June 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Issues and/or possible bugs in alsa
by Yomi Ogunwumi 03 Jul '15

03 Jul '15

Whoops, I didn't realize I sent that to you and you alone, Raymond. I'm going to take a guess and conclude that there must be bugs in the interface patch written for the realtek codecs. I happened to test two other laptops and they worked just fine. (Intel...) ...I guess in should report a bug related to that realtek patch... • Yomi On Mar 22, 2015 11:11 AM, "Yomi Ogunwumi" <abyomi0(a)gmail.com> wrote: > Is there anything I can do to resolve it? > > On Sun, Mar 22, 2015 at 11:09 AM, Yomi Ogunwumi <abyomi0(a)gmail.com> wrote: > >> There is one log where that snippet looks correct. >> >> https://gist.github.com/Yomi0/9bfc3e72588cf8a35811#file-alsainfo-headphones… >> >> >> https://gist.githubusercontent.com/Yomi0/9bfc3e72588cf8a35811/raw/77ff724d2… >> >> Codec: Realtek ALC269VB >> Address: 0 >> AFG Function Id: 0x1 (unsol 1) >> Vendor Id: 0x10ec0269 >> Subsystem Id: 0x1179fa22 >> Revision Id: 0x100100 >> 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 CLKSTOP EPSS >> Power: setting=D0, actual=D0 >> >> >> On Sun, Mar 15, 2015 at 11:58 AM, Raymond Yau < >> superquad.vortex2(a)gmail.com> wrote: >> >>> > >>> > Is there anything else I need to do? >>> >>> Codec: Realtek ALC269VB >>> Address: 0 >>> AFG Function Id: 0x1 (unsol 1) >>> Vendor Id: 0x10ec0269 >>> Subsystem Id: 0x1179fa22 >>> Revision Id: 0x100100 >>> No Modem Function Group found >>> Default PCM: >>> N/A >>> Default Amp-In caps: N/A >>> Default Amp-Out caps: N/A >>> State of AFG node 0x01: >>> Power: setting=UNKNOWN, actual=UNKNOWN, Error, Clock-stop-OK, >>> Setting-reset >>> Invalid AFG subtree >>> >>> Your problem seem related to power >>> >> -- >> *Yomi* >> > > > > -- > *Yomi* >

2 16

[alsa-devel] [PATCH] ASoC: omap: Fix typo in config option description
by Nik Nyby 02 Jul '15

02 Jul '15

This fixes a typo in the description for SND_OMAP_SOC_HDMI_AUDIO. Signed-off-by: Nik Nyby <nikolas(a)gnu.org> --- sound/soc/omap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 30d0109..5185a38 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,7 +24,7 @@ config SND_OMAP_SOC_HDMI_AUDIO component also under DSS HDMI device. Dummy codec is used as as codec component. The hdmi audio driver implements also the card and registers it under its own platform device. - The device for the dirver is registered by OMAPDSS hdmi + The device for the driver is registered by OMAPDSS hdmi driver. config SND_OMAP_SOC_N810 -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 01 Jul '15

01 Jul '15

When using snd_soc_dai_set_pll to set pll in machine driver, we should set pll in and pll out freq and ensure 5 < PLLN < 13, otherwise set pll will be failed. In order to support more formats and sample rates for a certain MCLK, if snd_soc_dai_set_pll failed, it will calculate a available pll out freq and set the pll again. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 160 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 126 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c46..9b17ca7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_dai *codec_dai, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -127,8 +130,9 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; - int bclk; + int freq_in; int sysclk; + int clk_id; struct wm8960_data pdata; }; @@ -565,6 +569,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +581,119 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + unsigned int sample_rate = params_rate(params); + unsigned int channels = params_channels(params); + unsigned int sysclk, bclk, pll_out, freq_in; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; } if (!wm8960->sysclk) { dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + bclk = snd_soc_params_to_bclk(params); + if (channels == 1) + bclk *= 2; + + sysclk = wm8960->sysclk; + + if (wm8960->clk_id == WM8960_SYSCLK_PLL) { + if (!wm8960->freq_in) { + dev_dbg(codec->dev, "No PLL input clock configured\n"); + return -EINVAL; + } + + pll_out = sysclk; + /* + * If the PLL input and output frequency are not available for + * wm8960 PLL, try to calculte a available pll out frequency and + * set pll again. + */ + if (!is_pll_freq_available(wm8960->freq_in, pll_out)) + goto get_pll_freq; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk /= sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) + if (sysclk == dac_divs[j] * sample_rate) + break; + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (j != ARRAY_SIZE(dac_divs) && k != ARRAY_SIZE(bclk_divs)) + break; + } + if (i != ARRAY_SIZE(sysclk_divs)) + goto configure_clock; + +get_pll_freq: + freq_in = wm8960->freq_in; + /* + * If the pll out frequcncy set from machine driver is not available, + * try to find a pll out frequcncy and set pll. + */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = sample_rate * dac_divs[j]; + pll_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, pll_out)) { + wm8960_set_pll(dai, freq_in, pll_out); break; + } else + continue; } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } +configure_clock: + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); /* * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC * pin is used as a frame clock for ADCs and DACs. */ if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -638,13 +703,8 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; - wm8960->bclk = snd_soc_params_to_bclk(params); - if (params_channels(params) == 1) - wm8960->bclk *= 2; - /* bit size */ switch (params_width(params)) { case 16: @@ -682,9 +742,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); - - return 0; + return wm8960_configure_clocking(substream, params, dai); } static int wm8960_mute(struct snd_soc_dai *dai, int mute) @@ -892,6 +950,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -916,7 +996,7 @@ static int pll_factors(unsigned int source, unsigned int target, pll_div->pre_div = 0; if ((Ndiv < 6) || (Ndiv > 12)) { - pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv); + pr_debug("WM8960 PLL: Unsupported N=%d\n", Ndiv); return -EINVAL; } @@ -943,8 +1023,8 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_dai *codec_dai, + unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -986,6 +1066,17 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + return wm8960_set_pll(codec_dai, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1048,6 +1139,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } -- 1.9.1

2 5

[alsa-devel] [V2 PATCH] ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
by Hui Wang 01 Jul '15

01 Jul '15

Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, they all belong to the cxt5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. This design refers to the similar solution for the realtek codec ALC269 family in the patch_realtek.c. Cc: Alex Hung <alex.hung(a)canonical.com> Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1..1f2717f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3244,9 +3244,29 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> +#include <acpi/acpi.h> static int (*led_set_func)(int, bool); +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + static void update_tpacpi_mute_led(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} -- 1.8.1.2

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[alsa-devel] [PATCH 0/4] ACPI: provide an override for _REV
by Dominik Brodowski 30 Jun '15

30 Jun '15

As promised, here are a few patches which allow to override the _REV (supported ACPI revision) setting of the ACPI implementation. With this patch in place (and possibly with other quirks added to drivers/acpi/osl.c -- Matthew mentioned the Inspiron 7437), the patch b1ef29725865 can be applied again (or the revert of b1ef29725865 be reverted). Also throw in a re-ordering of the kernel-parameters Documentation. Dominik Brodowski (4): acpi: use same type for acpi_predefined_names values as in definition acpi: allow for an override to set _REV acpi: add _REV quirk for Dell XPS 13 (2015) acpi: fix kernel-parameters ordering in Documentation Documentation/kernel-parameters.txt | 76 ++++++++++++++++++++++--------------- drivers/acpi/Kconfig | 15 ++++++++ drivers/acpi/bus.c | 2 + drivers/acpi/internal.h | 1 + drivers/acpi/osl.c | 60 ++++++++++++++++++++++++++++- include/acpi/acpiosxf.h | 2 +- 6 files changed, 123 insertions(+), 33 deletions(-) -- 2.1.4

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[alsa-devel] Intel Atom SST and low latency audio
by David Champlin 30 Jun '15

30 Jun '15

Hi Vinod, We're creating a machine driver between the Maxim 98090 codec and Intel Edison in support of a realtime audio application. So far, we have both record and playback working successfully. However, the latency on playback is higher than we would like. We've searched the alsa-devel messages but haven't been able to find a solution. We wondered if you might help us identify the best routing through the Atom's DSP for low-latency? We noticed that a "low latency" path is referenced in the Merrifield DPCM controls file (controls_v2_dpcm), but it doesn't appear to be implemented yet. What do we need to do to set up the SST to use its low latency paths? Would this be similar to configuring a media path (sending the command MMX_SET_MEDIA_PATH, SST command 119)? Thanks, David

1 0

[alsa-devel] [PATCH RFT] ASoC: ak4642: Fix up max_register setting
by Axel Lin 30 Jun '15

30 Jun '15

The max_register setting for ak4642, ak4643 and ak4648 are wrong, fix it. According to the datasheet: the maximum valid register for ak4642 is 0x1f the maximum valid register for ak4643 is 0x24 the maximum valid register for ak4648 is 0x27 The default settings for ak4642 and ak4643 are the same for 0x0 ~ 0x1f registers, so it's fine to use the same reg_default table with different num_reg_defaults setting. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have this h/w to test. I'd appreciate if someone can review and test this patch. Thanks, Axel sound/soc/codecs/ak4642.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7c0f6552..7f27345 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -64,12 +64,15 @@ #define FIL1_0 0x1c #define FIL1_1 0x1d #define FIL1_2 0x1e -#define FIL1_3 0x1f +#define FIL1_3 0x1f /* The maximum valid register for ak4642 */ #define PW_MGMT4 0x20 #define MD_CTL5 0x21 #define LO_MS 0x22 #define HP_MS 0x23 -#define SPK_MS 0x24 +#define SPK_MS 0x24 /* The maximum valid register for ak4643 */ +#define EQ_FBEQAB 0x25 +#define EQ_FBEQCD 0x26 +#define EQ_FBEQE 0x27 /* The maximum valid register for ak4648 */ /* PW_MGMT1*/ #define PMVCM (1 << 6) /* VCOM Power Management */ @@ -535,17 +538,25 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { static const struct regmap_config ak4642_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .max_register = FIL1_3, .reg_defaults = ak4642_reg, - .num_reg_defaults = ARRAY_SIZE(ak4642_reg), + .num_reg_defaults = FIL1_3 + 1, +}; + +static const struct regmap_config ak4643_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = SPK_MS, + .reg_defaults = ak4642_reg, + .num_reg_defaults = SPK_MS + 1, }; static const struct regmap_config ak4648_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, - .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .num_reg_defaults = EQ_FBEQE + 1, }; static const struct ak4642_drvdata ak4642_drvdata = { @@ -553,7 +564,7 @@ static const struct ak4642_drvdata ak4642_drvdata = { }; static const struct ak4642_drvdata ak4643_drvdata = { - .regmap_config = &ak4642_regmap, + .regmap_config = &ak4643_regmap, }; static const struct ak4642_drvdata ak4648_drvdata = { -- 2.1.0

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[alsa-devel] Help on snd_pcm_hw_params() for ulaw on plugin
by E.Richiardone 30 Jun '15

30 Jun '15

Sorry for void subject of before Hi all, I am building a solution on a Debian platform that should playback an audio recorded in G711 ulaw format. I am using Alsa to interfacing with the audio device, it's a snd_cs5535audio. Below there is a fragment of the code I am using. I don't have any problems accessing the device using the "plughw:0,0" device, where I can correctly set directly the ulaw 8000Hz format. Now I was needing an multiband equalizer, and found that Alsa supports some ladspa plugins, I choosed mbeq. But if I use a device name different from "plughw:0,0", as for example the "default" device or "plug", the final function snd_pcm_hw_params() returns an error: "cannot set hw params (Invalid argument)" I was thinking that maybe my problem is that only the "plughw" device accept directly the G711 ulaw format. Anyone can confirm my thought? There is a way to access a device directly in G711 ulaw format that can be equalized with a plugin? My initialisation source code: ----------8<---------- // #define AUDIODEV "plughw:0,0" #define AUDIODEV "default" [...] pcm_name = strdup(AUDIODEV); if((err = snd_pcm_open(&playback_handle, pcm_name, stream, 0)) < 0){ printf(" E> cannot open audio device %s (%s)\n", pcm_name, snd_strerror(err)); closeall(1); } if((err = snd_pcm_hw_params_any(playback_handle, hw_params)) < 0){ printf(" E> cannot initialize hardware parameter structure (%s)\n", snd_strerror(err)); closeall(1); } if((err = snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0){ printf(" E> cannot set access type (%s)\n", snd_strerror(err)); closeall(1); } err = snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_MU_LAW); if(err < 0){ printf(" E> cannot set ulaw format (%s)\n", snd_strerror(err)); closeall(1); } val = 8000; if((err = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &val, 0)) < 0){ printf(" E> cannot set rate (%s)\n", snd_strerror(err)); closeall(1); } [...] if((err = snd_pcm_hw_params(playback_handle, hw_params)) < 0){ printf(" E> cannot set hw params (%s)\n", snd_strerror(err)); closeall(1); } ---------->8---------- My /etc/asound.conf : ----------8<---------- defaults.ctl.card 0 defaults.ctl.device 0 defaults.pcm.card 0 defaults.pcm.device 0 ctl.equal { type equal; library "/usr/lib/ladspa/mbeq_1197.so"; module "mbeq"; controls "/usr/local/webif/.alsaequal.bin"; } pcm.plugequal { type equal; slave.pcm "plughw:0,0"; library "/usr/lib/ladspa/mbeq_1197.so"; module "mbeq"; controls "/usr/local/webif/.alsaequal.bin"; } pcm.!default { type plug; slave.pcm plugequal; hint { show on description "Default equalized" } } pcm.dsp0 { type plug slave.pcm plugequal; } ---------->8---------- Many thanks Best Regards -- E.Richiardone

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[alsa-devel] [RFC PATCH 1/2] regmap: add configurable lock class key for lockdep
by Nicolas Boichat 30 Jun '15

30 Jun '15

From: Antti Palosaari <crope(a)iki.fi> Lockdep validator complains about recursive locking and deadlock when two different regmap instances are called in a nested order. That happens anytime a regmap read/write call needs to access another regmap. This is because, for performance reason, lockdep groups all locks initialized by the same mutex_init() in the same lock class. Therefore all regmap mutexes are in the same lock class, leading to lockdep "nested locking" warnings if a regmap accesses another regmap. However, depending on the specifics of the driver, this can be perfectly safe (e.g. if there is a clear hierarchy between a "master" regmap that uses another "slave" regmap). In these cases, the warning is false and should be silenced. As a solution, add configuration option to pass custom lock class key for lockdep validator, to be used in the regmap that needs to access another regmap. This removes the need for uglier workarounds in drivers, just to silence this warning (e.g. add custom mutex lock/unlock functions). Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Antti Palosaari <crope(a)iki.fi> [drinkcat(a)chromium.org: Reworded the commit message and regmap.h documentation] Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- I'm trying to revive Antti's patch [1], as we are hitting similar issue with rt5677 driver. I only updated the commit message and documentation, I kept both Signed-off-by and From lines, with a small note highlighting my changes, let me know if that's not appropriate. One issue that was raised by Mark at the time is that a per-regmap_config lock class might not be enough (Mark mentioned clocks as an example). The current implementation should be good enough as long as the clock regmaps do not access each other. If this is a problem, maybe we should consider replacing lockdep_lock_class_key by a boolean use_own_lock_class that would allocate/use a per regmap instance lock class, or have devm_regmap_init take an extra parameter specifying the lock class. [1] https://www.mail-archive.com/linux-media%40vger.kernel.org/msg83490.html drivers/base/regmap/regmap.c | 3 +++ include/linux/regmap.h | 7 +++++++ 2 files changed, 10 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6273ff0..f5d1131 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -560,6 +560,9 @@ struct regmap *regmap_init(struct device *dev, mutex_init(&map->mutex); map->lock = regmap_lock_mutex; map->unlock = regmap_unlock_mutex; + if (config->lockdep_lock_class_key) + lockdep_set_class(&map->mutex, + config->lockdep_lock_class_key); } map->lock_arg = map; } diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 116655d..09aaaf5 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -135,6 +135,12 @@ typedef void (*regmap_unlock)(void *); * @lock_arg: this field is passed as the only argument of lock/unlock * functions (ignored in case regular lock/unlock functions * are not overridden). + * @lock_class_key: Custom lock class key for lockdep validator. Use that to + * silence false lockdep nested locking warning, when this + * regmap needs to access another regmap during read/write + * operations (directly in read/write functions, or + * indirectly, e.g. through bus accesses). + * Valid only when regmap default mutex locking is used. * @reg_read: Optional callback that if filled will be used to perform * all the reads from the registers. Should only be provided for * devices whose read operation cannot be represented as a simple @@ -198,6 +204,7 @@ struct regmap_config { regmap_lock lock; regmap_unlock unlock; void *lock_arg; + struct lock_class_key *lockdep_lock_class_key; int (*reg_read)(void *context, unsigned int reg, unsigned int *val); int (*reg_write)(void *context, unsigned int reg, unsigned int val); -- 2.4.3.573.g4eafbef

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[alsa-devel] (no subject)
by E.Richiardone 30 Jun '15

30 Jun '15

Hi all, I am building a solution on a Debian platform that should playback an audio recorded in G711 ulaw format. I am using Alsa to interfacing with the audio device, it's a snd_cs5535audio. Below there is a fragment of the code I am using. I don't have any problems accessing the device using the "plughw:0,0" device, where I can correctly set directly the ulaw 8000Hz format. Now I was needing an multiband equalizer, and found that Alsa supports some ladspa plugins, I choosed mbeq. But if I use a device name different from "plughw:0,0", as for example the "default" device or "plug", the final function snd_pcm_hw_params() returns an error: "cannot set hw params (Invalid argument)" I was thinking that maybe my problem is that only the "plughw" device accept directly the G711 ulaw format. Anyone can confirm my thought? There is a way to access a device directly in G711 ulaw format that can be equalized with a plugin? My initialisation source code: ----------8<---------- // #define AUDIODEV "plughw:0,0" #define AUDIODEV "default" [...] pcm_name = strdup(AUDIODEV); if((err = snd_pcm_open(&playback_handle, pcm_name, stream, 0)) < 0){ printf(" E> cannot open audio device %s (%s)\n", pcm_name, snd_strerror(err)); closeall(1); } if((err = snd_pcm_hw_params_any(playback_handle, hw_params)) < 0){ printf(" E> cannot initialize hardware parameter structure (%s)\n", snd_strerror(err)); closeall(1); } if((err = snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0){ printf(" E> cannot set access type (%s)\n", snd_strerror(err)); closeall(1); } err = snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_MU_LAW); if(err < 0){ printf(" E> cannot set ulaw format (%s)\n", snd_strerror(err)); closeall(1); } val = 8000; if((err = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &val, 0)) < 0){ printf(" E> cannot set rate (%s)\n", snd_strerror(err)); closeall(1); } [...] if((err = snd_pcm_hw_params(playback_handle, hw_params)) < 0){ printf(" E> cannot set hw params (%s)\n", snd_strerror(err)); closeall(1); } ---------->8---------- My /etc/asound.conf : ----------8<---------- defaults.ctl.card 0 defaults.ctl.device 0 defaults.pcm.card 0 defaults.pcm.device 0 ctl.equal { type equal; library "/usr/lib/ladspa/mbeq_1197.so"; module "mbeq"; controls "/usr/local/webif/.alsaequal.bin"; } pcm.plugequal { type equal; slave.pcm "plughw:0,0"; library "/usr/lib/ladspa/mbeq_1197.so"; module "mbeq"; controls "/usr/local/webif/.alsaequal.bin"; } pcm.!default { type plug; slave.pcm plugequal; hint { show on description "Default equalized" } } pcm.dsp0 { type plug slave.pcm plugequal; } ---------->8---------- Many thanks Best Regards -- E.Richiardone

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