdsr = 0x01800000; /* 6 - 1/6 */ - bsisr = 0x00100060 ;/* 6 - 1/6 */ - break; + goto convert_rate_err; } + /* + * E3 need to overwrite + */ + if (soc) + switch (rsnd_mod_id(mod)) { + case 0: + case 4: + chptn = chan222222; + } + + for (idx = 0; idx < ARRAY_SIZE(chan222222); idx++) + if (chptn[idx] & (1 << chan)) + break; + + if (chan > 8 || + idx >= ARRAY_SIZE(chan222222)) + goto convert_rate_err; + /* BUSIF_MODE */ tmp = rsnd_get_busif_shift(io, mod); i_busif = ( is_play ? tmp : 0) | 1; @@ -234,8 +334,8 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, rsnd_mod_write(mod, SRC_IFSCR, ifscr); rsnd_mod_write(mod, SRC_IFSVR, fsrate); rsnd_mod_write(mod, SRC_SRCCR, cr); - rsnd_mod_write(mod, SRC_BSDSR, bsdsr); - rsnd_mod_write(mod, SRC_BSISR, bsisr); + rsnd_mod_write(mod, SRC_BSDSR, bsdsr_table[idx]); + rsnd_mod_write(mod, SRC_BSISR, bsisr_table[idx]); rsnd_mod_write(mod, SRC_SRCIR, 0); /* cancel initialize */ rsnd_mod_write(mod, SRC_I_BUSIF_MODE, i_busif); @@ -244,6 +344,11 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, rsnd_mod_write(mod, SRC_BUSIF_DALIGN, rsnd_get_dalign(mod, io)); rsnd_adg_set_src_timesel_gen2(mod, io, fin, fout); + + return; + +convert_rate_err: + dev_err(dev, "unknown BSDSR/BSDIR settings\n"); } static int rsnd_src_irq(struct rsnd_mod *mod, -- 2.7.4

2 1

Re: [alsa-devel] [PATCH] sound: atom: fix a missing check of snd_pcm_lib_malloc_pages
by Mark Brown 07 Jan '19

07 Jan '19

On Tue, Dec 25, 2018 at 08:29:48PM -0600, Kangjie Lu wrote: > snd_pcm_lib_malloc_pages() may fail, so let's check its status and > return its error code upstream. Please use subject lines matching the style for the subsystem. This makes it easier for people to identify relevant patches.

1 0

[alsa-devel] [PATCH] ASoC: rt274: Variable "buf" in function rt274_jack_detect() could be uninitialized
by Yizhuo 07 Jan '19

07 Jan '19

In function rt274_jack_detect(), local variable "buf" could be uninitialized if function regmap_read() returns -EINVAL. However, it will be used to calculate "hp" and "mic" and make their value unpredictable while those value are used in the caller. This is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/soc/codecs/rt274.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index cd048df76232..a2c1a6df8df6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -353,6 +353,7 @@ static void rt274_index_sync(struct snd_soc_codec *codec) static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) { unsigned int buf; + int ret = 0; *hp = false; *mic = false; @@ -360,9 +361,15 @@ static int rt274_jack_detect(struct rt274_priv *rt274, bool *hp, bool *mic) if (!rt274->codec) return -EINVAL; - regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + ret = regmap_read(rt274->regmap, RT274_GET_HP_SENSE, &buf); + if (ret) + return ret; + *hp = buf & 0x80000000; - regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + ret = regmap_read(rt274->regmap, RT274_GET_MIC_SENSE, &buf); + if (ret) + return ret; + *mic = buf & 0x80000000; pr_debug("*hp = %d *mic = %d\n", *hp, *mic); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt298: Variable "val" and "buf" in rt298_jack_detect() could be uninitialized
by Yizhuo 07 Jan '19

07 Jan '19

In function rt298_jack_detect(), local variable "val" and "buf" are supposed to get initialized by regmap_read(). However, they could be leave uninitialized if regmap_read() returns -EINVAL. Those two variables are used in control flow or update the argument, which could lead to undefined behavior and thus unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/soc/codecs/rt298.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index ce963768449f..7f74349c17f3 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -223,6 +223,7 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) { struct snd_soc_dapm_context *dapm; unsigned int val, buf; + int ret = 0; *hp = false; *mic = false; @@ -233,7 +234,10 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) dapm = snd_soc_codec_get_dapm(rt298->codec); if (rt298->pdata.cbj_en) { - regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + ret = regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + if (ret) + return ret; + *hp = buf & 0x80000000; if (*hp == rt298->is_hp_in) return -1; @@ -260,16 +264,19 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) regmap_update_bits(rt298->regmap, RT298_CBJ_CTRL1, 0xfcc0, 0xd400); msleep(300); - regmap_read(rt298->regmap, RT298_CBJ_CTRL2, &val); - + ret = regmap_read(rt298->regmap, RT298_CBJ_CTRL2, &val); + if (ret) + return ret; if (0x0070 == (val & 0x0070)) { *mic = true; } else { regmap_update_bits(rt298->regmap, RT298_CBJ_CTRL1, 0xfcc0, 0xe400); msleep(300); - regmap_read(rt298->regmap, + ret = regmap_read(rt298->regmap, RT298_CBJ_CTRL2, &val); + if (ret) + return ret; if (0x0070 == (val & 0x0070)) *mic = true; else @@ -285,9 +292,14 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) RT298_CBJ_CTRL1, 0x0400, 0x0000); } } else { - regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + ret = regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + if (ret) + return ret; + *hp = buf & 0x80000000; - regmap_read(rt298->regmap, RT298_GET_MIC1_SENSE, &buf); + ret = regmap_read(rt298->regmap, RT298_GET_MIC1_SENSE, &buf); + if (ret) + return ret; *mic = buf & 0x80000000; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Fix error handling of device_property_read_u32_array
by Aditya Pakki 07 Jan '19

07 Jan '19

In rt5663_parse_dp, the function device_property_read_u32_array() can return an error. This fix adds a check to the latter as well as to the callsite of rt5663_parse_dp in rt5663_i2c_probe. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> --- sound/soc/codecs/rt5663.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index da6647015708..2678842d8cd4 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3480,7 +3480,7 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); - device_property_read_u32_array(dev, + return device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); } @@ -3507,8 +3507,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, if (pdata) rt5663->pdata = *pdata; - else - rt5663_parse_dp(rt5663, &i2c->dev); + else { + ret = rt5663_parse_dp(rt5663, &i2c->dev); + if (ret) + return ret; + } for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) rt5663->supplies[i].supply = rt5663_supply_names[i]; -- 2.17.1

2 1

[alsa-devel] [PATCH ucm/conf: 0/5] Factor out rt5645 + rt5650 shared UCM sequences
by russell.parker7＠gmail.com 07 Jan '19

07 Jan '19

Hi, When I submitted a patch to create a UCM config for chtrt5650 it was suggested that I also factor out command sequences that it shares with chtrt5645. The following patchset introduces a new component directory codecs/rt5645-common/ and refactors UCM profiles in: - chtrt5645 - chtrt5645-mono-speaker-analog-mic - chtrt5650 - ASUSTeKCOMPUTERINC.-T100HAN-1.0-T100HAN - LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 I compiled and tested playback with the rt5650 configs but I don't have the devices to test the other configurations. Suggestions and feedback are welcome, I'm not very familiar with UCM configuration so there may definitely be better approaches for some of these changes. - Russell

3 7

[alsa-devel] [PATCH v6 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Sinan Kaya 07 Jan '19

07 Jan '19

After 'commit 5d32a66541c4 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were satisfied implicitly through dependencies on CONFIG_ACPI have to be specified directly. This code relies on IOSF_MBI and IOSF_MBI depends on PCI. For this reason, add a direct dependency on CONFIG_PCI to the IOSF_MBI driver. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 99a62ba409df..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.19.0

3 5

[alsa-devel] [PATCH 1/2] ASoC: cs4341: Add driver for CS4341 DAC
by Alexander Shiyan 07 Jan '19

07 Jan '19

This patch adds Cirrus Logic CS4341. This is a very simple, playback only, stereo DAC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4341.c | 346 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 355 insertions(+) create mode 100644 sound/soc/codecs/cs4341.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f1848c9b..cd0f571f004d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -64,6 +64,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS43130 if I2C + select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 select SND_SOC_CS53L30 if I2C @@ -536,6 +537,12 @@ config SND_SOC_CS43130 tristate "Cirrus Logic CS43130 CODEC" depends on I2C +config SND_SOC_CS4341 + tristate "Cirrus Logic CS4341 CODEC" + depends on I2C || SPI_MASTER + select REGMAP_I2C if I2C + select REGMAP_SPI if SPI_MASTER + # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c8dbfa..63ac31c51356 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -59,6 +59,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs43130-objs := cs43130.o +snd-soc-cs4341-objs := cs4341.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o snd-soc-cs53l30-objs := cs53l30.o @@ -324,6 +325,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o +obj-$(CONFIG_SND_SOC_CS4341) += snd-soc-cs4341.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c new file mode 100644 index 000000000000..24ecadca5b25 --- /dev/null +++ b/sound/soc/codecs/cs4341.c @@ -0,0 +1,346 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Cirrus Logic CS4341A ALSA SoC Codec Driver + * Author: Alexander Shiyan <shc_work(a)mail.ru> + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define CS4341_REG_MODE1 0x00 +#define CS4341_REG_MODE2 0x01 +#define CS4341_REG_MIX 0x02 +#define CS4341_REG_VOLA 0x03 +#define CS4341_REG_VOLB 0x04 + +#define CS4341_MODE2_DIF (7 << 4) +#define CS4341_MODE2_DIF_I2S_24 (0 << 4) +#define CS4341_MODE2_DIF_I2S_16 (1 << 4) +#define CS4341_MODE2_DIF_LJ_24 (2 << 4) +#define CS4341_MODE2_DIF_RJ_24 (3 << 4) +#define CS4341_MODE2_DIF_RJ_16 (5 << 4) +#define CS4341_VOLX_MUTE (1 << 7) + +struct cs4341_priv { + unsigned int fmt; + struct regmap *regmap; + struct regmap_config regcfg; +}; + +static const struct reg_default cs4341_reg_defaults[] = { + { CS4341_REG_MODE1, 0x00 }, + { CS4341_REG_MODE2, 0x82 }, + { CS4341_REG_MIX, 0x49 }, + { CS4341_REG_VOLA, 0x80 }, + { CS4341_REG_VOLB, 0x80 }, +}; + +static int cs4341_set_fmt(struct snd_soc_dai *dai, unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct cs4341_priv *cs4341 = snd_soc_component_get_drvdata(component); + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4341->fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4341_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs4341_priv *cs4341 = snd_soc_component_get_drvdata(component); + unsigned int mode = 0; + int b24 = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + b24 = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(component->dev, "Unsupported PCM format 0x%08x.\n", + params_format(params)); + return -EINVAL; + } + + switch (cs4341->fmt) { + case SND_SOC_DAIFMT_I2S: + mode = b24 ? CS4341_MODE2_DIF_I2S_24 : CS4341_MODE2_DIF_I2S_16; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = CS4341_MODE2_DIF_LJ_24; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode = b24 ? CS4341_MODE2_DIF_RJ_24 : CS4341_MODE2_DIF_RJ_16; + break; + default: + dev_err(component->dev, "Unsupported DAI format 0x%08x.\n", + cs4341->fmt); + return -EINVAL; + } + + return snd_soc_component_update_bits(component, CS4341_REG_MODE2, + CS4341_MODE2_DIF, mode); +} + +static int cs4341_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = snd_soc_component_update_bits(component, CS4341_REG_VOLA, + CS4341_VOLX_MUTE, + mute ? CS4341_VOLX_MUTE : 0); + if (ret < 0) + return ret; + + return snd_soc_component_update_bits(component, CS4341_REG_VOLB, + CS4341_VOLX_MUTE, + mute ? CS4341_VOLX_MUTE : 0); +} + +static DECLARE_TLV_DB_SCALE(out_tlv, -9000, 100, 0); + +static const char * const deemph[] = { + "None", "44.1k", "48k", "32k", +}; + +static const struct soc_enum deemph_enum = + SOC_ENUM_SINGLE(CS4341_REG_MODE2, 2, 4, deemph); + +static const char * const srzc[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static const struct soc_enum srzc_enum = + SOC_ENUM_SINGLE(CS4341_REG_MIX, 5, 4, srzc); + + +static const struct snd_soc_dapm_widget cs4341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OutA"), + SND_SOC_DAPM_OUTPUT("OutB"), +}; + +static const struct snd_soc_dapm_route cs4341_routes[] = { + { "OutA", NULL, "HiFi DAC" }, + { "OutB", NULL, "HiFi DAC" }, + { "DAC Playback", NULL, "OutA" }, + { "DAC Playback", NULL, "OutB" }, +}; + +static const struct snd_kcontrol_new cs4341_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4341_REG_VOLA, CS4341_REG_VOLB, 0, 90, 1, out_tlv), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", srzc_enum), + SOC_SINGLE("Auto-Mute Switch", CS4341_REG_MODE2, 7, 1, 0), + SOC_SINGLE("Popguard Transient Switch", CS4341_REG_MODE2, 1, 1, 0), +}; + +static const struct snd_soc_dai_ops cs4341_dai_ops = { + .set_fmt = cs4341_set_fmt, + .hw_params = cs4341_hw_params, + .digital_mute = cs4341_digital_mute, +}; + +static struct snd_soc_dai_driver cs4341_dai = { + .name = "cs4341a-hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &cs4341_dai_ops, + .symmetric_rates = 1, +}; + +static const struct snd_soc_component_driver soc_component_cs4341 = { + .controls = cs4341_controls, + .num_controls = ARRAY_SIZE(cs4341_controls), + .dapm_widgets = cs4341_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4341_dapm_widgets), + .dapm_routes = cs4341_routes, + .num_dapm_routes = ARRAY_SIZE(cs4341_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct of_device_id __maybe_unused cs4341_dt_ids[] = { + { .compatible = "cirrus,cs4341a", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4341_dt_ids); + +static int cs4341_probe(struct device *dev) +{ + struct cs4341_priv *cs4341 = dev_get_drvdata(dev); + int i; + + for (i = 0; i < ARRAY_SIZE(cs4341_reg_defaults); i++) + regmap_write(cs4341->regmap, cs4341_reg_defaults[i].reg, + cs4341_reg_defaults[i].def); + + return devm_snd_soc_register_component(dev, &soc_component_cs4341, + &cs4341_dai, 1); +} + +#if defined(CONFIG_I2C) +static int cs4341_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct cs4341_priv *cs4341; + + cs4341 = devm_kzalloc(&i2c->dev, sizeof(*cs4341), GFP_KERNEL); + if (!cs4341) + return -ENOMEM; + + i2c_set_clientdata(i2c, cs4341); + + cs4341->regcfg.reg_bits = 8; + cs4341->regcfg.val_bits = 8; + cs4341->regcfg.max_register = CS4341_REG_VOLB; + cs4341->regcfg.cache_type = REGCACHE_FLAT; + cs4341->regcfg.reg_defaults = cs4341_reg_defaults; + cs4341->regcfg.num_reg_defaults = ARRAY_SIZE(cs4341_reg_defaults); + cs4341->regmap = devm_regmap_init_i2c(i2c, &cs4341->regcfg); + if (IS_ERR(cs4341->regmap)) + return PTR_ERR(cs4341->regmap); + + return cs4341_probe(&i2c->dev); +} + +static const struct i2c_device_id cs4341_i2c_id[] = { + { "cs4341", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs4341_i2c_id); + +static struct i2c_driver cs4341_i2c_driver = { + .driver = { + .name = "cs4341-i2c", + .of_match_table = of_match_ptr(cs4341_dt_ids), + }, + .probe = cs4341_i2c_probe, + .id_table = cs4341_i2c_id, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static bool cs4341_reg_readable(struct device *dev, unsigned int reg) +{ + return false; +} + +static int cs4341_spi_probe(struct spi_device *spi) +{ + struct cs4341_priv *cs4341; + int ret; + + cs4341 = devm_kzalloc(&spi->dev, sizeof(*cs4341), GFP_KERNEL); + if (!cs4341) + return -ENOMEM; + + if (!spi->bits_per_word) + spi->bits_per_word = 8; + if (!spi->max_speed_hz) + spi->max_speed_hz = 6000000; + ret = spi_setup(spi); + if (ret) + return ret; + + spi_set_drvdata(spi, cs4341); + + cs4341->regcfg.reg_bits = 16; + cs4341->regcfg.val_bits = 8; + cs4341->regcfg.write_flag_mask = 0x20; + cs4341->regcfg.max_register = CS4341_REG_VOLB; + cs4341->regcfg.cache_type = REGCACHE_FLAT; + cs4341->regcfg.readable_reg = cs4341_reg_readable; + cs4341->regcfg.reg_defaults = cs4341_reg_defaults; + cs4341->regcfg.num_reg_defaults = ARRAY_SIZE(cs4341_reg_defaults); + cs4341->regmap = devm_regmap_init_spi(spi, &cs4341->regcfg); + if (IS_ERR(cs4341->regmap)) + return PTR_ERR(cs4341->regmap); + + return cs4341_probe(&spi->dev); +} + +static struct spi_driver cs4341_spi_driver = { + .driver = { + .name = "cs4341-spi", + .of_match_table = of_match_ptr(cs4341_dt_ids), + }, + .probe = cs4341_spi_probe, +}; +#endif + +static int __init cs4341_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) + ret = i2c_add_driver(&cs4341_i2c_driver); + if (ret) + return ret; +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4341_spi_driver); +#endif + + return ret; +} +module_init(cs4341_init); + +static void __exit cs4341_exit(void) +{ +#if defined(CONFIG_I2C) + i2c_del_driver(&cs4341_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4341_spi_driver); +#endif +} +module_exit(cs4341_exit); + +MODULE_AUTHOR("Alexander Shiyan <shc_work(a)mail.ru>"); +MODULE_DESCRIPTION("Cirrus Logic CS4341 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- 2.13.0

2 4

[alsa-devel] [PATCH 2/2] ASoC: cs4341: Add DT bindings documentation for CS4341 DAC
by Alexander Shiyan 07 Jan '19

07 Jan '19

This patch adds DT bindings documentation for Cirrus Logic CS4341 DAC. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- Documentation/devicetree/bindings/sound/cs4341.txt | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt diff --git a/Documentation/devicetree/bindings/sound/cs4341.txt b/Documentation/devicetree/bindings/sound/cs4341.txt new file mode 100644 index 000000000000..12b4aa8ef0db --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4341.txt @@ -0,0 +1,22 @@ +Cirrus Logic CS4341 audio DAC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "cirrus,cs4341a" + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +For required properties on I2C-bus, please consult +Documentation/devicetree/bindings/i2c/i2c.txt +For required properties on SPI-bus, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Example: + codec: cs4341@0 { + #sound-dai-cells = <0>; + compatible = "cirrus,cs4341a"; + reg = <0>; + spi-max-frequency = <6000000>; + }; -- 2.13.0

2 4

[alsa-devel] [PATCH 0/9] ASoC: Intel: fix Coccinelle and Sparse warnings
by Pierre-Louis Bossart 07 Jan '19

07 Jan '19

Make minor warnings go away to make sure Intel/SOF CI checks don't fail and new issues are detected. Sparse checks are not complete, the legacy stuff is just too messy to fix, we only check for soc/sof/intel/boards since it's the only part reused by SOF. Pierre-Louis Bossart (9): ASoC: Intel: Skylake: remove useless cast ASoC: Intel: Skylake: simplify boolean tests ASoC: Intel: Haswell: remove unneeded semicolon ASoC: Intel: Haswell: assign booleans to true/false ASoC: Intel: Baytrail: remove unneeded variable ASoC: Intel: Baytrail: simplify boolean test ASoC: Intel: Atom: simplify boolean tests ASoC: Intel: Atom: work-around Coccinelle false-alarm ASoC: Intel: boards: use snd_mask_set_format in all machine drivers sound/soc/intel/atom/sst-atom-controls.c | 4 ++-- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +-- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 4 ++-- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 2 +- 14 files changed, 17 insertions(+), 18 deletions(-) base-commit: ba26f635bf7253b42b08135152b18a2c8082abb7 -- 2.17.1

3 39

[alsa-devel] Applied "ASoC: Intel: Baytrail: remove unneeded variable" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: Baytrail: remove unneeded variable 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 060d35be2dfa9202d37f967fd20f133c530505d2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Sun, 16 Dec 2018 16:49:06 -0600 Subject: [PATCH] ASoC: Intel: Baytrail: remove unneeded variable Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 260447da32b8..2cd8f9668b50 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -278,7 +278,6 @@ static int sst_byt_process_notification(struct sst_byt *byt, struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -298,7 +297,7 @@ static int sst_byt_process_notification(struct sst_byt *byt, break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() 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 a0c426fe143328760c9fd565cd203a37a7b4fde8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:42 +0900 Subject: [PATCH] ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() We will get DAI ID from "reg" property if it has on DT, otherwise get it by counting port/endpoint. But in below case, we need to get DAI ID = 0 via port reg = <0>, but current implementation returns ID = 1, because it can't judge ID = 0 was from "non reg" or "reg = <0>". Thus, it will count port/endpoint number as "non reg" case. of_graph_parse_endpoint() implementation itself is not a problem, but because asoc_simple_card_get_dai_id() need to count port/endpoint number when "non reg" case, it need to know ID = 0 was from "non reg" or "reg = <0>". This patch fix this issue. port { reg = <0>; xxxx: endpoint@0 { }; => xxxx: endpoint@1 { }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b807a47515eb..336895f7fd1e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -283,12 +283,20 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) /* use endpoint/port reg if exist */ ret = of_graph_parse_endpoint(ep, &info); if (ret == 0) { - if (info.id) + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) return info.id; - if (info.port) + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) return info.port; } - node = of_graph_get_port_parent(ep); /* -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add asoc_graph_card_get_conversion()" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: audio-graph-card: add asoc_graph_card_get_conversion() 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 40dfae169ad047535d566a4791daae3b08f71c0c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:48 +0900 Subject: [PATCH] ASoC: audio-graph-card: add asoc_graph_card_get_conversion() audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_graph_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0d6144560a1e..c3e80bc27e80 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -169,6 +169,22 @@ static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_graph_card_get_conversion(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + + asoc_simple_card_parse_convert(dev, top, NULL, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, ports, NULL, adata); + asoc_simple_card_parse_convert(dev, port, NULL, adata); + asoc_simple_card_parse_convert(dev, ep, NULL, adata); +} + static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -194,11 +210,7 @@ static int asoc_graph_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - asoc_simple_card_parse_convert(dev, top, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ports, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, port, NULL, &dai_props->adata); - asoc_simple_card_parse_convert(dev, ep, NULL, &dai_props->adata); + asoc_graph_card_get_conversion(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc 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 e4f4fdfc57d9c846862ea6109e356b3a4542df5b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:27 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/audio-graph-scu-card.txt | 123 ------------------ 1 file changed, 123 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt deleted file mode 100644 index 62d42768a00b..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ /dev/null @@ -1,123 +0,0 @@ -Audio-Graph-SCU-Card: - -Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". - -It is based on common bindings for device graphs. -see ${LINUX}/Documentation/devicetree/bindings/graph.txt - -Basically, Audio-Graph-SCU-Card property is same as -Simple-Card / Simple-SCU-Card / Audio-Graph-Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt - ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt - -Below are same as Simple-Card / Audio-Graph-Card. - -- label -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- dai-tdm-slot-num -- dai-tdm-slot-width -- clocks / system-clock-frequency - -Below are same as Simple-SCU-Card. - -- convert-rate -- convert-channels -- prefix -- routing - -Required properties: - -- compatible : "audio-graph-scu-card"; -- dais : list of CPU DAI port{s} - -Example 1. Sampling Rate Conversion - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - -Example 2. 2 CPU 1 Codec (Mixing) - - sound_card { - compatible = "audio-graph-scu-card"; - - label = "sound-card"; - routing = "codec Playback", "DAI0 Playback", - "codec Playback", "DAI1 Playback", - "DAI0 Capture", "codec Capture"; - - dais = <&cpu_port0 - &cpu_port1>; - }; - - audio-codec { - ... - - audio-graph-card,prefix = "codec"; - audio-graph-card,convert-rate = <48000>; - port { - codec_endpoint0: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: remove audio-graph-scu-card" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: audio-graph-scu-card: remove audio-graph-scu-card 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 61c263ac27a307cdf7f46aaee4810619103effad Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 18 Dec 2018 11:50:32 +0900 Subject: [PATCH] ASoC: audio-graph-scu-card: remove audio-graph-scu-card It is already merged into audio-graph-card. audio-graph-scu-card is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/Kconfig | 9 - sound/soc/generic/Makefile | 2 - sound/soc/generic/audio-graph-scu-card.c | 501 ----------------------- 3 files changed, 512 deletions(-) delete mode 100644 sound/soc/generic/audio-graph-scu-card.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 92c2cf06f40a..59190f42fc08 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -24,12 +24,3 @@ config SND_AUDIO_GRAPH_CARD This option enables generic simple sound card support with OF-graph DT bindings. It also support DPCM of multi CPU single Codec ststem. - -config SND_AUDIO_GRAPH_SCU_CARD - tristate "ASoC Audio Graph SCU sound card support" - depends on OF - select SND_SIMPLE_CARD_UTILS - help - This option enables generic simple SCU sound card support - with OF-graph DT bindings. - It supports DPCM of multi CPU single Codec ststem. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9dec293a4c4d..9fbfdd524b24 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -3,10 +3,8 @@ snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o -snd-soc-audio-graph-scu-card-objs := audio-graph-scu-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o -obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD) += snd-soc-audio-graph-scu-card.o diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c deleted file mode 100644 index e1b192ea147b..000000000000 --- a/sound/soc/generic/audio-graph-scu-card.c +++ /dev/null @@ -1,501 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// ASoC audio graph SCU sound card support -// -// Copyright (C) 2017 Renesas Solutions Corp. -// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> -// -// based on -// ${LINUX}/sound/soc/generic/simple-scu-card.c -// ${LINUX}/sound/soc/generic/audio-graph-card.c - -#include <linux/clk.h> -#include <linux/device.h> -#include <linux/gpio.h> -#include <linux/module.h> -#include <linux/of.h> -#include <linux/of_device.h> -#include <linux/of_gpio.h> -#include <linux/of_graph.h> -#include <linux/platform_device.h> -#include <linux/string.h> -#include <sound/jack.h> -#include <sound/simple_card_utils.h> - -struct graph_card_data { - struct snd_soc_card snd_card; - struct graph_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; - struct snd_soc_dai_link_component platform; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - } *dai_props; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; -}; - -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - -#define PREFIX "audio-graph-card," - -static int asoc_graph_card_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - -static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - -static const struct snd_soc_ops asoc_graph_card_ops = { - .startup = asoc_graph_card_startup, - .shutdown = asoc_graph_card_shutdown, -}; - -static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - -static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - /* overwrite by top level adata if exist */ - asoc_simple_card_convert_fixup(&priv->adata, params); - - return 0; -} - -static int asoc_graph_card_dai_link_of(struct device_node *cpu_ep, - struct device_node *codec_ep, - struct graph_card_data *priv, - int *dai_idx, int link_idx, - int *conf_idx, int is_fe) -{ - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, link_idx); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, link_idx); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *ep = is_fe ? cpu_ep : codec_ep; - struct device_node *node = of_graph_get_port_parent(ep); - struct asoc_simple_dai *dai; - int ret; - - if (is_fe) { - struct snd_soc_dai_link_component *codecs; - - /* BE is dummy */ - codecs = dai_link->codecs; - codecs->of_node = NULL; - codecs->dai_name = "snd-soc-dummy-dai"; - codecs->name = "snd-soc-dummy"; - - /* FE settings */ - dai_link->dynamic = 1; - dai_link->dpcm_merged_format = 1; - - dai = - dai_props->cpu_dai = &priv->dais[(*dai_idx)++]; - - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); - if (ret) - return ret; - - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); - if (ret < 0) - return ret; - - /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); - } else { - struct snd_soc_codec_conf *cconf; - - /* FE is dummy */ - dai_link->cpu_of_node = NULL; - dai_link->cpu_dai_name = "snd-soc-dummy-dai"; - dai_link->cpu_name = "snd-soc-dummy"; - - /* BE settings */ - dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = asoc_graph_card_be_hw_params_fixup; - - dai = - dai_props->codec_dai = &priv->dais[(*dai_idx)++]; - - cconf = - dai_props->codec_conf = &priv->codec_conf[(*conf_idx)++]; - - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - dai_link->codecs->dai_name); - if (ret < 0) - return ret; - - /* check "prefix" from top node */ - snd_soc_of_parse_audio_prefix(card, cconf, - dai_link->codecs->of_node, - "prefix"); - /* check "prefix" from each node if top doesn't have */ - if (!cconf->of_node) - snd_soc_of_parse_node_prefix(node, cconf, - dai_link->codecs->of_node, - PREFIX "prefix"); - } - - asoc_simple_card_parse_convert(dev, node, PREFIX, &dai_props->adata); - - ret = asoc_simple_card_of_parse_tdm(ep, dai); - if (ret) - return ret; - - ret = asoc_simple_card_canonicalize_dailink(dai_link); - if (ret < 0) - return ret; - - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); - if (ret < 0) - return ret; - - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; - dai_link->ops = &asoc_graph_card_ops; - dai_link->init = asoc_graph_card_dai_init; - - return 0; -} - -static int asoc_graph_card_parse_of(struct graph_card_data *priv) -{ - struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_card *card = graph_priv_to_card(priv); - struct device_node *node = dev->of_node; - 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 dai_idx, link_idx, conf_idx, ret; - int rc, codec; - - if (!node) - return -EINVAL; - - /* - * we need to consider "widgets", "mclk-fs" around here - * see simple-card - */ - - ret = asoc_simple_card_of_parse_routing(card, NULL); - if (ret < 0) - return ret; - - asoc_simple_card_parse_convert(dev, node, NULL, &priv->adata); - - /* - * it supports multi CPU, single CODEC only here - * see asoc_graph_get_dais_count - */ - - link_idx = 0; - dai_idx = 0; - conf_idx = 0; - codec_port_old = NULL; - for (codec = 0; codec < 2; codec++) { - /* - * To listup valid sounds continuously, - * detect all CPU-dummy first, and - * detect all dummy-Codec second - */ - 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_ep); - of_node_put(codec_ep); - of_node_put(cpu_port); - of_node_put(codec_port); - it.node = NULL; - - if (codec) { - if (codec_port_old == codec_port) - continue; - - codec_port_old = codec_port; - } - - ret = asoc_graph_card_dai_link_of(cpu_ep, codec_ep, - priv, &dai_idx, - link_idx++, &conf_idx, - !codec); - if (ret < 0) - goto parse_of_err; - } - } - - ret = asoc_simple_card_parse_card_name(card, NULL); - if (ret) - goto parse_of_err; - - if ((card->num_links != link_idx) || - (card->num_configs != conf_idx)) { - dev_err(dev, "dai_link or codec_config wrong (%d/%d, %d/%d)\n", - card->num_links, link_idx, card->num_configs, conf_idx); - ret = -EINVAL; - goto parse_of_err; - } - - ret = 0; - -parse_of_err: - return ret; -} - -static void asoc_graph_get_dais_count(struct device *dev, - int *link_num, - int *dais_num, - int *ccnf_num) -{ - struct of_phandle_iterator it; - struct device_node *node = dev->of_node; - 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; - struct device_node *codec_port_old2; - int rc; - - /* - * link_num : number of links. - * CPU-Codec / CPU-dummy / dummy-Codec - * dais_num : number of DAIs - * ccnf_num : number of codec_conf - * same number for dummy-Codec - * - * ex1) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 7 - * CPU2 -/ ccnf : 1 - * CPU3 --- Codec2 - * - * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec - * => 7 DAIs = 4xCPU + 3xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex2) - * CPU0 --- Codec0 link : 5 - * CPU1 --- Codec1 dais : 6 - * CPU2 -/ ccnf : 1 - * CPU3 -/ - * - * => 5 links = 1xCPU-Codec + 3xCPU-dummy + 1xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 1 ccnf = 1xdummy-Codec - * - * ex3) - * CPU0 --- Codec0 link : 6 - * CPU1 -/ dais : 6 - * CPU2 --- Codec1 ccnf : 2 - * CPU3 -/ - * - * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec - * => 6 DAIs = 4xCPU + 2xCodec - * => 2 ccnf = 2xdummy-Codec - */ - codec_port_old = NULL; - codec_port_old2 = 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_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - - (*link_num)++; - (*dais_num)++; - - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } - - codec_port_old2 = codec_port_old; - continue; - } - - (*dais_num)++; - codec_port_old = codec_port; - } -} - -static int asoc_graph_card_probe(struct platform_device *pdev) -{ - struct graph_card_data *priv; - struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; - struct asoc_simple_dai *dais; - struct device *dev = &pdev->dev; - struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; - int lnum = 0, dnum = 0, cnum = 0; - int ret, i; - - /* Allocate the private data and the DAI link array */ - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - asoc_graph_get_dais_count(dev, &lnum, &dnum, &cnum); - if (!lnum || !dnum) - return -EINVAL; - - dai_props = devm_kcalloc(dev, lnum, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, lnum, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, dnum, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, cnum, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < lnum; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - /* Init snd_soc_card */ - card = graph_priv_to_card(priv); - card->owner = THIS_MODULE; - card->dev = dev; - card->dai_link = priv->dai_link; - card->num_links = lnum; - card->codec_conf = cconf; - card->num_configs = cnum; - - ret = asoc_graph_card_parse_of(priv); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - goto err; - } - - snd_soc_card_set_drvdata(card, priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - return 0; -err: - asoc_simple_card_clean_reference(card); - - return ret; -} - -static int asoc_graph_card_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - return asoc_simple_card_clean_reference(card); -} - -static const struct of_device_id asoc_graph_of_match[] = { - { .compatible = "audio-graph-scu-card", }, - {}, -}; -MODULE_DEVICE_TABLE(of, asoc_graph_of_match); - -static struct platform_driver asoc_graph_card = { - .driver = { - .name = "asoc-audio-graph-scu-card", - .pm = &snd_soc_pm_ops, - .of_match_table = asoc_graph_of_match, - }, - .probe = asoc_graph_card_probe, - .remove = asoc_graph_card_remove, -}; -module_platform_driver(asoc_graph_card); - -MODULE_ALIAS("platform:asoc-audio-graph-scu-card"); -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("ASoC Audio Graph SCU Sound Card"); -MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 2.20.1

1 0

[alsa-devel] [PATCH 0/4] ASoC: remove audio-graph-scu / simple-scu
by Kuninori Morimoto 07 Jan '19

07 Jan '19

Hi Mark In previous "audio-graph / simple" card merging patch-set, I mentioned I will post "remove scu" patch in next version for more safety. But I was confused. There is no xx-scu-card user on upstream (only me, locally). So, we can remove it immediately Kuninori Morimoto (4): ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc ASoC: audio-graph-scu-card: remove audio-graph-scu-card ASoC: simple-scu-card: remove simple-scu-card on Doc ASoC: simple-scu-card: remove simple-scu-card .../bindings/sound/audio-graph-scu-card.txt | 123 ----- .../devicetree/bindings/sound/simple-scu-card.txt | 94 ---- sound/soc/generic/Kconfig | 17 - sound/soc/generic/Makefile | 4 - sound/soc/generic/audio-graph-scu-card.c | 501 --------------------- sound/soc/generic/simple-scu-card.c | 474 ------------------- 6 files changed, 1213 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt delete mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt delete mode 100644 sound/soc/generic/audio-graph-scu-card.c delete mode 100644 sound/soc/generic/simple-scu-card.c -- 2.7.4

2 12

[alsa-devel] Applied "dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings 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 e3e12ec09a18ad779b637f4a006a908cb6045aa7 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:27 +0530 Subject: [PATCH] dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings Added documentation for audio formatter IP core DT bindings. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/xlnx,audio-formatter.txt | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt diff --git a/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt new file mode 100644 index 000000000000..cbc93c8f4963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt @@ -0,0 +1,29 @@ +Device-Tree bindings for Xilinx PL audio formatter + +The IP core supports DMA, data formatting(AES<->PCM conversion) +of audio samples. + +Required properties: + - compatible: "xlnx,audio-formatter-1.0" + - interrupt-names: Names specified to list of interrupts in same + order mentioned under "interrupts". + List of supported interrupt names are: + "irq_mm2s" : interrupt from MM2S block + "irq_s2mm" : interrupt from S2MM block + - interrupts-parent: Phandle for interrupt controller. + - interrupts: List of Interrupt numbers. + - reg: Base address and size of the IP core instance. + - clock-names: List of input clocks. + Required elements: "s_axi_lite_aclk", "aud_mclk" + - clocks: Input clock specifier. Refer to common clock bindings. + +Example: + audio_ss_0_audio_formatter_0: audio_formatter@80010000 { + compatible = "xlnx,audio-formatter-1.0"; + interrupt-names = "irq_mm2s", "irq_s2mm"; + interrupt-parent = <&gic>; + interrupts = <0 104 4>, <0 105 4>; + reg = <0x0 0x80010000 0x0 0x1000>; + clock-names = "s_axi_lite_aclk", "aud_mclk"; + clocks = <&clk 71>, <&clk_wiz_1 0>; + }; -- 2.20.1

1 0

[alsa-devel] [PATCH v2 0/3] add Xilinx audio formatter driver
by Maruthi Srinivas Bayyavarapu 07 Jan '19

07 Jan '19

Audio formatter IP supports two streaming interfaces - MM2S for playback and S2MM for capture. The driver enables DMA functionality for both the interfaces. Patchset includes devicetree bindings documentation, driver and build enablement. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,audio-formatter: Document audio formatter bindings ASoC: xlnx: add pcm formatter platform driver ASoC: xlnx: enable audio formatter driver build .../bindings/sound/xlnx,audio-formatter.txt | 29 ++ sound/soc/xilinx/Kconfig | 7 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_formatter_pcm.c | 565 +++++++++++++++++++++ 4 files changed, 603 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt create mode 100644 sound/soc/xilinx/xlnx_formatter_pcm.c -- 2.7.4

2 7

[alsa-devel] Applied "ASoC: xlnx: enable audio formatter driver build" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: xlnx: enable audio formatter driver build 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 b31daa15af760747b91dbb76c80306d77d8ae05f Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Date: Fri, 21 Dec 2018 14:27:29 +0530 Subject: [PATCH] ASoC: xlnx: enable audio formatter driver build Enable audio formatter driver build. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/xilinx/Kconfig | 7 +++++++ sound/soc/xilinx/Makefile | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 723a583a8d57..ac48d6a00c36 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -6,3 +6,10 @@ config SND_SOC_XILINX_I2S mode, IP receives audio in AES format, extracts PCM and sends PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. + +config SND_SOC_XILINX_AUDIO_FORMATTER + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index 6c1209b9ee75..432693b1cc79 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,2 +1,4 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o +obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: audio-graph-card: add 1 CPU : 1 Codec support again 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 de2949fe262197298036989924d05f5de6b9815a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:45:54 +0900 Subject: [PATCH] ASoC: audio-graph-card: add 1 CPU : 1 Codec support again audio-graph-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= audio-graph-scu) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged audio-graph-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 44 ++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c3e80bc27e80..638333cdac66 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -408,6 +408,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *codec_ep = NULL; struct device_node *codec_port = NULL; struct device_node *codec_port_old = NULL; + struct asoc_simple_card_data adata; int rc, ret; int link_idx, dai_idx, conf_idx; int cpu; @@ -453,7 +454,13 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) dev_dbg(dev, "%pOFf <-> %pOFf\n", cpu_ep, codec_ep); - if (of_get_child_count(codec_port) > 1) { + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); + + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || + adata.convert_channels) { /* * for DPCM sound */ @@ -495,7 +502,7 @@ static void asoc_graph_get_dais_count(struct device *dev, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old; - struct device_node *codec_port_old2; + struct asoc_simple_card_data adata; int rc; /* @@ -534,9 +541,17 @@ static void asoc_graph_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ codec_port_old = NULL; - codec_port_old2 = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = NULL; @@ -554,17 +569,22 @@ static void asoc_graph_get_dais_count(struct device *dev, (*link_num)++; (*dais_num)++; - if (codec_port_old == codec_port) { - if (codec_port_old2 != codec_port_old) { - (*link_num)++; - (*ccnf_num)++; - } + memset(&adata, 0, sizeof(adata)); + asoc_graph_card_get_conversion(dev, codec_ep, &adata); + asoc_graph_card_get_conversion(dev, cpu_ep, &adata); - codec_port_old2 = codec_port_old; - continue; - } + if ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels) { - (*dais_num)++; + if (codec_port_old == codec_port) + continue; + + (*link_num)++; + (*ccnf_num)++; + (*dais_num)++; + } else { + (*dais_num)++; + } codec_port_old = codec_port; } } -- 2.20.1

1 0

[alsa-devel] [PATCH 00/13 v2] ASoC: audio/simple card tidyup/cleanup
by Kuninori Morimoto 07 Jan '19

07 Jan '19

Hi Mark As I mentioned on off-list mail, I noticed that previous patch-set included many exchanges in 1 patch. Because of this, the reviewing was very difficult. I had break downed it into few more small patches. I hope it becomes more review-able 1) : for "reg" property issue, again. 2) - 7) : missing 1CPU-1Codec DPCM case, and cleanup for audio-graph 8) - 13) : missing 1CPU-1Codec DPCM case, and cleanup for simple-card Kuninori Morimoto (13): 1) ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() 2) ASoC: audio-graph-card: add asoc_graph_card_get_conversion() 3) ASoC: audio-graph-card: add 1 CPU : 1 Codec support again 4) ASoC: audio-graph-card: add link_info 5) ASoC: audio-graph-card: cleanup DAI link loop method - step1 6) ASoC: audio-graph-card: cleanup DAI link loop method - step2 7) ASoC: audio-graph-card: reduce naming prefix 8) ASoC: simple-card: add asoc_simple_card_get_conversion() 9) ASoC: simple-card: add 1 CPU : 1 Codec support again 10) ASoC: simple-card: add link_info 11) ASoC: simple-card: cleanup DAI link loop method - step1 12) ASoC: simple-card: cleanup DAI link loop method - step2 13) ASoC: simple-card: reduce naming prefix sound/soc/generic/audio-graph-card.c | 465 +++++++++++++++++++--------------- sound/soc/generic/simple-card-utils.c | 14 +- sound/soc/generic/simple-card.c | 454 ++++++++++++++++++++------------- 3 files changed, 553 insertions(+), 380 deletions(-) -- 2.7.4

2 45

[alsa-devel] Applied "ASoC: simple-card: add asoc_simple_card_get_conversion()" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: simple-card: add asoc_simple_card_get_conversion() 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 7e5e1f8bbaa82e6877d8bc121cb1b44cb1ce7ddf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:42 +0900 Subject: [PATCH] ASoC: simple-card: add asoc_simple_card_get_conversion() simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(), it need to judge whether it is DPCM by checking convert-rate/channel. For this purpose, this patch adds asoc_simple_card_get_conversion() as preparation Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001cf9cd1..52048069b25a 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -165,6 +165,21 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static void asoc_simple_card_get_conversion(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) +{ + struct device_node *top = dev->of_node; + struct device_node *node = of_get_parent(np); + + asoc_simple_card_parse_convert(dev, top, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, PREFIX, adata); + asoc_simple_card_parse_convert(dev, node, NULL, adata); + asoc_simple_card_parse_convert(dev, np, NULL, adata); + + of_node_put(node); +} + static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, struct device_node *node, struct device_node *np, @@ -260,9 +275,7 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, "prefix"); } - asoc_simple_card_parse_convert(dev, top, PREFIX, &dai_props->adata); - asoc_simple_card_parse_convert(dev, node, prefix, &dai_props->adata); - asoc_simple_card_parse_convert(dev, np, NULL, &dai_props->adata); + asoc_simple_card_get_conversion(dev, np, &dai_props->adata); ret = asoc_simple_card_of_parse_tdm(np, dai); if (ret) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: add 1 CPU : 1 Codec support again" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: simple-card: add 1 CPU : 1 Codec support again 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 7adee60ee2732f23f703ff83ee35caad561490ba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 20 Dec 2018 10:46:47 +0900 Subject: [PATCH] ASoC: simple-card: add 1 CPU : 1 Codec support again simple-card is now supporting normal sound and DPCM sound. For DPCM sound, original sound card (= simple-scu-card) had been supported 1 CPU : 1 Codec connection which uses hw_params_fixup() for convert-rate/channel. But, merged simple-card is completely forgeting about it. This patch re-support it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 52048069b25a..b15651409c7f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -453,6 +453,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) struct device_node *node; struct device_node *np; struct device_node *codec; + struct asoc_simple_card_data adata; bool is_fe; int ret, loop; int dai_idx, link_idx, conf_idx; @@ -480,8 +481,13 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + /* DPCM */ - if (of_get_child_count(node) > 2) { + if (of_get_child_count(node) > 2 || + adata.convert_rate || adata.convert_channels) { for_each_child_of_node(node, np) { codec = of_get_child_by_name(node, loop ? "codec" : @@ -495,14 +501,16 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) top, node, np, codec, priv, &dai_idx, link_idx++, &conf_idx, is_fe, !loop); + if (ret < 0) + return ret; } } else { ret = asoc_simple_card_dai_link_of( top, node, priv, &dai_idx, link_idx++, !loop); + if (ret < 0) + return ret; } - if (ret < 0) - return ret; node = of_get_next_child(top, node); } while (loop && node); @@ -523,6 +531,8 @@ static void asoc_simple_card_get_dais_count(struct device *dev, { struct device_node *top = dev->of_node; struct device_node *node; + struct device_node *np; + struct asoc_simple_card_data adata; int loop; int num; @@ -562,6 +572,15 @@ static void asoc_simple_card_get_dais_count(struct device *dev, * => 6 links = 0xCPU-Codec + 4xCPU-dummy + 2xdummy-Codec * => 6 DAIs = 4xCPU + 2xCodec * => 2 ccnf = 2xdummy-Codec + * + * ex4) + * CPU0 --- Codec0 (convert-rate) link : 3 + * CPU1 --- Codec1 dais : 4 + * ccnf : 1 + * + * => 3 links = 1xCPU-Codec + 1xCPU-dummy + 1xdummy-Codec + * => 4 DAIs = 2xCPU + 2xCodec + * => 1 ccnf = 1xdummy-Codec */ if (!top) { (*link_num) = 1; @@ -578,9 +597,14 @@ static void asoc_simple_card_get_dais_count(struct device *dev, } do { + memset(&adata, 0, sizeof(adata)); + for_each_child_of_node(node, np) + asoc_simple_card_get_conversion(dev, np, &adata); + num = of_get_child_count(node); (*dais_num) += num; - if (num > 2) { + if (num > 2 || + adata.convert_rate || adata.convert_channels) { (*link_num) += num; (*ccnf_num)++; } else { -- 2.20.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: rockchip: support ACODEC for rk3328
by Katsuhiro Suzuki 07 Jan '19

07 Jan '19

This patch adds support for audio CODEC core of rk3328. Rockchip does not publish detail specification of this core but driver source code is opened on their GitHub repository. https://github.com/rockchip-linux/kernel So I ported this code to linux-next and added some trivial fixes. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 23 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3328_codec.c | 517 ++++++++++++++++++ sound/soc/codecs/rk3328_codec.h | 210 +++++++ 5 files changed, 757 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 sound/soc/codecs/rk3328_codec.c create mode 100644 sound/soc/codecs/rk3328_codec.h diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt new file mode 100644 index 000000000000..2469588c7ccb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -0,0 +1,23 @@ +* Rockchip Rk3328 internal codec + +Required properties: + +- compatible: "rockchip,rk3328-codec" +- reg: physical base address of the controller and length of memory mapped + region. +- rockchip,grf: the phandle of the syscon node for GRF register. +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should be "pclk". +- spk-depop-time-ms: speak depop time msec. + +Example for rk3328 internal codec: + +codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0x0 0xff410000 0x0 0x1000>; + rockchip,grf = <&grf>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + spk-depop-time-ms = 100; + status = "disabled"; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62bdb7e333b8..19a5168e0b89 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,6 +129,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RK3328 select SND_SOC_RT274 if I2C select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C @@ -799,6 +800,10 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_RK3328 + tristate "Rockchip RK3328 audio CODEC" + select REGMAP_MMIO + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 66f55d185620..0d924efbc8ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -132,6 +132,7 @@ snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-rk3328-objs := rk3328_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1305-objs := rt1305.o @@ -398,6 +399,7 @@ obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c new file mode 100644 index 000000000000..71f3fc2d970c --- /dev/null +++ b/sound/soc/codecs/rk3328_codec.c @@ -0,0 +1,517 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk3328 ALSA SoC Audio driver +// +// Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/mfd/syscon.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include "rk3328_codec.h" + +/* + * volume setting + * 0: -39dB + * 26: 0dB + * 31: 6dB + * Step: 1.5dB + */ +#define OUT_VOLUME (0x18) +#define RK3328_GRF_SOC_CON2 (0x0408) +#define RK3328_GRF_SOC_CON10 (0x0428) +#define INITIAL_FREQ (11289600) + +struct rk3328_codec_priv { + struct regmap *regmap; + struct regmap *grf; + struct clk *mclk; + struct clk *pclk; + unsigned int sclk; + int spk_depop_time; /* msec */ +}; + +static const struct reg_default rk3328_codec_reg_defaults[] = { + { CODEC_RESET, 0x03 }, + { DAC_INIT_CTRL1, 0x00 }, + { DAC_INIT_CTRL2, 0x50 }, + { DAC_INIT_CTRL3, 0x0e }, + { DAC_PRECHARGE_CTRL, 0x01 }, + { DAC_PWR_CTRL, 0x00 }, + { DAC_CLK_CTRL, 0x00 }, + { HPMIX_CTRL, 0x00 }, + { HPOUT_CTRL, 0x00 }, + { HPOUTL_GAIN_CTRL, 0x00 }, + { HPOUTR_GAIN_CTRL, 0x00 }, + { HPOUT_POP_CTRL, 0x11 }, +}; + +static int rk3328_codec_reset(struct rk3328_codec_priv *rk3328) +{ + regmap_write(rk3328->regmap, CODEC_RESET, 0x00); + mdelay(10); + regmap_write(rk3328->regmap, CODEC_RESET, 0x03); + + return 0; +} + +static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + val = PIN_DIRECTION_IN | DAC_I2S_MODE_SLAVE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + val = PIN_DIRECTION_OUT | DAC_I2S_MODE_MASTER; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL1, + PIN_DIRECTION_MASK | DAC_I2S_MODE_MASK, val); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + val = DAC_MODE_PCM; + break; + case SND_SOC_DAIFMT_I2S: + val = DAC_MODE_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = DAC_MODE_RJM; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = DAC_MODE_LJM; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL2, + DAC_MODE_MASK, val); + + return 0; +} + +static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute) +{ + unsigned int val = BIT(17); + + if (mute) + val |= BIT(1); + + regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val); +} + +static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val; + + if (mute) + val = HPOUTL_MUTE | HPOUTR_MUTE; + else + val = HPOUTL_UNMUTE | HPOUTR_UNMUTE; + + regmap_update_bits(rk3328->regmap, HPOUT_CTRL, + HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, val); + + return 0; +} + +static int rk3328_codec_power_on(struct rk3328_codec_priv *rk3328, int wait_ms) +{ + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_XCHARGE_MASK, DAC_CHARGE_PRECHARGE); + mdelay(10); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_ALL_ON); + mdelay(wait_ms); + + return 0; +} + +static int rk3328_codec_power_off(struct rk3328_codec_priv *rk3328, int wait_ms) +{ + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_XCHARGE_MASK, DAC_CHARGE_DISCHARGE); + mdelay(10); + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_ALL_ON); + mdelay(wait_ms); + + return 0; +} + +static const struct rk3328_reg_msk_val playback_open_list[] = { + { DAC_PWR_CTRL, DAC_PWR_MASK, DAC_PWR_ON }, + { DAC_PWR_CTRL, DACL_PATH_REFV_MASK | DACR_PATH_REFV_MASK, + DACL_PATH_REFV_ON | DACR_PATH_REFV_ON }, + { DAC_PWR_CTRL, HPOUTL_ZERO_CROSSING_MASK | HPOUTR_ZERO_CROSSING_MASK, + HPOUTL_ZERO_CROSSING_ON | HPOUTR_ZERO_CROSSING_ON }, + { HPOUT_POP_CTRL, HPOUTR_POP_MASK | HPOUTL_POP_MASK, + HPOUTR_POP_WORK | HPOUTL_POP_WORK }, + { HPMIX_CTRL, HPMIXL_MASK | HPMIXR_MASK, HPMIXL_EN | HPMIXR_EN }, + { HPMIX_CTRL, HPMIXL_INIT_MASK | HPMIXR_INIT_MASK, + HPMIXL_INIT_EN | HPMIXR_INIT_EN }, + { HPOUT_CTRL, HPOUTL_MASK | HPOUTR_MASK, HPOUTL_EN | HPOUTR_EN }, + { HPOUT_CTRL, HPOUTL_INIT_MASK | HPOUTR_INIT_MASK, + HPOUTL_INIT_EN | HPOUTR_INIT_EN }, + { DAC_CLK_CTRL, DACL_REFV_MASK | DACR_REFV_MASK, + DACL_REFV_ON | DACR_REFV_ON }, + { DAC_CLK_CTRL, DACL_CLK_MASK | DACR_CLK_MASK, + DACL_CLK_ON | DACR_CLK_ON }, + { DAC_CLK_CTRL, DACL_MASK | DACR_MASK, DACL_ON | DACR_ON }, + { DAC_CLK_CTRL, DACL_INIT_MASK | DACR_INIT_MASK, + DACL_INIT_ON | DACR_INIT_ON }, + { DAC_SELECT, DACL_SELECT_MASK | DACR_SELECT_MASK, + DACL_SELECT | DACR_SELECT }, + { HPMIX_CTRL, HPMIXL_INIT2_MASK | HPMIXR_INIT2_MASK, + HPMIXL_INIT2_EN | HPMIXR_INIT2_EN }, + { HPOUT_CTRL, HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, + HPOUTL_UNMUTE | HPOUTR_UNMUTE }, +}; + +static int rk3328_codec_open_playback(struct rk3328_codec_priv *rk3328) +{ + int i; + + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_I); + + for (i = 0; i < ARRAY_SIZE(playback_open_list); i++) { + regmap_update_bits(rk3328->regmap, + playback_open_list[i].reg, + playback_open_list[i].msk, + playback_open_list[i].val); + mdelay(1); + } + + msleep(rk3328->spk_depop_time); + rk3328_analog_output(rk3328, 1); + + regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, + HPOUTL_GAIN_MASK, OUT_VOLUME); + regmap_update_bits(rk3328->regmap, HPOUTR_GAIN_CTRL, + HPOUTR_GAIN_MASK, OUT_VOLUME); + + return 0; +} + +static const struct rk3328_reg_msk_val playback_close_list[] = { + { HPMIX_CTRL, HPMIXL_INIT2_MASK | HPMIXR_INIT2_MASK, + HPMIXL_INIT2_DIS | HPMIXR_INIT2_DIS }, + { DAC_SELECT, DACL_SELECT_MASK | DACR_SELECT_MASK, + DACL_UNSELECT | DACR_UNSELECT }, + { HPOUT_CTRL, HPOUTL_MUTE_MASK | HPOUTR_MUTE_MASK, + HPOUTL_MUTE | HPOUTR_MUTE }, + { HPOUT_CTRL, HPOUTL_INIT_MASK | HPOUTR_INIT_MASK, + HPOUTL_INIT_DIS | HPOUTR_INIT_DIS }, + { HPOUT_CTRL, HPOUTL_MASK | HPOUTR_MASK, HPOUTL_DIS | HPOUTR_DIS }, + { HPMIX_CTRL, HPMIXL_MASK | HPMIXR_MASK, HPMIXL_DIS | HPMIXR_DIS }, + { DAC_CLK_CTRL, DACL_MASK | DACR_MASK, DACL_OFF | DACR_OFF }, + { DAC_CLK_CTRL, DACL_CLK_MASK | DACR_CLK_MASK, + DACL_CLK_OFF | DACR_CLK_OFF }, + { DAC_CLK_CTRL, DACL_REFV_MASK | DACR_REFV_MASK, + DACL_REFV_OFF | DACR_REFV_OFF }, + { HPOUT_POP_CTRL, HPOUTR_POP_MASK | HPOUTL_POP_MASK, + HPOUTR_POP_XCHARGE | HPOUTL_POP_XCHARGE }, + { DAC_PWR_CTRL, DACL_PATH_REFV_MASK | DACR_PATH_REFV_MASK, + DACL_PATH_REFV_OFF | DACR_PATH_REFV_OFF }, + { DAC_PWR_CTRL, DAC_PWR_MASK, DAC_PWR_OFF }, + { HPMIX_CTRL, HPMIXL_INIT_MASK | HPMIXR_INIT_MASK, + HPMIXL_INIT_DIS | HPMIXR_INIT_DIS }, + { DAC_CLK_CTRL, DACL_INIT_MASK | DACR_INIT_MASK, + DACL_INIT_OFF | DACR_INIT_OFF }, +}; + +static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328) +{ + size_t i; + + rk3328_analog_output(rk3328, 0); + + regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, + HPOUTL_GAIN_MASK, 0); + regmap_update_bits(rk3328->regmap, HPOUTR_GAIN_CTRL, + HPOUTR_GAIN_MASK, 0); + + for (i = 0; i < ARRAY_SIZE(playback_close_list); i++) { + regmap_update_bits(rk3328->regmap, + playback_close_list[i].reg, + playback_close_list[i].msk, + playback_close_list[i].val); + mdelay(1); + } + + regmap_update_bits(rk3328->regmap, DAC_PRECHARGE_CTRL, + DAC_CHARGE_CURRENT_ALL_MASK, + DAC_CHARGE_CURRENT_I); + + return 0; +} + +static int rk3328_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + unsigned int val = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val = DAC_VDL_16BITS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = DAC_VDL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = DAC_VDL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val = DAC_VDL_32BITS; + break; + default: + return -EINVAL; + } + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL2, DAC_VDL_MASK, val); + + val = DAC_WL_32BITS | DAC_RST_DIS; + regmap_update_bits(rk3328->regmap, DAC_INIT_CTRL3, + DAC_WL_MASK | DAC_RST_MASK, val); + + return 0; +} + +static int rk3328_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + + return rk3328_codec_open_playback(rk3328); +} + +static void rk3328_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(dai->component); + + rk3328_codec_close_playback(rk3328); +} + +static const struct snd_soc_dai_ops rk3328_dai_ops = { + .hw_params = rk3328_hw_params, + .set_fmt = rk3328_set_dai_fmt, + .digital_mute = rk3328_digital_mute, + .startup = rk3328_pcm_startup, + .shutdown = rk3328_pcm_shutdown, +}; + +static struct snd_soc_dai_driver rk3328_dai[] = { + { + .name = "rk3328-hifi", + .id = RK3328_HIFI, + .playback = { + .stream_name = "HIFI Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .capture = { + .stream_name = "HIFI Capture", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &rk3328_dai_ops, + }, +}; + +static int rk3328_codec_probe(struct snd_soc_component *component) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(component); + + rk3328_codec_reset(rk3328); + rk3328_codec_power_on(rk3328, 0); + + return 0; +} + +static void rk3328_codec_remove(struct snd_soc_component *component) +{ + struct rk3328_codec_priv *rk3328 = + snd_soc_component_get_drvdata(component); + + rk3328_codec_close_playback(rk3328); + rk3328_codec_power_off(rk3328, 0); +} + +static const struct snd_soc_component_driver soc_codec_rk3328 = { + .probe = rk3328_codec_probe, + .remove = rk3328_codec_remove, +}; + +static bool rk3328_codec_write_read_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CODEC_RESET: + case DAC_INIT_CTRL1: + case DAC_INIT_CTRL2: + case DAC_INIT_CTRL3: + case DAC_PRECHARGE_CTRL: + case DAC_PWR_CTRL: + case DAC_CLK_CTRL: + case HPMIX_CTRL: + case DAC_SELECT: + case HPOUT_CTRL: + case HPOUTL_GAIN_CTRL: + case HPOUTR_GAIN_CTRL: + case HPOUT_POP_CTRL: + return true; + default: + return false; + } +} + +static bool rk3328_codec_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CODEC_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_config rk3328_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = HPOUT_POP_CTRL, + .writeable_reg = rk3328_codec_write_read_reg, + .readable_reg = rk3328_codec_write_read_reg, + .volatile_reg = rk3328_codec_volatile_reg, + .reg_defaults = rk3328_codec_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk3328_codec_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +static int rk3328_platform_probe(struct platform_device *pdev) +{ + struct device_node *rk3328_np = pdev->dev.of_node; + struct rk3328_codec_priv *rk3328; + struct resource *res; + struct regmap *grf; + void __iomem *base; + int ret = 0; + + rk3328 = devm_kzalloc(&pdev->dev, sizeof(*rk3328), GFP_KERNEL); + if (!rk3328) + return -ENOMEM; + + grf = syscon_regmap_lookup_by_phandle(rk3328_np, + "rockchip,grf"); + if (IS_ERR(grf)) { + dev_err(&pdev->dev, "missing 'rockchip,grf'\n"); + return PTR_ERR(grf); + } + rk3328->grf = grf; + /* enable i2s_acodec_en */ + regmap_write(grf, RK3328_GRF_SOC_CON2, + (BIT(14) << 16 | BIT(14))); + + ret = of_property_read_u32(rk3328_np, "spk-depop-time-ms", + &rk3328->spk_depop_time); + if (ret < 0) { + dev_info(&pdev->dev, "spk_depop_time use default value.\n"); + rk3328->spk_depop_time = 200; + } + + rk3328_analog_output(rk3328, 0); + + rk3328->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk3328->mclk)) + return PTR_ERR(rk3328->mclk); + + ret = clk_prepare_enable(rk3328->mclk); + if (ret) + return ret; + clk_set_rate(rk3328->mclk, INITIAL_FREQ); + + rk3328->pclk = devm_clk_get(&pdev->dev, "pclk"); + if (IS_ERR(rk3328->pclk)) { + dev_err(&pdev->dev, "can't get acodec pclk\n"); + return PTR_ERR(rk3328->pclk); + } + + ret = clk_prepare_enable(rk3328->pclk); + if (ret < 0) { + dev_err(&pdev->dev, "failed to enable acodec pclk\n"); + return ret; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &rk3328_codec_regmap_config); + if (IS_ERR(rk3328->regmap)) + return PTR_ERR(rk3328->regmap); + + platform_set_drvdata(pdev, rk3328); + + return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + rk3328_dai, + ARRAY_SIZE(rk3328_dai)); +} + +static const struct of_device_id rk3328_codec_of_match[] = { + { .compatible = "rockchip,rk3328-codec", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk3328_codec_of_match); + +static struct platform_driver rk3328_codec_driver = { + .driver = { + .name = "rk3328-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rk3328_codec_of_match), + }, + .probe = rk3328_platform_probe, +}; +module_platform_driver(rk3328_codec_driver); + +MODULE_AUTHOR("Sugar Zhang <sugar.zhang(a)rock-chips.com>"); +MODULE_DESCRIPTION("ASoC rk3328 codec driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk3328_codec.h b/sound/soc/codecs/rk3328_codec.h new file mode 100644 index 000000000000..655103586241 --- /dev/null +++ b/sound/soc/codecs/rk3328_codec.h @@ -0,0 +1,210 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rk3328 ALSA SoC Audio driver + * + * Copyright (c) 2017, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + */ + +#ifndef _RK3328_CODEC_H +#define _RK3328_CODEC_H + +#include <linux/bitfield.h> + +/* codec register */ +#define CODEC_RESET (0x00 << 2) +#define DAC_INIT_CTRL1 (0x03 << 2) +#define DAC_INIT_CTRL2 (0x04 << 2) +#define DAC_INIT_CTRL3 (0x05 << 2) +#define DAC_PRECHARGE_CTRL (0x22 << 2) +#define DAC_PWR_CTRL (0x23 << 2) +#define DAC_CLK_CTRL (0x24 << 2) +#define HPMIX_CTRL (0x25 << 2) +#define DAC_SELECT (0x26 << 2) +#define HPOUT_CTRL (0x27 << 2) +#define HPOUTL_GAIN_CTRL (0x28 << 2) +#define HPOUTR_GAIN_CTRL (0x29 << 2) +#define HPOUT_POP_CTRL (0x2a << 2) + +/* REG00: CODEC_RESET */ +#define PWR_RST_BYPASS_DIS (0x0 << 6) +#define PWR_RST_BYPASS_EN (0x1 << 6) +#define DIG_CORE_RST (0x0 << 1) +#define DIG_CORE_WORK (0x1 << 1) +#define SYS_RST (0x0 << 0) +#define SYS_WORK (0x1 << 0) + +/* REG03: DAC_INIT_CTRL1 */ +#define PIN_DIRECTION_MASK BIT(5) +#define PIN_DIRECTION_IN (0x0 << 5) +#define PIN_DIRECTION_OUT (0x1 << 5) +#define DAC_I2S_MODE_MASK BIT(4) +#define DAC_I2S_MODE_SLAVE (0x0 << 4) +#define DAC_I2S_MODE_MASTER (0x1 << 4) + +/* REG04: DAC_INIT_CTRL2 */ +#define DAC_I2S_LRP_MASK BIT(7) +#define DAC_I2S_LRP_NORMAL (0x0 << 7) +#define DAC_I2S_LRP_REVERSAL (0x1 << 7) +#define DAC_VDL_MASK GENMASK(6, 5) +#define DAC_VDL_16BITS (0x0 << 5) +#define DAC_VDL_20BITS (0x1 << 5) +#define DAC_VDL_24BITS (0x2 << 5) +#define DAC_VDL_32BITS (0x3 << 5) +#define DAC_MODE_MASK GENMASK(4, 3) +#define DAC_MODE_RJM (0x0 << 3) +#define DAC_MODE_LJM (0x1 << 3) +#define DAC_MODE_I2S (0x2 << 3) +#define DAC_MODE_PCM (0x3 << 3) +#define DAC_LR_SWAP_MASK BIT(2) +#define DAC_LR_SWAP_DIS (0x0 << 2) +#define DAC_LR_SWAP_EN (0x1 << 2) + +/* REG05: DAC_INIT_CTRL3 */ +#define DAC_WL_MASK GENMASK(3, 2) +#define DAC_WL_16BITS (0x0 << 2) +#define DAC_WL_20BITS (0x1 << 2) +#define DAC_WL_24BITS (0x2 << 2) +#define DAC_WL_32BITS (0x3 << 2) +#define DAC_RST_MASK BIT(1) +#define DAC_RST_EN (0x0 << 1) +#define DAC_RST_DIS (0x1 << 1) +#define DAC_BCP_MASK BIT(0) +#define DAC_BCP_NORMAL (0x0 << 0) +#define DAC_BCP_REVERSAL (0x1 << 0) + +/* REG22: DAC_PRECHARGE_CTRL */ +#define DAC_CHARGE_XCHARGE_MASK BIT(7) +#define DAC_CHARGE_DISCHARGE (0x0 << 7) +#define DAC_CHARGE_PRECHARGE (0x1 << 7) +#define DAC_CHARGE_CURRENT_64I_MASK BIT(6) +#define DAC_CHARGE_CURRENT_64I (0x1 << 6) +#define DAC_CHARGE_CURRENT_32I_MASK BIT(5) +#define DAC_CHARGE_CURRENT_32I (0x1 << 5) +#define DAC_CHARGE_CURRENT_16I_MASK BIT(4) +#define DAC_CHARGE_CURRENT_16I (0x1 << 4) +#define DAC_CHARGE_CURRENT_08I_MASK BIT(3) +#define DAC_CHARGE_CURRENT_08I (0x1 << 3) +#define DAC_CHARGE_CURRENT_04I_MASK BIT(2) +#define DAC_CHARGE_CURRENT_04I (0x1 << 2) +#define DAC_CHARGE_CURRENT_02I_MASK BIT(1) +#define DAC_CHARGE_CURRENT_02I (0x1 << 1) +#define DAC_CHARGE_CURRENT_I_MASK BIT(0) +#define DAC_CHARGE_CURRENT_I (0x1 << 0) +#define DAC_CHARGE_CURRENT_ALL_MASK GENMASK(6, 0) +#define DAC_CHARGE_CURRENT_ALL_OFF 0x00 +#define DAC_CHARGE_CURRENT_ALL_ON 0x7f + +/* REG23: DAC_PWR_CTRL */ +#define DAC_PWR_MASK BIT(6) +#define DAC_PWR_OFF (0x0 << 6) +#define DAC_PWR_ON (0x1 << 6) +#define DACL_PATH_REFV_MASK BIT(5) +#define DACL_PATH_REFV_OFF (0x0 << 5) +#define DACL_PATH_REFV_ON (0x1 << 5) +#define HPOUTL_ZERO_CROSSING_MASK BIT(4) +#define HPOUTL_ZERO_CROSSING_OFF (0x0 << 4) +#define HPOUTL_ZERO_CROSSING_ON (0x1 << 4) +#define DACR_PATH_REFV_MASK BIT(1) +#define DACR_PATH_REFV_OFF (0x0 << 1) +#define DACR_PATH_REFV_ON (0x1 << 1) +#define HPOUTR_ZERO_CROSSING_MASK BIT(0) +#define HPOUTR_ZERO_CROSSING_OFF (0x0 << 0) +#define HPOUTR_ZERO_CROSSING_ON (0x1 << 0) + +/* REG24: DAC_CLK_CTRL */ +#define DACL_REFV_MASK BIT(7) +#define DACL_REFV_OFF (0x0 << 7) +#define DACL_REFV_ON (0x1 << 7) +#define DACL_CLK_MASK BIT(6) +#define DACL_CLK_OFF (0x0 << 6) +#define DACL_CLK_ON (0x1 << 6) +#define DACL_MASK BIT(5) +#define DACL_OFF (0x0 << 5) +#define DACL_ON (0x1 << 5) +#define DACL_INIT_MASK BIT(4) +#define DACL_INIT_OFF (0x0 << 4) +#define DACL_INIT_ON (0x1 << 4) +#define DACR_REFV_MASK BIT(3) +#define DACR_REFV_OFF (0x0 << 3) +#define DACR_REFV_ON (0x1 << 3) +#define DACR_CLK_MASK BIT(2) +#define DACR_CLK_OFF (0x0 << 2) +#define DACR_CLK_ON (0x1 << 2) +#define DACR_MASK BIT(1) +#define DACR_OFF (0x0 << 1) +#define DACR_ON (0x1 << 1) +#define DACR_INIT_MASK BIT(0) +#define DACR_INIT_OFF (0x0 << 0) +#define DACR_INIT_ON (0x1 << 0) + +/* REG25: HPMIX_CTRL*/ +#define HPMIXL_MASK BIT(6) +#define HPMIXL_DIS (0x0 << 6) +#define HPMIXL_EN (0x1 << 6) +#define HPMIXL_INIT_MASK BIT(5) +#define HPMIXL_INIT_DIS (0x0 << 5) +#define HPMIXL_INIT_EN (0x1 << 5) +#define HPMIXL_INIT2_MASK BIT(4) +#define HPMIXL_INIT2_DIS (0x0 << 4) +#define HPMIXL_INIT2_EN (0x1 << 4) +#define HPMIXR_MASK BIT(2) +#define HPMIXR_DIS (0x0 << 2) +#define HPMIXR_EN (0x1 << 2) +#define HPMIXR_INIT_MASK BIT(1) +#define HPMIXR_INIT_DIS (0x0 << 1) +#define HPMIXR_INIT_EN (0x1 << 1) +#define HPMIXR_INIT2_MASK BIT(0) +#define HPMIXR_INIT2_DIS (0x0 << 0) +#define HPMIXR_INIT2_EN (0x1 << 0) + +/* REG26: DAC_SELECT */ +#define DACL_SELECT_MASK BIT(4) +#define DACL_UNSELECT (0x0 << 4) +#define DACL_SELECT (0x1 << 4) +#define DACR_SELECT_MASK BIT(0) +#define DACR_UNSELECT (0x0 << 0) +#define DACR_SELECT (0x1 << 0) + +/* REG27: HPOUT_CTRL */ +#define HPOUTL_MASK BIT(7) +#define HPOUTL_DIS (0x0 << 7) +#define HPOUTL_EN (0x1 << 7) +#define HPOUTL_INIT_MASK BIT(6) +#define HPOUTL_INIT_DIS (0x0 << 6) +#define HPOUTL_INIT_EN (0x1 << 6) +#define HPOUTL_MUTE_MASK BIT(5) +#define HPOUTL_MUTE (0x0 << 5) +#define HPOUTL_UNMUTE (0x1 << 5) +#define HPOUTR_MASK BIT(4) +#define HPOUTR_DIS (0x0 << 4) +#define HPOUTR_EN (0x1 << 4) +#define HPOUTR_INIT_MASK BIT(3) +#define HPOUTR_INIT_DIS (0x0 << 3) +#define HPOUTR_INIT_EN (0x1 << 3) +#define HPOUTR_MUTE_MASK BIT(2) +#define HPOUTR_MUTE (0x0 << 2) +#define HPOUTR_UNMUTE (0x1 << 2) + +/* REG28: HPOUTL_GAIN_CTRL */ +#define HPOUTL_GAIN_MASK GENMASK(4, 0) + +/* REG29: HPOUTR_GAIN_CTRL */ +#define HPOUTR_GAIN_MASK GENMASK(4, 0) + +/* REG2a: HPOUT_POP_CTRL */ +#define HPOUTR_POP_MASK GENMASK(5, 4) +#define HPOUTR_POP_XCHARGE (0x1 << 4) +#define HPOUTR_POP_WORK (0x2 << 4) +#define HPOUTL_POP_MASK GENMASK(1, 0) +#define HPOUTL_POP_XCHARGE (0x1 << 0) +#define HPOUTL_POP_WORK (0x2 << 0) + +#define RK3328_HIFI 0 + +struct rk3328_reg_msk_val { + unsigned int reg; + unsigned int msk; + unsigned int val; +}; + +#endif -- 2.19.2

2 9

[alsa-devel] [PATCH] ASoC: xlnx: fix error handling in xlnx_formatter_pcm_probe
by Gustavo A. R. Silva 07 Jan '19

07 Jan '19

Currently, if platform_get_irq_byname() fails, the returned error turns into a huge value, once it is being store into a variable of type unsigned int, hence never actually reporting any error and causing unexpected behavior when using the values stored in aud_drv_data->s2mm_irq and aud_drv_data->mm2s_irq. Fix this by changing the type of variables s2mm_irq and mm2s_irq in structure xlnx_pcm_drv_data from unsigned int to int. Addresses-Coverity-ID: 1476096 ("Unsigned compared against 0") Fixes: 796175a94a7f ("ASoC: xlnx: add pcm formatter platform driver") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index f7235f7664d7..d2194da928e7 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -76,8 +76,8 @@ struct xlnx_pcm_drv_data { void __iomem *mmio; bool s2mm_presence; bool mm2s_presence; - unsigned int s2mm_irq; - unsigned int mm2s_irq; + int s2mm_irq; + int mm2s_irq; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; struct clk *axi_clk; -- 2.20.1

2 4

[alsa-devel] [PATCH] ASoC: rockchip: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 07 Jan '19

07 Jan '19

From: kbuild test robot <fengguang.wu(a)intel.com> sound/soc/codecs/rk3328_codec.c:508:6-11: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Fixes: dc1b33660692 ("ASoC: rockchip: support ACODEC for rk3328") CC: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> Signed-off-by: kbuild test robot <fengguang.wu(a)intel.com> --- tree: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master head: 6a1d293238c1774cef23c25b5188b0c134c29a34 commit: dc1b33660692441867a8ec5be147915dfa3bc8cb [11432/12209] ASoC: rockchip: support ACODEC for rk3328 rk3328_codec.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -505,7 +505,6 @@ MODULE_DEVICE_TABLE(of, rk3328_codec_of_ static struct platform_driver rk3328_codec_driver = { .driver = { .name = "rk3328-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rk3328_codec_of_match), }, .probe = rk3328_platform_probe,

2 4

[alsa-devel] [PATCH 00/10] ASoC: bytcht-es8316 fixes and improvements
by Hans de Goede 07 Jan '19

07 Jan '19

Hi All, Here is a patch series which makes all the sound out and inputs on the 1 BYT + ES8316 and 2 CHT + ES8316 devices which I have fully work. As usual this was an interesting journey, with things like needing a GPIO to enable the speaker and the mono speaker on this devices being connected between the left and right outputs of the HP amplifier of the ES8316. Daniel and other Endless people I believe that Endless has at least one CHT devices with an ES8316, it could be good if someone from Endless can test your device with this series, combined with the ALSA ucm patches which I will also Cc you on. Note that I've chosen to make the mono speaker setup the defaults since because of its differential setup it does not work very well when setup for stereo speakers. So you may need to pass a quirk parameter and submit a DMI based quirk upstream to get stereo instead of mono on your device (you will get audio on both speakers, but it will be a mono mix output on both speakers). Regards, Hans

3 34

[alsa-devel] [PATCH v2 1/3] ASoC: Intel: sst: Simplify is_byt_cr()
by Stephan Gerhold 07 Jan '19

07 Jan '19

is_byt_cr() and its usage can be simplified by returning the bool directly, instead of through a pointer. This works because the return value is just treated as bytcr = false and is not used otherwise. This patch also removes the extra check of IS_ENABLED(CONFIG_IOSF_MBI) in favor of checking iosf_mbi_available() directly. The header already takes care of returning false if the config option is not enabled. No functional change. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- New patch in v2 to keep the necessary cleanup separate from the next patch. sound/soc/intel/atom/sst/sst_acpi.c | 33 ++++++++++++----------------- 1 file changed, 14 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3a95ebbfc45d..9eaac450f864 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -255,18 +255,15 @@ static int is_byt(void) return status; } -static int is_byt_cr(struct device *dev, bool *bytcr) +static bool is_byt_cr(struct device *dev) { int status = 0; - if (IS_ENABLED(CONFIG_IOSF_MBI)) { - u32 bios_status; - - if (!is_byt() || !iosf_mbi_available()) { - /* bail silently */ - return status; - } + if (!is_byt()) + return false; + if (iosf_mbi_available()) { + u32 bios_status; status = iosf_mbi_read(BT_MBI_UNIT_PMC, /* 0x04 PUNIT */ MBI_REG_READ, /* 0x10 */ 0x006, /* BIOS_CONFIG */ @@ -278,15 +275,17 @@ static int is_byt_cr(struct device *dev, bool *bytcr) /* bits 26:27 mirror PMIC options */ bios_status = (bios_status >> 26) & 3; - if ((bios_status == 1) || (bios_status == 3)) - *bytcr = true; - else - dev_info(dev, "BYT-CR not detected\n"); + if (bios_status == 1 || bios_status == 3) { + dev_info(dev, "Detected Baytrail-CR platform\n"); + return true; + } + + dev_info(dev, "BYT-CR not detected\n"); } } else { - dev_info(dev, "IOSF_MBI not enabled, no BYT-CR detection\n"); + dev_info(dev, "IOSF_MBI not available, no BYT-CR detection\n"); } - return status; + return false; } @@ -301,7 +300,6 @@ static int sst_acpi_probe(struct platform_device *pdev) struct platform_device *plat_dev; struct sst_platform_info *pdata; unsigned int dev_id; - bool bytcr = false; id = acpi_match_device(dev->driver->acpi_match_table, dev); if (!id) @@ -333,10 +331,7 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - ret = is_byt_cr(dev, &bytcr); - if (!(ret < 0 || !bytcr)) { - dev_info(dev, "Detected Baytrail-CR platform\n"); - + if (is_byt_cr(dev)) { /* override resource info */ byt_rvp_platform_data.res_info = &bytcr_res_info; } -- 2.20.1

3 8

[alsa-devel] Applied "ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C)" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) 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 51a13e401a83ef37aa98c049c2c30bba885676c2 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold <stephan(a)gerhold.net> Date: Wed, 2 Jan 2019 20:39:08 +0100 Subject: [PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) Add quirks to select the correct input map, jack-detect options and channel map to make sound work on the ASUS MeMO Pad 7 (ME176C). Note: Although sound works out of the box, jack detection currently requires overriding the ACPI DSDT table. This is necessary because the rt5640 ACPI device (10EC5640) has the wrong GPIO listed as interrupt (one of the Bluetooth GPIOs). The correct GPIO is GPO2 0x0004 (listed as the first GPIO in the Intel(R) Audio Machine Driver - AMCR0F28 device). Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a22366ce33c4..ca8b4d5ff70f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -428,6 +428,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: No need for IS_MODULE/BUILTIN check for pcm driver
by Peter Ujfalusi 07 Jan '19

07 Jan '19

Since the platform drivers are selected by the DAI drivers (including McASP) there is no longer a need to check whether the modules are built-in or module. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index eeda6d5565bc..ee9ab58d4d0c 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2149,26 +2149,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = davinci_mcasp_get_dma_type(mcasp); switch (ret) { case PCM_EDMA: -#if IS_BUILTIN(CONFIG_SND_SOC_TI_EDMA_PCM) || \ - (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ - IS_MODULE(CONFIG_SND_SOC_TI_EDMA_PCM)) ret = edma_pcm_platform_register(&pdev->dev); -#else - dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); - ret = -EINVAL; - goto err; -#endif break; case PCM_SDMA: -#if IS_BUILTIN(CONFIG_SND_SOC_TI_SDMA_PCM) || \ - (IS_MODULE(CONFIG_SND_SOC_DAVINCI_MCASP) && \ - IS_MODULE(CONFIG_SND_SOC_TI_SDMA_PCM)) ret = sdma_pcm_platform_register(&pdev->dev, NULL, NULL); -#else - dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); - ret = -EINVAL; - goto err; -#endif break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 3

[alsa-devel] [PATCH 0/3] ASoC: ti: davinci-mcasp: GPIO mode support for pins
by Peter Ujfalusi 07 Jan '19

07 Jan '19

Hi, Each McASP pins can be configured individually to be used as McASP or GPIO pin. This series will allow client drivers to use pins as GPIO (optional) in case the board wired up them to be used in this mode. Regards, Peter --- Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks bindings: sound: davinci-mcasp: Document GPIO support ASoC: ti: davinci-mcasp: Add support for GPIO mode of the pins .../bindings/sound/davinci-mcasp-audio.txt | 17 + sound/soc/ti/davinci-mcasp.c | 295 +++++++++++++----- 2 files changed, 237 insertions(+), 75 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 12

[alsa-devel] Applied "ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C)" to the asoc tree
by Mark Brown 07 Jan '19

07 Jan '19

The patch ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) 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 51a13e401a83ef37aa98c049c2c30bba885676c2 Mon Sep 17 00:00:00 2001 From: Stephan Gerhold <stephan(a)gerhold.net> Date: Wed, 2 Jan 2019 20:39:08 +0100 Subject: [PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) Add quirks to select the correct input map, jack-detect options and channel map to make sound work on the ASUS MeMO Pad 7 (ME176C). Note: Although sound works out of the box, jack detection currently requires overriding the ACPI DSDT table. This is necessary because the rt5640 ACPI device (10EC5640) has the wrong GPIO listed as interrupt (one of the Bluetooth GPIOs). The correct GPIO is GPO2 0x0004 (listed as the first GPIO in the Intel(R) Audio Machine Driver - AMCR0F28 device). Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a22366ce33c4..ca8b4d5ff70f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -428,6 +428,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.20.1

1 0

[alsa-devel] [RFC 0/2] conf/ucm: bytcht-es8316 support
by Hans de Goede 07 Jan '19

07 Jan '19

Hi All, Here is the UCM profile to match the kernel patches which I just submitted upstream. Note this series is RFC for now, since it depends on the name of a few controls added by the kernel series, as well as on the long_name set by the kernel series. Once the kernel series has been merged into -next, this UCM profile series can be merged as well. I will send another mail when the kernel series has been accepted. Regards, Hans

2 4

[alsa-devel] [PATCH] [v2] ALSA: isa: fix a missing check of snd_ctl_add
by Aditya Pakki 07 Jan '19

07 Jan '19

snd_ctl_add() could fail, so let's check its return value and return its error code upstream upon failure. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> --- sound/isa/sb/sb16_main.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 37e6ce7b0b13..981d65d122b6 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -879,10 +879,14 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops); - if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) - snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip)); - else + if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) { + err = snd_ctl_add(card, snd_ctl_new1( + &snd_sb16_dma_control, chip)); + if (err) + return err; + } else { pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; + } snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), -- 2.17.1

2 1

[alsa-devel] [PATCH] [v2] ALSA: line6: fix check on snd_card_register
by Aditya Pakki 07 Jan '19

07 Jan '19

The fix checks if snd_card_register() fails, and if so logs the error via dev_err() consistent with other patches. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> --- sound/usb/line6/pod.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index 020c81818951..ce45b6dab651 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -320,7 +320,8 @@ static void pod_startup4(struct work_struct *work) line6_read_serial_number(&pod->line6, &pod->serial_number); /* ALSA audio interface: */ - snd_card_register(line6->card); + if (snd_card_register(line6->card)) + dev_err(line6->ifcdev, "Failed to register POD card.\n"); } /* POD special files: */ -- 2.17.1

2 1

[alsa-devel] [PATCH] [v2] ALSA: ice1712: fix a missing check of snd_i2c_sendbytes
by Aditya Pakki 07 Jan '19

07 Jan '19

snd_i2c_sendbytes could fail. The fix checks its return value: if it fails, issues an error message and returns with its error code. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> --- sound/pci/ice1712/ews.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index b8af747ecb43..7646c93e8268 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -826,7 +826,12 @@ static int snd_ice1712_6fire_read_pca(struct snd_ice1712 *ice, unsigned char reg snd_i2c_lock(ice->i2c); byte = reg; - snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1); + if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1)) { + snd_i2c_unlock(ice->i2c); + dev_err(ice->card->dev, "cannot send pca\n"); + return -EIO; + } + byte = 0; if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1) != 1) { snd_i2c_unlock(ice->i2c); -- 2.17.1

2 1

[alsa-devel] [PATCH 1/1] ALSA: oxygen: initialize spdif_playback_enable to 0
by tom.ty89＠gmail.com 07 Jan '19

07 Jan '19

From: Tom Yan <tom.ty89(a)gmail.com> There's no reason for us to do that while we initialize dac_mute to 1. Also oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE bit anyway. Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b4ef5804212d..6a743c878415 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -373,7 +373,7 @@ static void oxygen_init(struct oxygen *chip) for (i = 0; i < 8; ++i) chip->dac_volume[i] = chip->model.dac_volume_min; chip->dac_mute = 1; - chip->spdif_playback_enable = 1; + chip->spdif_playback_enable = 0; chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL | (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT); chip->spdif_pcm_bits = chip->spdif_bits; -- 2.20.1

2 1

[alsa-devel] [PATCH 1/1] ALSA: virtuoso: add de-emphasis control
by tom.ty89＠gmail.com 07 Jan '19

07 Jan '19

From: Tom Yan <tom.ty89(a)gmail.com> Add control for the de-emphasis filter in the PCM179x DACs Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 24109d37ca09..0014c55ecc3d 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -331,7 +331,7 @@ static void pcm1796_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = - PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; + PCM1796_FMT_24_I2S | PCM1796_ATLD; if (!data->broken_i2c) data->pcm1796_regs[0][18 - PCM1796_REG_BASE] |= PCM1796_MUTE; data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = @@ -621,6 +621,33 @@ static void update_pcm1796_oversampling(struct oxygen *chip) pcm1796_write_cached(chip, i, 20, reg); } +static void update_pcm1796_deemph(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 reg; + + reg = data->pcm1796_regs[0][18 - PCM1796_REG_BASE] & ~PCM1796_DMF_MASK; + switch (data->current_rate) { + case 32000: + case 64000: + reg |= PCM1796_DMF_32; + break; + case 44100: + case 88200: + case 176400: + reg |= PCM1796_DMF_441; + break; + case 48000: + case 96000: + case 192000: + reg |= PCM1796_DMF_48; + break; + } + for (i = 0; i < data->dacs; ++i) + pcm1796_write_cached(chip, i, 18, reg); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -629,6 +656,7 @@ static void set_pcm1796_params(struct oxygen *chip, msleep(1); data->current_rate = params_rate(params); update_pcm1796_oversampling(chip); + update_pcm1796_deemph(chip); } static void update_pcm1796_volume(struct oxygen *chip) @@ -653,9 +681,11 @@ static void update_pcm1796_mute(struct oxygen *chip) unsigned int i; u8 value; - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_I2S | PCM1796_ATLD; + value = data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; if (chip->dac_mute) value |= PCM1796_MUTE; + else + value &= ~PCM1796_MUTE; for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 18, value); } @@ -777,6 +807,50 @@ static const struct snd_kcontrol_new rolloff_control = { .put = rolloff_put, }; +static int deemph_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.integer.value[0] = + (data->pcm1796_regs[0][18 - PCM1796_REG_BASE] & + PCM1796_DME) != PCM1796_DMF_DISABLED; + return 0; +} + +static int deemph_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; + if (!value->value.integer.value[0]) + reg &= ~PCM1796_DME; + else + reg |= PCM1796_DME; + changed = reg != data->pcm1796_regs[0][18 - PCM1796_REG_BASE]; + if (changed) { + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 18, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new deemph_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "De-emphasis Switch", + .info = snd_ctl_boolean_mono_info, + .get = deemph_get, + .put = deemph_put, +}; + static const struct snd_kcontrol_new hdav_hdmi_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "HDMI Playback Switch", @@ -1011,6 +1085,10 @@ static int add_pcm1796_controls(struct oxygen *chip) snd_ctl_new1(&rolloff_control, chip)); if (err < 0) return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&deemph_control, chip)); + if (err < 0) + return err; } return 0; } -- 2.20.0

4 7

[alsa-devel] New Dell platform support headset mode
by Kailang 07 Jan '19

07 Jan '19

Hi Takashi, Attach patch will enable headset mode for new platform. BR, Kailang

2 1

[alsa-devel] [PATCH 0/3] alsa-utils: axfer: fulfill scheduling model section for manual
by Takashi Sakamoto 07 Jan '19

07 Jan '19

Hi, This patchset fulfills a section about scheduling model for axfer manual. This is my attempt to describe well-known issue for 'no-period-wakeup' mode of PCM runtime, discussed in some recent Audio miniconferences[1]. [1] http://www.alsa-project.org/main/index.php/Miniconf_2018#PCM_improvements Takashi Sakamoto (3): axfer: add an explanation about IRQ-based scheduling model axfer: add an explanation about Timer-based scheduling model axfer: add an explanation about advantages/issues of Timer-based scheduling model axfer/axfer-transfer.1 | 112 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 111 insertions(+), 1 deletion(-) -- 2.19.1

2 4

[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Always check descriptor sizes in parser code
by Takashi Iwai 07 Jan '19

07 Jan '19

There are a few places where we access the data without checking the actual object size from the USB audio descriptor. This may result in OOB access, as recently reported. This patch addresses these missing checks. Most of added codes are simple bLength checks in the caller side. For the input and output terminal parsers, we put the length check in the parser functions. For the input terminal, a new argument is added to distinguish between UAC1 and the rest, as they treat different objects. Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Reported-by: Hui Peng <benquike(a)163.com> Tested-by: Hui Peng <benquike(a)163.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 +- sound/usb/mixer.c | 4 ++++ sound/usb/stream.c | 36 +++++++++++++++++++++++++----------- 3 files changed, 30 insertions(+), 12 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index a105947eaf55..746a72e23cf9 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -246,7 +246,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) h1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, UAC_HEADER); - if (!h1) { + if (!h1 || h1->bLength < sizeof(*h1)) { dev_err(&dev->dev, "cannot find UAC_HEADER\n"); return -EINVAL; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index dfd918891e69..e7d441d0e839 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2075,11 +2075,15 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, if (state->mixer->protocol == UAC_VERSION_2) { struct uac2_input_terminal_descriptor *d_v2 = raw_desc; + if (d_v2->bLength < sizeof(*d_v2)) + return -EINVAL; control = UAC2_TE_CONNECTOR; term_id = d_v2->bTerminalID; bmctls = le16_to_cpu(d_v2->bmControls); } else if (state->mixer->protocol == UAC_VERSION_3) { struct uac3_input_terminal_descriptor *d_v3 = raw_desc; + if (d_v3->bLength < sizeof(*d_v3)) + return -EINVAL; control = UAC3_TE_INSERTION; term_id = d_v3->bTerminalID; bmctls = le32_to_cpu(d_v3->bmControls); diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 67cf849aa16b..d9e3de495c16 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -596,12 +596,8 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, csep = snd_usb_find_desc(alts->extra, alts->extralen, NULL, USB_DT_CS_ENDPOINT); if (!csep || csep->bLength < 7 || - csep->bDescriptorSubtype != UAC_EP_GENERAL) { - usb_audio_warn(chip, - "%u:%d : no or invalid class specific endpoint descriptor\n", - iface_no, altsd->bAlternateSetting); - return 0; - } + csep->bDescriptorSubtype != UAC_EP_GENERAL) + goto error; if (protocol == UAC_VERSION_1) { attributes = csep->bmAttributes; @@ -609,6 +605,8 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct uac2_iso_endpoint_descriptor *csep2 = (struct uac2_iso_endpoint_descriptor *) csep; + if (csep2->bLength < sizeof(*csep2)) + goto error; attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX; /* emulate the endpoint attributes of a v1 device */ @@ -618,12 +616,20 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, struct uac3_iso_endpoint_descriptor *csep3 = (struct uac3_iso_endpoint_descriptor *) csep; + if (csep3->bLength < sizeof(*csep3)) + goto error; /* emulate the endpoint attributes of a v1 device */ if (le32_to_cpu(csep3->bmControls) & UAC2_CONTROL_PITCH) attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL; } return attributes; + + error: + usb_audio_warn(chip, + "%u:%d : no or invalid class specific endpoint descriptor\n", + iface_no, altsd->bAlternateSetting); + return 0; } /* find an input terminal descriptor (either UAC1 or UAC2) with the given @@ -631,13 +637,17 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, */ static void * snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) + int terminal_id, bool uac23) { struct uac2_input_terminal_descriptor *term = NULL; + size_t minlen = uac23 ? sizeof(struct uac2_input_terminal_descriptor) : + sizeof(struct uac_input_terminal_descriptor); while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_INPUT_TERMINAL))) { + if (term->bLength < minlen) + continue; if (term->bTerminalID == terminal_id) return term; } @@ -655,7 +665,8 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, ctrl_iface->extralen, term, UAC_OUTPUT_TERMINAL))) { - if (term->bTerminalID == terminal_id) + if (term->bLength >= sizeof(*term) && + term->bTerminalID == terminal_id) return term; } @@ -729,7 +740,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, format = le16_to_cpu(as->wFormatTag); /* remember the format value */ iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + false); if (iterm) { num_channels = iterm->bNrChannels; chconfig = le16_to_cpu(iterm->wChannelConfig); @@ -764,7 +776,8 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, * to extract the clock */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + true); if (input_term) { clock = input_term->bCSourceID; if (!chconfig && (num_channels == input_term->bNrChannels)) @@ -998,7 +1011,8 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * to extract the clock */ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, - as->bTerminalLink); + as->bTerminalLink, + true); if (input_term) { clock = input_term->bCSourceID; goto found_clock; -- 2.19.2

1 1

[alsa-devel] [PATCH] ASoC: AMD: Add delay before starting to capture
by Agrawal, Akshu 07 Jan '19

07 Jan '19

On capture through dmic we observe a glitch at the start of record. This is because we start capturing even before dmic is ready to send out data. The glitch seen last for ~20msec. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- sound/soc/amd/acp-da7219-max98357a.c | 28 ++++++++++++++++++++++++++-- 1 file changed, 26 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a5daad9..72b1cf4 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -44,6 +44,8 @@ #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +#define ADAU7002_DELAY_MS 20 + static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; @@ -213,6 +215,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + int ret; /* * On this platform for PCM device we support stereo @@ -225,7 +228,17 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + ret = da7219_clk_enable(substream); + if (ret) + return ret; + /* + * On some platforms, it takes ~20 msec for PDM DMICs and ADAU7002 + * to settle and start producing proper audio data. + */ + msleep(ADAU7002_DELAY_MS); + /* Delay in frames for 48Khz, 16bit, 2 channel */ + runtime->delay = ADAU7002_DELAY_MS * 48; + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -234,6 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + int ret; /* * On this platform for PCM device we support stereo @@ -247,7 +261,17 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + ret = da7219_clk_enable(substream); + if (ret) + return ret; + /* + * On some platforms, it takes ~20 msec for PDM DMICs and ADAU7002 + * to settle and start producing proper audio data. + */ + msleep(ADAU7002_DELAY_MS); + /* Delay in frames for 48Khz, 16bit, 2 channel */ + runtime->delay = ADAU7002_DELAY_MS * 48; + return 0; } static void cz_dmic_shutdown(struct snd_pcm_substream *substream) -- 1.9.1

2 2

[alsa-devel] [PATCH 1/2] ASoC: refine ASoC hdmi audio suspend/resume
by libin.yang＠intel.com 07 Jan '19

07 Jan '19

From: Libin Yang <libin.yang(a)intel.com> hdmi_codec_prepare() will trigger hdmi runtime resume, which will set the bitmask of hdev->addr. And skl_suspend() will clear the bitmask of HDA_CODEC_IDX_CONTROLLER. HDMI codec idx is not the same as HDA_CODEC_IDX_CONTROLLER, which means i915 power will not be released when suspend. On the other hand, hdmi_codec_prepare() don't need to call pm_runtime_get_sync() to wake up the audio subsystem (HDMI auido) for setting the codec registers. Turning display power on with snd_hdac_display_power() is enough. Let's use S3 without playback as an example: hdmi_codec_prepare() invokes the runtime resume of codec => snd_hdac_display_power(bus, hdev->addr, true) skl runtime resume skl_suspend() => snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); THis means hdev->addr will never release the display power when suspend. The new sequence will be: hdmi_codec_prepare() => snd_hdac_display_power(bus, hdev->addr, true) snd_hdac_display_power(bus, hdev->addr, false) skl runtime resume skl suspned Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 6 ++++-- sound/soc/intel/skylake/skl.c | 7 ------- 2 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3ab2949..782b323 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1895,7 +1895,7 @@ static int hdmi_codec_prepare(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); - pm_runtime_get_sync(&hdev->dev); + snd_hdac_display_power(hdev->bus, hdev->addr, true); /* * Power down afg. @@ -1906,6 +1906,7 @@ static int hdmi_codec_prepare(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hdac_display_power(hdev->bus, hdev->addr, false); return 0; } @@ -1915,6 +1916,7 @@ static void hdmi_codec_complete(struct device *dev) struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + snd_hdac_display_power(hdev->bus, hdev->addr, true); /* Power up afg */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); @@ -1930,7 +1932,7 @@ static void hdmi_codec_complete(struct device *dev) */ hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - pm_runtime_put_sync(&hdev->dev); + snd_hdac_display_power(hdev->bus, hdev->addr, false); } #else #define hdmi_codec_prepare NULL diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 60c9483..89f4d66 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -336,9 +336,6 @@ static int skl_suspend(struct device *dev) skl->skl_sst->fw_loaded = false; } - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - return 0; } @@ -350,10 +347,6 @@ static int skl_resume(struct device *dev) struct hdac_ext_link *hlink = NULL; int ret; - /* Turned OFF in HDMI codec driver after codec reconfiguration */ - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); - /* * resume only when we are not in suspend active, otherwise need to * restore the device -- 2.7.4

2 2

[alsa-devel] [snd_hda_intel] snd_hda_intel causes high CPU lockup and system instability if mic disabled in BIOS on Lenovo P50
by Alexander Kappner 07 Jan '19

07 Jan '19

My Lenovo P50 laptop has a BIOS option to disable the microphone. When this option gets chosen, the snd_hda_intel driver causes high CPU load on a single kworker thread, spinning on "process_unsol_events", and system instability. This behavior occurs from the time that the snd_hda_intel module is loaded, irrespective of whether anything is attempting to access the mic. The sound output still works. When in this state, the module cannot be removed cleanly; attempting to remove it (even without rmmod -f) triggers an oops. I have attached two exemplary dmesg outputs. Strangely enough, the exact location of the oops varies, but further up the call chain, I always see process_unsol_events. When the mic is not disabled in the BIOS, the module works stable, regardless whether or not the mic is muted in ALSA. I wasn't able to pinpoint the root cause. Any pointers on where to start? Much appreciated. Best regards Alexander First dmesg: [31592.691573] BUG: unable to handle kernel NULL pointer dereference at 000000000000020c [31592.691600] IP: mux_select+0x1d/0x190 [snd_hda_codec_generic] [31592.691601] PGD 0 P4D 0 [31592.691604] Oops: 0000 [#1] PREEMPT SMP [31592.691606] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel(-) snd_hda_codec snd_hwdep snd_hda_core snd_pcm snd_timer zfs(PO) zunicode(PO) zlua(PO) nvidia_drm(PO) nvidia_modeset(PO) nvidia(PO) iwlmvm mac80211 qmi_wwan iwlwifi cfg80211 thinkpad_acpi rfkill zcommon(PO) znvpair(PO) zavl(PO) icp(PO) spl(O) sdhci_pci cqhci sdhci mmc_block rtsx_pci_sdmmc rtsx_pci mfd_core [31592.691639] CPU: 3 PID: 11044 Comm: kworker/3:1 Tainted: P O 4.16.0-rc7-00206-gf461da3-dirty #33 [31592.691640] Hardware name: LENOVO 20ENCTO1WW/20ENCTO1WW, BIOS N1EET62W (1.35 ) 11/10/2016 [31592.691643] Workqueue: events process_unsol_events [snd_hda_core] [31592.691647] RIP: 0010:mux_select+0x1d/0x190 [snd_hda_codec_generic] [31592.691648] RSP: 0018:ffffc9000f0dfe08 EFLAGS: 00010292 [31592.691650] RAX: 0000000000000001 RBX: ffff880e44737300 RCX: 00000000ffffffff [31592.691651] RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffff880ee6368800 [31592.691652] RBP: 0000000000000000 R08: 0000000000000f09 R09: 0000000000000000 [31592.691653] R10: 0000000000000018 R11: 0000000000000003 R12: ffff880ee6368800 [31592.691655] R13: 0000000000000000 R14: ffff881053ce6100 R15: 0ffff881053ce610 [31592.691656] FS: 0000000000000000(0000) GS:ffff881053cc0000(0000) knlGS:0000000000000000 [31592.691658] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [31592.691659] CR2: 000000000000020c CR3: 000000000220a004 CR4: 00000000003606e0 [31592.691661] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [31592.691662] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 [31592.691662] Call Trace: [31592.691668] ? snd_hda_jack_detect_state+0x4b/0x60 [snd_hda_codec] [31592.691672] call_jack_callback.isra.1+0x20/0xa0 [snd_hda_codec] [31592.691676] snd_hda_jack_unsol_event+0x62/0x70 [snd_hda_codec] [31592.691679] process_unsol_events+0x64/0x80 [snd_hda_core] [31592.691682] process_one_work+0x13e/0x380 [31592.691684] worker_thread+0x62/0x3e0 [31592.691686] ? rescuer_thread+0x2d0/0x2d0 [31592.691689] kthread+0x10d/0x140 [31592.691690] ? kthread_bind+0x10/0x10 [31592.691693] ret_from_fork+0x35/0x40 [31592.691695] Code: e2 01 e9 77 fb ff ff c3 66 0f 1f 44 00 00 0f 1f 44 00 00 41 57 41 56 41 55 41 54 49 89 fc 55 53 48 83 ec 08 48 8b af 08 04 00 00 <8b> 85 0c 02 00 00 85 c0 74 73 89 d3 41 89 f5 8d 50 ff 39 d8 89 [31592.691721] RIP: mux_select+0x1d/0x190 [snd_hda_codec_generic] RSP: ffffc9000f0dfe08 [31592.691723] CR2: 000000000000020c [31592.691725] ---[ end trace dee155e55b20eab8 ]--- Second dmesg: [115121.084841] BUG: unable to handle kernel NULL pointer dereference at 00000000000005f8 [115121.084844] PGD 0 P4D 0 [115121.084847] Oops: 0000 [#1] PREEMPT SMP [115121.084850] CPU: 2 PID: 12228 Comm: kworker/2:1 Tainted: P O 4.18.0-rc2-00223-g1904148-dirty #39 [115121.084851] Hardware name: LENOVO 20ENCTO1WW/20ENCTO1WW, BIOS N1EET62W (1.35 ) 11/10/2016 [115121.084856] Workqueue: events process_unsol_events [snd_hda_core] [115121.084859] RIP: 0010:set_path_power+0x24/0x120 [snd_hda_codec_generic] [115121.084860] Code: 66 0f 1f 44 00 00 0f 1f 44 00 00 41 57 41 89 f7 41 56 41 55 41 54 55 48 89 fd 53 48 83 ec 08 4c 8b b7 08 04 00 00 89 54 24 04 <41> 8b be f8 05 00 00 49 8b b6 08 06 00 00 85 ff 0f 84 ce 00 00 00 [115121.084901] RSP: 0018:ffffc900243e3de8 EFLAGS: 00010296 [115121.084903] RAX: 0000000000000000 RBX: 0000000000000018 RCX: 00000000ffffffff [115121.084904] RDX: 0000000000000000 RSI: 0000000000000018 RDI: ffff880f993a2800 [115121.084906] RBP: ffff880f993a2800 R08: 0000000000000f09 R09: 0000000000000000 [115121.084907] R10: 0000000000000019 R11: 0000000000000000 R12: 0000000000000000 [115121.084908] R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000018 [115121.084910] FS: 0000000000000000(0000) GS:ffff881053c80000(0000) knlGS:0000000000000000 [115121.084911] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [115121.084912] CR2: 00000000000005f8 CR3: 000000000260a004 CR4: 00000000003606e0 [115121.084914] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [115121.084915] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 [115121.084916] Call Trace: [115121.084922] pin_power_callback+0x4f/0x80 [snd_hda_codec_generic] [115121.084926] call_jack_callback.isra.1+0x20/0xa0 [snd_hda_codec] [115121.084930] snd_hda_jack_unsol_event+0x62/0x70 [snd_hda_codec] [115121.084933] process_unsol_events+0x64/0x80 [snd_hda_core] [115121.084937] process_one_work+0x157/0x390 [115121.084939] worker_thread+0x64/0x3e0 [115121.084942] ? rescuer_thread+0x2e0/0x2e0 [115121.084944] kthread+0x10d/0x140 [115121.084946] ? kthread_bind+0x10/0x10 [115121.084948] ret_from_fork+0x35/0x40 [115121.084951] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel(-) snd_hda_codec snd_hwdep snd_hda_core snd_pcm snd_timer tun nvidia_modeset(PO) nvidia(PO) iwlmvm mac80211 iwlwifi btusb btrtl btbcm btintel bluetooth cfg80211 qmi_wwan ecdh_generic thinkpad_acpi rfkill sdhci_pci cqhci sdhci mmc_block rtsx_pci_sdmmc rtsx_pci mfd_core [115121.084969] CR2: 00000000000005f8 [115121.084971] ---[ end trace 010d684599325728 ]--- [115121.084975] RIP: 0010:set_path_power+0x24/0x120 [snd_hda_codec_generic] [115121.084976] Code: 66 0f 1f 44 00 00 0f 1f 44 00 00 41 57 41 89 f7 41 56 41 55 41 54 55 48 89 fd 53 48 83 ec 08 4c 8b b7 08 04 00 00 89 54 24 04 <41> 8b be f8 05 00 00 49 8b b6 08 06 00 00 85 ff 0f 84 ce 00 00 00 [115121.085000] RSP: 0018:ffffc900243e3de8 EFLAGS: 00010296 [115121.085001] RAX: 0000000000000000 RBX: 0000000000000018 RCX: 00000000ffffffff [115121.085002] RDX: 0000000000000000 RSI: 0000000000000018 RDI: ffff880f993a2800 [115121.085003] RBP: ffff880f993a2800 R08: 0000000000000f09 R09: 0000000000000000 [115121.085004] R10: 0000000000000019 R11: 0000000000000000 R12: 0000000000000000 [115121.085005] R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000018 [115121.085006] FS: 0000000000000000(0000) GS:ffff881053c80000(0000) knlGS:0000000000000000 [115121.085007] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [115121.085008] CR2: 00000000000005f8 CR3: 000000000260a004 CR4: 00000000003606e0 [115121.085009] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [115121.085010] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400

2 6

[alsa-devel] ALSA:usb audio Higher sample rates on usb audio no longer working.
by Con Kolivas 06 Jan '19

06 Jan '19

Upon switching from 4.19.0 to 4.20.0, pulseaudio started complaining that sinks that previously worked are no longer supported. On 4.19.0 trying 24 bit 88200, 176400, and 192000 I get the following output from pulse. resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.sink-input.c: Created input 15 "Playback Stream" on alsa_output.usb-DSPeaker_Anti-Mode_X4-00.iec958-stereo with sample spec float32le 2ch 176400Hz and channel map front-left,front-right Switching to 4.20 gives me: alsa-sink.c: Sink does not support sample rate of 176400 Hz and alsa-sink.c: Sink does not support sample rate of 88200 Hz and alsa-sink.c: Sink does not support sample rate of 192000 Hz Sample rates of 44100, 48000, and 96000 work fine, but 88200, 176400, and 192000 no longer work Switching back to 4.19 immediately fixes the issue. I tried looking through the alsa changelogs but there were too many to give an obvious culprit, and haven't had time to do a git bisect. If there's an obvious choice patch to back out I'd be grateful for the heads up. Regards, Con Output from 4.19 dmesg grep snd: [ 4.941164] snd_hda_intel 0000:02:00.1: Disabling MSI [ 4.957534] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC892: line_outs=4 (0x14/0x15/0x16/0x17/0x0) type:line [ 4.957536] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 4.957537] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 4.957538] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 4.957538] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x11/0x1e [ 4.957539] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 4.957540] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 4.957541] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 4.957542] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 6.215818] usbcore: registered new interface driver snd-usb-audio Output from 4.20 dmesg grep snd (identical): [ 4.960455] snd_hda_intel 0000:02:00.1: Disabling MSI [ 4.986848] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC892: line_outs=4 (0x14/0x15/0x16/0x17/0x0) type:line [ 4.986850] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 4.986852] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 4.986853] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 4.986854] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x11/0x1e [ 4.986855] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 4.986856] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 4.986857] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 4.986858] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 6.211079] usbcore: registered new interface driver snd-usb-audio Output from usb connected audio device 4.19: [ 998.567015] usb 2-1.8.4: new high-speed USB device number 19 using ehci-pci [ 998.682497] usb 2-1.8.4: New USB device found, idVendor=19fb, idProduct=2040, bcdDevice= 2.00 [ 998.682502] usb 2-1.8.4: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 998.682505] usb 2-1.8.4: Product: Anti-Mode X4 [ 998.682507] usb 2-1.8.4: Manufacturer: DSPeaker 4.20 (identical): [ 98.428175] usb 2-1.8.4: new high-speed USB device number 19 using ehci-pci [ 98.543175] usb 2-1.8.4: New USB device found, idVendor=19fb, idProduct=2040, bcdDevice= 2.00 [ 98.543177] usb 2-1.8.4: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 98.543178] usb 2-1.8.4: Product: Anti-Mode X4 [ 98.543179] usb 2-1.8.4: Manufacturer: DSPeaker lspci output: 00:00.0 Host bridge: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 DMI2 (rev 04) 00:01.0 PCI bridge: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 PCI Express Root Port 1a (rev 04) 00:02.0 PCI bridge: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 PCI Express Root Port 2a (rev 04) 00:03.0 PCI bridge: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 PCI Express Root Port 3a (rev 04) 00:05.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 VTd/Memory Map/Misc (rev 04) 00:05.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 IIO RAS (rev 04) 00:05.4 PIC: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 IOAPIC (rev 04) 00:11.0 PCI bridge: Intel Corporation C600/X79 series chipset PCI Express Virtual Root Port (rev 06) 00:16.0 Communication controller: Intel Corporation C600/X79 series chipset MEI Controller #1 (rev 05) 00:1a.0 USB controller: Intel Corporation C600/X79 series chipset USB2 Enhanced Host Controller #2 (rev 06) 00:1b.0 Audio device: Intel Corporation C600/X79 series chipset High Definition Audio Controller (rev 06) 00:1c.0 PCI bridge: Intel Corporation C600/X79 series chipset PCI Express Root Port 1 (rev b6) 00:1c.2 PCI bridge: Intel Corporation C600/X79 series chipset PCI Express Root Port 3 (rev b6) 00:1c.3 PCI bridge: Intel Corporation C600/X79 series chipset PCI Express Root Port 4 (rev b6) 00:1c.4 PCI bridge: Intel Corporation C600/X79 series chipset PCI Express Root Port 5 (rev b6) 00:1c.5 PCI bridge: Intel Corporation C600/X79 series chipset PCI Express Root Port 6 (rev b6) 00:1d.0 USB controller: Intel Corporation C600/X79 series chipset USB2 Enhanced Host Controller #1 (rev 06) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev a6) 00:1f.0 ISA bridge: Intel Corporation C600/X79 series chipset LPC Controller (rev 06) 00:1f.2 SATA controller: Intel Corporation C600/X79 series chipset 6-Port SATA AHCI Controller (rev 06) 00:1f.3 SMBus: Intel Corporation C600/X79 series chipset SMBus Host Controller (rev 06) 02:00.0 VGA compatible controller: NVIDIA Corporation GP106 [GeForce GTX 1060 3GB] (rev a1) 02:00.1 Audio device: NVIDIA Corporation GP106 High Definition Audio Controller (rev a1) 06:00.0 USB controller: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller 07:00.0 USB controller: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller 08:00.0 SATA controller: ASMedia Technology Inc. ASM1062 Serial ATA Controller (rev 01) 09:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168/8411 PCI Express Gigabit Ethernet Controller (rev 09) ff:08.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 QPI Link 0 (rev 04) ff:09.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 QPI Link 1 (rev 04) ff:0a.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Power Control Unit 0 (rev 04) ff:0a.1 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Power Control Unit 1 (rev 04) ff:0a.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Power Control Unit 2 (rev 04) ff:0a.3 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Power Control Unit 3 (rev 04) ff:0b.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 UBOX Registers (rev 04) ff:0b.3 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 UBOX Registers (rev 04) ff:0c.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Unicast Registers (rev 04) ff:0c.1 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Unicast Registers (rev 04) ff:0c.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Unicast Registers (rev 04) ff:0d.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Unicast Registers (rev 04) ff:0d.1 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Unicast Registers (rev 04) ff:0d.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Unicast Registers (rev 04) ff:0e.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Home Agent 0 (rev 04) ff:0e.1 Performance counters: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Home Agent 0 (rev 04) ff:0f.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 0 Target Address/Thermal Registers (rev 04) ff:0f.1 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 0 RAS Registers (rev 04) ff:0f.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 0 Channel Target Address Decoder Registers (rev 04) ff:0f.3 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 0 Channel Target Address Decoder Registers (rev 04) ff:0f.4 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 0 Channel Target Address Decoder Registers (rev 04) ff:0f.5 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 0 Channel Target Address Decoder Registers (rev 04) ff:10.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 Thermal Control 0 (rev 04) ff:10.1 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 Thermal Control 1 (rev 04) ff:10.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 ERROR Registers 0 (rev 04) ff:10.3 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 ERROR Registers 1 (rev 04) ff:10.4 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 Thermal Control 2 (rev 04) ff:10.5 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 Thermal Control 3 (rev 04) ff:10.6 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 ERROR Registers 2 (rev 04) ff:10.7 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Integrated Memory Controller 1 Channel 0-3 ERROR Registers 3 (rev 04) ff:13.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 R2PCIe (rev 04) ff:13.1 Performance counters: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 R2PCIe (rev 04) ff:13.4 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 QPI Ring Registers (rev 04) ff:13.5 Performance counters: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 QPI Ring Performance Ring Monitoring (rev 04) ff:16.0 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 System Address Decoder (rev 04) ff:16.1 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Broadcast Registers (rev 04) ff:16.2 System peripheral: Intel Corporation Xeon E7 v2/Xeon E5 v2/Core i7 Broadcast Registers (rev 04) lsusb: Bus 002 Device 014: ID 19fb:2040 Bus 002 Device 010: ID 05e3:0610 Genesys Logic, Inc. 4-port hub Bus 002 Device 007: ID 1a40:0101 Terminus Technology Inc. Hub Bus 002 Device 016: ID 0627:0001 Adomax Technology Co., Ltd Bus 002 Device 012: ID 04d9:0169 Holtek Semiconductor, Inc. Bus 002 Device 009: ID 046d:09a2 Logitech, Inc. QuickCam Communicate Deluxe/S7500 Bus 002 Device 005: ID 2109:2811 VIA Labs, Inc. Hub Bus 002 Device 013: ID 046d:0a29 Logitech, Inc. H600 [Wireless Headset] Bus 002 Device 017: ID 04d9:0198 Holtek Semiconductor, Inc. Bus 002 Device 015: ID 1235:8202 Focusrite-Novation Bus 002 Device 011: ID 1a40:0201 Terminus Technology Inc. FE 2.1 7-port Hub Bus 002 Device 008: ID 0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode) Bus 002 Device 006: ID 04b8:012c Seiko Epson Corp. GT-X900 [Perfection V700/V750 Photo] Bus 002 Device 004: ID 1a40:0201 Terminus Technology Inc. FE 2.1 7-port Hub Bus 002 Device 003: ID 1759:5000 LucidPort Technology, Inc. Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 006 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub

6 13

[alsa-devel] Need help fixing pop/click artifacts in an ASOC driver
by Dimitris Papavasiliou 05 Jan '19

05 Jan '19

Hi, A (commercial) DAC board, I use with a Raspberry Pi is plagued by clicks and pops and I'm trying to fix it at the driver level. I have mostly fixed it, but I could use some guidance on the problem described below. I apologize in advance for the long read, but the situation is a bit convoluted (or my understanding of it is). The board (a HifiBerry DAC+ Pro) uses a PCM5122 chip which is driven by the pcm512x codec driver, but the Pro version also includes a set of external clocks on the board (one each for 48Khz and 44.1Khz sample rates and multiples thereof). There are two kinds of clicks and pops: on the one hand the board clicks when the device is opened/closed, mostly because the pcm512x codec driver doesn't support the digital_mute operation. These are relatively mild clicks, which are an annoyance at most and I have a separate patch for the pcm512x, that addresses them, which I'm currently testing and will (hopefully) submit soon. A separate case is a pop that's generated when the clock source is changed. This pop is independent of the digital volume setting and very loud. As far as I could determine, when the hw_params callback specified in the snd_soc_dai_link struct is called, it figures out what clock is needed for the giver sample rate and powers the appropriate clock through a GPIO on the PCM5122. This essentially means that the external clock source fed into the 5122 is stopped and restarted at a different rate, potentially while the DAC chip is operating. Later the hw_params callback defined in the pcm512x driver is called and it reconfigure the clock dividers for the new rate. It is at this point that the pop seems to be generated, provided that the card is not powered down or muted. If I comment out either the clock power-up/down in the dai_link hw_params, or the clock divider reconfiguration in the codec's hw_params, no pop is generated. Now, the digital_mute patch described before, also happens to fix this loud pop. I'm not sure whether that's a happy coincidence that might cease to be the case in the future though and was considering fixing the pop problem separately, at its source: the clock change. One way to fix the problem is to switch the 5122 into standby before changing the clock source in the dai_link hw_params callback and switch it back into normal operation afterwards. This seems to work regardless of whether digital_mute is implemented or not, but has the following problem: The register that's used to put the chip into/out of standby, is also used by the pcm512x driver, in the set_bias_level operation. Since the hw_params operation is, as far as I can tell, not atomic, I'm concerned that race conditions may arise, causing the card to be left powered up, when it should be off and vice-versa. Is there some way to synchronize the use of common chip registers in the DAI and codec drivers? Is the digital_mute somehow guaranteed to fix the pop originating at the DAI/machine driver level (whatever the hw_params operation in the dai_link struct is supposed to be)? Am I perhaps missing something entirely? Any help or guidance is welcome. Thanks, Dimitris

5 41

[alsa-devel] am57xx-evm: linux-4.14.y b8b55f8fd937 build Failure
by Naresh Kamboju 05 Jan '19

05 Jan '19

Build failure log of arm - BeagleBoard-X15 board while make modules Git branch: linux-4.14.y Git URL: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable-rc.git Git commit: b8b55f8fd937348f867eb0e4098b6d658959d075 In file included from sound/pci/hda/patch_realtek.c:33:0: sound/pci/hda/patch_realtek.c:6381:48: error: 'ALC294_FIXUP_ASUS_SPK' undeclared here (not in a function); did you mean 'ALC256_FIXUP_ASUS_MIC'? SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), ^ include/sound/core.h:415:43: note: in definition of macro 'SND_PCI_QUIRK' {_SND_PCI_QUIRK_ID(vend, dev), .value = (val)} ^~~ CC [M] drivers/cpufreq/highbank-cpufreq.o scripts/Makefile.build:326: recipe for target 'sound/pci/hda/patch_realtek.o' failed make[5]: *** [sound/pci/hda/patch_realtek.o] Error 1 - Naresh

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6asm: fix typo
by Matteo Croce 04 Jan '19

04 Jan '19

Fix spelling mistake: "lenght" -> "length" Signed-off-by: Matteo Croce <mcroce(a)redhat.com> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 4f85cb19a309..e8141a33a55e 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -1194,7 +1194,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * q6asm_write_async() - non blocking write * * @ac: audio client pointer - * @len: lenght in bytes + * @len: length in bytes * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw * @wflags: flags associated with write -- 2.20.1

1 0

Re: [alsa-devel] [PATCH] ASoC: rt5677: use more of the volume range from DACs
by Ross Zwisler 04 Jan '19

04 Jan '19

On Thu, Dec 20, 2018 at 04:25:55PM -0700, Ross Zwisler wrote: > From: Dylan Reid <dgreid(a)chromium.org> > > The DACs volume can go over 0, both according to the data sheet and > real world testing. The control can go up to +30dB. > > This was tested by playing audio at full volume on a samus chromebook. > > Signed-off-by: Dylan Reid <dgreid(a)chromium.org> > Reviewed-by: Hsinyu Chao <hychao(a)chromium.org> > Signed-off-by: Ross Zwisler <zwisler(a)google.com> Ping on this patch.

2 1

[alsa-devel] [PATCH 1/1] conf/ucm: chtrt5650: Add UCM config for chtrt5650
by Russell Parker 04 Jan '19

04 Jan '19

Add a UCM configuration for the rt5650 codec. Tested on a Samsung Chromebook 3. Adapted with minor modifications from GitHub user evan-a-a's gist: https://gist.github.com/evan-a-a/86b2a698708074530e2d0ee7c6498767 Signed-off-by: Russell Parker <russell.parker7(a)gmail.com> diff --git a/configure.ac b/configure.ac index d0cab2d6..d524ff4b 100644 --- a/configure.ac +++ b/configure.ac @@ -737,6 +737,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/chtnau8824/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ + src/conf/ucm/chtrt5650/Makefile \ src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ src/conf/ucm/DB410c/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 85c15090..985c460c 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -21,6 +21,7 @@ bytcr-rt5651-stereo-spk-in12-mic \ chtnau8824 \ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ +chtrt5650 \ cube-i1_TF-Defaultstring-CherryTrailCR \ DAISY-I2S \ DB410c \ diff --git a/src/conf/ucm/chtrt5650/HiFi.conf b/src/conf/ucm/chtrt5650/HiFi.conf new file mode 100644 index 00000000..cc3d64bf --- /dev/null +++ b/src/conf/ucm/chtrt5650/HiFi.conf @@ -0,0 +1,203 @@ + +SectionVerb { + # ALSA PCM + Value { + TQ "HiFi" + + # ALSA PCM device for HiFi + PlaybackPCM "hw:chtrt5650" + CapturePCM "hw:chtrt5650" + } + + EnableSequence [ + cdev "hw:chtrt5650" + + <platforms/bytcr/PlatformEnableSeq.conf> + + # Output Configuration + cset "name='DAC1 L Mux' IF1 DAC" + cset "name='DAC1 R Mux' IF1 DAC" + cset "name='DAC1 MIXL DAC1 Switch' 1" + cset "name='DAC1 MIXR DAC1 Switch' 1" + cset "name='Stereo DAC MIXL DAC L1 Switch' 1" + cset "name='Stereo DAC MIXR DAC R1 Switch' 1" + cset "name='Speaker HWEQ' 1,164,237,135,1,165,0,0,1,166,237,135,1,167,0,0,1,192,30,196,1,193,0,0,1,194,30,196,1,195,0,0,1,196,31,180,1,197,0,75,1,198,31,180,1,199,31,180,1,200,0,75,1,201,31,180,0,177,51,224" + cset "name='DAC1 Playback Volume' 77,77" + + cset "name='Speaker ClassD Playback Volume' 4" + + cset "name='DAC L2 Mux' IF2 DAC" + cset "name='DAC R2 Mux' IF2 DAC" + cset "name='Mono DAC MIXL DAC L2 Switch' on" + cset "name='Mono DAC MIXR DAC R2 Switch' on" + cset "name='DAC2 Playback Switch' on" + + cset "name='HPOVOL MIXL DAC1 Switch' on" + cset "name='HPOVOL MIXR DAC1 Switch' on" + cset "name='HPOVOL MIXL DAC2 Switch' on" + cset "name='HPOVOL MIXR DAC2 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='HPOVOL L Switch' on" + cset "name='HPOVOL R Switch' on" + + cset "name='SPK MIXL DAC L1 Switch' on" + cset "name='SPK MIXR DAC R1 Switch' on" + cset "name='SPK MIXL DAC L2 Switch' on" + cset "name='SPK MIXR DAC R2 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" + cset "name='SPOR MIX SPKVOL R Switch' on" + cset "name='SPKVOL L Switch' on" + cset "name='SPKVOL R Switch' on" + + # Input Configuration + cset "name='Stereo1 DMIC Mux' 0" + cset "name='Stereo1 ADC2 Mux' 1" + cset "name='ADC Capture Switch' on" + cset "name='ADC Capture Volume' 74" + cset "name='ADC Boost Capture Volume' 3" + cset "name='Mono ADC Capture Volume' 74" + cset "name='Mono ADC Boost Capture Volume' 2" + cset "name='IN Capture Volume' 63" + cset "name='RT5650 IF1 ADC Mux' 0" + cset "name='I2S2 Func Switch' off" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + # Disable audio output path + cset "name='codec_out1 mix 0 pcm0_in Switch' off" + cset "name='media0_out mix 0 media1_in Switch' off" + + cset "name='media1_in Gain 0 Switch' off" + cset "name='pcm0_in Gain 0 Switch' off" + cset "name='codec_out1 Gain 0 Switch' off" + + # Disable audio input path + cset "name='pcm1_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop2_out mix 0 codec_in0 Switch' off" + + cset "name='media_loop2_out Gain 0 Switch' off" + cset "name='pcm1_out Gain 0 Switch' off" + cset "name='codec_in0 Gain 0 Switch' off" + ] +} + +SectionDevice."Speaker" { + Comment "Speaker" + + Value { + PlaybackChannels "2" + } + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headphone Switch' off" + cset "name='Headphone Channel Switch' off" + + cset "name='Ext Spk Switch' on" + cset "name='Speaker Channel Switch' on" + cset "name='Speaker Playback Volume' 31" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Ext Spk Switch' off" + cset "name='Speaker Channel Switch' off" + ] +} + +SectionDevice."Headphones" { + Comment "Headphones" + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Ext Spk Switch' off" + cset "name='Speaker Channel Switch' off" + + cset "name='Headphone Switch' on" + cset "name='Headphone Channel Switch' on" + cset "name='Headphone Playback Volume' 31" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headphone Switch' off" + cset "name='Headphone Channel Switch' off" + ] +} + +SectionDevice."Mic".0 { + Comment "Internal Microphone" + + Value { + CaptureChannels "2" + CapturePriority "150" + } + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Int Mic Switch' on" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + cset "name='Int Mic Switch' off" + ] +} + +SectionDevice."HSMic".0 { + Comment "Headset Microphone" + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + JackHWMute "Mic" + } + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + cset "name='IN1 Boost' 1" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + cset "name='IN1 Boost' 0" + ] +} diff --git a/src/conf/ucm/chtrt5650/Makefile.am b/src/conf/ucm/chtrt5650/Makefile.am new file mode 100644 index 00000000..f506cd77 --- /dev/null +++ b/src/conf/ucm/chtrt5650/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/chtrt5650 +ucm_DATA = chtrt5650.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/chtrt5650/chtrt5650.conf b/src/conf/ucm/chtrt5650/chtrt5650.conf new file mode 100644 index 00000000..dfc36c89 --- /dev/null +++ b/src/conf/ucm/chtrt5650/chtrt5650.conf @@ -0,0 +1,5 @@ +Comment "Intel SoC Audio Device" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.17.2 (Apple Git-113)

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[alsa-devel] Applied "ASoC: Intel: Baytrail: remove unneeded variable" to the asoc tree
by Mark Brown 04 Jan '19

04 Jan '19

The patch ASoC: Intel: Baytrail: remove unneeded variable 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 060d35be2dfa9202d37f967fd20f133c530505d2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Sun, 16 Dec 2018 16:49:06 -0600 Subject: [PATCH] ASoC: Intel: Baytrail: remove unneeded variable Detected with Coccinelle Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 260447da32b8..2cd8f9668b50 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -278,7 +278,6 @@ static int sst_byt_process_notification(struct sst_byt *byt, struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -298,7 +297,7 @@ static int sst_byt_process_notification(struct sst_byt *byt, break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context) -- 2.20.1

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