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[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 v2 0/13] topology support for alsa-lib
by Liam Girdwood 01 Jul '15

01 Jul '15

This patch series adds the topology text parser support to alsa-lib. The topology parser converts text topology data to the binary ABI format read by the kernel topology core. Changes since V1 :- o Terminate all strncpy() strings with NULL. o Use snd_pcm_format_value() instead of local lookup. o Set a maximum size for private data files. o Set capture/playback stream direction for new PCMs o Only set TLV access flags for TLV CTLs. Liam

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[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

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[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

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[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

4 25

[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

1 0

[alsa-devel] [PATCH 0/5] Add configuration options for tlv320aic32x4
by Kimmo Saarela 30 Jun '15

30 Jun '15

* Patch 1: Make tlv320aic32x4 module selectable on it's own on config system * Make codec visible in config tool, instead of magic selection via certain board option * Patch 2: Don't stop driver initialization if mlck is not defined * Only warn if mclk isn't in dtb file, such cases exist when other end of I2S bus provides the clock for the system and coded is not in control * Patch 3: Fix bus signal polarity configuration * Fix driver to use correct format polarity bits in config word, instead of magically setting polarity in format configuration * Patch 4: Add more devicetree options to driver * Add input-use and cmode array options to configure codec input pins properly. Current driver lets unused input pins to float. * Add swapdacs option, works similarly to old platform data option. When enabled, swaps output channels * Add powercfg config mask to pass codec power options, like: micbias off, configurable ldoin voltage, disable weak pull between AVDD and DVDD, configurable LDO voltages * Patch 5: Update codec documentation to meet changes above Kimmo Saarela (5): sound/soc/codecs: Make tlv320aic32x4 codec selectable on it's own ASoC: tlv320aic32x4: Make mclk optional ASoC: tlv320aic32x4: Move bit polarity config to specific switch statement ASoC: tlv320aic32x4: Change codec input config options ASoC: tlv320aic32x4: Update devicetree documentation .../devicetree/bindings/sound/tlv320aic32x4.txt | 27 ++- include/sound/tlv320aic32x4.h | 8 + sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/tlv320aic32x4.c | 244 ++++++++++++++++++--- sound/soc/codecs/tlv320aic32x4.h | 10 + 5 files changed, 258 insertions(+), 34 deletions(-) -- 1.9.1

1 5

[alsa-devel] [PATCH] ALSA: Fix uninintialized error return
by Colin King 30 Jun '15

30 Jun '15

From: Colin Ian King <colin.king(a)canonical.com> Static analysis with cppcheck found the following error: [sound/core/init.c:118]: (error) Uninitialized variable: err ..this was introduced by commit 2471b6c80a70e80de69f5ff4c37187c3912e5874 ("ALSA: info: Register proc entries recursively, too") where the call to snd_info_card_register was removed and no longer setting the error return in err. When snd_info_create_card_entry fails to allocate a an entry, the error path exits with garbage in err. Fix is to return -ENOMEM if entry fails to be allocated. Author: Takashi Iwai <tiwai(a)suse.de> Date: Mon May 18 09:20:24 2015 +0200 Fixes: 2471b6c80a ("ALSA: info: Register proc entries recursively, too") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/init.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 3e0ceba..20f37fb 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -109,13 +109,12 @@ static void snd_card_id_read(struct snd_info_entry *entry, static int init_info_for_card(struct snd_card *card) { - int err; struct snd_info_entry *entry; entry = snd_info_create_card_entry(card, "id", card->proc_root); if (!entry) { dev_dbg(card->dev, "unable to create card entry\n"); - return err; + return -ENOMEM; } entry->c.text.read = snd_card_id_read; card->proc_id = entry; -- 2.1.4

3 2

[alsa-devel] [PATCH lib v2] Replace unsafe characters with _ in card name
by Alexander E. Patrakov 29 Jun '15

29 Jun '15

Otherwise, they get misinterpreted as argument separators in USB-Audio PCM definitions, and thus prevent SPDIF blacklist entries from working. While at it, add my Logitec C910 webcam to the SPDIF blacklist. Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- v2: Fixed misplaced NULL check for tmp->u.string include/conf.h | 1 + src/conf.c | 32 ++++++++++++++++++++++++++++++++ src/conf/cards/USB-Audio.conf | 3 ++- src/confmisc.c | 2 +- 4 files changed, 36 insertions(+), 2 deletions(-) diff --git a/include/conf.h b/include/conf.h index ff270f6..087c05d 100644 --- a/include/conf.h +++ b/include/conf.h @@ -126,6 +126,7 @@ int snd_config_imake_integer(snd_config_t **config, const char *key, const long int snd_config_imake_integer64(snd_config_t **config, const char *key, const long long value); int snd_config_imake_real(snd_config_t **config, const char *key, const double value); int snd_config_imake_string(snd_config_t **config, const char *key, const char *ascii); +int snd_config_imake_safe_string(snd_config_t **config, const char *key, const char *ascii); int snd_config_imake_pointer(snd_config_t **config, const char *key, const void *ptr); snd_config_type_t snd_config_get_type(const snd_config_t *config); diff --git a/src/conf.c b/src/conf.c index bb256e7..c6a83ee 100644 --- a/src/conf.c +++ b/src/conf.c @@ -2228,6 +2228,38 @@ int snd_config_imake_string(snd_config_t **config, const char *id, const char *v return 0; } +int snd_config_imake_safe_string(snd_config_t **config, const char *id, const char *value) +{ + int err; + snd_config_t *tmp; + char *c; + + err = snd_config_make(&tmp, id, SND_CONFIG_TYPE_STRING); + if (err < 0) + return err; + if (value) { + tmp->u.string = strdup(value); + if (!tmp->u.string) { + snd_config_delete(tmp); + return -ENOMEM; + } + + for (c = tmp->u.string; *c; c++) { + if (*c == ' ' || *c == '-' || *c == '_' || + (*c >= '0' && *c <= '9') || + (*c >= 'a' && *c <= 'z') || + (*c >= 'A' && *c <= 'Z')) + continue; + *c = '_'; + } + } else { + tmp->u.string = NULL; + } + *config = tmp; + return 0; +} + + /** * \brief Creates a pointer configuration node with the given initial value. * \param[out] config The function puts the handle to the new node at diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 031bee0..e365f29 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -57,7 +57,8 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 - "USB Device 0x46d:0x992" 999 + "USB Device 0x46d_0x821" 999 + "USB Device 0x46d_0x992" 999 } # Second iec958 device number, if any. diff --git a/src/confmisc.c b/src/confmisc.c index 1fb4f28..ae0275f 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -935,7 +935,7 @@ int snd_func_card_name(snd_config_t **dst, snd_config_t *root, } err = snd_config_get_id(src, &id); if (err >= 0) - err = snd_config_imake_string(dst, id, + err = snd_config_imake_safe_string(dst, id, snd_ctl_card_info_get_name(info)); __error: if (ctl) -- 2.4.4

2 1

[alsa-devel] [PATCH 01/13] topology: uapi: Add UAPI headers for topology ABI
by Liam Girdwood 29 Jun '15

29 Jun '15

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/asoc.h | 387 +++++++++++++++++++++++++++++++++++++++++++++++++++ include/sound/tlv.h | 23 +++ 2 files changed, 410 insertions(+) create mode 100644 include/sound/asoc.h create mode 100644 include/sound/tlv.h diff --git a/include/sound/asoc.h b/include/sound/asoc.h new file mode 100644 index 0000000..498cd0c --- /dev/null +++ b/include/sound/asoc.h @@ -0,0 +1,387 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +/* + * Maximum number of channels topology kcontrol can represent. + */ +#define SND_SOC_TPLG_MAX_CHAN 8 + +/* + * Maximum number of PCM formats capability + */ +#define SND_SOC_TPLG_MAX_FORMATS 16 + +/* + * Maximum number of PCM stream configs + */ +#define SND_SOC_TPLG_STREAM_CONFIG_MAX 8 + +/* individual kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_CTL_VOLSW 1 +#define SND_SOC_TPLG_CTL_VOLSW_SX 2 +#define SND_SOC_TPLG_CTL_VOLSW_XR_SX 3 +#define SND_SOC_TPLG_CTL_ENUM 4 +#define SND_SOC_TPLG_CTL_BYTES 5 +#define SND_SOC_TPLG_CTL_ENUM_VALUE 6 +#define SND_SOC_TPLG_CTL_RANGE 7 +#define SND_SOC_TPLG_CTL_STROBE 8 + + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_DAPM_CTL_VOLSW 64 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE 65 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT 66 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE 67 +#define SND_SOC_TPLG_DAPM_CTL_PIN 68 + +/* DAPM widget types - add new items to the end */ +#define SND_SOC_TPLG_DAPM_INPUT 0 +#define SND_SOC_TPLG_DAPM_OUTPUT 1 +#define SND_SOC_TPLG_DAPM_MUX 2 +#define SND_SOC_TPLG_DAPM_MIXER 3 +#define SND_SOC_TPLG_DAPM_PGA 4 +#define SND_SOC_TPLG_DAPM_OUT_DRV 5 +#define SND_SOC_TPLG_DAPM_ADC 6 +#define SND_SOC_TPLG_DAPM_DAC 7 +#define SND_SOC_TPLG_DAPM_SWITCH 8 +#define SND_SOC_TPLG_DAPM_PRE 9 +#define SND_SOC_TPLG_DAPM_POST 10 +#define SND_SOC_TPLG_DAPM_AIF_IN 11 +#define SND_SOC_TPLG_DAPM_AIF_OUT 12 +#define SND_SOC_TPLG_DAPM_DAI_IN 13 +#define SND_SOC_TPLG_DAPM_DAI_OUT 14 +#define SND_SOC_TPLG_DAPM_DAI_LINK 15 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* Max size of TLV data */ +#define SND_SOC_TPLG_TLV_SIZE 32 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_TYPE_MIXER 1 +#define SND_SOC_TPLG_TYPE_BYTES 2 +#define SND_SOC_TPLG_TYPE_ENUM 3 +#define SND_SOC_TPLG_TYPE_DAPM_GRAPH 4 +#define SND_SOC_TPLG_TYPE_DAPM_WIDGET 5 +#define SND_SOC_TPLG_TYPE_DAI_LINK 6 +#define SND_SOC_TPLG_TYPE_PCM 7 +#define SND_SOC_TPLG_TYPE_MANIFEST 8 +#define SND_SOC_TPLG_TYPE_CODEC_LINK 9 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 +#define SND_SOC_TPLG_TYPE_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_TYPE_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_TYPEVENDOR_CODEC 1003 + +#define SND_SOC_TPLG_STREAM_PLAYBACK 0 +#define SND_SOC_TPLG_STREAM_CAPTURE 1 + +/* + * Block Header. + * This header precedes all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; /* magic number */ + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_TYPE_ */ + __le32 size; /* size of this structure */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 payload_size; /* data bytes, excluding this header */ + __le32 index; /* identifier for block */ + __le32 count; /* number of elements in block */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 size; /* in bytes of private data */ + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 size; /* in bytes aligned to 4 */ + __le32 numid; /* control element numeric identification */ + __le32 count; /* number of elem in data array */ + __le32 data[SND_SOC_TPLG_TLV_SIZE]; +} __attribute__((packed)); + +/* + * Kcontrol channel data + */ +struct snd_soc_tplg_channel { + __le32 size; /* in bytes of this structure */ + __le32 reg; + __le32 shift; + __le32 id; /* ID maps to Left, Right, LFE etc */ +} __attribute__((packed)); + +/* + * Kcontrol Operations IDs + */ +struct snd_soc_tplg_kcontrol_ops_id { + __le32 get; + __le32 put; + __le32 info; +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_ctl_hdr { + __le32 size; /* in bytes of this structure */ + __le32 type; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 access; + struct snd_soc_tplg_kcontrol_ops_id ops; + __le32 tlv_size; /* non zero means control has TLV data */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ + __le32 rate_min; /* min rate */ + __le32 rate_max; /* max rate */ + __le32 channels_min; /* min channels */ + __le32 channels_max; /* max channels */ + __le32 periods_min; /* min number of periods */ + __le32 periods_max; /* max number of periods */ + __le32 period_size_min; /* min period size bytes */ + __le32 period_size_max; /* max period size bytes */ + __le32 buffer_size_min; /* min buffer size bytes */ + __le32 buffer_size_max; /* max buffer size bytes */ +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le32 size; /* in bytes of this structure */ + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global component resource allocations. + * + * File block representation for manifest :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_manifest | 1 | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_manifest { + __le32 size; /* in bytes of this structure */ + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Mixer kcontrol. + * + * File block representation for mixer kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_mixer_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 min; + __le32 max; + __le32 platform_max; + __le32 invert; + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + struct snd_soc_tplg_ctl_tlv tlv; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + * + * File block representation for enum kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_enum_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + __le32 items; + __le32 mask; + __le32 count; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + * + * File block representation for bytes kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_bytes_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_bytes_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 max; + __le32 mask; + __le32 base; + __le32 num_regs; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + * + * File block representation for DAPM graph elements :- + * +-------------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+----+ + * | struct snd_soc_tplg_dapm_graph_elem | N | + * +-------------------------------------+----+ + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + * + * File block representation for DAPM widget :- + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_dapm_widget | N | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_enum_control | 0|1 | + * | struct snd_soc_tplg_mixer_control | 0|N | + * +-------------------------------------+-----+ + * + * Optional enum or mixer control can be appended to the end of each widget + * in the block. + */ +struct snd_soc_tplg_dapm_widget { + __le32 size; /* in bytes of this structure */ + __le32 id; /* SND_SOC_DAPM_CTL */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __le32 subseq; /* sort within widget type */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +struct snd_soc_tplg_pcm_cfg_caps { + struct snd_soc_tplg_stream_caps caps; + struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; + __le32 num_configs; /* number of configs */ +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + * + * File block representation for PCM/DAI-Link :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_dapm_pcm_dai | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_pcm_dai { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ +} __attribute__((packed)); + +#endif diff --git a/include/sound/tlv.h b/include/sound/tlv.h new file mode 100644 index 0000000..33d747d --- /dev/null +++ b/include/sound/tlv.h @@ -0,0 +1,23 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __UAPI_SOUND_TLV_H +#define __UAPI_SOUND_TLV_H + +#define SNDRV_CTL_TLVT_CONTAINER 0 /* one level down - group of TLVs */ +#define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ +#define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ +#define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ + +#endif -- 2.1.4

2 15

[alsa-devel] [PATCH lib] Replace unsafe characters with _ in card name
by Alexander E. Patrakov 29 Jun '15

29 Jun '15

Otherwise, they get misinterpreted as argument separators in USB-Audio PCM definitions, and thus prevent SPDIF blacklist entries from working. While at it, add my Logitec C910 webcam to the SPDIF blacklist. Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- I did not send this patch earlier because of hesitation whether to send it as it is or to implement proper escaping, and then forgot about it. include/conf.h | 1 + src/conf.c | 32 ++++++++++++++++++++++++++++++++ src/conf/cards/USB-Audio.conf | 3 ++- src/confmisc.c | 2 +- 4 files changed, 36 insertions(+), 2 deletions(-) diff --git a/include/conf.h b/include/conf.h index ff270f6..087c05d 100644 --- a/include/conf.h +++ b/include/conf.h @@ -126,6 +126,7 @@ int snd_config_imake_integer(snd_config_t **config, const char *key, const long int snd_config_imake_integer64(snd_config_t **config, const char *key, const long long value); int snd_config_imake_real(snd_config_t **config, const char *key, const double value); int snd_config_imake_string(snd_config_t **config, const char *key, const char *ascii); +int snd_config_imake_safe_string(snd_config_t **config, const char *key, const char *ascii); int snd_config_imake_pointer(snd_config_t **config, const char *key, const void *ptr); snd_config_type_t snd_config_get_type(const snd_config_t *config); diff --git a/src/conf.c b/src/conf.c index bb256e7..e72a58a 100644 --- a/src/conf.c +++ b/src/conf.c @@ -2228,6 +2228,38 @@ int snd_config_imake_string(snd_config_t **config, const char *id, const char *v return 0; } +int snd_config_imake_safe_string(snd_config_t **config, const char *id, const char *value) +{ + int err; + snd_config_t *tmp; + char *c; + + err = snd_config_make(&tmp, id, SND_CONFIG_TYPE_STRING); + if (err < 0) + return err; + if (value) { + tmp->u.string = strdup(value); + for (c = tmp->u.string; *c; c++) { + if (*c == ' ' || *c == '-' || *c == '_' || + (*c >= '0' && *c <= '9') || + (*c >= 'a' && *c <= 'z') || + (*c >= 'A' && *c <= 'Z')) + continue; + *c = '_'; + } + + if (!tmp->u.string) { + snd_config_delete(tmp); + return -ENOMEM; + } + } else { + tmp->u.string = NULL; + } + *config = tmp; + return 0; +} + + /** * \brief Creates a pointer configuration node with the given initial value. * \param[out] config The function puts the handle to the new node at diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 031bee0..e365f29 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -57,7 +57,8 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 - "USB Device 0x46d:0x992" 999 + "USB Device 0x46d_0x821" 999 + "USB Device 0x46d_0x992" 999 } # Second iec958 device number, if any. diff --git a/src/confmisc.c b/src/confmisc.c index 1fb4f28..ae0275f 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -935,7 +935,7 @@ int snd_func_card_name(snd_config_t **dst, snd_config_t *root, } err = snd_config_get_id(src, &id); if (err >= 0) - err = snd_config_imake_string(dst, id, + err = snd_config_imake_safe_string(dst, id, snd_ctl_card_info_get_name(info)); __error: if (ctl) -- 2.4.4

2 3

[alsa-devel] [PATCH 1/3] ucm: docs: fix doxygen exclude patch for UCM local header
by Liam Girdwood 29 Jun '15

29 Jun '15

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- doc/doxygen.cfg.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/doc/doxygen.cfg.in b/doc/doxygen.cfg.in index f4499d6..043e75b 100644 --- a/doc/doxygen.cfg.in +++ b/doc/doxygen.cfg.in @@ -94,7 +94,7 @@ EXCLUDE = @top_srcdir@/src/control/control_local.h \ @top_srcdir@/src/mixer/mixer_local.h \ @top_srcdir@/src/rawmidi/rawmidi_local.h \ @top_srcdir@/src/seq/seq_local.h \ - @top_srcdir@/src/seq/ucm_local.h + @top_srcdir@/src/ucm/ucm_local.h RECURSIVE = YES FILE_PATTERNS = *.c *.h EXAMPLE_PATH = @top_srcdir@/test -- 2.1.4

2 4

[alsa-devel] [PATCH] ALSA: hda - Fix the dock headphone output on Fujitsu Lifebook E780
by Takashi Iwai 29 Jun '15

29 Jun '15

Fujitsu Lifebook E780 sets the sequence number 0x0f to only only of the two headphones, thus the driver tries to assign another as the line-out, and this results in the inconsistent mapping between the created jack ctl and the actual I/O. Due to this, PulseAudio doesn't handle it properly and gets the silent output. The fix is to ignore the non-HP sequencer checks. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99681 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cab09c42598b..dfa3852a3476 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4464,6 +4464,7 @@ enum { ALC269_FIXUP_LIFEBOOK, ALC269_FIXUP_LIFEBOOK_EXTMIC, ALC269_FIXUP_LIFEBOOK_HP_PIN, + ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT, ALC269_FIXUP_AMIC, ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, @@ -4634,6 +4635,10 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + }, [ALC269_FIXUP_AMIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5208,6 +5213,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), -- 2.4.4

2 1

[alsa-devel] [PATCH] ALSA: hda: Delete an unnecessary check before the function call "snd_info_free_entry"
by SF Markus Elfring 29 Jun '15

29 Jun '15

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 28 Jun 2015 11:15:28 +0200 The snd_info_free_entry() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f852734..2f24338 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -591,7 +591,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) static void eld_proc_free(struct hdmi_spec_per_pin *per_pin) { - if (!per_pin->codec->bus->shutdown && per_pin->proc_entry) { + if (!per_pin->codec->bus->shutdown) { snd_info_free_entry(per_pin->proc_entry); per_pin->proc_entry = NULL; } -- 2.4.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Add a fixup for Dell E7450
by Takashi Iwai 29 Jun '15

29 Jun '15

Dell E7450 [0128:062e] needs the same quirk as other E7xx models. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=100571 Cc: <stable(a)vger.kernel.org> # v4.1 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dfa3852a3476..b3b44681d3cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5126,6 +5126,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), + SND_PCI_QUIRK(0x1028, 0x062e, "Dell Latitude E7450", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.4.4

1 0

[alsa-devel] ASoC: 88pm860x: Question about pm860x_set_dai_sysclk implementation
by Axel Lin 29 Jun '15

29 Jun '15

Hi Haojian, In sound/soc/codecs/88pm860x-codec.c (pm860x_set_dai_sysclk): When dir != PM860X_CLK_DIR_OUT, current code set pm860x->dir = PM860X_CLK_DIR_IN; and then return -EINVAL. This looks confusion. Since there is no comment for that, I'm wondering if the return -EINVAL is intentional or not. Regards, Axel

2 3

[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 28 Jun '15

28 Jun '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e79dc16..d043c7c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

1 0

[alsa-devel] [PATCH] ASoC: fsl_ssi: fix AC'97 mode
by Maciej S. Szmigiero 28 Jun '15

28 Jun '15

Currently the AC'97 mode in fsl_ssi driver isn't functional. This patch implements the following changes to make it work properly: * IPG clock have to be enabled during AC'97 CODEC register access, * AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params, * AC'97 bus can support asymmetric playback/capture rates, * Check whether setting AC'97 ops succeeded and clean them on removal so the driver can be reloaded, * AC'97 CODEC will be instantiated in AC'97 mode, * Set DAI format function small fixes in AC'97 mode. Tested on UDOO board. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9a63df2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; } @@ -1101,6 +1103,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -1127,10 +1130,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1151,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1163,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1182,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; } @@ -1291,6 +1313,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; + bool newbinding; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1320,7 +1343,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1357,7 +1384,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } @@ -1405,7 +1434,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ - if (!of_get_property(np, "codec-handle", NULL)) + newbinding = !of_get_property(np, "codec-handle", NULL); + if (newbinding) goto done; /* Trigger the machine driver's probe function. The platform driver @@ -1434,6 +1464,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (newbinding && fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card: @@ -1458,6 +1509,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

3 6

[alsa-devel] PulseAudio and SNDRV_PCM_INFO_BATCH
by Arun Raghavan 28 Jun '15

28 Jun '15

Hi folks, I'd like to bring this one up again, since we are currently in the sub-optimal position of forcing ~100 ms latency on USB devices. The original thread is here -- http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/069666.h… I see two flags that are possibly of consequence here: SNDRV_PCM_INFO_BATCH and SNDRV_PCM_INFO_BLOCK_TRANSFER. I'm not sure what these mean -- the documentation mentions "double buffering" for the batch flag, and just that the block transfer means "block transfer". :-) We've spoken about batch meaning either transfers in period size chunks, or some fixed chunk size. It seems that it would make more sense for it to mean the former, and block transfer to mean the latter. So I guess the first thing that would be nice to have is a clear meaning of these two flags. With this done, we could potentially get to the API to report the transfer size from the driver. I did notice that there is a snd_pcm_hw_params_get_fifo_size(). Is this something we could use for the purpose of transfer size reporting, by any chance? Cheers, Arun

8 38

[alsa-devel] snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 4.
by Adam Nielsen 27 Jun '15

27 Jun '15

Hi all, I'm trying to track down a problem with my media player (xmms2) where it has stopped advancing to the next track after reaching the end of the song. I have the same distro with the same package versions on two slightly different computers, and it works fine on one but not the other. One difference between the two is that although both have Intel HD Audio, each is provided by a different type of chip. I was asked to try pulseaudio to see if the problem persisted there and it did, however upon loading the pulseaudio server it told me this: E: [alsa-source-ALC892 Analog] alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 4. E: [alsa-source-ALC892 Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: [alsa-source-ALC892 Analog] alsa-util.c: snd_pcm_dump(): E: [alsa-source-ALC892 Analog] alsa-util.c: Direct Snoop PCM E: [alsa-source-ALC892 Analog] alsa-util.c: Its setup is: E: [alsa-source-ALC892 Analog] alsa-util.c: stream : CAPTURE E: [alsa-source-ALC892 Analog] alsa-util.c: access : MMAP_INTERLEAVED E: [alsa-source-ALC892 Analog] alsa-util.c: format : S32_LE E: [alsa-source-ALC892 Analog] alsa-util.c: subformat : STD E: [alsa-source-ALC892 Analog] alsa-util.c: channels : 2 E: [alsa-source-ALC892 Analog] alsa-util.c: rate : 48000 E: [alsa-source-ALC892 Analog] alsa-util.c: exact rate : 48000 (48000/1) E: [alsa-source-ALC892 Analog] alsa-util.c: msbits : 32 E: [alsa-source-ALC892 Analog] alsa-util.c: buffer_size : 16384 E: [alsa-source-ALC892 Analog] alsa-util.c: period_size : 1024 E: [alsa-source-ALC892 Analog] alsa-util.c: period_time : 21333 E: [alsa-source-ALC892 Analog] alsa-util.c: tstamp_mode : ENABLE E: [alsa-source-ALC892 Analog] alsa-util.c: period_step : 1 E: [alsa-source-ALC892 Analog] alsa-util.c: avail_min : 15943 E: [alsa-source-ALC892 Analog] alsa-util.c: period_event : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: start_threshold : -1 E: [alsa-source-ALC892 Analog] alsa-util.c: stop_threshold : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_threshold: 0 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_size : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: boundary : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: Hardware PCM card 1 'HDA Intel PCH' device 0 subdevice 0 E: [alsa-source-ALC892 Analog] alsa-util.c: Its setup is: E: [alsa-source-ALC892 Analog] alsa-util.c: stream : CAPTURE E: [alsa-source-ALC892 Analog] alsa-util.c: access : MMAP_INTERLEAVED E: [alsa-source-ALC892 Analog] alsa-util.c: format : S32_LE E: [alsa-source-ALC892 Analog] alsa-util.c: subformat : STD E: [alsa-source-ALC892 Analog] alsa-util.c: channels : 2 E: [alsa-source-ALC892 Analog] alsa-util.c: rate : 48000 E: [alsa-source-ALC892 Analog] alsa-util.c: exact rate : 48000 (48000/1) E: [alsa-source-ALC892 Analog] alsa-util.c: msbits : 32 E: [alsa-source-ALC892 Analog] alsa-util.c: buffer_size : 16384 E: [alsa-source-ALC892 Analog] alsa-util.c: period_size : 1024 E: [alsa-source-ALC892 Analog] alsa-util.c: period_time : 21333 E: [alsa-source-ALC892 Analog] alsa-util.c: tstamp_mode : ENABLE E: [alsa-source-ALC892 Analog] alsa-util.c: period_step : 1 E: [alsa-source-ALC892 Analog] alsa-util.c: avail_min : 1024 E: [alsa-source-ALC892 Analog] alsa-util.c: period_event : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: start_threshold : 1 E: [alsa-source-ALC892 Analog] alsa-util.c: stop_threshold : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_threshold: 0 E: [alsa-source-ALC892 Analog] alsa-util.c: silence_size : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: boundary : 4611686018427387904 E: [alsa-source-ALC892 Analog] alsa-util.c: appl_ptr : 0 E: [alsa-source-ALC892 Analog] alsa-util.c: hw_ptr : 8236 So I am reporting this in the hope that someone understands what the problem is and knows how to fix it. The wiki said to include the output of aadebug with a bug report, so here's that: ALSA Audio Debug v0.2.0 - Sat Jun 27 17:26:44 AEST 2015 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/agpl-3.0.txt Kernel ---------------------------------------------------- Linux korath.teln.shikadi.net 4.0.6-1-ARCH #1 SMP PREEMPT Tue Jun 23 14:25:08 CEST 2015 x86_64 GNU/Linux Advanced Linux Sound Architecture Driver Version k4.0.6-1-ARCH. Loaded Modules -------------------------------------------- snd_hrtimer 16384 1 snd_seq 61440 1 snd_seq_device 16384 1 snd_seq snd_hda_codec_hdmi 53248 1 snd_hda_codec_realtek 65536 1 snd_hda_codec_generic 65536 1 snd_hda_codec_realtek snd_hda_intel 28672 9 snd_hda_controller 28672 1 snd_hda_intel snd_hda_codec 114688 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller snd_hwdep 16384 1 snd_hda_codec snd_pcm 90112 6 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller snd_timer 28672 5 snd_hrtimer,snd_pcm,snd_seq snd 69632 25 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec_generic,snd_hda_codec,snd_hda_intel,snd_seq_device Proc Asound ----------------------------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7c34000 irq 30 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c30000 irq 28 1: : sequencer 2: [ 1] : control 3: [ 1- 0]: digital audio playback 4: [ 1- 0]: digital audio capture 5: [ 1- 1]: digital audio playback 6: [ 1- 2]: digital audio capture 7: [ 1- 0]: hardware dependent 8: [ 0] : control 9: [ 0- 3]: digital audio playback 10: [ 0- 7]: digital audio playback 11: [ 0- 0]: hardware dependent 33: : timer 01-00: HDA Codec 0 00-00: HDA Codec 0 00-03: HDMI 0 : HDMI 0 : playback 1 00-07: HDMI 1 : HDMI 1 : playback 1 01-00: ALC892 Analog : ALC892 Analog : playback 1 : capture 1 01-01: ALC892 Digital : ALC892 Digital : playback 1 01-02: ALC892 Alt Analog : ALC892 Alt Analog : capture 1 Client info cur clients : 2 peak clients : 2 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 128 : "TiMidity" [User] Port 0 : "TiMidity port 0" (-We-) Port 1 : "TiMidity port 1" (-We-) Port 2 : "TiMidity port 2" (-We-) Port 3 : "TiMidity port 3" (-We-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Input pool : Pool size : 1000 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Dev Snd --------------------------------------------------- total 0 drwxr-xr-x 2 root root 80 Jun 27 13:31 by-path crw-rw----+ 1 root audio 116, 8 Jun 27 13:31 controlC0 crw-rw----+ 1 root audio 116, 2 Jun 27 13:31 controlC1 crw-rw----+ 1 root audio 116, 11 Jun 27 13:31 hwC0D0 crw-rw----+ 1 root audio 116, 7 Jun 27 13:31 hwC1D0 crw-rw----+ 1 root audio 116, 9 Jun 27 17:12 pcmC0D3p crw-rw----+ 1 root audio 116, 10 Jun 27 17:12 pcmC0D7p crw-rw----+ 1 root audio 116, 4 Jun 27 17:12 pcmC1D0c crw-rw----+ 1 root audio 116, 3 Jun 27 17:13 pcmC1D0p crw-rw----+ 1 root audio 116, 5 Jun 27 13:31 pcmC1D1p crw-rw----+ 1 root audio 116, 6 Jun 27 13:31 pcmC1D2c crw-rw----+ 1 root audio 116, 1 Jun 27 13:31 seq crw-rw----+ 1 root audio 116, 33 Jun 27 13:31 timer CPU ------------------------------------------------------- model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3097.500 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3000.000 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3295.664 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3120.703 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3001.406 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3000.351 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3645.585 model name : Intel(R) Core(TM) i7-4770K CPU @ 3.50GHz cpu MHz : 3000.468 RAM ------------------------------------------------------- MemTotal: 16362900 kB SwapTotal: 0 kB Hardware -------------------------------------------------- 00:02.0 VGA compatible controller: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller (rev 06) 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 04) Interupts ------------------------------------------------- CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 0: 19 0 0 0 0 0 0 0 IO-APIC-edge timer 1: 1 0 0 0 0 0 0 1 IO-APIC-edge i8042 7: 9 0 0 0 0 0 0 0 IO-APIC-edge 8: 0 0 0 1 0 0 0 0 IO-APIC-edge rtc0 9: 1 0 0 0 1 0 0 1 IO-APIC-fasteoi acpi 16: 20 0 0 0 4 1 3 1 IO-APIC 16-fasteoi ehci_hcd:usb1 23: 23 0 5 0 2 1 1 1 IO-APIC 23-fasteoi ehci_hcd:usb2 24: 236257 13391 10344 8635 7919 6578 6237 5429 PCI-MSI-edge xhci_hcd 25: 27200 5286 3884 3536 10293 3395 2433 1864 PCI-MSI-edge 0000:00:1f.2 26: 3 6 0 0 0 2 0 2 PCI-MSI-edge mei_me 27: 231790 38059 33089 33480 38333 18853 15356 13578 PCI-MSI-edge eth0 28: 549701 20320 18918 16545 17439 14054 12933 11245 PCI-MSI-edge snd_hda_intel 29: 1936 799 524 368 520 541 356 265 PCI-MSI-edge i915 30: 25 272 13 2 477 200 8 2 PCI-MSI-edge snd_hda_intel NMI: 15 14 14 14 9 7 9 10 Non-maskable interrupts LOC: 1051971 756503 703396 726403 198333 221752 198703 213191 Local timer interrupts SPU: 0 0 0 0 0 0 0 0 Spurious interrupts PMI: 15 14 14 14 9 7 9 10 Performance monitoring interrupts IWI: 0 0 0 0 0 0 1 0 IRQ work interrupts RTR: 7 0 0 0 0 0 0 0 APIC ICR read retries RES: 13573 10138 7672 8132 3954 3624 3500 2935 Rescheduling interrupts CAL: 32694 13897 13827 14113 968 1025 1057 1107 Function call interrupts TLB: 7503 7559 9116 13064 9259 6773 5330 4805 TLB shootdowns TRM: 0 0 0 0 0 0 0 0 Thermal event interrupts THR: 0 0 0 0 0 0 0 0 Threshold APIC interrupts MCE: 0 0 0 0 0 0 0 0 Machine check exceptions MCP: 48 48 48 48 48 48 48 48 Machine check polls HYP: 0 0 0 0 0 0 0 0 Hypervisor callback interrupts ERR: 9 MIS: 0 Please let me know if I can provide any other info. Many thanks, Adam.

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[alsa-devel] [PATCH] ALSA: hda - Add headset support to Acer Aspire V5
by Takashi Iwai 27 Jun '15

27 Jun '15

Acer Aspire V5 with ALC282 codec needs the similar quirk like Dell laptops to support the headset mic. The headset mic pin is 0x19 and it's not exposed by BIOS, thus we need to fix the pincfg as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96201 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 78dfec9d34c8..cab09c42598b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4484,6 +4484,7 @@ enum { ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, + ALC269_FIXUP_ASPIRE_HEADSET_MIC, ALC269_FIXUP_ASUS_X101_FUNC, ALC269_FIXUP_ASUS_X101_VERB, ALC269_FIXUP_ASUS_X101, @@ -4761,6 +4762,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_no_hp_mic, }, + [ALC269_FIXUP_ASPIRE_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* headset mic w/o jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC286_FIXUP_SONY_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5096,6 +5106,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), + SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), -- 2.4.4

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[alsa-devel] Sennheiser HDVD 800 Support
by Armin Ranjbar 26 Jun '15

26 Jun '15

Dear everyone, i have noticed that alas doesn't support HDVD-800 USB DAC, i can not hear anything. note that this device requires it's own driver under windows/mac. this is my alsa-info output: http://www.alsa-project.org/db/?f=fe56e65add73248baf5307a04b6b9deb2b236004 output of pasuspender -- aplay -Dplughw:2 some_file.wav: ALSA lib pcm_hw.c:1667:(_snd_pcm_hw_open) Invalid value for card main:722: audio open error: No such file or directory output of aplay -v -fdat --dump-hw-params -d1 -Dhw:S20 /dev/zero: http://pastebin.com/p2Daxza6 speaker-test -c2 -twav -Dplughw:S20 starts OK, but i can't hear anything/ product page: http://en-us.sennheiser.com/headphone-audio-amplification-hdvd-800-acoustic I'm open to help if you need anything to be done. --- Armin ranjbar

1 1

[alsa-devel] Audio Driver Works In Ubuntu but Crash in Debian After 2nd Run
by Azizul Hakim 26 Jun '15

26 Jun '15

Dear altruists, I'm writing a sound card driver based on ALSA API. My driver works perfectly on Ubuntu 12.04, 13.10 and 14.04. There is no problem on how many times I play a sound file using my driver. Everything works great. But same driver has some issue on debian distribution. I've been trying to test it on BeagleBone and during the first run the driver works pretty well. But as soon as I play a second sound file, it crash. I thought may be I did something wrong in releasing resources after 1st run. But so far from my debugging it looks like there is no problem with resource release. Also same driver works on Ubuntu as many times as I test. I even tested with the same sound file. On debian, I get the following message is kernel log- Apr 23 20:26:41 beaglebone kernel: [ 183.631905] ALSA sound/core/pcm_lib.c:1910 playback write error (DMA or IRQ trouble?) It would be highly appreciated if I can get some suggestions on which direction I should try to resolve this issue. Is there any known issue with debian distribution or anyone faced similar kind of problem before? Thanks Azizul Hakim

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[alsa-devel] Using plughw device and samplerate conversion with an ALSA driver not supporting memory mapped access
by Christian Gruber 26 Jun '15

26 Jun '15

Hello, I have problems using the plughw device with samplerate conversion on an embedded board having an ALSA driver, which does not support memory mapped access. The audio codec only supports 48 kHz sampling rate. The problem occurs, when I want to playback for instance a 16 kHz file via plughw device, which is the default device. Interestingly the behaviour of the ALSA system is as follows: 1. aplay -D default audio_16kHz.wav does not work 2. aplay -D default -M audio_16_kHz.wav works correctly This is quite strange, since although the ALSA driver does not support memory mapped access, using the option -M works, but the default read write access method without option -M does not, although the driver supports read write access. In this case the function snd_pcm_hw_params() returns with an error (EINVAL). I started debugging this and could see, that in the second case, when using option -M, the plughw device establishes the following signal processing chain: plughw -> rate -> mmap_emul -> hw This is what I expected from the plughw device. It first introduces a samplerate converter for conversion from 16 kHz to 48 kHz, since the codec only supports 48 kHz. And second it introduces a memory mapped access emulator to be able to use memory mapped access method on the application side and read write access method on the driver side (hw device). In the first case without using option -M the plughw device establishes the following processing chain: plughw -> rate -> hw This means, it does not introduce a memory mapped access emulator, which is comprehensible as well, since the application already uses read write access method, the same as the ALSA driver does. But nevertheless an error occurs. Further debugging showed, that the error occurs in the function hw_refine_call() (pcm_hw.c) during the call to _snd_pcm_hw_params_internal() within the function snd_pcm_plug_hw_params() (pcm_plug.c). For some reason the rate plugin wants to use memory mapped access only on the hw device. So I ask myself, if this is the expected behaviour of the rate plugin. Does the rate plugin only support memory mapped access? If yes, why is no mmap_emul plugin introduced into the signal processing chain in this case? If not, why does the rate plugin choose memory mapped access method? Is the ALSA driver maybe buggy giving wrong information about its hw capabilities? Can anybody help me? Regards, Christian

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix deadlock on auto-disable mux controls
by Charles Keepax 26 Jun '15

26 Jun '15

The commit 02aa78abec6e ("ASoC: DAPM: Add APIs to create individual DAPM controls.") added locking to the snd_soc_dapm_new_control function but did not update the call to snd_soc_dapm_new_control in the auto-disable mux code, this appears to be because the patches were sent at fairly similar times. This patch change the call in the auto-disable mux code to use the new snd_soc_dapm_new_control_unlocked function instead. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..a47a8ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -389,8 +389,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->value = template.on_val; - data->widget = snd_soc_dapm_new_control(widget->dapm, - &template); + data->widget = snd_soc_dapm_new_control_unlocked( + widget->dapm, &template); if (!data->widget) { ret = -ENOMEM; goto err_name; -- 1.7.2.5

2 1

[alsa-devel] DPCM: BE to BE cpu DAIs question‏
by srv_kaichieh.chaung 26 Jun '15

26 Jun '15

Hi, Our platform use ASoC structure. Currently, we want to refactor our driver to use DPCM structure. I followed the documentation of DPCM, but have some trouble implementing hostless BE to BE dai links. My question is: 1. Is there any code example for this(hostless BE to BE)? Since I didn't see any existing code on linux kernel. 2. From documentation, i have to assgin a snd_soc_pcm_stream for BE dais. It seems that the BE<->BE dai links can only support one spec (format/rates/ch). Or am i getting it wrong, does it support dynamically change of the spec? Thanks in advanced! -------------------------------------------------------------------------------------- I have tried some implementations, but failed. My linux version is 3.18. Illustration, PCM -----------------> BE1 ↑ -------< BE2 Code snippet, static const struct snd_soc_dapm_route machine_routes[] = { {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, FE_STNAME}, {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, BE2_CAPTURE_CPU_DAI_STNAME}, }; static const struct snd_soc_pcm_stream dai_params = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, }; static struct snd_soc_dai_link mt_soc_dai_common[] = { { .name = FE_PCM, .stream_name = FE_STNAME, .cpu_dai_name = FE_CPUDAINAME, // cannot use dummy!!!! .platform_name = FE_PCMNAME, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .init = mt_soc_audio_init3, .ops = &kc_snd_soc_ops, .dynamic = 1, .dpcm_playback = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, /* BE DAIs */ { .name = "BE2", .cpu_dai_name = BE2_PLAYBACK_CPU_DAI, //"snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, { .name = BE1_PLAYBACK, .cpu_dai_name = BE1_PLAYBACK_CPU_DAI, .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, 1. when only routes PCM-->BE1, If i add .params in BE1 soc dai link, when i start PCM, it will crash. and get "Unable to handle kernel NULL pointer dereference at virtual address 000000d0" Seems to crash in dpcm_be_dai_startup() If i removed .params, it will not crash, and play normally. 2. when only with BE2--> BE1, BE2 is never trigger, where did i get wrong? Thanks. ************* Email Confidentiality Notice ******************** The information contained in this e-mail message (including any attachments) may be confidential, proprietary, privileged, or otherwise exempt from disclosure under applicable laws. It is intended to be conveyed only to the designated recipient(s). Any use, dissemination, distribution, printing, retaining or copying of this e-mail (including its attachments) by unintended recipient(s) is strictly prohibited and may be unlawful. If you are not an intended recipient of this e-mail, or believe that you have received this e-mail in error, please notify the sender immediately (by replying to this e-mail), delete any and all copies of this e-mail (including any attachments) from your system, and do not disclose the content of this e-mail to any other person. Thank you!

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Re: [alsa-devel] ALSA: hdac_ext: add hdac extended controller
by Dan Carpenter 26 Jun '15

26 Jun '15

Hello Jeeja KP, The patch 0b00a5615dc4: "ALSA: hdac_ext: add hdac extended controller" from Jun 11, 2015, leads to the following static checker warning: sound/hda/ext/hdac_ext_controller.c:47 snd_hdac_ext_bus_parse_capabilities() warn: unsigned 'offset' is never less than zero. sound/hda/ext/hdac_ext_controller.c:54 snd_hdac_ext_bus_parse_capabilities() warn: unsigned 'cur_cap' is never less than zero. sound/hda/ext/hdac_ext_controller.c 38 int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *ebus) 39 { 40 unsigned int cur_cap; 41 unsigned int offset; 42 struct hdac_bus *bus = &ebus->bus; 43 unsigned int counter = 0; 44 45 offset = snd_hdac_chip_readl(bus, LLCH); 46 47 if (offset < 0) ^^^^^^^^^^ 48 return -EIO; 49 50 /* Lets walk the linked capabilities list */ 51 do { 52 cur_cap = _snd_hdac_chip_read(l, bus, offset); 53 54 if (cur_cap < 0) ^^^^^^^^^^^ 55 return -EIO; regards, dan carpenter

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Re: [alsa-devel] Need help with Behringer Firepower FCA610 profile
by Raymond Yau 26 Jun '15

26 Jun '15

>> > >> > I'd prefer a multi-purpose configuration with multiple >> sinks/sources: >> > >> > - Source 'DAW': aux1,aux2,aux3,aux4,aux5,aux6 >> > - Source 'Mic/Instrument': channel 0/1 = left/right >> > - Source 'Line': channel 2/3 = left/right >> > - Source 'S/PDIF' channel 4/5 = left/right >> > - Source 'TV' channel 4/5 = left/right >> (=S/PDIF) >> > >> > - Sink 'DAW': >> aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9,aux10 >> > - Sink 'Stereo1/Headphone A': channel 0/1 = left/right >> > - Sink 'Stereo2/Headphone B': channel 2/3 = left/right >> > - Sink 'Stereo3': channel 4/5 = left/right >> > - Sink 'Stereo4': channel 6/7 = left/right >> > - Sink 'Stereo5/SPDIF': channel 8/9 = left/right >> > - Sink 'Surround 7.1': front-left, front-right, >> rear-left, rear-right, center, lfe, side-left, side-right >> > - Sink 'Surround 7.1 -> 5.0': front-left, front-right, >> rear-left, rear-right, center, lfe -> front-left + front-right, >> side-left -> front-left + rear-left, side-right -> front-right + >> rear-right >> >> http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/modules/alsa/mi… >> [1] >> >> >> Pulseaudio only support different device >> >> If your driver only have simgle device for 10 channel playback and 6 >> channel capture > > > ALSA provides the following hardware-devices/plugs: > > - sysdefault:CARD=FCA610 > - front:CARD=FCA610,DEV=0 > - surround21:CARD=FCA610,DEV=0 > - surround40:CARD=FCA610,DEV=0 > - surround41:CARD=FCA610,DEV=0 > - surround50:CARD=FCA610,DEV=0 > - surround51:CARD=FCA610,DEV=0 > - surround71:CARD=FCA610,DEV=0 > - iec958:CARD=FCA610,DEV=0 > - dmix:CARD=FCA610,DEV=0 > - dsnoop:CARD=FCA610,DEV=0 > - hw:CARD=FCA610,DEV=0 > - plughw:CARD=FCA610,DEV=0 Are you sure that those devices really work ?

1 0

[alsa-devel] [PATCH] ALSA: hda - restore the MIC FIXUP for some Dell machines
by Hui Wang 26 Jun '15

26 Jun '15

Those FIXUPs were applied to the machines through pin quirks, but recently the PCI_QUIRK makes them can't apply to the machines. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8e02cdf..78dfec9d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4522,6 +4522,8 @@ enum { ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13_GPIO6, + ALC288_FIXUP_DELL_XPS_13, + ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -5053,9 +5055,23 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC288_FIXUP_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC288_FIXUP_DELL_XPS_13_GPIO6 + }, + [ALC288_FIXUP_DELL_XPS_13] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_dell_xps13, + .chained = true, + .chain_id = ALC288_FIXUP_DISABLE_AAMIX + }, [ALC292_FIXUP_DISABLE_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE }, [ALC292_FIXUP_DELL_E7X] = { .type = HDA_FIXUP_FUNC, @@ -5096,7 +5112,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC292_FIXUP_DELL_E7X), + SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: jack: Fix endless loop at unique index detection
by Takashi Iwai 26 Jun '15

26 Jun '15

While the commit [d0a601c278de: ALSA: jack: Fix the id uniqueness check] fixes the wrong string check, it leads to a worse result -- the loop in get_available_index() goes into an endless loop. The cause is that snd_ctl_find_id() returns the object assigned to the numid if it's set. Thus it points to the previous entry again. This patch clears the numid field for the next call properly. Reported-and-tested-by: Tomáš Pružina <pruzinat(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/ctljack.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c index 9149a4aefa95..84a3cd683068 100644 --- a/sound/core/ctljack.c +++ b/sound/core/ctljack.c @@ -41,8 +41,11 @@ static int get_available_index(struct snd_card *card, const char *name) sid.iface = SNDRV_CTL_ELEM_IFACE_CARD; strlcpy(sid.name, name, sizeof(sid.name)); - while (snd_ctl_find_id(card, &sid)) + while (snd_ctl_find_id(card, &sid)) { sid.index++; + /* reset numid; otherwise snd_ctl_find_id() hits this again */ + sid.numid = 0; + } return sid.index; } -- 2.4.4

1 0

[alsa-devel] DPCM: BE to BE cpu DAIs question‏
by srv_kaichieh.chaung 26 Jun '15

26 Jun '15

Hi, Our platform use ASoC structure. Currently, we want to refactor our driver to use DPCM structure. I followed the documentation of DPCM, but have some trouble implementing hostless BE to BE dai links. My question is: 1. Is there any code example for this(hostless BE to BE)? Since I didn't see any existing code on linux kernel. 2. From documentation, i have to assgin a snd_soc_pcm_stream for BE dais. It seems that the BE<->BE dai links can only support one spec (format/rates/ch). Or am i getting it wrong, does it support dynamically change of the spec? Thanks in advanced! -------------------------------------------------------------------------------------- I have tried some implementations, but failed. My linux version is 3.18. Illustration, PCM -----------------> BE1 ↑ -------< BE2 Code snippet, static const struct snd_soc_dapm_route machine_routes[] = { {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, FE_STNAME}, {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, BE2_CAPTURE_CPU_DAI_STNAME}, }; static const struct snd_soc_pcm_stream dai_params = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, }; static struct snd_soc_dai_link mt_soc_dai_common[] = { { .name = FE_PCM, .stream_name = FE_STNAME, .cpu_dai_name = FE_CPUDAINAME, // cannot use dummy!!!! .platform_name = FE_PCMNAME, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .init = mt_soc_audio_init3, .ops = &kc_snd_soc_ops, .dynamic = 1, .dpcm_playback = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, /* BE DAIs */ { .name = "BE2", .cpu_dai_name = BE2_PLAYBACK_CPU_DAI, //"snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, { .name = BE1_PLAYBACK, .cpu_dai_name = BE1_PLAYBACK_CPU_DAI, .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, 1. when only routes PCM-->BE1, If i add .params in BE1 soc dai link, when i start PCM, it will crash. and get "Unable to handle kernel NULL pointer dereference at virtual address 000000d0" Seems to crash in dpcm_be_dai_startup() If i removed .params, it will not crash, and play normally. 2. when only with BE2--> BE1, BE2 is never trigger, where did i get wrong? Thanks. ************* Email Confidentiality Notice ******************** The information contained in this e-mail message (including any attachments) may be confidential, proprietary, privileged, or otherwise exempt from disclosure under applicable laws. It is intended to be conveyed only to the designated recipient(s). Any use, dissemination, distribution, printing, retaining or copying of this e-mail (including its attachments) by unintended recipient(s) is strictly prohibited and may be unlawful. If you are not an intended recipient of this e-mail, or believe that you have received this e-mail in error, please notify the sender immediately (by replying to this e-mail), delete any and all copies of this e-mail (including any attachments) from your system, and do not disclose the content of this e-mail to any other person. Thank you!

1 0

[alsa-devel] DPCM: BE to BE cpu DAIs question
by Chuang Eric 26 Jun '15

26 Jun '15

Hi, Our platform use ASoC structure. Currently, we want to refactor our driver to use DPCM structure.I followed the documentation of DPCM, but have some trouble implementing hostless BE to BE dai links. My question is:1. Is there any code example for this(hostless BE to BE)?Since I didn't see any existing code on linux kernel.2. From documentation, i have to assgin a snd_soc_pcm_stream for BE dais. It seems that the BE<->BE dai links can only support one spec (format/rates/ch). Or am i getting it wrong, does it support dynamically change of the spec? Thanks in advanced! -------------------------------------------------------------------------------------- I have tried some implementations, but failed. My linux version is 3.18. Illustration,PCM -----------------> BE1 ↑ -------< BE2 Code snippet, static const struct snd_soc_dapm_route machine_routes[] = { {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, FE_STNAME}, {BE1_PLAYBACK_CPU_DAI_STNAME, NULL, BE2_CAPTURE_CPU_DAI_STNAME},}; static const struct snd_soc_pcm_stream dai_params = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2,}; static struct snd_soc_dai_link mt_soc_dai_common[] = { { .name = FE_PCM, .stream_name = FE_STNAME, .cpu_dai_name = FE_CPUDAINAME, // cannot use dummy!!!! .platform_name = FE_PCMNAME, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .init = mt_soc_audio_init3, .ops = &kc_snd_soc_ops, .dynamic = 1, .dpcm_playback = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, }, /* BE DAIs */ { .name = "BE2", .cpu_dai_name = BE2_PLAYBACK_CPU_DAI, //"snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, { .name = BE1_PLAYBACK, .cpu_dai_name = BE1_PLAYBACK_CPU_DAI, .platform_name = "snd-soc-dummy", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ignore_suspend = 1, .ignore_pmdown_time = 1, .params = &dai_params, }, 1. when only routes PCM-->BE1, If i add .params in BE1 soc dai link, when i start PCM, it will crash. and get "Unable to handle kernel NULL pointer dereference at virtual address 000000d0" Seems to crash in dpcm_be_dai_startup() If i removed .params, it will not crash, and play normally. 2. when only with BE2--> BE1, BE2 is never trigger, where did i get wrong? Thanks.

1 0

[alsa-devel] [PATCH 1/2] ASoC: gtm601: Fix modalias
by Axel Lin 25 Jun '15

25 Jun '15

The MODULE_ALIAS needs to match the driver name to make module auto-loading work. Also move MODULE_ALIAS close to other MODULE_* macro. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/gtm601.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/gtm601.c b/sound/soc/codecs/gtm601.c index 1b77ff2..a696e3e 100644 --- a/sound/soc/codecs/gtm601.c +++ b/sound/soc/codecs/gtm601.c @@ -70,8 +70,6 @@ static int gtm601_platform_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:gtm601_codec_audio"); - #if defined(CONFIG_OF) static const struct of_device_id gtm601_codec_of_match[] = { { .compatible = "option,gtm601", }, @@ -96,3 +94,4 @@ module_platform_driver(gtm601_codec_driver); MODULE_DESCRIPTION("ASoC gtm601 driver"); MODULE_AUTHOR("Marek Belisko <marek(a)goldelico.com>"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:gtm601"); -- 2.1.0

2 6

[alsa-devel] [PATCH] surround41/50.conf: Use chmap syntax for better flexibility
by David Henningsson 25 Jun '15

25 Jun '15

In case the hardware only supports a specific channel map, this change would allow surround41/50 to select the correct channel map and channel count in this situation. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/pcm/surround41.conf | 11 +++++------ src/conf/pcm/surround50.conf | 11 +++++------ 2 files changed, 10 insertions(+), 12 deletions(-) diff --git a/src/conf/pcm/surround41.conf b/src/conf/pcm/surround41.conf index 10e486e..2f82381 100644 --- a/src/conf/pcm/surround41.conf +++ b/src/conf/pcm/surround41.conf @@ -53,12 +53,11 @@ pcm.!surround41 { ] } } - slave.channels 6 - ttable.0.0 1 - ttable.1.1 1 - ttable.2.2 1 - ttable.3.3 1 - ttable.4.5 1 + ttable.0.FL 1 + ttable.1.FR 1 + ttable.2.RL 1 + ttable.3.RR 1 + ttable.4.LFE 1 hint { description "4.1 Surround output to Front, Rear and Subwoofer speakers" device $DEV diff --git a/src/conf/pcm/surround50.conf b/src/conf/pcm/surround50.conf index 7b7b17e..dc95c17 100644 --- a/src/conf/pcm/surround50.conf +++ b/src/conf/pcm/surround50.conf @@ -53,12 +53,11 @@ pcm.!surround50 { ] } } - slave.channels 6 - ttable.0.0 1 - ttable.1.1 1 - ttable.2.2 1 - ttable.3.3 1 - ttable.4.4 1 + ttable.0.FL 1 + ttable.1.FR 1 + ttable.2.RL 1 + ttable.3.RR 1 + ttable.4.FC 1 hint { description "5.0 Surround output to Front, Center and Rear speakers" device $DEV -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix Dock Headphone on Thinkpad X250 seen as a Line Out
by David Henningsson 25 Jun '15

25 Jun '15

Thinkpad X250, when attached to a dock, has two headphone outs but no line out. Make sure we don't try to turn this into one headphone and one line out (since that disables the headphone amp on the dock). Alsa-info at http://www.alsa-project.org/db/?f=36f8764e1d782397928feec715d0ef90dfddd4c1 Cc: stable(a)vger.kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 431a20b..8e02cdf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4511,6 +4511,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPT440_DOCK2, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4960,6 +4961,12 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HEADSET_MODE }, [ALC292_FIXUP_TPT440_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK2 + }, + [ALC292_FIXUP_TPT440_DOCK2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x16, 0x21211010 }, /* dock headphone */ -- 1.9.1

3 8

[alsa-devel] [PATCH] hda: set proper caps for newer AMD hda audio in KB/KV
by Alex Deucher 25 Jun '15

25 Jun '15

Fixes audio problems on newer asics. Noticed by: Kelly Anderson <kelly(a)xilka.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a8a1e14..479ee25 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2056,6 +2056,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* ATI HDMI */ + { PCI_DEVICE(0x1002, 0x1308), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), @@ -2064,6 +2066,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, + { PCI_DEVICE(0x1002, 0x9840), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0xaa08), -- 1.8.3.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Disable widget power-save for VIA codecs
by Takashi Iwai 25 Jun '15

25 Jun '15

The widget power-save that was enabled in 4.1 kernel seems resulting in the silent output on VIA codecs by some reason. Some widgets get wrong power states. As a quick fix, turn this flag off while keeping power_down_unused flag. This will bring back to the state of 4.0.x. Fixes: 688b12cc3ca8 ('ALSA: hda - Use the new power control for VIA codecs') Reported-and-tested-by: Harald Dunkel <harri(a)afaics.de> Cc: <stable(a)vger.kernel.org> # v4.1 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_via.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0521be8d46a8..da5366405eda 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -241,7 +241,9 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = codec->power_save_node; + struct via_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->gen.power_down_unused; return 0; } @@ -252,9 +254,9 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; bool val = !!ucontrol->value.enumerated.item[0]; - if (val == codec->power_save_node) + if (val == spec->gen.power_down_unused) return 0; - codec->power_save_node = val; + /* codec->power_save_node = val; */ /* widget PM seems yet broken */ spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; -- 2.4.4

1 0

[alsa-devel] [PATCH] Revert "ALSA: hda - Add mute-LED mode control to Thinkpad"
by Takashi Iwai 24 Jun '15

24 Jun '15

This reverts commit 7290006d8c0900c56d8c58428134f02c35109d17. Through the regression report, it was revealed that the tpacpi_led_set() call to thinkpad_acpi helper doesn't only toggle the mute LED but actually mutes the sound. This is contradiction to the expectation, and rather confuses user. According to Henrique, it's not trivial to judge which TP model behaves "LED-only" and which model does whatever more intrusive, as Lenovo's implementations vary model by model. So, from the safety reason, we should revert the patch for now. Reported-by: Martin Steigerwald <martin(a)lichtvoll.de> Cc: Pali Rohár <pali.rohar(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/thinkpad_helper.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index d51703e30523..0a4ad5feb82e 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -72,7 +72,6 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; - spec->vmaster_mute_enum = 1; removefunc = false; } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { -- 2.4.1

3 3

[alsa-devel] [PATCH 1/1] SPI : spi-pxa2xx : fix spi init of WM510205 codec via ACPI (resend)
by Christian Hartmann 24 Jun '15

24 Jun '15

Hello, I have one patch that enables the spi initialization of the WM510205 audio codec Without the patch, this message indicates that the codec chip fails with the current chipselect number in spi-pxa2xx: [ 0.296256] pxa2xx-spi 80860F0E:00: cs1 >= max 1 [ 0.296270] spi_master spi32766: failed to add SPI device WM510205:00 from ACPI So raise num_chipselect by one to enable the codec. The WM510202 is on chipselect = 1. With the patch applied I got in dmesg now: [ 6.056829] pxa2xx-spi 80860F0E:00: no DMA channels available, using PIO [ 6.067035] pxa2xx-spi 80860F0E:00: registered master spi32766 (dynamic) [ 6.067214] spi spi-WM510205:00: 8333333 Hz actual, PIO [ 6.067221] spi spi-WM510205:00: setup mode 0, 8 bits/w, 8000000 Hz max --> 0 [ 6.067287] pxa2xx-spi 80860F0E:00: registered child spi-WM510205:00 Signed-off-by: Christian Hartmann <cornogle(a)googlemail.com> --- drivers/spi/spi-pxa2xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c index e3223ac..7b54b04 100644 --- a/drivers/spi/spi-pxa2xx.c +++ b/drivers/spi/spi-pxa2xx.c @@ -1279,7 +1279,7 @@ pxa2xx_spi_acpi_get_pdata(struct platform_device *pdev) if (adev->pnp.unique_id && !kstrtoint(adev->pnp.unique_id, 0, &devid)) ssp->port_id = devid; - pdata->num_chipselect = 1; + pdata->num_chipselect = 2; pdata->enable_dma = true; return pdata; -- 1.8.3.1 would be nice to hear some feedback or comments about this patch. Notice: That is my first attempt to make the codec chip via ACPI (DSDT) configuration working on Linux systems and some other patches are needed to let the codec work together with the intel_sst_acpi driver under baytrail systems. Currently I am trying to write the also needed machine driver which binds to this codec chip. Also the arizona-core / arizona-spi drivers have to be patched, but that patches will be landing later... cheers Christian Hartmann

2 1

[alsa-devel] [PATCH] cplay: check return value of fread() function and compare it with "number of blocks to read" rather than "size of each block".
by Manish Sogi 24 Jun '15

24 Jun '15

fread() function on success returns the number of blocks which it reads, so return value should be compared with "1" because only 1 block having size equal to sizeof(header) needs to be read. On success fread will return 1 else error case. Signed-off-by: Manish Sogi <manish.sogi(a)samsung.com> --- src/utils/cplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/utils/cplay.c b/src/utils/cplay.c index 2d1a176..aeea5bb 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -218,7 +218,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, } read = fread(&header, sizeof(header), 1, file); - if (read != sizeof(header)) { + if (read != 1) { fprintf(stderr, "Unable to read header \n"); fclose(file); exit(EXIT_FAILURE); -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: mediatek: Fix unbalanced calls to runtime suspend/resume
by Koro Chen 23 Jun '15

23 Jun '15

This adds call to runtime suspend in dev remove. It fixs the problem that suspend is not called in the case of CONFIG_PM=n. It also fixs build warning when CONFIG_PM=n. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index cc228db..9863da7 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -1199,6 +1199,8 @@ err_pm_disable: static int mtk_afe_pcm_dev_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + mtk_afe_runtime_suspend(&pdev->dev); snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); return 0; -- 1.8.1.1.dirty

2 1

[alsa-devel] [PATCH 0/6] ASoC: Intel: Sklyake - add IPC routines
by Vinod Koul 23 Jun '15

23 Jun '15

The SKL aDSP needs to send and receive IPCs to and from aDSP. To aid this we add IPC and aDSP management routines as a library for driver to use. For most of IPC handling we use the common IPC Intel code in place and add the messaging and message parsing bits on top. First 4 patches are against common IPC code and can be applied independently Last two add IPC handling and aDSP management routines, they are not dependent upon anything and can be applied independently but they are generated against SKL PCM driver series so Makefile updates expect those to be there Subhransu S. Prusty (6): ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Define SKL ADSP IPC and general purpose registers ASoC: Intel: Reorganize the common dsp structure ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL sound/soc/intel/common/sst-dsp-priv.h | 20 +- sound/soc/intel/common/sst-dsp.c | 51 +++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-sst-dsp.c | 338 +++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 124 ++++++ sound/soc/intel/skylake/skl-sst-ipc.c | 751 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 126 ++++++ 8 files changed, 1417 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h -- 1.9.1

1 6

[alsa-devel] ASoC updates for v4.2
by Mark Brown 23 Jun '15

23 Jun '15

The following changes since commit 0f57d86787d8b1076ea8f9cbdddda2a46d534a27: Linux 4.1-rc8 (2015-06-14 15:51:10 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.2-2 for you to fetch changes up to c99d49a8f81fb35e67b0ffa45f320a75e0b5639d: Merge remote-tracking branch 'asoc/topic/wm8995' into asoc-next (2015-06-22 10:24:37 +0100) ---------------------------------------------------------------- ASoC: Further updates for v4.2 There's a bunch of additional updates and fixes that came in since my orignal pull request here, including DT support for rt5645 and fairly large serieses of cleanups and improvements to tas2552 and rcar. ---------------------------------------------------------------- Alexander Stein (2): ASoC: sgtl5000: Use specific variable for lo_vag ASoC: sgtl5000: Calculate Lineout Channel Output Level Alexandre Belloni (4): ASoC: atmel: simplify Kconfig ASoC: atmel: compile pcm driver in snd-soc-atmel_ssc_dai ASoC: atmel: Revert previous fix for Kconfig limations ASoC: atmel: properly select dma driver state Anatol Pomozov (1): ASoC: rt5677: Add reset-gpio dts option Antonio Ospite (1): ASoC: adau1977: fix typo s/Substraction/Subtraction/ Arnaud Pouliquen (1): ASoC: simple card: Add mclk-fs property in dai-link Arnd Bergmann (3): ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency, again ASoC: qcom: remove incorrect dependencies ASoC: qcom: fix STORM board Kconfig Axel Lin (11): ASoC: da7210: Fix dependency to allow build with SND_SOC_I2C_AND_SPI ASoC: sta32x: Use devm_gpiod_get_optional at appropriate place ASoC: sta350: Use devm_gpiod_get_optional at appropriate place ASoC: wm8737: Fixup setting VMID Impedance control register ASoC: wm8903: Fix define for WM8903_VMID_RES_250K ASoC: wm8955: Fix setting wrong register for WM8955_K_8_0_MASK bits ASoC: max98925: Fix mask for setting DAI invert mode ASoC: tas2552: Use devm_gpiod_get_optional for enable-gpio ASoC: rt5645: Use devm_gpiod_get_optional for hp-detect ASoC: ml26124: Remove duplicate code ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case Bard Liao (10): ASoC: rt5645: Adds push button support for rt5650 ASoC: rt5677: add i2s asrc clk src selection ASoC: rt5645: fix implicit declaration error ASoC: rt5645: fix wrong mask for button report ASoC: rt5645: remove RT5645_I2S_BCLK_MS1 control ASoC: rt5645: fix PLL source register definitions ASoC: rt5645: add TDM slot control into dapm route ASoC: rt5645: make volume TLV closer to reality ASoC: rt5645: fix add missing widget ASoC: rt5645: remove unused field in pdata Charles Keepax (22): ASoC: wm_adsp: Split out adsp1 & 2 setup algorithms ASoC: wm_adsp: Improve variable naming ASoC: wm_adsp: Remove len field from wm_adsp_alg_region ASoC: wm_adsp: Limit firmware control name to ALSA control name size ASoC: wm_adsp: Move temporary control name to the stack ASoC: wm_adsp: Clean up low level control read/write functions ASoC: wm_adsp: Factor out creation of alg_regions ASoC: wm_adsp: Remove private field from wm_coeff_ctl ASoC: wm_adsp: Group all the ALSA control functions together ASoC: wm_adsp: Add basic support for rev 1 firmware file format ASoC: wm_adsp: Add support for rev 2 firmware file format ASoC: wm_adsp: Warn that firmware file format 0 is depreciated ASoC: wm_adsp: Use __leXX for little endian data ASoC: wm_adsp: Add support for DSP control flags ASoC: dapm: Remove local OOM error message ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro ASoC: dapm: Append "Autodisable" to autodisable widget names ASoC: dapm: Add support for autodisable mux controls ASoC: arizona: Use auto disable muxes for routing ASoC: dapm: Add missing mutex unlock ASoC: dapm: Break out of widget search when source and sink are located ASoC: dapm: Add cache to speed up adding of routes Dan Carpenter (2): ASoC: wm8741: check for error returns from wm8741_set_pdata() ASoC: imx-wm8962: Add a missing error check Dylan Reid (2): ASoC: Add jack types to dt-bindings ASoC: jack - add_gpiods accepts filled descriptors Fabio Estevam (8): ASoC: fsl_ssi: No need call of_device_is_available() ASoC: fsl: fsl_dma: Use true/false for boolean init ASoC: fsl: imx-mc13783: Simplify trivial if-return sequence ASoC: wm8996: Pass the IRQF_ONESHOT flag ASoC: wm5100: Pass the IRQF_ONESHOT flag ASoC: wm8994: Pass the IRQF_ONESHOT flag ASoC: twl6040: Pass the IRQF_ONESHOT flag ASoC: max98095: Pass the IRQF_ONESHOT flag Fang, Yang A (13): ASoC: Intel: Add support rt5650 in sst driver ASoC: rt5645: set platform data base on DMI ASoC: rt5645: fixed kbuild err ASoC: Intel: Add support max98090 in sst driver ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_max98090_ti ASoC: Intel: Support rt5650 codec for Cherrytrail & Braswell ASoC: Intel: Fixed kbuild warnings ASoC: Intel: Enabled button jack for BSW platform with rt5650 codec ASoC: max98090: read micbias from device property ASoC: ts3a227e: use device property api ASoC: Intel: Fixed incorrect sst pcm_delay ASoC: Intel: Added I2C dependency to two BSW machine drivers ASoC: Intel: fixed TI button detection Geert Uytterhoeven (2): ASoC: rsnd: Use generic names for device nodes ASoC: rsnd: Document r8a7778-specific binding Jie Yang (6): ASoC: Intel: remove unused function hsw_pcm_free_modules() ASoC: Intel: don't need compress offload for broadwell ASoC: intel: Revert "ASoC: Intel: remove unused function hsw_pcm_free_modules()" ASoC: intel: Revert "ASoC: Intel: fix broadwell module removing failed issue" ASoC: Intel: check and clear runtime module pointer ASoC: Intel: handle haswell pcm suspend including runtime modules freeing Jin Yao (1): ASoC: Intel: Remove invalid kfree of devm allocated data John Lin (5): ASoC: rt5645: improve headphone depop function ASoC: rt5645: remove unnecessary power in JD function ASoC: rt5645: fix IRQ error in jack detection ASoC: rt5645: fix jack type detect error ASoC: rt5645: fix kernel hang when call rt5645_set_jack_detect() Julian Scheel (1): ASoC: wm8523: Set bclk ratio for master mode Jun Nie (5): ASoC: simple-card: support platform in dts parse ASoC: zx: Add zx296702 SPDIF support ASoC: zx: Add ZTE zx296702 I2S DAI driver dt: Add documentation for the ZTE SPDIF controller dt: Add documentation for the ZTE I2S controller Jyri Sarha (5): ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule ASoC: davinci-mcasp: Channel count constraints for multi-serializer case ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule ASoC: core: If component doesn't have of_node use parent's node instead ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller Kevin Cernekee (4): ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver ASoC: tas571x: Eliminate redundant dev->of_node NULL check Koro Chen (3): ASoC: mediatek: Add AFE platform driver ASoC: mediatek: Add machine driver for MAX98090 codec ASoC: mediatek: Add machine driver for rt5650 rt5676 codec Krzysztof Kozlowski (4): ASoC: au1x: Constify platform_device_id ASoC: bt-sco: Constify platform_device_id ASoC: fsl: Constify platform_device_id ASoC: samsung: Constify platform_device_id Kuninori Morimoto (34): ASoC: rsnd: add rsnd_dai_to_priv() macro ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call ASoC: rsnd: tidyup SSI parent related function/macro names ASoC: rsnd: make sure SSI parent/child uses same number of sound channel. ASoC: rsnd: care snd_kcontrol's index ASoC: soc.h: tidyup struct snd_soc_dai_link definition order ASoC: rsnd: revert lock for calls to rsnd_dai_call ASoC: rsnd: indicate unknown HW start ASoC: rsnd: add rsnd_dai_stream_quit() ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM dmaengine: rcar-dmac: fixup spinlock in rcar-dmac ASoC: rsnd: spin lock for interrupt handler ASoC: rsnd: don't care under/over run error when PIO ASoC: rsnd: don't call snd_pcm_period_elapsed() under spin lock ASoC: rsrc-card: used "fe.xxx"/"be.xxx" name for dai_link ASoC: rsrc-card: tidyup priv->snd_card setup timing ASoC: rsrc-card: enable multi cpu/codec for DPCM ASoC: rsrc-card: remove unused name ASoC: rsnd-card: tidyup rsrc_priv_to_() ASoC: rsrc-card: move rsrc_card_parse_daifmt() to upper side ASoC: rsrc-card: tidyup return value of clock error ASoC: rsrc-card: cleanup for DPCM ASoC: rsnd: count each mod (SSI/SRC/DVC) ASoC: rsnd: rsnd_mod has rsnd_priv ASoC: rsnd: adds struct rsnd_dai_stream as on each fuction as parameter ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working() ASoC: rsnd: remove io from rsnd_mod Lars-Peter Clausen (115): ASoC: at91sam9g20ek: Automatically disconnect non-connected pins ASoC: dmaengine_pcm: Make FLAG_NO_RESIDUE internal ASoC: Add helper function getting CODEC's DAPM context ASoC: Add helper functions bias level management ASoC: Route all bias level updates through the core ASoC: Move bias level update to the core ASoC: samsung: wolfson: Improve compile test coverage ASoC: lowland: Use card DAPM context to access widgets ASoC: smdk_wm8994: Use card DAPM context to access widgets ASoC: smartq: Remove unnecessary snd_soc_dapm_disable_pin() ASoC: smartq: Use card DAPM context to access widgets ASoC: speyside: Use snd_soc_codec_get_dapm() ASoC: ad1836: Replace direct snd_soc_codec dapm field access ASoC: adau17x1: Replace direct snd_soc_codec dapm field access ASoC: adau1977: Replace direct snd_soc_codec dapm field access ASoC: adav80x: Replace direct snd_soc_codec dapm field access ASoC: ssm2518: Replace direct snd_soc_codec dapm field access ASoC: ssm2602: Replace direct snd_soc_codec dapm field access ASoC: ssm4567: Replace direct snd_soc_codec dapm field access ASoC: dapm: Add new widgets to the end of the widget list ASoC: dapm: Add new widgets to the end of the widget list ASoC: dapm: Add demux support ASoC: lm4857: Use DAPM demux ASoC: lm4857: Convert to component ASoC: omap-twl4030: Use card DAPM context to access widgets ASoC: brownstone: Automatically disconnect non-connected pins ASoC: poodle: Automatically disconnect non-connected pins ASoC: tosa: Automatically disconnect non-connected pins ASoC: z2: Automatically disconnect non-connected pins ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access ASoC: ab8500: Replace direct snd_soc_codec dapm field access ASoC: ak4641: Replace direct snd_soc_codec dapm field access ASoC: cx20442: Replace direct snd_soc_codec dapm field access ASoC: es8328: Replace direct snd_soc_codec dapm field access ASoC: jz4740: Replace direct snd_soc_codec dapm field access ASoC: ml26124: Replace direct snd_soc_codec dapm field access ASoC: uda134x: Replace direct snd_soc_codec dapm field access ASoC: uda1380: Replace direct snd_soc_codec dapm field access ASoC: sgtl5000: Replace direct snd_soc_codec dapm field access ASoC: sirf-audio-codec: Replace direct snd_soc_codec dapm field access ASoC: sn95031: Replace direct snd_soc_codec dapm field access ASoC: lm49453: Replace direct snd_soc_codec dapm field access ASoC: pcm512x: Replace direct snd_soc_codec dapm field access ASoC: tlv320aix31xx: Replace direct snd_soc_codec dapm field access ASoC: tlv320aic3x: Replace direct snd_soc_codec dapm field access ASoC: tlv320dac33: Replace direct snd_soc_codec dapm field access ASoC: twl4030: Replace direct snd_soc_codec dapm field access ASoC: twl6040: Replace direct snd_soc_codec dapm field access ASoC: Drop unnecessary bias level check on resume ASoC: sta32x: Replace direct snd_soc_codec dapm field access ASoC: sta350: Replace direct snd_soc_codec dapm field access ASoC: sta529: Replace direct snd_soc_codec dapm field access ASoC: da7213: Replace direct snd_soc_codec dapm field access ASoC: da732x: Replace direct snd_soc_codec dapm field access ASoC: da9055: Replace direct snd_soc_codec dapm field access ASoC: max98088: Replace direct snd_soc_codec dapm field access ASoC: max98090: Remove unnecessary snd_soc_dapm_sync() ASoC: max98090: Replace direct snd_soc_codec dapm field access ASoC: max98095: Replace direct snd_soc_codec dapm field access ASoC: max9850: Replace direct snd_soc_codec dapm field access ASoC: alc5623: Replace direct snd_soc_codec dapm field access ASoC: rt286: Replace direct snd_soc_codec dapm field access ASoC: rt5631: Replace direct snd_soc_codec dapm field access ASoC: rt5640: Replace direct snd_soc_codec dapm field access ASoC: rt5651: Replace direct snd_soc_codec dapm field access ASoC: rt5670: Replace direct snd_soc_codec dapm field access ASoC: rt5677: Replace direct snd_soc_codec dapm field access ASoC: cs42l52: Replace direct snd_soc_codec dapm field access ASoC: cs42l56: Replace direct snd_soc_codec dapm field access ASoC: cs42l73: Replace direct snd_soc_codec dapm field access ASoC: cs42xx8: Replace direct snd_soc_codec dapm field access ASoC: arizona: Replace direct snd_soc_codec dapm field access ASoC: wm0010: Replace direct snd_soc_codec dapm field access ASoC: wm5100: Replace direct snd_soc_codec dapm field access ASoC: wm5102: Replace direct snd_soc_codec dapm field access ASoC: wm5110: Replace direct snd_soc_codec dapm field access ASoC: wm8350: Replace direct snd_soc_codec dapm field access ASoC: wm8400: Replace direct snd_soc_codec dapm field access ASoC: wm8510: Replace direct snd_soc_codec dapm field access ASoC: wm8523: Replace direct snd_soc_codec dapm field access ASoC: wm8580: Replace direct snd_soc_codec dapm field access ASoC: wm8711: Replace direct snd_soc_codec dapm field access ASoC: wm8728: Replace direct snd_soc_codec dapm field access ASoC: wm8731: Replace direct snd_soc_codec dapm field access ASoC: wm8737: Replace direct snd_soc_codec dapm field access ASoC: wm8750: Replace direct snd_soc_codec dapm field access ASoC: wm8753: Replace direct snd_soc_codec dapm field access ASoC: wm8770: Replace direct snd_soc_codec dapm field access ASoC: wm8776: Replace direct snd_soc_codec dapm field access ASoC: wm8804: Replace direct snd_soc_codec dapm field access ASoC: wm8900: Replace direct snd_soc_codec dapm field access ASoC: wm8903: Replace direct snd_soc_codec dapm field access ASoC: wm8904: Replace direct snd_soc_codec dapm field access ASoC: wm8940: Replace direct snd_soc_codec dapm field access ASoC: wm8955: Replace direct snd_soc_codec dapm field access ASoC: wm8960: Replace direct snd_soc_codec dapm field access ASoC: wm8961: Replace direct snd_soc_codec dapm field access ASoC: wm8962: Replace direct snd_soc_codec dapm field access ASoC: wm8971: Replace direct snd_soc_codec dapm field access ASoC: wm8974: Replace direct snd_soc_codec dapm field access ASoC: wm8978: Replace direct snd_soc_codec dapm field access ASoC: wm8983: Replace direct snd_soc_codec dapm field access ASoC: wm8985: Replace direct snd_soc_codec dapm field access ASoC: wm8988: Replace direct snd_soc_codec dapm field access ASoC: wm8990: Replace direct snd_soc_codec dapm field access ASoC: wm8991: Replace direct snd_soc_codec dapm field access ASoC: wm8993: Replace direct snd_soc_codec dapm field access ASoC: wm8994: Replace direct snd_soc_codec dapm field access ASoC: wm8995: Replace direct snd_soc_codec dapm field access ASoC: wm8996: Replace direct snd_soc_codec dapm field access ASoC: wm8997: Replace direct snd_soc_codec dapm field access ASoC: wm9081: Replace direct snd_soc_codec dapm field access ASoC: wm9090: Replace direct snd_soc_codec dapm field access ASoC: wm_hubs: Replace direct snd_soc_codec dapm field access ASoC: rx51: Automatically disconnect non-connected pins Liam Girdwood (6): ASoC: Intel: fix broadwell module removing failed issue ASoC: DAPM: Add APIs to create individual DAPM controls. ASoC: topology: Add topology UAPI header ASoC: topology: Add topology core ALSA: topology: Export ID types for TLV controls. ASoC: dapm: fix build errors for missing snd_soc_dapm_new_control symbol Maciej S. Szmigiero (3): ASoC: ac97: Remove rate constraints ASoC: ac97: make selectable in config ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask Marek Belisko (2): ASoC: gtm601: Document GTM601 bindings ASoC: bt-sco: Add devicetree support for bt-sco codec Mark Brown (44): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-tas751x ASoC: tas751x: Factor setting of new bias level into the core Merge branch 'topic/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/dpcm' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'topic/rcar-dma' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: zte: Fix missing dev in devm operation Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fmtbit', 'asoc/fix/max98925', 'asoc/fix/rcar' and 'asoc/fix/ux500' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8737', 'asoc/fix/wm8903' and 'asoc/fix/wm8955' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ad1836', 'asoc/topic/ada1977', 'asoc/topic/adau1701' and 'asoc/topic/adau17x1' into asoc-next Merge remote-tracking branches 'asoc/topic/adau1977', 'asoc/topic/adav80x', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/au1x', 'asoc/topic/bt-sco', 'asoc/topic/cs35l32' and 'asoc/topic/da7210' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci' and 'asoc/topic/dpcm' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-spdif', 'asoc/topic/fsl-ssi' and 'asoc/topic/gpio-jack' into asoc-next Merge remote-tracking branches 'asoc/topic/gpiod-flags', 'asoc/topic/gtm601', 'asoc/topic/intel', 'asoc/topic/lm3857' and 'asoc/topic/max98090' into asoc-next Merge remote-tracking branches 'asoc/topic/max98095', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/qcom' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5677', 'asoc/topic/samsung' and 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branches 'asoc/topic/simple', 'asoc/topic/ssm2518', 'asoc/topic/ssm2602', 'asoc/topic/ssm4567' and 'asoc/topic/sta32x' into asoc-next Merge remote-tracking branches 'asoc/topic/sta350', 'asoc/topic/tas2552', 'asoc/topic/tas3a227e' and 'asoc/topic/tas571x' into asoc-next Merge remote-tracking branches 'asoc/topic/topology', 'asoc/topic/twl6040', 'asoc/topic/wm5100', 'asoc/topic/wm8741' and 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8994', 'asoc/topic/wm8996' and 'asoc/topic/zx' into asoc-next Merge tag 'asoc-v4.2' into asoc-rt5645 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fmtbit', 'asoc/fix/intel', 'asoc/fix/max98925', 'asoc/fix/rcar' and 'asoc/fix/ux500' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8737', 'asoc/fix/wm8903', 'asoc/fix/wm8955' and 'asoc/fix/wm8960' into asoc-linus Merge tag 'asoc-v4.2' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5670' into asoc-next Merge remote-tracking branches 'asoc/topic/adsp' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branches 'asoc/topic/intel' and 'asoc/topic/max98090' into asoc-next Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/ml26124' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/qcom', 'asoc/topic/rcar', 'asoc/topic/rt286' and 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/simple' into asoc-next Merge remote-tracking branch 'asoc/topic/tas2552' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8523' and 'asoc/topic/wm8741' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8995' into asoc-next Mathias Krause (4): ASoC: rt5640: Constify ACPI device ids and register defaults ASoC: max98090: Constify ACPI device ids and register defaults ASoC: rt5670: Constify ACPI device ids and register defaults ASoC: intel: Constify ACPI device ids Michele Curti (1): ASoC: rt5645: move RT5650 muxes to rt5650_specific_dapm_widgets Misael Lopez Cruz (1): ASoC: davinci-mcasp: Logic low for inactive output slots Nicolas Boichat (3): ASoC: rt5645: Rename HP control to Headphone ASoC: rt5645: Init jack_detect_work before registering irq ASoC: rt5645: Lock mutex in rt5645_enable_push_button_irq Nicolin Chen (1): ASoC: fsl_spdif: Don't try to round-up for clock divisor calculation Oder Chiou (6): ASoC: rt5677: Add DMIC ASRC detect function ASoC: rt5645: make RT5650_TDM_CTRL_4 readable ASoC: rt5645: add device tree support ASoC: rt5645: change gpio to gpiod APIs ASoC: rt286: Add RL6347A class device shared support for RT286 ASoC: rt5645: Add the device tree parser Pascal Huerst (1): ASoC: adau1701: add regulator consumer support Peter Ujfalusi (27): ASoC: tas2552: Make the enable-gpio really optional ASoC: tas2552: Fix kernel crash when the codec is loaded but not part of a card ASoC: tas2552: Fix kernel crash caused by wrong kcontrol entry ASoC: tas2552: Correct PDM configuration register bit definitions ASoC: tas2552: Correct CFG1 register bit definitions ASoC: tas2552: Simplify the tas2552_mute function ASoC: tas2552: Simplify and reverse the functionality of tas2552_sw_shutdown ASoC: tas2552: Rename mclk parameter to pll_clkin to match with the datasheet ASoC: tas2552: bindings header file for tas2552 codec ASoC: tas2552: Add support for pll and pdm source clock selection ASoC: tas2552: Correct dai format support ASoC: tas2552: Correct and clean up data format and BCLK/WCLK direction ASoC: tas2552: Add TDM support ASoC: tas2552: Clean up the Digital - Analog DAPM route definition ASoC: tas2552: Correct the Speaker Driver Playback Volume (PGA_GAIN) ASoC: tas2552: Implement startup/stop sequence as per TRM ASoC: tas2552: Add support for word length configuration ASoC: tas2552: Configure the WCLK frequency based on the stream ASoC: tas2552: Correct the PLL configuration ASoC: tas2552: Add control for selecting DIN source ASoC: tas2552: Correct Output Data register usage ASoC: tas2552: Correct Boost Auto-Pass Through Control register usage ASoC: tas2552: Code, define alignment changes for uniformity ASoC: tas2552: Call pm_runtime_disable when the module is removed ASoC: tas2552: Remove unneeded semicolon ASoC: tas2552: Use consistent name for 'struct tas2552_data' ASoC: tas2552: Update DT binding document regarding clock configuration Rajan Vaja (1): ASoC: mop500_ab8500: Add a NULL pointer check in mop500_ab8500_machine_init() Rasmus Villemoes (1): ASoC: Intel: mrfld: simplify sst_fill_widget_module_info Richard Fitzgerald (9): ASoC: arizona: Fix noise generator gain TLV ASoC: wm_adsp: Dump scratch registers on DSP shutdown ASoC: arizona: Export functions to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control regulator: arizona-ldo1: Do not control DVFS clocking from regulator ASoC: wm_adsp: Add codec_probe and codec_remove stubs ASoC: wm_adsp: create ALSA controls from wm_adsp driver ASoC: wm_adsp: Add basic debugfs entries Sebastian Andrzej Siewior (1): ASoC: Intel: sst: use ; instead of , at the of a C statement Sergej Sawazki (2): ASoC: wm8741: Add differential mono mode support ASoC: wm8741: Allow master clock switching Srinivas Kandagatla (17): ASoC: qcom: Remove redundant error check. ASoC: qcom: remove unnecessary header files ASoC: qcom: move ipq806x specific bits out of lpass driver. ASoC: qcom: remove hardcoded i2s port number ASoC: qcom: remove hardcoded dma channel ASoC: skip legacy dai naming if dai driver has all the information ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: make osr clock optional ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Document apq8016 bindings. ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: storm: allocate snd_soc_card struct dynamically. ASoC: qcom: storm: remove special casing EPROBE_DEFER ASoC: qcom: document apq8016 sbc machine driver bindings ASoC: qcom: add apq8016 sound card support Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size Sudip Mukherjee (1): ASoC: rt5645: fixed section mismatch Takashi Iwai (1): ASoC: Simplify format_register_str() without stack usages Uwe Kleine-König (3): ASoC: sta32x: use devm_gpiod_get_optional for optional reset gpio ASoC: max98357a: use flags argument of devm_gpiod_get to set direction ASoC: rx51: use flags argument of devm_gpiod_get to set direction Vinod Koul (6): ASoC: intel - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN ASoC: Intel: add frame and data polarity to ssp config ASoC: Intel: use local values for ssp configuration ASoC: Intel: load hw_defaults in hw_params of ssp be ASoC: intel: add support for specifying PCM format ASoC: Intel: add support for configuring TDM slots for SSP Vladimir Zapolskiy (1): ASoC: dapm: fix snd_soc_dapm_new_control() implicit declaration Wei Yongjun (1): ASoC: cs35l32: Remove unused including <linux/version.h> Zidan Wang (5): ASoC: fsl_sai: add sai master mode support ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock ASoC: wm8960: add 32 bit word length support ASoC: wm8960: the enum of "DAC Polarity" should be wm8960_enum[1] kbuild test robot (7): ASoC: adau1977: fix simple_return.cocci warnings ASoC: ssm2518: fix simple_return.cocci warnings ASoC: qcom: ipq806x_lpass_alloc_dma_channel() can be static ASoC: zx: zx_i2s_dai can be static ASoC: zx: fix platform_no_drv_owner.cocci warnings ASoC: zx: fix simple_return.cocci warnings ASoC: zx: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/adi,adau1701.txt | 4 + Documentation/devicetree/bindings/sound/bt-sco.txt | 13 + Documentation/devicetree/bindings/sound/gtm601.txt | 13 + .../devicetree/bindings/sound/max98090.txt | 6 + .../devicetree/bindings/sound/mt8173-max98090.txt | 13 + .../bindings/sound/mt8173-rt5650-rt5676.txt | 13 + .../devicetree/bindings/sound/mtk-afe-pcm.txt | 45 + .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 60 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 3 +- Documentation/devicetree/bindings/sound/rt5645.txt | 72 + Documentation/devicetree/bindings/sound/rt5677.txt | 2 + .../devicetree/bindings/sound/simple-card.txt | 6 +- .../devicetree/bindings/sound/tas2552.txt | 6 + .../devicetree/bindings/sound/tas571x.txt | 41 + Documentation/devicetree/bindings/sound/wm8741.txt | 11 + .../devicetree/bindings/sound/zte,zx-i2s.txt | 44 + .../devicetree/bindings/sound/zte,zx-spdif.txt | 28 + .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 6 + drivers/dma/sh/rcar-dmac.c | 37 +- drivers/regulator/arizona-ldo1.c | 5 - include/dt-bindings/sound/apq8016-lpass.h | 9 + include/dt-bindings/sound/audio-jack-events.h | 9 + include/dt-bindings/sound/tas2552.h | 18 + include/sound/dmaengine_pcm.h | 5 - include/sound/rt5645.h | 6 - include/sound/soc-dapm.h | 49 + include/sound/soc-topology.h | 168 ++ include/sound/soc.h | 118 +- include/sound/tlv.h | 15 +- include/uapi/sound/asoc.h | 388 +++++ include/uapi/sound/tlv.h | 31 + sound/soc/Kconfig | 2 + sound/soc/Makefile | 3 + sound/soc/atmel/Kconfig | 37 +- sound/soc/atmel/atmel-pcm-dma.c | 3 +- sound/soc/atmel/sam9g20_wm8731.c | 10 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/cirrus/ep93xx-pcm.c | 1 - sound/soc/codecs/88pm860x-codec.c | 19 +- sound/soc/codecs/Kconfig | 17 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ab8500-codec.c | 20 +- sound/soc/codecs/ac97.c | 8 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 126 +- sound/soc/codecs/adau1761.c | 27 +- sound/soc/codecs/adau1781.c | 10 +- sound/soc/codecs/adau17x1.c | 20 +- sound/soc/codecs/adau1977.c | 14 +- sound/soc/codecs/adav80x.c | 11 +- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 3 +- sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/alc5623.c | 3 +- sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/arizona.c | 174 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/bt-sco.c | 11 +- sound/soc/codecs/cq93vc.c | 1 - sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs42l52.c | 5 +- sound/soc/codecs/cs42l56.c | 5 +- sound/soc/codecs/cs42l73.c | 3 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cx20442.c | 6 +- sound/soc/codecs/da7213.c | 3 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 3 +- sound/soc/codecs/es8328.c | 3 +- sound/soc/codecs/isabelle.c | 2 - sound/soc/codecs/jz4740.c | 4 +- sound/soc/codecs/lm4857.c | 114 +- sound/soc/codecs/lm49453.c | 4 +- sound/soc/codecs/max98088.c | 3 +- sound/soc/codecs/max98090.c | 38 +- sound/soc/codecs/max98095.c | 24 +- sound/soc/codecs/max98357a.c | 3 +- sound/soc/codecs/max9850.c | 3 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/ml26124.c | 61 +- sound/soc/codecs/pcm512x.c | 8 +- sound/soc/codecs/rl6347a.c | 128 ++ sound/soc/codecs/rl6347a.h | 32 + sound/soc/codecs/rt286.c | 130 +- sound/soc/codecs/rt5631.c | 5 +- sound/soc/codecs/rt5640.c | 21 +- sound/soc/codecs/rt5645.c | 1112 +++++++++--- sound/soc/codecs/rt5645.h | 31 +- sound/soc/codecs/rt5651.c | 5 +- sound/soc/codecs/rt5670.c | 31 +- sound/soc/codecs/rt5677.c | 157 +- sound/soc/codecs/rt5677.h | 15 + sound/soc/codecs/sgtl5000.c | 56 +- sound/soc/codecs/sirf-audio-codec.c | 2 +- sound/soc/codecs/sn95031.c | 12 +- sound/soc/codecs/ssm2518.c | 9 +- sound/soc/codecs/ssm2602.c | 5 +- sound/soc/codecs/ssm4567.c | 9 +- sound/soc/codecs/sta32x.c | 19 +- sound/soc/codecs/sta350.c | 9 +- sound/soc/codecs/sta529.c | 8 +- sound/soc/codecs/stac9766.c | 3 +- sound/soc/codecs/tas2552.c | 430 +++-- sound/soc/codecs/tas2552.h | 153 +- sound/soc/codecs/tas571x.c | 514 ++++++ sound/soc/codecs/tas571x.h | 33 + sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 11 +- sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 10 +- sound/soc/codecs/tlv320dac33.c | 5 +- sound/soc/codecs/ts3a227e.c | 15 +- sound/soc/codecs/twl4030.c | 3 +- sound/soc/codecs/twl6040.c | 9 +- sound/soc/codecs/uda134x.c | 4 +- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm1250-ev1.c | 2 - sound/soc/codecs/wm5100.c | 12 +- sound/soc/codecs/wm5102.c | 68 +- sound/soc/codecs/wm5110.c | 25 +- sound/soc/codecs/wm8350.c | 3 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 3 +- sound/soc/codecs/wm8523.c | 29 +- sound/soc/codecs/wm8580.c | 3 +- sound/soc/codecs/wm8711.c | 3 +- sound/soc/codecs/wm8728.c | 3 +- sound/soc/codecs/wm8731.c | 8 +- sound/soc/codecs/wm8737.c | 11 +- sound/soc/codecs/wm8741.c | 190 +- sound/soc/codecs/wm8741.h | 10 + sound/soc/codecs/wm8750.c | 3 +- sound/soc/codecs/wm8753.c | 3 +- sound/soc/codecs/wm8770.c | 3 +- sound/soc/codecs/wm8776.c | 3 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 13 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8903.h | 2 +- sound/soc/codecs/wm8904.c | 5 +- sound/soc/codecs/wm8940.c | 6 +- sound/soc/codecs/wm8955.c | 7 +- sound/soc/codecs/wm8960.c | 127 +- sound/soc/codecs/wm8961.c | 6 +- sound/soc/codecs/wm8962.c | 21 +- sound/soc/codecs/wm8971.c | 3 +- sound/soc/codecs/wm8974.c | 3 +- sound/soc/codecs/wm8978.c | 7 +- sound/soc/codecs/wm8983.c | 3 +- sound/soc/codecs/wm8985.c | 3 +- sound/soc/codecs/wm8988.c | 3 +- sound/soc/codecs/wm8990.c | 5 +- sound/soc/codecs/wm8991.c | 3 +- sound/soc/codecs/wm8993.c | 12 +- sound/soc/codecs/wm8994.c | 68 +- sound/soc/codecs/wm8995.c | 8 +- sound/soc/codecs/wm8996.c | 23 +- sound/soc/codecs/wm8997.c | 18 +- sound/soc/codecs/wm9081.c | 4 +- sound/soc/codecs/wm9090.c | 6 +- sound/soc/codecs/wm9712.c | 3 +- sound/soc/codecs/wm9713.c | 7 +- sound/soc/codecs/wm_adsp.c | 1450 ++++++++++------ sound/soc/codecs/wm_adsp.h | 33 +- sound/soc/codecs/wm_hubs.c | 4 +- sound/soc/codecs/wmfw.h | 44 +- sound/soc/davinci/davinci-mcasp.c | 239 ++- sound/soc/davinci/davinci-mcasp.h | 5 +- sound/soc/fsl/fsl_dma.c | 4 +- sound/soc/fsl/fsl_sai.c | 144 +- sound/soc/fsl/fsl_sai.h | 9 +- sound/soc/fsl/fsl_spdif.c | 10 +- sound/soc/fsl/fsl_ssi.c | 7 - sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-mc13783.c | 6 +- sound/soc/fsl/imx-wm8962.c | 2 +- sound/soc/generic/simple-card.c | 34 +- sound/soc/intel/Kconfig | 19 +- sound/soc/intel/atom/sst-atom-controls.c | 187 +- sound/soc/intel/atom/sst-atom-controls.h | 9 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 47 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 + sound/soc/intel/atom/sst/sst.c | 4 +- sound/soc/intel/atom/sst/sst_acpi.c | 4 + sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 348 ++++ sound/soc/intel/boards/cht_bsw_rt5645.c | 118 +- sound/soc/intel/common/sst-acpi.c | 2 +- sound/soc/intel/common/sst-ipc.c | 34 +- sound/soc/intel/common/sst-ipc.h | 7 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 + sound/soc/intel/haswell/sst-haswell-pcm.c | 31 +- sound/soc/mediatek/Kconfig | 30 + sound/soc/mediatek/Makefile | 5 + sound/soc/mediatek/mt8173-max98090.c | 213 +++ sound/soc/mediatek/mt8173-rt5650-rt5676.c | 278 +++ sound/soc/mediatek/mtk-afe-common.h | 109 ++ sound/soc/mediatek/mtk-afe-pcm.c | 1233 +++++++++++++ sound/soc/omap/Kconfig | 5 +- sound/soc/omap/omap-twl4030.c | 3 +- sound/soc/omap/rx51.c | 40 +- sound/soc/pxa/brownstone.c | 25 +- sound/soc/pxa/poodle.c | 19 +- sound/soc/pxa/tosa.c | 13 +- sound/soc/pxa/z2.c | 9 +- sound/soc/qcom/Kconfig | 28 +- sound/soc/qcom/Makefile | 6 + sound/soc/qcom/apq8016_sbc.c | 198 +++ sound/soc/qcom/lpass-apq8016.c | 242 +++ sound/soc/qcom/lpass-cpu.c | 240 +-- sound/soc/qcom/lpass-ipq806x.c | 109 ++ .../{lpass-lpaif-ipq806x.h => lpass-lpaif-reg.h} | 92 +- sound/soc/qcom/lpass-platform.c | 202 ++- sound/soc/qcom/lpass.h | 51 +- sound/soc/qcom/storm.c | 26 +- sound/soc/samsung/Kconfig | 15 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 6 +- sound/soc/samsung/smdk_wm8994.c | 3 +- sound/soc/samsung/speyside.c | 2 +- sound/soc/sh/rcar/core.c | 139 +- sound/soc/sh/rcar/dma.c | 113 +- sound/soc/sh/rcar/dvc.c | 30 +- sound/soc/sh/rcar/rsnd.h | 113 +- sound/soc/sh/rcar/rsrc-card.c | 437 ++--- sound/soc/sh/rcar/src.c | 130 +- sound/soc/sh/rcar/ssi.c | 160 +- sound/soc/soc-core.c | 67 +- sound/soc/soc-dapm.c | 349 +++- sound/soc/soc-generic-dmaengine-pcm.c | 25 +- sound/soc/soc-jack.c | 9 +- sound/soc/soc-pcm.c | 47 +- sound/soc/soc-topology.c | 1826 ++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.c | 4 + sound/soc/ux500/ux500_pcm.c | 1 - sound/soc/zte/Kconfig | 17 + sound/soc/zte/Makefile | 2 + sound/soc/zte/zx296702-i2s.c | 436 +++++ sound/soc/zte/zx296702-spdif.c | 365 ++++ 248 files changed, 13076 insertions(+), 3099 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8173-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5645.txt create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 include/dt-bindings/sound/apq8016-lpass.h create mode 100644 include/dt-bindings/sound/audio-jack-events.h create mode 100644 include/dt-bindings/sound/tas2552.h create mode 100644 include/sound/soc-topology.h create mode 100644 include/uapi/sound/asoc.h create mode 100644 include/uapi/sound/tlv.h create mode 100644 sound/soc/codecs/rl6347a.c create mode 100644 sound/soc/codecs/rl6347a.h create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h create mode 100644 sound/soc/intel/boards/cht_bsw_max98090_ti.c create mode 100644 sound/soc/mediatek/Kconfig create mode 100644 sound/soc/mediatek/Makefile create mode 100644 sound/soc/mediatek/mt8173-max98090.c create mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5676.c create mode 100644 sound/soc/mediatek/mtk-afe-common.h create mode 100644 sound/soc/mediatek/mtk-afe-pcm.c create mode 100644 sound/soc/qcom/apq8016_sbc.c create mode 100644 sound/soc/qcom/lpass-apq8016.c create mode 100644 sound/soc/qcom/lpass-ipq806x.c rename sound/soc/qcom/{lpass-lpaif-ipq806x.h => lpass-lpaif-reg.h} (64%) create mode 100644 sound/soc/soc-topology.c create mode 100644 sound/soc/zte/Kconfig create mode 100644 sound/soc/zte/Makefile create mode 100644 sound/soc/zte/zx296702-i2s.c create mode 100644 sound/soc/zte/zx296702-spdif.c

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[alsa-devel] [PATCH] ALSA: pcm: Fix pcm_class sysfs output
by Takashi Iwai 23 Jun '15

23 Jun '15

The pcm_class sysfs of each PCM substream gives only "none" since the recent code change to embed the struct device. Fix the code to point directly to the embedded device object properly. Fixes: ef46c7af93f9 ('ALSA: pcm: Embed struct device') Cc: <stable(a)vger.kernel.org> # v4.0+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e53794319ef3..02bd96954dc4 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1027,7 +1027,8 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) static ssize_t show_pcm_class(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_pcm *pcm; + struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); + struct snd_pcm *pcm = pstr->pcm; const char *str; static const char *strs[SNDRV_PCM_CLASS_LAST + 1] = { [SNDRV_PCM_CLASS_GENERIC] = "generic", @@ -1036,8 +1037,7 @@ static ssize_t show_pcm_class(struct device *dev, [SNDRV_PCM_CLASS_DIGITIZER] = "digitizer", }; - if (! (pcm = dev_get_drvdata(dev)) || - pcm->dev_class > SNDRV_PCM_CLASS_LAST) + if (pcm->dev_class > SNDRV_PCM_CLASS_LAST) str = "none"; else str = strs[pcm->dev_class]; -- 2.4.4

1 0

[alsa-devel] Internal speakers not working on HP ProBook 455 G2
by Michel Dänzer 23 Jun '15

23 Jun '15

Hi, the built-in speakers of my HP ProBook 455 G2 are not working by default. I'm attaching the output of alsa-info.sh. In pavucontrol, the speakers appear as "Speakers (unavailable)", and in the GNOME sound settings they don't appear at all. After some playing around with HDAAnalyzer, I discovered that enabling the 'OUT' widget control of pin 0x14 makes the speakers emit sound, see the diff below. Let me know if you need any more information, and I'll happily test any patches to fix this problem. Diff for codec 1/0 (0x10ec0282): --- +++ @@ -120,39 +120,40 @@ bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 - Control: name="Internal Mic Phantom Jack", index=0, device=0 + Control: iface="card", name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, 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: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 - Control: name="Speaker Phantom Jack", index=0, device=0 + Control: iface="card", name="Speaker Phantom Jack", index=0, device=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=0x00, enabled=0 Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute= -- Earthling Michel Dänzer | http://www.amd.com Libre software enthusiast | Mesa and X developer

2 8

[alsa-devel] [asoc:topic/gtm601 2/3] sound/soc/codecs/gtm601.c:36:27: sparse: symbol 'gtm601_dai' was not declared. Should it be static?
by kbuild test robot 22 Jun '15

22 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/gtm601 head: ac6e7c29ac12e1e285b0740bb671039ef436cf91 commit: 600a2b098a87843d44ab2f54fde9eda421946b38 [2/3] ASoC: Add gtm601 codec driver reproduce: # apt-get install sparse git checkout 600a2b098a87843d44ab2f54fde9eda421946b38 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/gtm601.c:36:27: sparse: symbol 'gtm601_dai' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 1

[alsa-devel] [asoc:topic/gtm601 2/3] sound/soc/codecs/gtm601.c:86:4-9: No need to set .owner here. The core will do it.
by kbuild test robot 22 Jun '15

22 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/gtm601 head: ac6e7c29ac12e1e285b0740bb671039ef436cf91 commit: 600a2b098a87843d44ab2f54fde9eda421946b38 [2/3] ASoC: Add gtm601 codec driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/gtm601.c:86:4-9: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 1

[alsa-devel] [PATCH] ASoC: rsrc-card: Remove unused variable in rsrc_card_startup()
by Geert Uytterhoeven 22 Jun '15

22 Jun '15

sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup': sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable 'ret' [-Wunused-variable] int ret; ^ Fixes: 047000278da3a17f ("ASoC: rsrc-card: cleanup for DPCM") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/rcar/rsrc-card.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 8caca2e180c39122..84e935711e29e442 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -75,8 +75,6 @@ static int rsrc_card_startup(struct snd_pcm_substream *substream) struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, rtd - rtd->card->rtd); - int ret; - return clk_prepare_enable(dai_props->clk); } -- 1.9.1

5 7

[alsa-devel] [PATCH v2 0/2] ALSA: hda: add module flag snd_adsp_enable
by Vinod Koul 22 Jun '15

22 Jun '15

This is second attempt for creating a module option for hda core to select either legacy HDA driver or upcoming ASoC driver. This series still relies on module flag (with feedback from previous series baked in). The eventual plan is to rely on HW advertised capabilities and BIOS configuration to load either driver, but since ASoC driver has still some time to get fully merged and matured on various system, we will rely on module flag till the who ASoC HDA systems is completed and available is all distros Vinod Koul (2): ALSA: hda: add snd_adsp_enable module flag ALSA: hda: use snd_adsp_enable in HDA driver include/sound/hdaudio.h | 11 +++++++++++ sound/hda/hda_bus_type.c | 6 ++++++ sound/pci/hda/hda_controller.h | 1 + sound/pci/hda/hda_intel.c | 10 +++++++++- 4 files changed, 27 insertions(+), 1 deletion(-) -- 1.9.1

2 4

[alsa-devel] Post Audio codec sgtl5000 with ARM/Zynq under Linux
by Wojciech Zebrowski 22 Jun '15

22 Jun '15

Hi All I try solve problem with support SGTL5000 under Linux , ARM / ZYNQ ARCHITECTURE. Please add this post to "alsa" forum. If you need any others information i can send it . My post: I try use freescale audio chip sgtl5000 on my custom board with Zynq chip. I need help with instance driver in device tree and debug first execution. 1.*I use I2C* controler to configure sgtl5000 : / axi_iic_sgtl: i2c@41610000 {// // #address-cells = <1>;// // #size-cells = <0>;// // compatible = "xlnx,xps-iic-2.00.a";// // interrupt-parent = <&gic>;// // interrupts = <0 31 4>;// // reg = <0x41610000 0x10000>;// //// // codec: sgtl5000@0a {// // #sound-dai-cells = <0>;// // compatible = "fsl,sgtl5000";// // reg = <0x0a>;// ////// VDDA-supply = <&reg_3p3v>;// ////// VDDIO-supply = <&reg_3p3v>;// // clocks = <&clkc 19>;// // }; / How can i set comment lines ? 2. sound : / sound { compatible = "fsl,imx51-babbage-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx51-babbage-sgtl5000"; // ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT"; mux-int-port = <1>; mux-ext-port = <3>; };/ I want use I2C configuration driver not SPI . How can i set comment line ? 3. I2S driver looks like ok : / axi_i2s_adi_0: axi_i2s_adi@77600000 { compatible = "xlnx,axi-i2s-adi-1.0"; reg = <0x77600000 0x10000>; xlnx,bclk-pol = <0x0>; xlnx,dma-type = <0x1>; xlnx,has-rx = <0x1>; xlnx,has-tx = <0x1>; xlnx,lrclk-pol = <0x0>; xlnx,num-ch = <0x1>; xlnx,s-axi-min-size = <0x000001FF>; xlnx,slot-width = <0x18>; };/ PS. Right now i can compile device tree but audio is not detected. What kind of log do you need to check Linux boot problem ?

2 3

[alsa-devel] [PATCH v6 0/4] ASoC: intel - add skylake PCM driver
by Vinod Koul 22 Jun '15

22 Jun '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch This version removes the stream object creation, freeup, link freeup and buffer prealloc flag as they have moved to hdac now. Also irq management is improved upon. Please note this series is dependent on sound next having hda core patches and extended bus patches Jeeja KP (3): ASoC: Intel: add Skylake HDA platform driver ASoC: Intel - add Skylake HDA audio driver ASoC: Intel - add makefile support for SKL driver Vinod Koul (1): ASoC: intel - adds support for decoupled mode in skl driver sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 935 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 543 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 71 +++ 6 files changed, 1560 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl: fsl_asrc: Check for clk_prepare_enable() error
by Fabio Estevam 20 Jun '15

20 Jun '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_asrc.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c068494..37dde26 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -931,14 +931,22 @@ static int fsl_asrc_probe(struct platform_device *pdev) static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); - int i; + int i, ret; - clk_prepare_enable(asrc_priv->mem_clk); - clk_prepare_enable(asrc_priv->ipg_clk); + ret = clk_prepare_enable(asrc_priv->mem_clk); + if (ret) + return ret; + ret = clk_prepare_enable(asrc_priv->ipg_clk); + if (ret) + goto disable_mem_clk; for (i = 0; i < ASRC_CLK_MAX_NUM; i++) clk_prepare_enable(asrc_priv->asrck_clk[i]); return 0; + +disable_mem_clk: + clk_disable_unprepare(asrc_priv->mem_clk); + return ret; } static int fsl_asrc_runtime_suspend(struct device *dev) -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda-beep: Update authors dead email address
by Matt Ranostay 20 Jun '15

20 Jun '15

Signed-off-by: Matt Ranostay <mranostay(a)gmail.com> --- sound/pci/hda/hda_beep.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 46524ff..1052ad3 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -1,7 +1,7 @@ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> + * Author: Matt Ranostay <mranostay(a)gmail.com> * Copyright (c) 2008 Embedded Alley Solutions Inc * * This driver is free software; you can redistribute it and/or modify -- 1.9.1

2 1

Re: [alsa-devel] No sound from Mac Pro headphone port
by Raymond Yau 20 Jun '15

20 Jun '15

> > You will need to provide more specific instructions on what you want. I do not read code. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> >> Bingo: the pin-outs are very incorrect. >> >> This is the (original) tech specs of the machine. >> >> https://support.apple.com/kb/SP506?viewlocale=en_US&locale=en_US >> >> Note: >> >> 1 x internal speaker >> 1 x FRONT headphone port >> >> >> Refer to user manual , >> Headphone Jack >> Connect headphones, including Apple Earphone with remote and mic through >> the headphone mini jack >> >> Is node 0x18 headset mic ? >> need specific sequence number and parser_flags >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Try to find headset mic; >> mark seq number as 0xc to trigger */ >> >> 1 x REAR line out port >> >> 1 x REAR line in port >> 1 x REAR optical TOSLINK out port >> 1 x REAR optical TOSLINK in port >> >> The machine does not have a built-in mic port. >> >> >> you need to use hdajacksensetest to verify hp, line out , line in jacks >> are node 0x19, 0x14 and 0x1a by plug and unplug those jacks >> >> clear bit 8 of pin default of Line Out Jack if it return correct jack state >> >> >> >> Linux 4. Mac Pro 2009. ACL889A codec. >> >> No sound from headphone port. Seems to detect plug / unplug OK. >> >> Works OK when machine booted to OS X or Windows 7. >> >> Have found several bug repors on similar issues with Mac >> laptops but >> no fixes. >> >> Have written e-mail with David Henningsson on problem but >> unable to >> find a solution. >> >> ALSA-info output: >> >> http://www.alsa-project.org/db/?f=1dc7146eb34af5ad307d029868f775fac5e2764e >> >> >> Do your macpro spec support 5.1 ? >> >> Simple mixer control 'Channel Mode',0 >> Capabilities: enum >> Items: '2ch' '4ch' '6ch' >> Item0: '2ch' Channel mode is created when there are three or more jacks at same location Use hdajackretask to change location of HP and Mic from Ext Rear to Ext Front Advanced override and set as boot default http://git.alsa-project.org/?p=alsa-tools.git;a=blob;f=hdajackretask/README… http://voices.canonical.com/david.henningsson/2011/11/29/turn-your-mic-jack… http://git.alsa-project.org/?p=alsa-tools.git;a=tree;f=hdajacksensetest;hb=… Use hdajacksensetest -a to verify line out support jack detect clear MISC = NO_PRESENCE bit 8 of pin default of Line out by hdajackretask Pin Default 0x01010050: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x5, Sequence = 0x0 Most likely need vendor coefficient to support mono mic of apple earphones >> >> Pin default seem incorrect >> >> 1) Headphone at ext Rear >> 2) Line Out does not support jack detection >> 3) Mic jack ???? >> >> /sys/class/sound/hwC0D0/init_pin_configs: >> 0x14 0x01010150 >> 0x15 0x400000f0 >> 0x16 0x400000f0 >> 0x17 0x400000f0 >> 0x18 0x01a19070 >> 0x19 0x01214040 >> 0x1a 0x01813010 >> 0x1b 0x90100130 >> 0x1c 0x400000f0 >> 0x1d 0x400000f0 >> 0x1e 0x0145e160 >> 0x1f 0x01c5e120 >> >> /sys/class/sound/hwC0D0/driver_pin_configs: >> >> >> >> [ 4.057250] sound hdaudioC0D0: ALC889A: SKU not ready 0x400000f0 >> [ 4.057633] sound hdaudioC0D0: autoconfig for ALC889A: >> line_outs=1 >> (0x14/0x0/0x0/0x0/0x0) type:line >> [ 4.057636] sound hdaudioC0D0: speaker_outs=1 >> (0x1b/0x0/0x0/0x0/0x0) >> [ 4.057638] sound hdaudioC0D0: hp_outs=1 >> (0x19/0x0/0x0/0x0/0x0) >> [ 4.057639] sound hdaudioC0D0: mono: mono_out=0x0 >> [ 4.057641] sound hdaudioC0D0: dig-out=0x1e/0x0 >> [ 4.057642] sound hdaudioC0D0: inputs: >> [ 4.057645] sound hdaudioC0D0: Mic=0x18 >> [ 4.057646] sound hdaudioC0D0: Line=0x1a >> [ 4.057648] sound hdaudioC0D0: dig-in=0x1f >> >>

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[alsa-devel] Scarlett 18i8 hardware out of sync with AlsaMixer
by Adam Goode 19 Jun '15

19 Jun '15

Linux 4.0.2-300.fc22.x86_64 Hi, I have the Scarlett 18i8 USB device that is supported by the new scarlett_mixer.c code. I am happy to say that the mixer code works in most cases. But under some conditions, I cannot get any sound out of the device until I go and toggle the "Matrix 01 Input" and "Matrix 02 Input" enums up and then down (from PCM 1 and PCM 2). This appears to me to be some kind of cache invalidation bug, where the device is out of sync with kernel mixer state. I will take a look at the code myself at some point, but not for a while. But I am happy to try patches if anyone comes up with anything in the mean time. Thanks, Adam

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[alsa-devel] [PATCH] ASoC: wm_adsp: Move DSP Rate controls into the codec
by Richard Fitzgerald 19 Jun '15

19 Jun '15

The rate controls are codec-specific, it's not possible to generically say what the range or the meaning of each control is (or even if they exist at all) - that depends on the particular codec. This is currently being handled for Arizona codecs by putting an Arizona-specific table of controls inside the wm_adsp driver. This creates a dependency between wm_adsp and arizona.c, and is an awkward solution if the ADSP is used in another family of codecs Fix this by moving the Arizona-specific rate controls into the Arizona codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 27 ++++++++++++++++++++++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 5 ++++ sound/soc/codecs/wm5110.c | 6 +++++ sound/soc/codecs/wm_adsp.c | 52 +++++-------------------------------------- sound/soc/codecs/wm_adsp.h | 2 +- 7 files changed, 48 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 88f6df2..802e05e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2286,6 +2286,33 @@ int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) } EXPORT_SYMBOL_GPL(arizona_set_output_mode); +static const struct soc_enum arizona_adsp2_rate_enum[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; + +const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { + SOC_ENUM("DSP1 Rate", arizona_adsp2_rate_enum[0]), + SOC_ENUM("DSP2 Rate", arizona_adsp2_rate_enum[1]), + SOC_ENUM("DSP3 Rate", arizona_adsp2_rate_enum[2]), + SOC_ENUM("DSP4 Rate", arizona_adsp2_rate_enum[3]), +}; +EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 8824f5b..9881b6f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -217,6 +217,8 @@ extern const struct soc_enum arizona_ng_hold; extern const struct soc_enum arizona_in_hpf_cut_enum; extern const struct soc_enum arizona_in_dmic_osr[]; +extern const struct snd_kcontrol_new arizona_adsp2_rate_controls[]; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 5a9da28..c830832 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1555,7 +1555,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) wm2200->codec = codec; - ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c5ec519..d097f09e5 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1880,6 +1880,11 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = snd_soc_add_codec_controls(codec, + arizona_adsp2_rate_controls, 1); + if (ret) + return ret; + arizona_init_spk(codec); arizona_init_gpio(codec); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 5f032a3..709fcc6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1614,6 +1614,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) return ret; } + ret = snd_soc_add_codec_controls(codec, + arizona_adsp2_rate_controls, + WM5110_NUM_ADSP); + if (ret) + return ret; + snd_soc_dapm_disable_pin(dapm, "HAPTICS"); return 0; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9f90b0..0bb415a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -458,57 +458,17 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; -const struct snd_kcontrol_new wm_adsp1_fw_controls[] = { +const struct snd_kcontrol_new wm_adsp_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], + wm_adsp_fw_get, wm_adsp_fw_put), }; -EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls); - -static const struct soc_enum wm_adsp2_rate_enum[] = { - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), -}; - -static const struct snd_kcontrol_new wm_adsp2_fw_controls[4][2] = { - { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]), - }, - { - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]), - }, - { - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]), - }, - { - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]), - }, -}; +EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -2125,8 +2085,8 @@ int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) wm_adsp2_init_debugfs(dsp, codec); return snd_soc_add_codec_controls(codec, - wm_adsp2_fw_controls[dsp->num - 1], - ARRAY_SIZE(wm_adsp2_fw_controls[0])); + &wm_adsp_fw_controls[dsp->num - 1], + 1); } EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 5042cbd..579a635 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -83,7 +83,7 @@ struct wm_adsp { #define WM_ADSP2(wname, num) \ WM_ADSP2_E(wname, num, wm_adsp2_early_event) -extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: Intel Kconfig: add SND_SOC_INTEL_BOARDS option
by Michele Curti 19 Jun '15

19 Jun '15

Machine drivers under boards folder must be available for both SND_SST_MFLD_PLATFORM and SND_SOC_INTEL_SST. This patch adds a new kernel option SND_SOC_INTEL_BOARD to permit this. Signed-off-by: Michele Curti <michele.curti(a)gmail.com> --- sound/soc/intel/Kconfig | 5 +++++ sound/soc/intel/Makefile | 2 +- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f3060a4..c442cd7 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -10,7 +10,11 @@ config SND_MFLD_MACHINE Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BOARDS + tristate + config SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_BOARDS tristate config SND_SST_IPC @@ -28,6 +32,7 @@ config SND_SST_IPC_ACPI config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI + select SND_SOC_INTEL_BOARDS depends on (X86 || COMPILE_TEST) depends on DW_DMAC_CORE help diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 3853ec2..4c4bf74 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -7,4 +7,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ # Machine support -obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ +obj-$(CONFIG_SND_SOC_INTEL_BOARDS) += boards/ -- 2.4.3

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[alsa-devel] question about dapm-routes and uda1380.c
by Andrea Venturi 19 Jun '15

19 Jun '15

hello, i'm working on integrating on an Allwinner A20 based SBC an external codec based on NXP UDA1380 through I2S DAI; the SBC is Olime Olinuxino Lime2, the codec board is this one, as it's pretty cheap on usual china sites: http://www.wvshare.com/product/UDA1380-Board.htm i'm using mainline kernel, 4.0.x from here: https://github.com/jwrdegoede/linux-sunxi/tree/sunxi-wip i'm planning to just port the I2S DAI from the legacy allwinner linux-sunxi 3.4 and use a "simple-audio-card" approach. i have more or less the setup in place with my 3 modules and DTS someway hacked to work out the connection between SOC/CPU + DAI + CODEC. now i load the three modules in the kernel this way (i'm wondering if there's a way for autoloading, i'll look for when it's stable..): # insert the platform/DAI driver modprobe snd-soc-sunxi-i2s sleep 1 modprobe snd-soc-simple-card sleep 1 modprobe snd-soc-uda1380 sleep 1 then , i've a number of warning and error (it's normal as i'm in early debugging now..). the one for which i'm posting is related to "dapm routes" in the .../codecs/uda1380.c, these are errors in the log: [ 33.876179] uda1380-codec 1-0018: Control not supported for path Right PGA -> [Mic + Line R] -> Right ADC [ 33.885756] uda1380-codec 1-0018: ASoC: no dapm match for Right PGA --> Mic + Line R --> Right ADC [ 33.894719] uda1380-codec 1-0018: ASoC: Failed to add route Right PGA -> Mic + Line R -> Right ADC [ 33.903669] uda1380-codec 1-0018: Control not supported for path Right PGA -> [Line] -> Right ADC [ 33.912542] uda1380-codec 1-0018: ASoC: no dapm match for Right PGA --> Line --> Right ADC [ 33.920807] uda1380-codec 1-0018: ASoC: Failed to add route Right PGA -> Line -> Right ADC to clear them out, i had to patch the dapm_route structure of uda1380.c with: --- sound/soc/codecs/uda1380-orig.c 2015-05-26 12:41:58.650111434 +0200 +++ sound/soc/codecs/uda1380.c 2015-06-18 22:38:51.845892195 +0200 @@ -403,8 +403,11 @@ {"Input Mux", "Line", "Left PGA"}, /* right input */ - {"Right ADC", "Mic + Line R", "Right PGA"}, - {"Right ADC", "Line", "Right PGA"}, + {"Right ADC", NULL, "Input Mux"}, + {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Mic + Line R", "Right PGA"}, + {"Input Mux", "Line L", "Left PGA"}, + {"Input Mux", "Line", "Right PGA"}, /* inputs */ {"Mic LNA", NULL, "VINM"}, i added the same "Input Mux" DAPM widget as it's done on the left input of the codec, that's already bound to this "dapm widget" and is managing the couple of codec register bits at 0x22, SEL_LNA and SEL_MIC, that select one of the many input path (see the PDF to better understand.. i could add a shot but i don't know if it's policy permitted ) http://www.nxp.com/documents/data_sheet/UDA1380.pdf as far as i can see, it should be an issue independent by my "development" setup, but should happening, at module load time, to all the devices having such a codec and using a fairly recent mainline kernel in an ASOC environment; is that right? could you confirm this feeling? let me sincerely tell you that there's not that much documentation about Alsa ASoC, simple-audio-card, codec, DAIs, dapm, dts, up to date, or at least is scattered here and there. so it takes a bit of time to read, understand (maybe), code, test and debug this kind of drivers, really! i can understand there's lot's ongoing here, and API are on flux, for example i had to put together stuff from different card drivers as i'm willing to use also Dmaengine and regmap and don't remember whatever else, but many drivers are still frozen on previous API (i understand it would take lot's of janitoring effort and testing on regression cases..) of course, i still can't say this patch is good for something until i've worked out all the other issues and finally listened some good music out of this combo SOC+codec! :-) hope to post more about this setup when it's working.. bests

1 0

[alsa-devel] How to build alsa-driver ?
by Happy 18 Jun '15

18 Jun '15

Dear ALSA developers, I am trying to get into ALSA development. I need some help. I checked out to the head of the alsa-driver mirror branch (latest update 9 days ago). I made some changes to the usb/mixer_scarlett.c file. (just to test, change the text of the output channels only for the 18i20) Questions: How can I build this? How can I and update (Ubuntu 15.04) with this new build to this change? Thank you very much! Martin

1 0

[alsa-devel] About kernel tlv320aic32x4 driver
by Kimmo Saarela 18 Jun '15

18 Jun '15

I have some patches to that driver and recently I ported those changes to reasonably fresh kernel (4.1-rc7). There are some changes that are bit distractible to current workings of the driver. (mixer settings related) Current mainline driver does nothing to connect unused inputs to common mode (datasheet states that that shall be done) and in our testing that had big impact on recorded audio quality. I've added OF bindings that check which inputs to connect and where and then connect unused inputs to common. Should I just post my patches here or should I first test them against some other kernel version? - Kimmo S.

1 0

Re: [alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Raymond Yau 18 Jun '15

18 Jun '15

> > > > How do the other user know the mapping even if you implemented 5.1 ? > I don't know (?) https://rog.asus.com/forum/showthread.php?54685-Asus-G751JT-supports-5-1-an… Is the support of 5.1 done by realtek hda manager or asus audio wizard ?

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[alsa-devel] Default volume for all soundcards (-20dB by default?)
by Jo 17 Jun '15

17 Jun '15

Hi, It seems that the default volume for any new soundcard is set by default to -20dB. Do you know where is it hardcoded ? In which source file (I used package "alsa-utils" in Debian) or which configuration file? Thanks, Jo PS: here's the context : I woud like to set up the default sound volume once for all, for all ALSA devices that will be connected ever. Of course, I could do "amixer ..." or even "alsamixer" to modify the volume of _currently available_ soundcards. But I really want to modify the default volume even for future soundcards that will be added later. I've seen /var/lib/alsa/asound.state but the content is specific to _currently connected_ soundcards. What I want is a solution that will apply to any soundcard that will be connected in the future. Why do I want this? I'm providing a ready-to-use Debian image for my project http://www.samplerbox.org. User #1 might use the computer's built-in-soundcard, User #2 might have a USB DAC, User #3 might have another soundcard... I would like to provide a default -3dB volume that will work for any ALSA soundcard people could have...

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[alsa-devel] [PATCH] ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
by Axel Lin 17 Jun '15

17 Jun '15

Fix trivial typo. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Charles and Dimitris, This looks like a typo. Can you confirm? Thanks, Axel. sound/soc/codecs/wm8995.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 687c4dd..505b65f 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1930,7 +1930,7 @@ static int wm8995_set_dai_sysclk(struct snd_soc_dai *dai, dai->id + 1, freq); break; case WM8995_SYSCLK_MCLK2: - wm8995->sysclk[dai->id] = WM8995_SYSCLK_MCLK1; + wm8995->sysclk[dai->id] = WM8995_SYSCLK_MCLK2; wm8995->mclk[1] = freq; dev_dbg(dai->dev, "AIF%d using MCLK2 at %uHz\n", dai->id + 1, freq); -- 2.1.0

3 2

[alsa-devel] [PATCH v2] ASoC: wm8960: add dapm kcontrols to select left/right ADC input source
by Zidan Wang 17 Jun '15

17 Jun '15

Add dapm kcontrols to select left/right ADC input source, one to select the left ADC input source and one for the right ADC input source. In default, the left ADC will select the left input, and the right ADC will select the right input. When the left(right) ADC select the right(left) input, the left(right) input path will be powered down. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 023e898..3ea12b7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -141,6 +141,8 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adcl_src_text[] = {"Left", "Right"}; +static const char *wm8960_adcr_src_text[] = {"Right", "Left"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -150,6 +152,10 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), }; +static SOC_ENUM_SINGLE_DECL(wm8960_adcl_src_enum, + WM8960_ADDCTL1, 3, wm8960_adcl_src_text); +static SOC_ENUM_SINGLE_DECL(wm8960_adcr_src_enum, + WM8960_ADDCTL1, 2, wm8960_adcr_src_text); static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -281,6 +287,11 @@ SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), }; +static const struct snd_kcontrol_new wm8960_adcl_mux = + SOC_DAPM_ENUM("ADCL Source", wm8960_adcl_src_enum); +static const struct snd_kcontrol_new wm8960_adcr_mux = + SOC_DAPM_ENUM("ADCR Source", wm8960_adcr_src_enum); + static const struct snd_kcontrol_new wm8960_lin_boost[] = { SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0), SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0), @@ -344,6 +355,9 @@ SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0), SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), +SND_SOC_DAPM_MUX("ADCL Source", SND_SOC_NOPM, 0, 0, &wm8960_adcl_mux), +SND_SOC_DAPM_MUX("ADCR Source", SND_SOC_NOPM, 0, 0, &wm8960_adcr_mux), + SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0, &wm8960_loutput_mixer[0], ARRAY_SIZE(wm8960_loutput_mixer)), @@ -399,8 +413,14 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "Right Input Mixer", NULL, "RINPUT2" }, { "Right Input Mixer", NULL, "RINPUT3" }, - { "Left ADC", NULL, "Left Input Mixer" }, - { "Right ADC", NULL, "Right Input Mixer" }, + { "ADCL Source", "Left", "Left Input Mixer" }, + { "ADCL Source", "Right", "Right Input Mixer" }, + + { "ADCR Source", "Left", "Left Input Mixer" }, + { "ADCR Source", "Right", "Right Input Mixer" }, + + { "Left ADC", NULL, "ADCL Source" }, + { "Right ADC", NULL, "ADCR Source" }, { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" }, { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} , -- 1.9.1

2 3

[alsa-devel] [PATCH 1/3] ALSA: hda: add hdac_ext stream creation and cleanup routines
by Vinod Koul 17 Jun '15

17 Jun '15

HDAC extended core should create streams for an extended bus and also free up those on cleanup. So introduce snd_hdac_ext_stream_init_all and snd_hdac_stream_free_all routines Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hdaudio_ext.h | 3 +++ sound/hda/ext/hdac_ext_stream.c | 50 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 53 insertions(+) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 202350a8eddb..877661629322 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -87,6 +87,9 @@ struct hdac_ext_stream { void snd_hdac_ext_stream_init(struct hdac_ext_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag); +int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, + int num_stream, int dir); +void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus); struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *bus, struct snd_pcm_substream *substream, int type); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 0677bb65a494..f8ffbdbb450d 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -18,6 +18,7 @@ */ #include <linux/delay.h> +#include <linux/slab.h> #include <sound/pcm.h> #include <sound/hda_register.h> #include <sound/hdaudio_ext.h> @@ -54,6 +55,55 @@ void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); /** + * snd_hdac_ext_stream_init_all - create and initialize the stream objects + * for an extended hda bus + * @ebus: HD-audio ext core bus + * @start_idx: start index for streams + * @num_stream: number of streams to initialize + * @dir: direction of streams + */ +int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, + int num_stream, int dir) +{ + int stream_tag = 0; + int i, tag, idx = start_idx; + + for (i = 0; i < num_stream; i++) { + struct hdac_ext_stream *stream = + kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + tag = ++stream_tag; + snd_hdac_ext_stream_init(ebus, stream, idx, dir, tag); + idx++; + } + + return 0; + +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); + +/** + * snd_hdac_stream_free_all - free hdac extended stream objects + * + * @ebus: HD-audio ext core bus + */ +void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus) +{ + struct hdac_stream *s; + struct hdac_ext_stream *stream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + while (!list_empty(&bus->stream_list)) { + s = list_first_entry(&bus->stream_list, struct hdac_stream, list); + stream = stream_to_hdac_ext_stream(s); + list_del(&s->list); + kfree(stream); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); + +/** * snd_hdac_ext_stream_decouple - decouple the hdac stream * @ebus: HD-audio ext core bus * @stream: HD-audio ext core stream object to initialize -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: rsrc-card: remove unused ret
by Kuninori Morimoto 17 Jun '15

17 Jun '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> '047000278da3a17f8("ASoC: rsrc-card: cleanup for DPCM")' cleanuped rsrc-card driver, but then, unused ret was left. Below warning happen without this patch ${LINUX}/sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup': ${LINUX}/sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable \ 'ret' [-Wunused-variable] Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 8caca2e..84e9357 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -75,8 +75,6 @@ static int rsrc_card_startup(struct snd_pcm_substream *substream) struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, rtd - rtd->card->rtd); - int ret; - return clk_prepare_enable(dai_props->clk); } -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
by Vinod Koul 16 Jun '15

16 Jun '15

Since this is common option for HDA driver to specfiy pre-allocated buffer, we should make this option availble to all HDA driver by moving this to HDA core Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/Kconfig | 13 +++++++++++++ sound/pci/hda/Kconfig | 13 ------------- 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 6dc3914fd28b..3129546398d0 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -14,3 +14,16 @@ config SND_HDA_I915 config SND_HDA_EXT_CORE tristate select SND_HDA_CORE + +config SND_HDA_PREALLOC_SIZE + int "Pre-allocated buffer size for HD-audio driver" + range 0 32768 + default 64 + help + Specifies the default pre-allocated buffer-size in kB for the + HD-audio driver. A larger buffer (e.g. 2048) is preferred + for systems using PulseAudio. The default 64 is chosen just + for compatibility reasons. + + Note that the pre-allocation size can be changed dynamically + via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 98ced971cb43..e94cfd5c69f7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -38,19 +38,6 @@ config SND_HDA_TEGRA if SND_HDA -config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" - range 0 32768 - default 64 - help - Specifies the default pre-allocated buffer-size in kB for the - HD-audio driver. A larger buffer (e.g. 2048) is preferred - for systems using PulseAudio. The default 64 is chosen just - for compatibility reasons. - - Note that the pre-allocation size can be changed dynamically - via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. - config SND_HDA_HWDEP bool "Build hwdep interface for HD-audio driver" select SND_HWDEP -- 1.9.1

2 1

[alsa-devel] [PATCH v5 0/4] ASoC: intel - add skylake PCM driver
by Vinod Koul 16 Jun '15

16 Jun '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch This version removes the bus and match function as that is merged in HDA core now and we have extended HDA bus merged as well Please note this series is dependent on sound next having hda core patches and extended bus patches Jeeja KP (4): ASoC: Intel: add Skylake HDA platform driver ASoC: Intel - add Skylake HDA audio driver ASoC: Intel - add makefile support for SKL driver ASoC: intel - adds support for decoupled mode in skl driver sound/soc/intel/Kconfig | 17 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 936 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 646 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 74 +++ 6 files changed, 1677 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

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[alsa-devel] ASoC : WM5102 Audio Codec needs ACPI support and a baytrail machine driver
by Christian Hartmann 16 Jun '15

16 Jun '15

Hi list, I have a device "Lenovo Yoga 851 F" which have a WM510205 Audio codec on a Baytrail System. I have sent already one patch (spi-pxa2xx) raise num_chipselect by one, which fixes to bind the arizona WM5102 codec as a spi slave attachted on a pxa2xx spi master device. I have tried several times and added the ACPI id "WM510205" to the arizona-spi.c driver. The ACPI ID is extracted from the DSDT.dsl of this machine. But unfortunately it fails and I got only a NULL pointer deference error in arizona_spi_probe(). I want to request ACPI support for this device and I cannot fix currently the NULL pointer. I do not know yet, if I have to add this ACPI id to the arizona-spi driver (sure, thats what I am trying and failing) or to the NEW baytrail machine driver. In this case it should be the bytcr_wm5102.c and byt-wm5102.c, which does not exists yet. I have a branched tovalds/master and added (my first attempt) the above mentioned machine driver on a local branch. The result is only a dsp boot timeout error. Any help, ideas, hints, patches to try would be appreciate sincerely yours Christian Hartmann a snap of the dsdt Device (LPEA) { 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, "17AA7004") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_DEP, Package (0x01) // _DEP: Dependencies { ^SPI1.AUDI }) ... ... Device (AUDI) { Name (_HID, "WM510205") // _HID: Hardware ID Name (_CID, "WM510205") // _CID: Compatible ID Name (_DDN, "Wolfson Microelectronics Audio WM5102") // _DDN: DOS Device Name Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , ) GpioInt (Edge, ActiveLow, ExclusiveAndWake, PullNone, 0x0000, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.I2C7.PMIC", 0x00, ResourceConsumer, , ) { // Pin list 0x0003 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO1", 0x00, ResourceConsumer, , ) { // Pin list 0x0017 } })

1 1

[alsa-devel] [PATCH v2 0/3] ASoC: Mediatek: Add support for MT8173 SOC
by Koro Chen 16 Jun '15

16 Jun '15

This adds basic support for the Mediatek AFE (Audio Front End) unit of MT8173. This patch is based on Linux 4.1-rc1 and Sascha's SCPSYS power patch [1]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). [1] https://lkml.org/lkml/2015/6/9/172 changes since v1: - change Kconfig to tristate - remove unnecessary !! in the platform driver - let MODULE_DEVICE_TABLE follow the table - remove regulators from machine drivers changes since RFC: - Use DPCM to model AFE AFE memory interfaces such as DL1, VUL are modeled as frontend DAIs, and AFE IOs such as I2S, HDMI are backend DAIs. Which memif and IO to be used can be set in the machine driver. There is no need to indicate this in dts anymore so include/dt-bindings/sound/mtk-afe.h is removed. AFE connections are represented by DAPM widgets and routes, and set connection APIs/tables are no longer required. (mtk-afe-connection.c and mtk-afe-connection.h are removed)

2 6

[alsa-devel] [PATCH 0/21] ASoC: rsnd: prepare for MIXer support
by Kuninori Morimoto 16 Jun '15

16 Jun '15

Hi Mark These are prepare for MIXer support for Renesas sound card. Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths if it supports MIXer. In such case, mod <-> stream is no longer 1:1 relationship. this patch set don't use rsnd_mod_to_io() as prepare for MIXer support 1) - 2) bugfix patch for PIO mode. These are not critical bug since no platform is using PIO mode. 3) - 10) cleanup for DPCM related code for MIXer support 11) - 21) remove rsnd_mod_to_io() from Renesas sound driver Kuninori Morimoto (21): 1) ASoC: rsnd: don't care under/over run error when PIO 2) ASoC: rsnd: don't call snd_pcm_period_elapsed() under spin lock 3) ASoC: rsrc-card: used "fe.xxx"/"be.xxx" name for dai_link 4) ASoC: rsrc-card: tidyup priv->snd_card setup timing 5) ASoC: rsrc-card: enable multi cpu/codec for DPCM 6) ASoC: rsrc-card: remove unused name 7) ASoC: rsnd-card: tidyup rsrc_priv_to_() 8) ASoC: rsrc-card: move rsrc_card_parse_daifmt() to upper side 9) ASoC: rsrc-card: tidyup return value of clock error 10) ASoC: rsrc-card: cleanup for DPCM 11) ASoC: rsnd: count each mod (SSI/SRC/DVC) 12) ASoC: rsnd: rsnd_mod has rsnd_priv 13) ASoC: rsnd: adds struct rsnd_dai_stream as on each fuction as parameter 14) ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA 15) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr() 16) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx() 17) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx() 18) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx() 19) ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol 20) ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working() 21) ASoC: rsnd: remove io from rsnd_mod sound/soc/sh/rcar/core.c | 111 ++++++++++++++++++++++++--------- sound/soc/sh/rcar/dma.c | 113 ++++++++++++++++++++-------------- sound/soc/sh/rcar/dvc.c | 30 +++++---- sound/soc/sh/rcar/rsnd.h | 112 +++++++++++++++++++++------------ sound/soc/sh/rcar/rsrc-card.c | 438 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------------------------------------- sound/soc/sh/rcar/src.c | 121 ++++++++++++++++++++---------------- sound/soc/sh/rcar/ssi.c | 96 +++++++++++++++++------------ 7 files changed, 558 insertions(+), 463 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/4] ASoC: wm8960: codec mclk should be enabled early to avoid jack detect error
by Zidan Wang 16 Jun '15

16 Jun '15

It will playback from speaker in the first 2 seconds, then switch to headphone. Steps to reproduce this issue: 1. plug out headphone and playback a wav. 2. stop playback and wait for at least 5 seconds, then plug in headphone and playback a wav. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 73 +++++++++++++++++++++-------------------------- 1 file changed, 33 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 761418f..729205f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -491,6 +491,34 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) return 0; } +int wm8960_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + return 0; +} + +void wm8960_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); +} + static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -702,38 +730,14 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int ret; switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - switch (snd_soc_codec_get_bias_level(codec)) { - case SND_SOC_BIAS_STANDBY: - if (!IS_ERR(wm8960->mclk)) { - ret = clk_prepare_enable(wm8960->mclk); - if (ret) { - dev_err(codec->dev, - "Failed to enable MCLK: %d\n", - ret); - return ret; - } - } - - /* Set VMID to 2x50k */ - snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); - break; - - case SND_SOC_BIAS_ON: - if (!IS_ERR(wm8960->mclk)) - clk_disable_unprepare(wm8960->mclk); - break; - - default: - break; - } - + /* Set VMID to 2x50k */ + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_STANDBY: @@ -780,7 +784,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg, ret; + int reg; switch (level) { case SND_SOC_BIAS_ON: @@ -821,22 +825,9 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); - - if (!IS_ERR(wm8960->mclk)) { - ret = clk_prepare_enable(wm8960->mclk); - if (ret) { - dev_err(codec->dev, - "Failed to enable MCLK: %d\n", - ret); - return ret; - } - } break; case SND_SOC_BIAS_ON: - if (!IS_ERR(wm8960->mclk)) - clk_disable_unprepare(wm8960->mclk); - /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -1059,6 +1050,8 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8960_dai_ops = { + .startup = wm8960_startup, + .shutdown = wm8960_shutdown, .hw_params = wm8960_hw_params, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, -- 1.9.1

2 4

[alsa-devel] [PATCH 1/2] ASoC: soc-compress: add a config item for soc-compress
by Jie Yang 16 Jun '15

16 Jun '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, please add 'select SND_SOC_COMPRESS' to driver Kconfig when it is needed. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 7 +++++++ sound/soc/Kconfig | 6 +++++- sound/soc/Makefile | 3 ++- 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..03c53a9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,14 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#else +static inline int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + return -EPERM; +} +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index e2828e1..de743d9 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,11 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + tristate + select SND_COMPRESS_OFFLOAD + default n + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index a0e1ee6..184c1e6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,7 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-objs += soc-topology.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o -- 1.9.1

4 11

[alsa-devel] [PATCH 2/4] ASoC: wm8960: support configure headphone jack detect pin and polarity from device tree
by Zidan Wang 16 Jun '15

16 Jun '15

The ADCLRC/GPIO1, LINPUT3/JD2 and RINPUT3/JD3 pins can be selected as headphone jack detect inputs to automatically disable the speaker output and enable the headphone. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 11 +++++++++++ include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 22 ++++++++++++++++++++++ 3 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt index 2deb8a3..4f1aa8b 100644 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -21,6 +21,16 @@ Optional properties: enabled and disabled together with HP_L and HP_R pins in response to jack detect events. + - wlf,hp-det: ADCLRC/GPIO1, LINPUT3/JD2 and RINPUT3/JD3 pins can be selected as + headphone jack detect inputs to automatically disable the speaker output + and enable the headphone output. + hp-det = <hp-det-pin hp-det-polarity>; + hp-det-pin = 1: ADCLRC/GPIO1 used as detect pin + hp-det-pin = 2: LINPUT3/JD2 used as detect pin + hp-det-pin = 3: RINPUT3/JD3 used as detect pin + hp-det-polarity = 0: hp detect high for headphone + hp-det-polarity = 1: hp detect high for speaker + Example: codec: wm8960@1a { @@ -28,4 +38,5 @@ codec: wm8960@1a { reg = <0x1a>; wlf,shared-lrclk; + wlf,hp-det = <2 0> }; diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..a787376 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -19,6 +19,7 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ + u32 hp_det[2]; /* headphone detect pin and polarity */ }; #endif diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 729205f..c1be15c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1123,6 +1123,11 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + of_property_read_u32_array(np, "wlf,hp-det", pdata->hp_det, 2); + + if (pdata->hp_det[0] > 3) + pdata->hp_det[0] = 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1168,6 +1173,23 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, } } + /* + * ADCLRC/GPIO1 pin configured as GPIO1 to select jack + * detect input function + */ + if (wm8960->pdata.hp_det[0] == 1) + regmap_update_bits(wm8960->regmap, WM8960_IFACE2, 1<<6, 1<<6); + + /* Enable headphone jack detect */ + if (wm8960->pdata.hp_det[0] != 0) { + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 1<<6, 1<<6); + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 1<<5, + wm8960->pdata.hp_det[1]<<5); + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL4, 3<<2, + wm8960->pdata.hp_det[0]<<2); + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL1, 3, 3); + } + /* Latch the update bits */ regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100); regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100); -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: DAPM: fix build errors for missing snd_soc_dapm_new_control symbol
by Liam Girdwood 16 Jun '15

16 Jun '15

Fix the following error:- All error/warnings (new ones prefixed by >>): > > sound/built-in.o: In function `soc_tplg_dapm_widget_create': > >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control' Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 92d57a9..1b4a6eb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -53,7 +53,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink)); -static struct snd_soc_dapm_widget * +struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); @@ -3270,7 +3270,7 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -static struct snd_soc_dapm_widget * +struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { -- 2.1.4

3 4

[alsa-devel] [PATCH 0/4] ASoC: constification patches
by Mathias Krause 15 Jun '15

15 Jun '15

Hi Liam, Mark, this series constifies a few variables in the SoC code that are only read, namely struct reg_default and struct acpi_device_id. Please apply! Mathias Krause (4): ASoC: max98090: Constify ACPI device ids and register map ASoC: rt5640: Constify ACPI device ids and register map ASoC: rt5670: Constify ACPI device ids and register map ASoC: intel: Constify ACPI device ids sound/soc/codecs/max98090.c | 4 ++-- sound/soc/codecs/rt5640.c | 5 ++--- sound/soc/codecs/rt5670.c | 5 ++--- sound/soc/intel/common/sst-acpi.c | 2 +- 4 files changed, 7 insertions(+), 9 deletions(-) -- 1.7.10.4

3 10

[alsa-devel] [PATCH] ASoC: rt5645: move RT5650 muxes to rt5650_specific_dapm_widgets
by Oder Chiou 15 Jun '15

15 Jun '15

Developing a driver for an Asus X205TA laptop I get these dmesg errors: rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC1 Swap Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC2 Swap Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC3 Swap Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 L Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 R Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 L Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 R Mux has no paths so, move these muxes to the rt5650_specific_dapm_widgets[] list. Signed-off-by: Michele Curti <michele.curti(a)gmail.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++----------------- 1 file changed, 18 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4344043..9ce311e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1706,15 +1706,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if1_adc_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC1 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc1_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC2 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc2_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC3 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc3_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc_in_mux), - SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if2_adc_in_mux), @@ -1733,14 +1724,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { &rt5645_if1_dac2_tdm_sel_mux), SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, &rt5645_if1_dac3_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac0_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac1_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac2_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac3_tdm_sel_mux), SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1882,6 +1865,24 @@ static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { 0, 0, &rt5650_a_dac2_l_mux), SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, 0, 0, &rt5650_a_dac2_r_mux), + + SND_SOC_DAPM_MUX("RT5650 IF1 ADC1 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc1_in_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 ADC2 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 ADC3 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc3_in_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc_in_mux), + + SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac3_tdm_sel_mux), }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] [PATCH] mips/ad1843: convert time to jiffies HZ independent
by Nicholas Mc Guire 15 Jun '15

15 Jun '15

PI compliance scanning with coccinelle flagged: ./sound/mips/ad1843.c:503:2-32: WARNING: timeout (5) seems HZ dependent This was introduced in 'commit 862c2c0a61c5 ("ALSA: ALSA driver for SGI O2 audio board")'. schedule_timeout_interruptible() expects a timeout in jiffies so the numeric constant makes the effective timeout HZ dependent. Simply put it through msecs_to_jiffies() to make it HZ independent. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- The loop here is bounded by 500 jiffies and it does not seem reasonable that it would loop 10 times faster on HZ=1000 configs over HZ=100 configs. The conversion via msecs_to_jiffies(5) ensures that it will loop at most 100 times for all configs of HZ which seems reasonable (to me) but needs to be checked by someone that knows the details of this driver. Note that the CONFIG_SND_PCM_OSS dependency was not listed in the Depends on: in Kconfig (which it probably should) Patch was compile tested with ip32_defconfig + CONFIG_SOUND=m, CONFIG_SND=m, CONFIG_SND_PCM_OSS=y, CONFIG_SND_SGI_O2=m Patch is against 4.1-rc7 (localversion-next is -next-20150612) sound/mips/ad1843.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c index 5869075..ec22b55 100644 --- a/sound/mips/ad1843.c +++ b/sound/mips/ad1843.c @@ -500,7 +500,7 @@ int ad1843_init(struct snd_ad1843 *ad1843) "ad1843: AD1843 won't power up\n"); return -EIO; } - schedule_timeout_interruptible(5); + schedule_timeout_interruptible(msecs_to_jiffies(5)); } /* 5. Power up the clock generators and enable clock output pins. */ -- 1.7.10.4

3 5

[alsa-devel] [PATCH] ALSA: hda - Fix noisy outputs on Dell XPS13 (2015 model)
by Takashi Iwai 15 Jun '15

15 Jun '15

The new Dell XPS13 also requires the similar quirk for fixing the noisy outputs. (But, as the codec was changed, now the fixup for Latitude is used instead.) Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 919051d92a0b..6d010452c1f5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5072,6 +5072,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.4.3

1 0

[alsa-devel] [PATCH] ASoC: wm8960: add two kcontrols to select ADC left/right channel source
by Zidan Wang 15 Jun '15

15 Jun '15

Add two kcontrols to select ADC left/right channel source, one to select the left channel source and one for the right channel source. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 023e898..8f9aed1 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -141,6 +141,8 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adc_left_source[] = {"left ADC", "right ADC"}; +static const char *wm8960_adc_right_source[] = {"right ADC", "left ADC"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -149,6 +151,8 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), + SOC_ENUM_SINGLE(WM8960_ADDCTL1, 3, 2, wm8960_adc_left_source), + SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 2, wm8960_adc_right_source), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -279,6 +283,9 @@ SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", WM8960_BYPASS2, 4, 7, 1, bypass_tlv), SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), + +SOC_ENUM("ADC Left Channel Source", wm8960_enum[6]), +SOC_ENUM("ADC Right Channel Source", wm8960_enum[7]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: Add max98926 codec driver
by Anish Kumar 15 Jun '15

15 Jun '15

Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98926.txt | 32 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98926.c | 727 ++++++++++++++++++ sound/soc/codecs/max98926.h | 846 +++++++++++++++++++++ 5 files changed, 1611 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 sound/soc/codecs/max98926.c create mode 100644 sound/soc/codecs/max98926.h diff --git a/Documentation/devicetree/bindings/sound/max98926.txt b/Documentation/devicetree/bindings/sound/max98926.txt new file mode 100644 index 0000000..ee26f7d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98926.txt @@ -0,0 +1,32 @@ +max98926 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98926" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + + - imon-slot-no : slot number used to send current information + + - interleave-mode : When using two MAX98926 in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98926@1a { + compatible = "maxim,max98926"; + vmon-slot-no = <0>; + imon-slot-no = <2>; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..10c9c46 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX98925 if I2C + select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C @@ -465,6 +466,9 @@ config SND_SOC_MAX98357A config SND_SOC_MAX98925 tristate +config SND_SOC_MAX98926 + tristate + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..8768c42 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -66,6 +66,7 @@ snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o snd-soc-max98925-objs := max98925.o +snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o @@ -251,6 +252,7 @@ obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o +obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c new file mode 100644 index 0000000..c70cbc8 --- /dev/null +++ b/sound/soc/codecs/max98926.c @@ -0,0 +1,727 @@ +/* + * max98926.c -- ALSA SoC MAX98926 driver + * Copyright 2013-15 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98926.h" + +static const char * const max98926_boost_voltage_txt[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static const char *const max98926_dai_txt[] = { + "Left", "Right", "LeftRight", "LeftRightDiv2", +}; + +static const char *const pdm_text[] = { + "Current", "Voltage", +}; + +static const char *const max98926_hpf_cutoff_txt[] = { + "hpf_disable", "hpf_dc_block", "hpf_enable_100", + "hpf_enable_200", "hpf_enable_400", "hpf_enable_800", +}; + +static const char *const max98926_input_txt[] = { + "Pcm", "Analog", +}; + +static struct reg_default max98926_reg[] = { + { 0x00, 0x00 }, /* Battery Voltage Data */ + { 0x01, 0x00 }, /* Boost Voltage Data */ + { 0x02, 0x00 }, /* Live Status0 */ + { 0x03, 0x00 }, /* Live Status1 */ + { 0x04, 0x00 }, /* Live Status2 */ + { 0x05, 0x00 }, /* State0 */ + { 0x06, 0x00 }, /* State1 */ + { 0x07, 0x00 }, /* State2 */ + { 0x08, 0x00 }, /* Flag0 */ + { 0x09, 0x00 }, /* Flag1 */ + { 0x0A, 0x00 }, /* Flag2 */ + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x04 }, /* DAI Clock Mode 1 */ + { 0x1B, 0x00 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ + { 0xFF, 0x50 }, /* Revision ID */ +}; + +static int max98926_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(max98926->regmap, + MAX98926_BLOCK_ENABLE, MAX98926_BST_EN_MASK | + MAX98926_ADC_IMON_EN_MASK | + MAX98926_ADC_VMON_EN_MASK, + MAX98926_BST_EN_MASK | + MAX98926_ADC_IMON_EN_MASK | + MAX98926_ADC_VMON_EN_MASK); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98926->regmap, + MAX98926_BLOCK_ENABLE, MAX98926_BST_EN_MASK | + MAX98926_ADC_IMON_EN_MASK | + MAX98926_ADC_VMON_EN_MASK, 0); + break; + default: + return 0; + } + return 0; +} + +static const struct snd_soc_dapm_widget max98926_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("Amp Enable", NULL, MAX98926_BLOCK_ENABLE, + MAX98926_SPK_EN_SHIFT, 0, max98926_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98926_GLOBAL_ENABLE, + MAX98926_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98926_audio_map[] = { + {"Amp Enable", NULL, "DAI_OUT"}, + {"BE_OUT", NULL, "Global Enable"}, + {"BE_OUT", NULL, "Amp Enable"}, +}; + +static bool max98926_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98926_VBAT_DATA: + case MAX98926_VBST_DATA: + case MAX98926_LIVE_STATUS0: + case MAX98926_LIVE_STATUS1: + case MAX98926_LIVE_STATUS2: + case MAX98926_STATE0: + case MAX98926_STATE1: + case MAX98926_STATE2: + case MAX98926_FLAG0: + case MAX98926_FLAG1: + case MAX98926_FLAG2: + case MAX98926_VERSION: + return true; + default: + return false; + } +} + +static bool max98926_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98926_IRQ_CLEAR0: + case MAX98926_IRQ_CLEAR1: + case MAX98926_IRQ_CLEAR2: + case MAX98926_ALC_HOLD_RLS: + return false; + default: + return true; + } +}; + +DECLARE_TLV_DB_SCALE(max98926_spk_tlv, -600, 100, 0); + +static int pdm_ch_zero; +static int pdm_ch_one; + +int max98926_common_pdm(struct max98926_priv *max98926, int channel, int source) +{ + int ret = 0; + int reg = MAX98926_DAI_CLK_DIV_N_LSBS; + + /* enable channel */ + if (channel == 0) { + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_CHANNEL_0_MASK, + MAX98926_PDM_CHANNEL_0_MASK); + } else { + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_CHANNEL_1_MASK, + MAX98926_PDM_CHANNEL_1_MASK); + } + switch (source) { + case 0: + /* enable current */ + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_CURRENT_MASK, + MAX98926_PDM_CURRENT_MASK); + /* enable source*/ + if (channel == 0) + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_SOURCE_0_MASK, + MAX98926_PDM_SOURCE_0_MASK); + else + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_SOURCE_1_MASK, + MAX98926_PDM_SOURCE_1_MASK); + break; + case 1: + /* enable voltage */ + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_VOLTAGE_MASK, + MAX98926_PDM_VOLTAGE_MASK); + /* enable source*/ + if (channel == 0) + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_SOURCE_0_MASK, 0); + else + regmap_update_bits(max98926->regmap, + reg, MAX98926_PDM_SOURCE_1_MASK, 0); + break; + default: + return -EINVAL; + } + return ret; +} + +static int max98926_get_pdm_ch_zero(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = pdm_ch_zero; + return 0; +} + +static int max98926_put_pdm_ch_zero(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + int ret; + + pdm_ch_zero = ucontrol->value.integer.value[0]; + ret = max98926_common_pdm(max98926, 0, pdm_ch_zero); + if (ret < 0) + return -EINVAL; + return 0; +} + +static int max98926_get_pdm_ch_one(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = pdm_ch_one; + return 0; +} + +static int max98926_put_pdm_ch_one(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + int ret; + + pdm_ch_one = ucontrol->value.integer.value[0]; + ret = max98926_common_pdm(max98926, 1, pdm_ch_one); + if (ret < 0) + return -EINVAL; + return 0; +} + +static const struct soc_enum max98926_global_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(pdm_text), pdm_text), +}; + +static SOC_ENUM_SINGLE_DECL(max98926_dac_hpf_cutoff, + MAX98926_FILTERS, MAX98926_DAC_HPF_SHIFT, + max98926_hpf_cutoff_txt); + +static SOC_ENUM_SINGLE_DECL(max98926_input_sel, + MAX98926_SPK_AMP, MAX98926_INSELECT_MODE_SHIFT, + max98926_input_txt); + +static SOC_ENUM_SINGLE_DECL(max98926_dai_input_sel, + MAX98926_GAIN, + MAX98926_DAC_IN_SEL_SHIFT, + max98926_dai_txt); + +static SOC_ENUM_SINGLE_DECL(max98926_boost_voltage, + MAX98926_CONFIGURATION, MAX98926_BST_VOUT_SHIFT, + max98926_boost_voltage_txt); + +static const struct snd_kcontrol_new max98926_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98926_GAIN, + MAX98926_SPK_GAIN_SHIFT, (1<<MAX98926_SPK_GAIN_WIDTH)-1, 0, + max98926_spk_tlv), + SOC_SINGLE("Ramp Switch", MAX98926_GAIN_RAMPING, + MAX98926_SPK_RMP_EN_SHIFT, 1, 0), + SOC_SINGLE("ZCD Switch", MAX98926_GAIN_RAMPING, + MAX98926_SPK_ZCD_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Switch", MAX98926_THRESHOLD, + MAX98926_ALC_EN_SHIFT, 1, 0), + SOC_SINGLE("ALC Threshold", MAX98926_THRESHOLD, MAX98926_ALC_TH_SHIFT, + (1<<MAX98926_ALC_TH_WIDTH)-1, 0), + SOC_ENUM("Boost Output Voltage", max98926_boost_voltage), + SOC_ENUM("Input Sel", max98926_input_sel), + SOC_ENUM("DAC HPF Cutoff", max98926_dac_hpf_cutoff), + SOC_ENUM("DAI Input Sel", max98926_dai_input_sel), + SOC_ENUM_EXT("PDM CH0 Enable", max98926_global_enum[0], + max98926_get_pdm_ch_zero, max98926_put_pdm_ch_zero), + SOC_ENUM_EXT("PDM CH1 Enable", max98926_global_enum[0], + max98926_get_pdm_ch_one, max98926_put_pdm_ch_one), +}; + +static const struct { + int rate; + int sr; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + }, + { + .rate = 11025, + .sr = 1, + }, + { + .rate = 12000, + .sr = 2, + }, + { + .rate = 16000, + .sr = 3, + }, + { + .rate = 22050, + .sr = 4, + }, + { + .rate = 24000, + .sr = 5, + }, + { + .rate = 32000, + .sr = 6, + }, + { + .rate = 44100, + .sr = 7, + }, + { + .rate = 48000, + .sr = 8, + }, +}; + +static void max98926_set_sense_data(struct max98926_priv *max98926) +{ + if (max98926->intrleave_mode) { + /* set VMON slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VMON, + MAX98926_DAI_VMON_EN_MASK, MAX98926_DAI_VMON_EN_MASK); + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VMON, + MAX98926_DAI_VMON_SLOT_MASK, + max98926->v_slot); + /* set IMON slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_IMON, + MAX98926_DAI_IMON_EN_MASK, MAX98926_DAI_IMON_EN_MASK); + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_IMON, + MAX98926_DAI_IMON_SLOT_MASK, + max98926->i_slot); + } else { + /* enable interleave mode */ + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_INTERLEAVE_MASK, + MAX98926_DAI_INTERLEAVE_MASK); + /* set interleave slots */ + regmap_update_bits(max98926->regmap, + MAX98926_DOUT_CFG_VBAT, + MAX98926_DAI_INTERLEAVE_SLOT_MASK, + max98926->v_slot); + } +} + +static int max98926_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98926_set_sense_data(max98926); + break; + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + default: + dev_err(codec->dev, "DAI format unsupported, fmt:0x%x", fmt); + return -EINVAL; + } + dev_dbg(codec->dev, "DAI format supported, fmt:0x%x", fmt); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = MAX98926_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = MAX98926_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = MAX98926_DAI_BCI_MASK | MAX98926_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98926->regmap, MAX98926_FORMAT, + MAX98926_DAI_BCI_MASK, invert); + return 0; +} + +static int max98926_set_clock(struct max98926_priv *max98926, + struct snd_pcm_hw_params *params) +{ + int dai_sr = -EINVAL, clock; + int rate = params_rate(params), i; + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98926->ch_size; + + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, MAX98926_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, MAX98926_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_BSEL_MASK, MAX98926_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (max98926->sysclk) { + case 6000000: + clock = 0; + break; + case 11289600: + clock = 1; + break; + case 12000000: + clock = 0; + break; + case 12288000: + clock = 2; + break; + default: + dev_info(max98926->codec->dev, "unsupported sysclk %d\n", + max98926->sysclk); + return -EINVAL; + } + + /* find the closest rate */ + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + dai_sr = rate_table[i].sr; + break; + } + } + if (dai_sr < 0) + return -EINVAL; + + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98926->regmap, + MAX98926_DAI_CLK_MODE2, + MAX98926_DAI_SR_MASK, dai_sr << MAX98926_DAI_SR_SHIFT); + return 0; +} + +static int max98926_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_16); + max98926->ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_24); + max98926->ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98926->regmap, + MAX98926_FORMAT, + MAX98926_DAI_CHANSZ_MASK, + MAX98926_DAI_CHANSZ_32); + max98926->ch_size = 32; + break; + default: + dev_dbg(codec->dev, "format unsupported %d", + params_format(params)); + return -EINVAL; + } + return max98926_set_clock(max98926, params); +} + +static int max98926_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + + max98926->sysclk = freq; + return 0; +} + +#define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98926_dai_ops = { + .set_sysclk = max98926_dai_set_sysclk, + .set_fmt = max98926_dai_set_fmt, + .hw_params = max98926_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98926_dai[] = { + { + .name = "max98926-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98926_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98926_FORMATS, + }, + .ops = &max98926_dai_ops, + } +}; + +static int max98926_probe(struct snd_soc_codec *codec) +{ + int reg, ret; + struct max98926_priv *max98926 = snd_soc_codec_get_drvdata(codec); + + max98926->codec = codec; + ret = regmap_read(max98926->regmap, + MAX98926_VERSION, &reg); + if ((ret < 0) || + ((reg != MAX98926_CHIP_VERSION) && + (reg != MAX98926_CHIP_VERSION1))) { + dev_err(codec->dev, "Failed to read: %x\n", reg); + return -EINVAL; + } + codec->control_data = max98926->regmap; + regmap_write(max98926->regmap, MAX98926_GLOBAL_ENABLE, 0x00); + /* It's not the default but we need to set DAI_DLY */ + regmap_write(max98926->regmap, + MAX98926_FORMAT, MAX98926_DAI_DLY_MASK); + regmap_write(max98926->regmap, MAX98926_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98926->regmap, MAX98926_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98926->regmap, MAX98926_FILTERS, 0xD8); + regmap_write(max98926->regmap, MAX98926_ALC_CONFIGURATION, 0xF8); + regmap_write(max98926->regmap, MAX98926_CONFIGURATION, 0xF0); + /* Disable ALC muting */ + regmap_write(max98926->regmap, MAX98926_BOOST_LIMITER, 0xF8); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98926 = { + .probe = max98926_probe, + .controls = max98926_snd_controls, + .num_controls = ARRAY_SIZE(max98926_snd_controls), + .dapm_routes = max98926_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98926_audio_map), + .dapm_widgets = max98926_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98926_dapm_widgets), +}; + +static struct regmap_config max98926_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98926_VERSION, + .reg_defaults = max98926_reg, + .num_reg_defaults = ARRAY_SIZE(max98926_reg), + .volatile_reg = max98926_volatile_register, + .readable_reg = max98926_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98926_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret; + u32 value; + struct max98926_priv *max98926; + + max98926 = devm_kzalloc(&i2c->dev, + sizeof(*max98926), GFP_KERNEL); + if (!max98926) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98926); + max98926->regmap = devm_regmap_init_i2c(i2c, &max98926_regmap); + if (IS_ERR(max98926->regmap)) { + ret = PTR_ERR(max98926->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err_out; + } + if (of_property_read_bool(i2c->dev.of_node, "interleave-mode")) + max98926->intrleave_mode = true; + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { + if (value > MAX98926_DAI_VMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "vmon slot number is wrong:\n"); + return -EINVAL; + } + max98926->v_slot = value; + } + if (!of_property_read_u32(i2c->dev.of_node, "imon-slot-no", &value)) { + if (value > MAX98926_DAI_IMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "imon slot number is wrong:\n"); + return -EINVAL; + } + max98926->i_slot = value; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98926, + max98926_dai, ARRAY_SIZE(max98926_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); +err_out: + return ret; +} + +static int max98926_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98926_i2c_id[] = { + { "max98926", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98926_i2c_id); + +static const struct of_device_id max98926_of_match[] = { + { .compatible = "maxim,max98926", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98926_of_match); + +static struct i2c_driver max98926_i2c_driver = { + .driver = { + .name = "max98926", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98926_of_match), + .pm = NULL, + }, + .probe = max98926_i2c_probe, + .remove = max98926_i2c_remove, + .id_table = max98926_i2c_id, +}; + +module_i2c_driver(max98926_i2c_driver) +MODULE_DESCRIPTION("ALSA SoC MAX98926 driver"); +MODULE_AUTHOR("Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98926.h b/sound/soc/codecs/max98926.h new file mode 100644 index 0000000..95bd0b5 --- /dev/null +++ b/sound/soc/codecs/max98926.h @@ -0,0 +1,846 @@ +/* + * max98926.h -- MAX98926 ALSA SoC Audio driver + * Copyright 2013-2015 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98926_H +#define _MAX98926_H + +#define MAX98926_CHIP_VERSION 0x40 +#define MAX98926_CHIP_VERSION1 0x50 + +#define MAX98926_VBAT_DATA 0x00 +#define MAX98926_VBST_DATA 0x01 +#define MAX98926_LIVE_STATUS0 0x02 +#define MAX98926_LIVE_STATUS1 0x03 +#define MAX98926_LIVE_STATUS2 0x04 +#define MAX98926_STATE0 0x05 +#define MAX98926_STATE1 0x06 +#define MAX98926_STATE2 0x07 +#define MAX98926_FLAG0 0x08 +#define MAX98926_FLAG1 0x09 +#define MAX98926_FLAG2 0x0A +#define MAX98926_IRQ_ENABLE0 0x0B +#define MAX98926_IRQ_ENABLE1 0x0C +#define MAX98926_IRQ_ENABLE2 0x0D +#define MAX98926_IRQ_CLEAR0 0x0E +#define MAX98926_IRQ_CLEAR1 0x0F +#define MAX98926_IRQ_CLEAR2 0x10 +#define MAX98926_MAP0 0x11 +#define MAX98926_MAP1 0x12 +#define MAX98926_MAP2 0x13 +#define MAX98926_MAP3 0x14 +#define MAX98926_MAP4 0x15 +#define MAX98926_MAP5 0x16 +#define MAX98926_MAP6 0x17 +#define MAX98926_MAP7 0x18 +#define MAX98926_MAP8 0x19 +#define MAX98926_DAI_CLK_MODE1 0x1A +#define MAX98926_DAI_CLK_MODE2 0x1B +#define MAX98926_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98926_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98926_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98926_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98926_FORMAT 0x20 +#define MAX98926_TDM_SLOT_SELECT 0x21 +#define MAX98926_DOUT_CFG_VMON 0x22 +#define MAX98926_DOUT_CFG_IMON 0x23 +#define MAX98926_DOUT_CFG_VBAT 0x24 +#define MAX98926_DOUT_CFG_VBST 0x25 +#define MAX98926_DOUT_CFG_FLAG 0x26 +#define MAX98926_DOUT_HIZ_CFG1 0x27 +#define MAX98926_DOUT_HIZ_CFG2 0x28 +#define MAX98926_DOUT_HIZ_CFG3 0x29 +#define MAX98926_DOUT_HIZ_CFG4 0x2A +#define MAX98926_DOUT_DRV_STRENGTH 0x2B +#define MAX98926_FILTERS 0x2C +#define MAX98926_GAIN 0x2D +#define MAX98926_GAIN_RAMPING 0x2E +#define MAX98926_SPK_AMP 0x2F +#define MAX98926_THRESHOLD 0x30 +#define MAX98926_ALC_ATTACK 0x31 +#define MAX98926_ALC_ATTEN_RLS 0x32 +#define MAX98926_ALC_HOLD_RLS 0x33 +#define MAX98926_ALC_CONFIGURATION 0x34 +#define MAX98926_BOOST_CONVERTER 0x35 +#define MAX98926_BLOCK_ENABLE 0x36 +#define MAX98926_CONFIGURATION 0x37 +#define MAX98926_GLOBAL_ENABLE 0x38 +#define MAX98926_BOOST_LIMITER 0x3A +#define MAX98926_VERSION 0xFF + +#define MAX98926_REG_CNT (MAX98926_R03A_BOOST_LIMITER+1) + +#define MAX98926_PDM_CURRENT_MASK (1<<7) +#define MAX98926_PDM_CURRENT_SHIFT 7 +#define MAX98926_PDM_VOLTAGE_MASK (1<<3) +#define MAX98926_PDM_VOLTAGE_SHIFT 3 +#define MAX98926_PDM_CHANNEL_0_MASK (1<<2) +#define MAX98926_PDM_CHANNEL_0_SHIFT 2 +#define MAX98926_PDM_CHANNEL_1_MASK (1<<6) +#define MAX98926_PDM_CHANNEL_1_SHIFT 6 +#define MAX98926_PDM_SOURCE_0_SHIFT 0 +#define MAX98926_PDM_SOURCE_0_MASK (1<<0) +#define MAX98926_PDM_SOURCE_1_MASK (1<<4) +#define MAX98926_PDM_SOURCE_1_SHIFT 4 + +/* MAX98926 Register Bit Fields */ + +/* MAX98926_R002_LIVE_STATUS0 */ +#define MAX98926_THERMWARN_STATUS_MASK (1<<3) +#define MAX98926_THERMWARN_STATUS_SHIFT 3 +#define MAX98926_THERMWARN_STATUS_WIDTH 1 +#define MAX98926_THERMSHDN_STATUS_MASK (1<<1) +#define MAX98926_THERMSHDN_STATUS_SHIFT 1 +#define MAX98926_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98926_R003_LIVE_STATUS1 */ +#define MAX98926_SPKCURNT_STATUS_MASK (1<<5) +#define MAX98926_SPKCURNT_STATUS_SHIFT 5 +#define MAX98926_SPKCURNT_STATUS_WIDTH 1 +#define MAX98926_WATCHFAIL_STATUS_MASK (1<<4) +#define MAX98926_WATCHFAIL_STATUS_SHIFT 4 +#define MAX98926_WATCHFAIL_STATUS_WIDTH 1 +#define MAX98926_ALCINFH_STATUS_MASK (1<<3) +#define MAX98926_ALCINFH_STATUS_SHIFT 3 +#define MAX98926_ALCINFH_STATUS_WIDTH 1 +#define MAX98926_ALCACT_STATUS_MASK (1<<2) +#define MAX98926_ALCACT_STATUS_SHIFT 2 +#define MAX98926_ALCACT_STATUS_WIDTH 1 +#define MAX98926_ALCMUT_STATUS_MASK (1<<1) +#define MAX98926_ALCMUT_STATUS_SHIFT 1 +#define MAX98926_ALCMUT_STATUS_WIDTH 1 +#define MAX98926_ACLP_STATUS_MASK (1<<0) +#define MAX98926_ACLP_STATUS_SHIFT 0 +#define MAX98926_ACLP_STATUS_WIDTH 1 + +/* MAX98926_R004_LIVE_STATUS2 */ +#define MAX98926_SLOTOVRN_STATUS_MASK (1<<6) +#define MAX98926_SLOTOVRN_STATUS_SHIFT 6 +#define MAX98926_SLOTOVRN_STATUS_WIDTH 1 +#define MAX98926_INVALSLOT_STATUS_MASK (1<<5) +#define MAX98926_INVALSLOT_STATUS_SHIFT 5 +#define MAX98926_INVALSLOT_STATUS_WIDTH 1 +#define MAX98926_SLOTCNFLT_STATUS_MASK (1<<4) +#define MAX98926_SLOTCNFLT_STATUS_SHIFT 4 +#define MAX98926_SLOTCNFLT_STATUS_WIDTH 1 +#define MAX98926_VBSTOVFL_STATUS_MASK (1<<3) +#define MAX98926_VBSTOVFL_STATUS_SHIFT 3 +#define MAX98926_VBSTOVFL_STATUS_WIDTH 1 +#define MAX98926_VBATOVFL_STATUS_MASK (1<<2) +#define MAX98926_VBATOVFL_STATUS_SHIFT 2 +#define MAX98926_VBATOVFL_STATUS_WIDTH 1 +#define MAX98926_IMONOVFL_STATUS_MASK (1<<1) +#define MAX98926_IMONOVFL_STATUS_SHIFT 1 +#define MAX98926_IMONOVFL_STATUS_WIDTH 1 +#define MAX98926_VMONOVFL_STATUS_MASK (1<<0) +#define MAX98926_VMONOVFL_STATUS_SHIFT 0 +#define MAX98926_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98926_R005_STATE0 */ +#define MAX98926_THERMWARN_END_STATE_MASK (1<<3) +#define MAX98926_THERMWARN_END_STATE_SHIFT 3 +#define MAX98926_THERMWARN_END_STATE_WIDTH 1 +#define MAX98926_THERMWARN_BGN_STATE_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_STATE_SHIFT 1 +#define MAX98926_THERMWARN_BGN_STATE_WIDTH 1 +#define MAX98926_THERMSHDN_END_STATE_MASK (1<<1) +#define MAX98926_THERMSHDN_END_STATE_SHIFT 1 +#define MAX98926_THERMSHDN_END_STATE_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_STATE_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_STATE_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98926_R006_STATE1 */ +#define MAX98926_SPRCURNT_STATE_MASK (1<<5) +#define MAX98926_SPRCURNT_STATE_SHIFT 5 +#define MAX98926_SPRCURNT_STATE_WIDTH 1 +#define MAX98926_WATCHFAIL_STATE_MASK (1<<4) +#define MAX98926_WATCHFAIL_STATE_SHIFT 4 +#define MAX98926_WATCHFAIL_STATE_WIDTH 1 +#define MAX98926_ALCINFH_STATE_MASK (1<<3) +#define MAX98926_ALCINFH_STATE_SHIFT 3 +#define MAX98926_ALCINFH_STATE_WIDTH 1 +#define MAX98926_ALCACT_STATE_MASK (1<<2) +#define MAX98926_ALCACT_STATE_SHIFT 2 +#define MAX98926_ALCACT_STATE_WIDTH 1 +#define MAX98926_ALCMUT_STATE_MASK (1<<1) +#define MAX98926_ALCMUT_STATE_SHIFT 1 +#define MAX98926_ALCMUT_STATE_WIDTH 1 +#define MAX98926_ALCP_STATE_MASK (1<<0) +#define MAX98926_ALCP_STATE_SHIFT 0 +#define MAX98926_ALCP_STATE_WIDTH 1 + +/* MAX98926_R007_STATE2 */ +#define MAX98926_SLOTOVRN_STATE_MASK (1<<6) +#define MAX98926_SLOTOVRN_STATE_SHIFT 6 +#define MAX98926_SLOTOVRN_STATE_WIDTH 1 +#define MAX98926_INVALSLOT_STATE_MASK (1<<5) +#define MAX98926_INVALSLOT_STATE_SHIFT 5 +#define MAX98926_INVALSLOT_STATE_WIDTH 1 +#define MAX98926_SLOTCNFLT_STATE_MASK (1<<4) +#define MAX98926_SLOTCNFLT_STATE_SHIFT 4 +#define MAX98926_SLOTCNFLT_STATE_WIDTH 1 +#define MAX98926_VBSTOVFL_STATE_MASK (1<<3) +#define MAX98926_VBSTOVFL_STATE_SHIFT 3 +#define MAX98926_VBSTOVFL_STATE_WIDTH 1 +#define MAX98926_VBATOVFL_STATE_MASK (1<<2) +#define MAX98926_VBATOVFL_STATE_SHIFT 2 +#define MAX98926_VBATOVFL_STATE_WIDTH 1 +#define MAX98926_IMONOVFL_STATE_MASK (1<<1) +#define MAX98926_IMONOVFL_STATE_SHIFT 1 +#define MAX98926_IMONOVFL_STATE_WIDTH 1 +#define MAX98926_VMONOVFL_STATE_MASK (1<<0) +#define MAX98926_VMONOVFL_STATE_SHIFT 0 +#define MAX98926_VMONOVFL_STATE_WIDTH 1 + +/* MAX98926_R008_FLAG0 */ +#define MAX98926_THERMWARN_END_FLAG_MASK (1<<3) +#define MAX98926_THERMWARN_END_FLAG_SHIFT 3 +#define MAX98926_THERMWARN_END_FLAG_WIDTH 1 +#define MAX98926_THERMWARN_BGN_FLAG_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_FLAG_SHIFT 2 +#define MAX98926_THERMWARN_BGN_FLAG_WIDTH 1 +#define MAX98926_THERMSHDN_END_FLAG_MASK (1<<1) +#define MAX98926_THERMSHDN_END_FLAG_SHIFT 1 +#define MAX98926_THERMSHDN_END_FLAG_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_FLAG_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98926_R009_FLAG1 */ +#define MAX98926_SPKCURNT_FLAG_MASK (1<<5) +#define MAX98926_SPKCURNT_FLAG_SHIFT 5 +#define MAX98926_SPKCURNT_FLAG_WIDTH 1 +#define MAX98926_WATCHFAIL_FLAG_MASK (1<<4) +#define MAX98926_WATCHFAIL_FLAG_SHIFT 4 +#define MAX98926_WATCHFAIL_FLAG_WIDTH 1 +#define MAX98926_ALCINFH_FLAG_MASK (1<<3) +#define MAX98926_ALCINFH_FLAG_SHIFT 3 +#define MAX98926_ALCINFH_FLAG_WIDTH 1 +#define MAX98926_ALCACT_FLAG_MASK (1<<2) +#define MAX98926_ALCACT_FLAG_SHIFT 2 +#define MAX98926_ALCACT_FLAG_WIDTH 1 +#define MAX98926_ALCMUT_FLAG_MASK (1<<1) +#define MAX98926_ALCMUT_FLAG_SHIFT 1 +#define MAX98926_ALCMUT_FLAG_WIDTH 1 +#define MAX98926_ALCP_FLAG_MASK (1<<0) +#define MAX98926_ALCP_FLAG_SHIFT 0 +#define MAX98926_ALCP_FLAG_WIDTH 1 + +/* MAX98926_R00A_FLAG2 */ +#define MAX98926_SLOTOVRN_FLAG_MASK (1<<6) +#define MAX98926_SLOTOVRN_FLAG_SHIFT 6 +#define MAX98926_SLOTOVRN_FLAG_WIDTH 1 +#define MAX98926_INVALSLOT_FLAG_MASK (1<<5) +#define MAX98926_INVALSLOT_FLAG_SHIFT 5 +#define MAX98926_INVALSLOT_FLAG_WIDTH 1 +#define MAX98926_SLOTCNFLT_FLAG_MASK (1<<4) +#define MAX98926_SLOTCNFLT_FLAG_SHIFT 4 +#define MAX98926_SLOTCNFLT_FLAG_WIDTH 1 +#define MAX98926_VBSTOVFL_FLAG_MASK (1<<3) +#define MAX98926_VBSTOVFL_FLAG_SHIFT 3 +#define MAX98926_VBSTOVFL_FLAG_WIDTH 1 +#define MAX98926_VBATOVFL_FLAG_MASK (1<<2) +#define MAX98926_VBATOVFL_FLAG_SHIFT 2 +#define MAX98926_VBATOVFL_FLAG_WIDTH 1 +#define MAX98926_IMONOVFL_FLAG_MASK (1<<1) +#define MAX98926_IMONOVFL_FLAG_SHIFT 1 +#define MAX98926_IMONOVFL_FLAG_WIDTH 1 +#define MAX98926_VMONOVFL_FLAG_MASK (1<<0) +#define MAX98926_VMONOVFL_FLAG_SHIFT 0 +#define MAX98926_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98926_R00B_IRQ_ENABLE0 */ +#define MAX98926_THERMWARN_END_EN_MASK (1<<3) +#define MAX98926_THERMWARN_END_EN_SHIFT 3 +#define MAX98926_THERMWARN_END_EN_WIDTH 1 +#define MAX98926_THERMWARN_BGN_EN_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_EN_SHIFT 2 +#define MAX98926_THERMWARN_BGN_EN_WIDTH 1 +#define MAX98926_THERMSHDN_END_EN_MASK (1<<1) +#define MAX98926_THERMSHDN_END_EN_SHIFT 1 +#define MAX98926_THERMSHDN_END_EN_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_EN_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_EN_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98926_R00C_IRQ_ENABLE1 */ +#define MAX98926_SPKCURNT_EN_MASK (1<<5) +#define MAX98926_SPKCURNT_EN_SHIFT 5 +#define MAX98926_SPKCURNT_EN_WIDTH 1 +#define MAX98926_WATCHFAIL_EN_MASK (1<<4) +#define MAX98926_WATCHFAIL_EN_SHIFT 4 +#define MAX98926_WATCHFAIL_EN_WIDTH 1 +#define MAX98926_ALCINFH_EN_MASK (1<<3) +#define MAX98926_ALCINFH_EN_SHIFT 3 +#define MAX98926_ALCINFH_EN_WIDTH 1 +#define MAX98926_ALCACT_EN_MASK (1<<2) +#define MAX98926_ALCACT_EN_SHIFT 2 +#define MAX98926_ALCACT_EN_WIDTH 1 +#define MAX98926_ALCMUT_EN_MASK (1<<1) +#define MAX98926_ALCMUT_EN_SHIFT 1 +#define MAX98926_ALCMUT_EN_WIDTH 1 +#define MAX98926_ALCP_EN_MASK (1<<0) +#define MAX98926_ALCP_EN_SHIFT 0 +#define MAX98926_ALCP_EN_WIDTH 1 + +/* MAX98926_R00D_IRQ_ENABLE2 */ +#define MAX98926_SLOTOVRN_EN_MASK (1<<6) +#define MAX98926_SLOTOVRN_EN_SHIFT 6 +#define MAX98926_SLOTOVRN_EN_WIDTH 1 +#define MAX98926_INVALSLOT_EN_MASK (1<<5) +#define MAX98926_INVALSLOT_EN_SHIFT 5 +#define MAX98926_INVALSLOT_EN_WIDTH 1 +#define MAX98926_SLOTCNFLT_EN_MASK (1<<4) +#define MAX98926_SLOTCNFLT_EN_SHIFT 4 +#define MAX98926_SLOTCNFLT_EN_WIDTH 1 +#define MAX98926_VBSTOVFL_EN_MASK (1<<3) +#define MAX98926_VBSTOVFL_EN_SHIFT 3 +#define MAX98926_VBSTOVFL_EN_WIDTH 1 +#define MAX98926_VBATOVFL_EN_MASK (1<<2) +#define MAX98926_VBATOVFL_EN_SHIFT 2 +#define MAX98926_VBATOVFL_EN_WIDTH 1 +#define MAX98926_IMONOVFL_EN_MASK (1<<1) +#define MAX98926_IMONOVFL_EN_SHIFT 1 +#define MAX98926_IMONOVFL_EN_WIDTH 1 +#define MAX98926_VMONOVFL_EN_MASK (1<<0) +#define MAX98926_VMONOVFL_EN_SHIFT 0 +#define MAX98926_VMONOVFL_EN_WIDTH 1 + +/* MAX98926_R00E_IRQ_CLEAR0 */ +#define MAX98926_THERMWARN_END_CLR_MASK (1<<3) +#define MAX98926_THERMWARN_END_CLR_SHIFT 3 +#define MAX98926_THERMWARN_END_CLR_WIDTH 1 +#define MAX98926_THERMWARN_BGN_CLR_MASK (1<<2) +#define MAX98926_THERMWARN_BGN_CLR_SHIFT 2 +#define MAX98926_THERMWARN_BGN_CLR_WIDTH 1 +#define MAX98926_THERMSHDN_END_CLR_MASK (1<<1) +#define MAX98926_THERMSHDN_END_CLR_SHIFT 1 +#define MAX98926_THERMSHDN_END_CLR_WIDTH 1 +#define MAX98926_THERMSHDN_BGN_CLR_MASK (1<<0) +#define MAX98926_THERMSHDN_BGN_CLR_SHIFT 0 +#define MAX98926_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98926_R00F_IRQ_CLEAR1 */ +#define MAX98926_SPKCURNT_CLR_MASK (1<<5) +#define MAX98926_SPKCURNT_CLR_SHIFT 5 +#define MAX98926_SPKCURNT_CLR_WIDTH 1 +#define MAX98926_WATCHFAIL_CLR_MASK (1<<4) +#define MAX98926_WATCHFAIL_CLR_SHIFT 4 +#define MAX98926_WATCHFAIL_CLR_WIDTH 1 +#define MAX98926_ALCINFH_CLR_MASK (1<<3) +#define MAX98926_ALCINFH_CLR_SHIFT 3 +#define MAX98926_ALCINFH_CLR_WIDTH 1 +#define MAX98926_ALCACT_CLR_MASK (1<<2) +#define MAX98926_ALCACT_CLR_SHIFT 2 +#define MAX98926_ALCACT_CLR_WIDTH 1 +#define MAX98926_ALCMUT_CLR_MASK (1<<1) +#define MAX98926_ALCMUT_CLR_SHIFT 1 +#define MAX98926_ALCMUT_CLR_WIDTH 1 +#define MAX98926_ALCP_CLR_MASK (1<<0) +#define MAX98926_ALCP_CLR_SHIFT 0 +#define MAX98926_ALCP_CLR_WIDTH 1 + +/* MAX98926_R010_IRQ_CLEAR2 */ +#define MAX98926_SLOTOVRN_CLR_MASK (1<<6) +#define MAX98926_SLOTOVRN_CLR_SHIFT 6 +#define MAX98926_SLOTOVRN_CLR_WIDTH 1 +#define MAX98926_INVALSLOT_CLR_MASK (1<<5) +#define MAX98926_INVALSLOT_CLR_SHIFT 5 +#define MAX98926_INVALSLOT_CLR_WIDTH 1 +#define MAX98926_SLOTCNFLT_CLR_MASK (1<<4) +#define MAX98926_SLOTCNFLT_CLR_SHIFT 4 +#define MAX98926_SLOTCNFLT_CLR_WIDTH 1 +#define MAX98926_VBSTOVFL_CLR_MASK (1<<3) +#define MAX98926_VBSTOVFL_CLR_SHIFT 3 +#define MAX98926_VBSTOVFL_CLR_WIDTH 1 +#define MAX98926_VBATOVFL_CLR_MASK (1<<2) +#define MAX98926_VBATOVFL_CLR_SHIFT 2 +#define MAX98926_VBATOVFL_CLR_WIDTH 1 +#define MAX98926_IMONOVFL_CLR_MASK (1<<1) +#define MAX98926_IMONOVFL_CLR_SHIFT 1 +#define MAX98926_IMONOVFL_CLR_WIDTH 1 +#define MAX98926_VMONOVFL_CLR_MASK (1<<0) +#define MAX98926_VMONOVFL_CLR_SHIFT 0 +#define MAX98926_VMONOVFL_CLR_WIDTH 1 + +/* MAX98926_R011_MAP0 */ +#define MAX98926_ER_THERMWARN_EN_MASK (1<<7) +#define MAX98926_ER_THERMWARN_EN_SHIFT 7 +#define MAX98926_ER_THERMWARN_EN_WIDTH 1 +#define MAX98926_ER_THERMWARN_MAP_MASK (0x07<<4) +#define MAX98926_ER_THERMWARN_MAP_SHIFT 4 +#define MAX98926_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98926_R012_MAP1 */ +#define MAX98926_ER_ALCMUT_EN_MASK (1<<7) +#define MAX98926_ER_ALCMUT_EN_SHIFT 7 +#define MAX98926_ER_ALCMUT_EN_WIDTH 1 +#define MAX98926_ER_ALCMUT_MAP_MASK (0x07<<4) +#define MAX98926_ER_ALCMUT_MAP_SHIFT 4 +#define MAX98926_ER_ALCMUT_MAP_WIDTH 3 +#define MAX98926_ER_ALCP_EN_MASK (1<<3) +#define MAX98926_ER_ALCP_EN_SHIFT 3 +#define MAX98926_ER_ALCP_EN_WIDTH 1 +#define MAX98926_ER_ALCP_MAP_MASK (0x07<<0) +#define MAX98926_ER_ALCP_MAP_SHIFT 0 +#define MAX98926_ER_ALCP_MAP_WIDTH 3 + +/* MAX98926_R013_MAP2 */ +#define MAX98926_ER_ALCINFH_EN_MASK (1<<7) +#define MAX98926_ER_ALCINFH_EN_SHIFT 7 +#define MAX98926_ER_ALCINFH_EN_WIDTH 1 +#define MAX98926_ER_ALCINFH_MAP_MASK (0x07<<4) +#define MAX98926_ER_ALCINFH_MAP_SHIFT 4 +#define MAX98926_ER_ALCINFH_MAP_WIDTH 3 +#define MAX98926_ER_ALCACT_EN_MASK (1<<3) +#define MAX98926_ER_ALCACT_EN_SHIFT 3 +#define MAX98926_ER_ALCACT_EN_WIDTH 1 +#define MAX98926_ER_ALCACT_MAP_MASK (0x07<<0) +#define MAX98926_ER_ALCACT_MAP_SHIFT 0 +#define MAX98926_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98926_R014_MAP3 */ +#define MAX98926_ER_SPKCURNT_EN_MASK (1<<7) +#define MAX98926_ER_SPKCURNT_EN_SHIFT 7 +#define MAX98926_ER_SPKCURNT_EN_WIDTH 1 +#define MAX98926_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define MAX98926_ER_SPKCURNT_MAP_SHIFT 4 +#define MAX98926_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98926_R015_MAP4 */ +/* RESERVED */ + +/* MAX98926_R016_MAP5 */ +#define MAX98926_ER_IMONOVFL_EN_MASK (1<<7) +#define MAX98926_ER_IMONOVFL_EN_SHIFT 7 +#define MAX98926_ER_IMONOVFL_EN_WIDTH 1 +#define MAX98926_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define MAX98926_ER_IMONOVFL_MAP_SHIFT 4 +#define MAX98926_ER_IMONOVFL_MAP_WIDTH 3 +#define MAX98926_ER_VMONOVFL_EN_MASK (1<<3) +#define MAX98926_ER_VMONOVFL_EN_SHIFT 3 +#define MAX98926_ER_VMONOVFL_EN_WIDTH 1 +#define MAX98926_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define MAX98926_ER_VMONOVFL_MAP_SHIFT 0 +#define MAX98926_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98926_R017_MAP6 */ +#define MAX98926_ER_VBSTOVFL_EN_MASK (1<<7) +#define MAX98926_ER_VBSTOVFL_EN_SHIFT 7 +#define MAX98926_ER_VBSTOVFL_EN_WIDTH 1 +#define MAX98926_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define MAX98926_ER_VBSTOVFL_MAP_SHIFT 4 +#define MAX98926_ER_VBSTOVFL_MAP_WIDTH 3 +#define MAX98926_ER_VBATOVFL_EN_MASK (1<<3) +#define MAX98926_ER_VBATOVFL_EN_SHIFT 3 +#define MAX98926_ER_VBATOVFL_EN_WIDTH 1 +#define MAX98926_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define MAX98926_ER_VBATOVFL_MAP_SHIFT 0 +#define MAX98926_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98926_R018_MAP7 */ +#define MAX98926_ER_INVALSLOT_EN_MASK (1<<7) +#define MAX98926_ER_INVALSLOT_EN_SHIFT 7 +#define MAX98926_ER_INVALSLOT_EN_WIDTH 1 +#define MAX98926_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define MAX98926_ER_INVALSLOT_MAP_SHIFT 4 +#define MAX98926_ER_INVALSLOT_MAP_WIDTH 3 +#define MAX98926_ER_SLOTCNFLT_EN_MASK (1<<3) +#define MAX98926_ER_SLOTCNFLT_EN_SHIFT 3 +#define MAX98926_ER_SLOTCNFLT_EN_WIDTH 1 +#define MAX98926_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define MAX98926_ER_SLOTCNFLT_MAP_SHIFT 0 +#define MAX98926_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98926_R019_MAP8 */ +#define MAX98926_ER_SLOTOVRN_EN_MASK (1<<3) +#define MAX98926_ER_SLOTOVRN_EN_SHIFT 3 +#define MAX98926_ER_SLOTOVRN_EN_WIDTH 1 +#define MAX98926_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define MAX98926_ER_SLOTOVRN_MAP_SHIFT 0 +#define MAX98926_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98926_R01A_DAI_CLK_MODE1 */ +#define MAX98926_DAI_CLK_SOURCE_MASK (1<<6) +#define MAX98926_DAI_CLK_SOURCE_SHIFT 6 +#define MAX98926_DAI_CLK_SOURCE_WIDTH 1 +#define MAX98926_MDLL_MULT_MASK (0x0F<<0) +#define MAX98926_MDLL_MULT_SHIFT 0 +#define MAX98926_MDLL_MULT_WIDTH 4 + +#define MAX98926_MDLL_MULT_MCLKx8 6 +#define MAX98926_MDLL_MULT_MCLKx16 8 + +/* MAX98926_R01B_DAI_CLK_MODE2 */ +#define MAX98926_DAI_SR_MASK (0x0F<<4) +#define MAX98926_DAI_SR_SHIFT 4 +#define MAX98926_DAI_SR_WIDTH 4 +#define MAX98926_DAI_MAS_MASK (1<<3) +#define MAX98926_DAI_MAS_SHIFT 3 +#define MAX98926_DAI_MAS_WIDTH 1 +#define MAX98926_DAI_BSEL_MASK (0x07<<0) +#define MAX98926_DAI_BSEL_SHIFT 0 +#define MAX98926_DAI_BSEL_WIDTH 3 + +#define MAX98926_DAI_BSEL_32 (0 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_48 (1 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_64 (2 << MAX98926_DAI_BSEL_SHIFT) +#define MAX98926_DAI_BSEL_256 (6 << MAX98926_DAI_BSEL_SHIFT) + +/* MAX98926_R01C_DAI_CLK_DIV_M_MSBS */ +#define MAX98926_DAI_M_MSBS_MASK (0xFF<<0) +#define MAX98926_DAI_M_MSBS_SHIFT 0 +#define MAX98926_DAI_M_MSBS_WIDTH 8 + +/* MAX98926_R01D_DAI_CLK_DIV_M_LSBS */ +#define MAX98926_DAI_M_LSBS_MASK (0xFF<<0) +#define MAX98926_DAI_M_LSBS_SHIFT 0 +#define MAX98926_DAI_M_LSBS_WIDTH 8 + +/* MAX98926_R01E_DAI_CLK_DIV_N_MSBS */ +#define MAX98926_DAI_N_MSBS_MASK (0x7F<<0) +#define MAX98926_DAI_N_MSBS_SHIFT 0 +#define MAX98926_DAI_N_MSBS_WIDTH 7 + +/* MAX98926_R01F_DAI_CLK_DIV_N_LSBS */ +#define MAX98926_DAI_N_LSBS_MASK (0xFF<<0) +#define MAX98926_DAI_N_LSBS_SHIFT 0 +#define MAX98926_DAI_N_LSBS_WIDTH 8 + +/* MAX98926_R020_FORMAT */ +#define MAX98926_DAI_CHANSZ_MASK (0x03<<6) +#define MAX98926_DAI_CHANSZ_SHIFT 6 +#define MAX98926_DAI_CHANSZ_WIDTH 2 +#define MAX98926_DAI_INTERLEAVE_MASK (1<<5) +#define MAX98926_DAI_INTERLEAVE_SHIFT 5 +#define MAX98926_DAI_INTERLEAVE_WIDTH 1 +#define MAX98926_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define MAX98926_DAI_EXTBCLK_HIZ_SHIFT 4 +#define MAX98926_DAI_EXTBCLK_HIZ_WIDTH 1 +#define MAX98926_DAI_WCI_MASK (1<<3) +#define MAX98926_DAI_WCI_SHIFT 3 +#define MAX98926_DAI_WCI_WIDTH 1 +#define MAX98926_DAI_BCI_MASK (1<<2) +#define MAX98926_DAI_BCI_SHIFT 2 +#define MAX98926_DAI_BCI_WIDTH 1 +#define MAX98926_DAI_DLY_MASK (1<<1) +#define MAX98926_DAI_DLY_SHIFT 1 +#define MAX98926_DAI_DLY_WIDTH 1 +#define MAX98926_DAI_TDM_MASK (1<<0) +#define MAX98926_DAI_TDM_SHIFT 0 +#define MAX98926_DAI_TDM_WIDTH 1 + +#define MAX98926_DAI_CHANSZ_16 (1 << MAX98926_DAI_CHANSZ_SHIFT) +#define MAX98926_DAI_CHANSZ_24 (2 << MAX98926_DAI_CHANSZ_SHIFT) +#define MAX98926_DAI_CHANSZ_32 (3 << MAX98926_DAI_CHANSZ_SHIFT) + +/* MAX98926_R021_TDM_SLOT_SELECT */ +#define MAX98926_DAI_DO_EN_MASK (1<<7) +#define MAX98926_DAI_DO_EN_SHIFT 7 +#define MAX98926_DAI_DO_EN_WIDTH 1 +#define MAX98926_DAI_DIN_EN_MASK (1<<6) +#define MAX98926_DAI_DIN_EN_SHIFT 6 +#define MAX98926_DAI_DIN_EN_WIDTH 1 +#define MAX98926_DAI_INR_SOURCE_MASK (0x07<<3) +#define MAX98926_DAI_INR_SOURCE_SHIFT 3 +#define MAX98926_DAI_INR_SOURCE_WIDTH 3 +#define MAX98926_DAI_INL_SOURCE_MASK (0x07<<0) +#define MAX98926_DAI_INL_SOURCE_SHIFT 0 +#define MAX98926_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98926_R022_DOUT_CFG_VMON */ +#define MAX98926_DAI_VMON_EN_MASK (1<<5) +#define MAX98926_DAI_VMON_EN_SHIFT 5 +#define MAX98926_DAI_VMON_EN_WIDTH 1 +#define MAX98926_DAI_VMON_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_VMON_SLOT_SHIFT 0 +#define MAX98926_DAI_VMON_SLOT_WIDTH 5 + +#define MAX98926_DAI_VMON_SLOT_00_01 (0 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_01_02 (1 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_02_03 (2 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_03_04 (3 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_04_05 (4 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_05_06 (5 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_06_07 (6 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_07_08 (7 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_08_09 (8 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_09_0A (9 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0A_0B (10 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0B_0C (11 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0C_0D (12 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0D_0E (13 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0E_0F (14 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_0F_10 (15 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_10_11 (16 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_11_12 (17 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_12_13 (18 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_13_14 (19 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_14_15 (20 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_15_16 (21 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_16_17 (22 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_17_18 (23 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_18_19 (24 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_19_1A (25 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1A_1B (26 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1B_1C (27 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1C_1D (28 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1D_1E (29 << MAX98926_DAI_VMON_SLOT_SHIFT) +#define MAX98926_DAI_VMON_SLOT_1E_1F (30 << MAX98926_DAI_VMON_SLOT_SHIFT) + +/* MAX98926_R023_DOUT_CFG_IMON */ +#define MAX98926_DAI_IMON_EN_MASK (1<<5) +#define MAX98926_DAI_IMON_EN_SHIFT 5 +#define MAX98926_DAI_IMON_EN_WIDTH 1 +#define MAX98926_DAI_IMON_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_IMON_SLOT_SHIFT 0 +#define MAX98926_DAI_IMON_SLOT_WIDTH 5 + +#define MAX98926_DAI_IMON_SLOT_00_01 (0 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_01_02 (1 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_02_03 (2 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_03_04 (3 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_04_05 (4 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_05_06 (5 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_06_07 (6 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_07_08 (7 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_08_09 (8 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_09_0A (9 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0A_0B (10 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0B_0C (11 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0C_0D (12 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0D_0E (13 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0E_0F (14 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_0F_10 (15 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_10_11 (16 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_11_12 (17 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_12_13 (18 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_13_14 (19 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_14_15 (20 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_15_16 (21 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_16_17 (22 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_17_18 (23 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_18_19 (24 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_19_1A (25 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1A_1B (26 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1B_1C (27 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1C_1D (28 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1D_1E (29 << MAX98926_DAI_IMON_SLOT_SHIFT) +#define MAX98926_DAI_IMON_SLOT_1E_1F (30 << MAX98926_DAI_IMON_SLOT_SHIFT) + +/* MAX98926_R024_DOUT_CFG_VBAT */ +#define MAX98926_DAI_INTERLEAVE_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_INTERLEAVE_SLOT_SHIFT 0 +#define MAX98926_DAI_INTERLEAVE_SLOT_WIDTH 5 + +/* MAX98926_R025_DOUT_CFG_VBST */ +#define MAX98926_DAI_VBST_EN_MASK (1<<5) +#define MAX98926_DAI_VBST_EN_SHIFT 5 +#define MAX98926_DAI_VBST_EN_WIDTH 1 +#define MAX98926_DAI_VBST_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_VBST_SLOT_SHIFT 0 +#define MAX98926_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98926_R026_DOUT_CFG_FLAG */ +#define MAX98926_DAI_FLAG_EN_MASK (1<<5) +#define MAX98926_DAI_FLAG_EN_SHIFT 5 +#define MAX98926_DAI_FLAG_EN_WIDTH 1 +#define MAX98926_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define MAX98926_DAI_FLAG_SLOT_SHIFT 0 +#define MAX98926_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98926_R027_DOUT_HIZ_CFG1 */ +#define MAX98926_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98926_R028_DOUT_HIZ_CFG2 */ +#define MAX98926_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98926_R029_DOUT_HIZ_CFG3 */ +#define MAX98926_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98926_R02A_DOUT_HIZ_CFG4 */ +#define MAX98926_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define MAX98926_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define MAX98926_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98926_R02B_DOUT_DRV_STRENGTH */ +#define MAX98926_DAI_OUT_DRIVE_MASK (0x03<<0) +#define MAX98926_DAI_OUT_DRIVE_SHIFT 0 +#define MAX98926_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98926_R02C_FILTERS */ +#define MAX98926_ADC_DITHER_EN_MASK (1<<7) +#define MAX98926_ADC_DITHER_EN_SHIFT 7 +#define MAX98926_ADC_DITHER_EN_WIDTH 1 +#define MAX98926_IV_DCB_EN_MASK (1<<6) +#define MAX98926_IV_DCB_EN_SHIFT 6 +#define MAX98926_IV_DCB_EN_WIDTH 1 +#define MAX98926_DAC_DITHER_EN_MASK (1<<4) +#define MAX98926_DAC_DITHER_EN_SHIFT 4 +#define MAX98926_DAC_DITHER_EN_WIDTH 1 +#define MAX98926_DAC_FILTER_MODE_MASK (1<<3) +#define MAX98926_DAC_FILTER_MODE_SHIFT 3 +#define MAX98926_DAC_FILTER_MODE_WIDTH 1 +#define MAX98926_DAC_HPF_MASK (0x07<<0) +#define MAX98926_DAC_HPF_SHIFT 0 +#define MAX98926_DAC_HPF_WIDTH 3 +#define MAX98926_DAC_HPF_DISABLE (0 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_DC_BLOCK (1 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_100 (2 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_200 (3 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_400 (4 << MAX98926_DAC_HPF_SHIFT) +#define MAX98926_DAC_HPF_EN_800 (5 << MAX98926_DAC_HPF_SHIFT) + +/* MAX98926_R02D_GAIN */ +#define MAX98926_DAC_IN_SEL_MASK (0x03<<5) +#define MAX98926_DAC_IN_SEL_SHIFT 5 +#define MAX98926_DAC_IN_SEL_WIDTH 2 +#define MAX98926_SPK_GAIN_MASK (0x1F<<0) +#define MAX98926_SPK_GAIN_SHIFT 0 +#define MAX98926_SPK_GAIN_WIDTH 5 + +#define MAX98926_DAC_IN_SEL_LEFT_DAI (0 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_RIGHT_DAI (1 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_SUMMED_DAI (2 << MAX98926_DAC_IN_SEL_SHIFT) +#define MAX98926_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << MAX98926_DAC_IN_SEL_SHIFT) + +/* MAX98926_R02E_GAIN_RAMPING */ +#define MAX98926_SPK_RMP_EN_MASK (1<<1) +#define MAX98926_SPK_RMP_EN_SHIFT 1 +#define MAX98926_SPK_RMP_EN_WIDTH 1 +#define MAX98926_SPK_ZCD_EN_MASK (1<<0) +#define MAX98926_SPK_ZCD_EN_SHIFT 0 +#define MAX98926_SPK_ZCD_EN_WIDTH 1 + +/* MAX98926_R02F_SPK_AMP */ +#define MAX98926_SPK_MODE_MASK (1<<0) +#define MAX98926_SPK_MODE_SHIFT 0 +#define MAX98926_SPK_MODE_WIDTH 1 +#define MAX98926_INSELECT_MODE_MASK (1<<1) +#define MAX98926_INSELECT_MODE_SHIFT 1 +#define MAX98926_INSELECT_MODE_WIDTH 1 + +/* MAX98926_R030_THRESHOLD */ +#define MAX98926_ALC_EN_MASK (1<<5) +#define MAX98926_ALC_EN_SHIFT 5 +#define MAX98926_ALC_EN_WIDTH 1 +#define MAX98926_ALC_TH_MASK (0x1F<<0) +#define MAX98926_ALC_TH_SHIFT 0 +#define MAX98926_ALC_TH_WIDTH 5 + +/* MAX98926_R031_ALC_ATTACK */ +#define MAX98926_ALC_ATK_STEP_MASK (0x0F<<4) +#define MAX98926_ALC_ATK_STEP_SHIFT 4 +#define MAX98926_ALC_ATK_STEP_WIDTH 4 +#define MAX98926_ALC_ATK_RATE_MASK (0x7<<0) +#define MAX98926_ALC_ATK_RATE_SHIFT 0 +#define MAX98926_ALC_ATK_RATE_WIDTH 3 + +/* MAX98926_R032_ALC_ATTEN_RLS */ +#define MAX98926_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define MAX98926_ALC_MAX_ATTEN_SHIFT 4 +#define MAX98926_ALC_MAX_ATTEN_WIDTH 4 +#define MAX98926_ALC_RLS_RATE_MASK (0x7<<0) +#define MAX98926_ALC_RLS_RATE_SHIFT 0 +#define MAX98926_ALC_RLS_RATE_WIDTH 3 + +/* MAX98926_R033_ALC_HOLD_RLS */ +#define MAX98926_ALC_RLS_TGR_MASK (1<<0) +#define MAX98926_ALC_RLS_TGR_SHIFT 0 +#define MAX98926_ALC_RLS_TGR_WIDTH 1 + +/* MAX98926_R034_ALC_CONFIGURATION */ +#define MAX98926_ALC_MUTE_EN_MASK (1<<7) +#define MAX98926_ALC_MUTE_EN_SHIFT 7 +#define MAX98926_ALC_MUTE_EN_WIDTH 1 +#define MAX98926_ALC_MUTE_DLY_MASK (0x07<<4) +#define MAX98926_ALC_MUTE_DLY_SHIFT 4 +#define MAX98926_ALC_MUTE_DLY_WIDTH 3 +#define MAX98926_ALC_RLS_DBT_MASK (0x07<<0) +#define MAX98926_ALC_RLS_DBT_SHIFT 0 +#define MAX98926_ALC_RLS_DBT_WIDTH 3 + +/* MAX98926_R035_BOOST_CONVERTER */ +#define MAX98926_BST_SYNC_MASK (1<<7) +#define MAX98926_BST_SYNC_SHIFT 7 +#define MAX98926_BST_SYNC_WIDTH 1 +#define MAX98926_BST_PHASE_MASK (0x03<<4) +#define MAX98926_BST_PHASE_SHIFT 4 +#define MAX98926_BST_PHASE_WIDTH 2 +#define MAX98926_BST_SKIP_MODE_MASK (0x03<<0) +#define MAX98926_BST_SKIP_MODE_SHIFT 0 +#define MAX98926_BST_SKIP_MODE_WIDTH 2 + +/* MAX98926_R036_BLOCK_ENABLE */ +#define MAX98926_BST_EN_MASK (1<<7) +#define MAX98926_BST_EN_SHIFT 7 +#define MAX98926_BST_EN_WIDTH 1 +#define MAX98926_WATCH_EN_MASK (1<<6) +#define MAX98926_WATCH_EN_SHIFT 6 +#define MAX98926_WATCH_EN_WIDTH 1 +#define MAX98926_CLKMON_EN_MASK (1<<5) +#define MAX98926_CLKMON_EN_SHIFT 5 +#define MAX98926_CLKMON_EN_WIDTH 1 +#define MAX98926_SPK_EN_MASK (1<<4) +#define MAX98926_SPK_EN_SHIFT 4 +#define MAX98926_SPK_EN_WIDTH 1 +#define MAX98926_ADC_VBST_EN_MASK (1<<3) +#define MAX98926_ADC_VBST_EN_SHIFT 3 +#define MAX98926_ADC_VBST_EN_WIDTH 1 +#define MAX98926_ADC_VBAT_EN_MASK (1<<2) +#define MAX98926_ADC_VBAT_EN_SHIFT 2 +#define MAX98926_ADC_VBAT_EN_WIDTH 1 +#define MAX98926_ADC_IMON_EN_MASK (1<<1) +#define MAX98926_ADC_IMON_EN_SHIFT 1 +#define MAX98926_ADC_IMON_EN_WIDTH 1 +#define MAX98926_ADC_VMON_EN_MASK (1<<0) +#define MAX98926_ADC_VMON_EN_SHIFT 0 +#define MAX98926_ADC_VMON_EN_WIDTH 1 + +/* MAX98926_R037_CONFIGURATION */ +#define MAX98926_BST_VOUT_MASK (0x0F<<4) +#define MAX98926_BST_VOUT_SHIFT 4 +#define MAX98926_BST_VOUT_WIDTH 4 +#define MAX98926_THERMWARN_LEVEL_MASK (0x03<<2) +#define MAX98926_THERMWARN_LEVEL_SHIFT 2 +#define MAX98926_THERMWARN_LEVEL_WIDTH 2 +#define MAX98926_WATCH_TIME_MASK (0x03<<0) +#define MAX98926_WATCH_TIME_SHIFT 0 +#define MAX98926_WATCH_TIME_WIDTH 2 + +/* MAX98926_R038_GLOBAL_ENABLE */ +#define MAX98926_EN_MASK (1<<7) +#define MAX98926_EN_SHIFT 7 +#define MAX98926_EN_WIDTH 1 + +/* MAX98926_R03A_BOOST_LIMITER */ +#define MAX98926_BST_ILIM_MASK (0x1F<<3) +#define MAX98926_BST_ILIM_SHIFT 3 +#define MAX98926_BST_ILIM_WIDTH 5 + +/* MAX98926_R0FF_VERSION */ +#define MAX98926_REV_ID_MASK (0xFF<<0) +#define MAX98926_REV_ID_SHIFT 0 +#define MAX98926_REV_ID_WIDTH 8 + +struct max98926_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + unsigned int sysclk; + unsigned int v_slot; + unsigned int i_slot; + unsigned int ch_size; + unsigned int intrleave_mode; +}; +#endif -- 1.9.3

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[alsa-devel] [asoc:topic/topology 104/105] :undefined reference to `snd_soc_dapm_new_control'
by kbuild test robot 15 Jun '15

15 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/topology head: 932ae8809469770a07ce19d6967d2ce303befa08 commit: 8a9782346dccd82cf912552735bda619de4efd8c [104/105] ASoC: topology: Add topology core config: arm-at91_dt_defconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8a9782346dccd82cf912552735bda619de4efd8c # save the attached .config to linux build tree make.cross ARCH=arm All error/warnings (new ones prefixed by >>): sound/built-in.o: In function `soc_tplg_dapm_widget_create': >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control' --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

4 4

Re: [alsa-devel] linux-next: Tree for Jun 15 (sound/soc/codecs/wm_adsp.c)
by Randy Dunlap 15 Jun '15

15 Jun '15

On 06/15/15 02:05, Michael Ellerman wrote: > Hi all, > > Changes since 20150612: > on x86_64: WARNING: "arizona_rate_text" [sound/soc/codecs/snd-soc-wm-adsp.ko] undefined! WARNING: "arizona_rate_val" [sound/soc/codecs/snd-soc-wm-adsp.ko] undefined! Full randconfig is attached. -- ~Randy

1 0

[alsa-devel] AIC3106 problem with driver (MIC BIAS)
by Timur Karaldin 15 Jun '15

15 Jun '15

Hello, I'm using linux 3.12 kernel for TI SITARA ARM (am335x) with TI AIC3106 codec ( tlv320aic3x.c driver). There is a mic bias support through device tree option aic3x-micbias-vg. But in fact it's not working correctly. During kernel booting it's setting (after 2-3 seconds) but after finish booting it's cleared. Even if I change the default regs values in aic3x_reg, it's cleared after booting. Could somebody point me to solution? Another thing I need to do is to get mic/headset detection in mixer. There is some code in driver ("Detection" output) but actually I don't understand right way of using it. Best regards, Tim Karaldin

1 0

Re: [alsa-devel] Audio crackles with 4.1-rc1
by Takashi Iwai 15 Jun '15

15 Jun '15

At Wed, 29 Apr 2015 12:28:59 +0100, Jonathan McDowell wrote: > > On Tue, Apr 28, 2015 at 04:43:00PM +0200, Takashi Iwai wrote: > > At Tue, 28 Apr 2015 15:05:20 +0200, > > Takashi Iwai wrote: > > > At Tue, 28 Apr 2015 13:35:18 +0100, > > > Jonathan McDowell wrote: > > > > On Tue, Apr 28, 2015 at 02:00:17PM +0200, Takashi Iwai wrote: > > > > > At Tue, 28 Apr 2015 12:21:57 +0100, > > > > > Jonathan McDowell wrote: > > > > > > > > > > > Having upgraded to 4.1-rc1 from 4.0 I'm now hearing audio > > > > > > crackles at regular intervals. I'm fairly sure this is due to > > > > > > the HDA power save as once audio is playing things are fine, > > > > > > it's just when starting to play audio that I hear the crackle. > > > > > > > > > > > > System is a Dell Latitude E7240. I haven't tried a bisect yet > > > > > > but will attempt to find some time to do so in the next few > > > > > > days. It looks like there have been some changes in sound/hda/ > > > > > > between 4.0 + 4.1-rc1 so I'll concentrate on those first. > > > > > > > > > > > There are lots of code changes and enhancements wrt power saving > > > > > in 4.1, and bisection won't help so much, I'm afraid. > > > > > > > > > > First off, check the device status while you hear crackles. Is > > > > > the codec in runtime suspend (aka power save)? This can be seen > > > > > in /sys/bus/hdaudio/devices/*/power/runtime_status. Then check > > > > > the runtime status of the controller, too, found in > > > > > /sys/class/sound/card?/device/power/runtime_status. > > > > > > > > The cracking is definitely happening with the transition from suspended > > > > to active: > > > > > > > > /sys/class/sound/card0/device/power/runtime_status:suspended > > > > /sys/class/sound/card1/device/power/runtime_status:active > > > > /sys/bus/hdaudio/devices/hdaudioC0D0/power/runtime_status:suspended > > > > /sys/bus/hdaudio/devices/hdaudioC1D0/power/runtime_status:active > > > > > > > > card1 + C1D0 were suspended, I hit delete in a terminal to force a > > > > terminal bell, got the crackle and the state changed to active as above. > > > > > > Ah, so it's a click noise that happens only once per runtime PM > > > transition? I thought it were constant crackling noises from your > > > description ("regular intervals"). > > > > > > In anyway, please give alsa-info.sh output on both 4.0 and 4.1. > > > > Also, does the click noise occur only at powering up, and not at > > powering down? > > The click is only on the transition from suspended to active; not > constantly during playback and not when transitioning from active to > suspended. It doesn't appear 4.0 is doing the suspending at all; there > are no /sys/bus/hdaudio/devices/*/power/runtime_status files and the > card runtime_status is always active. alsa-info for 4.0 + 4.1-rc1 > attached. OK, thanks for clarification. One patch you can try (with or without power_save_node disablement) is below, it squashes the verb sequences at (runtime) PM resume as we did for 4.0. Let me know if this changes the behavior. Takashi --- diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 2a8aa9dfb83d..8e47d9cc369f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -79,6 +79,7 @@ struct hdac_device { bool lazy_cache:1; /* don't wake up for writes */ bool caps_overwriting:1; /* caps overwrite being in process */ bool cache_coef:1; /* cache COEF read/write too */ + bool cache_only:1; /* don't write actually registers */ }; /* device/driver type used for matching */ diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 7371e0c3926f..c5b3dcbbacdd 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -265,6 +265,9 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; + if (codec->cache_only) + return 0; + reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); verb = get_verb(reg); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e70a7fb393dd..8f79d649975b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3156,28 +3156,29 @@ static void hda_call_codec_resume(struct hda_codec *codec) { atomic_inc(&codec->core.in_pm); - if (codec->core.regmap) - regcache_mark_dirty(codec->core.regmap); - codec->power_jiffies = jiffies; hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); hda_exec_init_verbs(codec); snd_hda_jack_set_dirty_all(codec); + if (codec->core.regmap) { + regcache_mark_dirty(codec->core.regmap); + codec->core.cache_only = true; + } if (codec->patch_ops.resume) codec->patch_ops.resume(codec); - else { - if (codec->patch_ops.init) - codec->patch_ops.init(codec); - if (codec->core.regmap) - regcache_sync(codec->core.regmap); - } + else if (codec->patch_ops.init) + codec->patch_ops.init(codec); if (codec->jackpoll_interval) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); + if (codec->core.regmap) { + codec->core.cache_only = false; + regcache_sync(codec->core.regmap); + } atomic_dec(&codec->core.in_pm); } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3d2597b7037b..844d5e6ae1a1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5780,8 +5780,6 @@ int snd_hda_gen_init(struct hda_codec *codec) /* call init functions of standard auto-mute helpers */ update_automute_all(codec); - regcache_sync(codec->core.regmap); - if (spec->vmaster_mute.sw_kctl && spec->vmaster_mute.hook) snd_hda_sync_vmaster_hook(&spec->vmaster_mute); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5f44f60a6389..e4a774ddb1fc 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2211,7 +2211,6 @@ static int generic_hdmi_resume(struct hda_codec *codec) int pin_idx; codec->patch_ops.init(codec); - regcache_sync(codec->core.regmap); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b18b9c67b262..34219aa6a40e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -799,7 +799,6 @@ static int alc_resume(struct hda_codec *codec) if (!spec->no_depop_delay) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); - regcache_sync(codec->core.regmap); hda_call_check_power_status(codec, 0x01); return 0; } @@ -3059,7 +3058,6 @@ static int alc269_resume(struct hda_codec *codec) msleep(200); } - regcache_sync(codec->core.regmap); hda_call_check_power_status(codec, 0x01); /* on some machine, the BIOS will clear the codec gpio data when enter

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[alsa-devel] [PATCH 00/11] ALSA: bebob: improve support for DM1500/BeBoBv3
by Takashi Sakamoto 15 Jun '15

15 Jun '15

Hi, This patchset updates my previous one. [alsa-devel] [RFC][PATCH 00/11] improve support for DM1500/BeBoBv3 http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093320.html Changes since the RFC: * re-gain missing enumerator member * update commit comments I hope them to be merged into upstream. Takashi Sakamoto (11): ALSA: bebob: improve signal mode detection for clock source ALSA: bebob: apply new enumerator to normalize the type of clock source ALSA: bebob: preparation for replacing string literals by normalized representation for model-dependent structures ALSA: bebob: use normalized representation for the type of clock source ALSA: bebob: obsolete string literal expression for clock source ALSA: bebob: add SYT-Match support ALSA: bebob: add 'version' member for BeBoB protocol version ALSA: bebob: expand timeout for DM1500 quirk ALSA: bebob: loosen up severity of checking continuity for BeBoB v3 quirk ALSA: bebob: keep duplex streams always to keep internal multiplexer properly ALSA: bebob: add support for Behringer FCA 610/1616 sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 14 ++- sound/firewire/bebob/bebob.h | 20 +++-- sound/firewire/bebob/bebob_focusrite.c | 33 +++++--- sound/firewire/bebob/bebob_maudio.c | 23 +++-- sound/firewire/bebob/bebob_midi.c | 8 +- sound/firewire/bebob/bebob_pcm.c | 14 +-- sound/firewire/bebob/bebob_proc.c | 22 ++--- sound/firewire/bebob/bebob_stream.c | 150 ++++++++++++++++++++++++--------- sound/firewire/bebob/bebob_terratec.c | 30 +++++-- sound/firewire/bebob/bebob_yamaha.c | 20 ++++- 11 files changed, 235 insertions(+), 100 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: ml26124: Remove duplicate code
by Axel Lin 15 Jun '15

15 Jun '15

Current code has duplicate code for 16000, 32000 and 48000 sample rates. get_srate() returns negative error code for unsupported rate, so we can remove the duplicate code in the swith cases by calling get_srate() first. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ml26124.c | 58 +++++++++------------------------------------- 1 file changed, 11 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 62dda24..b74118e 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -341,6 +341,7 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); int i = get_coeff(priv->mclk, params_rate(hw_params)); + int srate; if (i < 0) return i; @@ -370,53 +371,16 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, BIT(0) | BIT(1), 0); } - switch (params_rate(hw_params)) { - case 16000: - snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, - get_srate(params_rate(hw_params))); - snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, - coeff_div[i].pllnl); - snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, - coeff_div[i].pllnh); - snd_soc_update_bits(codec, ML26124_PLLML, 0xff, - coeff_div[i].pllml); - snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, - coeff_div[i].pllmh); - snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, - coeff_div[i].plldiv); - break; - case 32000: - snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, - get_srate(params_rate(hw_params))); - snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, - coeff_div[i].pllnl); - snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, - coeff_div[i].pllnh); - snd_soc_update_bits(codec, ML26124_PLLML, 0xff, - coeff_div[i].pllml); - snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, - coeff_div[i].pllmh); - snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, - coeff_div[i].plldiv); - break; - case 48000: - snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, - get_srate(params_rate(hw_params))); - snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, - coeff_div[i].pllnl); - snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, - coeff_div[i].pllnh); - snd_soc_update_bits(codec, ML26124_PLLML, 0xff, - coeff_div[i].pllml); - snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, - coeff_div[i].pllmh); - snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, - coeff_div[i].plldiv); - break; - default: - pr_err("%s:this rate is no support for ml26124\n", __func__); - return -EINVAL; - } + srate = get_srate(params_rate(hw_params)); + if (srate < 0) + return srate; + + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, srate); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, coeff_div[i].plldiv); return 0; } -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: hda - adding a DAC/pin preference map for a HP Envy TS machine
by Hui Wang 15 Jun '15

15 Jun '15

On a HP Envy TouchSmart laptop, there are 2 speakers (main speaker and subwoofer speaker), 1 headphone and 2 DACs, without this fixup, the headphone will be assigned to a DAC and the 2 speakers will be assigned to another DAC, this assignment makes the surround-2.1 channels invalid. To fix it, here using a DAC/pin preference map to bind the main speaker to 1 DAC and the subwoofer speaker will be assigned to another DAC. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bdcda6a..dcc7fe9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_ENVY_BASS, STAC_HP_BNB13_EQ, STAC_HP_ENVY_TS_BASS, + STAC_HP_ENVY_TS_DAC_BIND, STAC_92HD83XXX_GPIO10_EAPD, STAC_92HD83XXX_MODELS }; @@ -2171,6 +2172,22 @@ static void stac92hd83xxx_fixup_gpio10_eapd(struct hda_codec *codec, spec->eapd_switch = 0; } +static void hp_envy_ts_fixup_dac_bind(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct sigmatel_spec *spec = codec->spec; + static hda_nid_t preferred_pairs[] = { + 0xd, 0x13, + 0 + }; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.preferred_dacs = preferred_pairs; +} + static const struct hda_verb hp_bnb13_eq_verbs[] = { /* 44.1KHz base */ { 0x22, 0x7A6, 0x3E }, @@ -2686,6 +2703,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_HP_ENVY_TS_DAC_BIND] = { + .type = HDA_FIXUP_FUNC, + .v.func = hp_envy_ts_fixup_dac_bind, + .chained = true, + .chain_id = STAC_HP_ENVY_TS_BASS, + }, [STAC_92HD83XXX_GPIO10_EAPD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_gpio10_eapd, @@ -2764,6 +2787,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e, "HP ENVY TS", STAC_HP_ENVY_TS_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1967, + "HP ENVY TS", STAC_HP_ENVY_TS_DAC_BIND), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940, "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: max98090: read micbias from device property
by yang.a.fang＠intel.com 15 Jun '15

15 Jun '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This patch reads max98090 micbias from acpi or dt Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/max98090.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c230626..27df17e 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2419,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec) struct max98090_cdata *cdata; enum max98090_type devtype; int ret = 0; + int err; + unsigned int micbias; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2503,8 +2505,13 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_MASK); + err = device_property_read_u32(codec->dev, "max,micbias", &micbias); + + if (err) + micbias = M98090_MBVSEL_2V8; + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, - M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + M98090_MBVSEL_MASK, micbias); max98090_add_widgets(codec); -- 1.7.9.5

5 20

[alsa-devel] [PATCH] ASoC: Intel: mrfld: simplify sst_fill_widget_module_info
by Rasmus Villemoes 15 Jun '15

15 Jun '15

Some tiny improvements, cutting 180 bytes off the generated code. - use strchr() for single-character needle - compute index using pointer subtraction instead of two strlen() calls - factor out the common check for whether the initial part of kctl->id.name (before the space) is identical to w->name. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/soc/intel/atom/sst-atom-controls.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 90aa5c0476f3..16670feeee6a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1280,36 +1280,32 @@ static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w, down_read(&card->controls_rwsem); list_for_each_entry(kctl, &card->controls, list) { - idx = strstr(kctl->id.name, " "); + idx = strchr(kctl->id.name, ' '); if (idx == NULL) continue; - index = strlen(kctl->id.name) - strlen(idx); + index = idx - (char*)kctl->id.name; + if (strncmp(kctl->id.name, w->name, index)) + continue; - if (strstr(kctl->id.name, "Volume") && - !strncmp(kctl->id.name, w->name, index)) + if (strstr(kctl->id.name, "Volume")) ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN); - else if (strstr(kctl->id.name, "params") && - !strncmp(kctl->id.name, w->name, index)) + else if (strstr(kctl->id.name, "params")) ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO); else if (strstr(kctl->id.name, "Switch") && - !strncmp(kctl->id.name, w->name, index) && strstr(kctl->id.name, "Gain")) { struct sst_gain_mixer_control *mc = (void *)kctl->private_value; mc->w = w; - } else if (strstr(kctl->id.name, "interleaver") && - !strncmp(kctl->id.name, w->name, index)) { + } else if (strstr(kctl->id.name, "interleaver")) { struct sst_enum *e = (void *)kctl->private_value; e->w = w; - } else if (strstr(kctl->id.name, "deinterleaver") && - !strncmp(kctl->id.name, w->name, index)) { - + } else if (strstr(kctl->id.name, "deinterleaver")) { struct sst_enum *e = (void *)kctl->private_value; e->w = w; -- 2.1.3

3 2

[alsa-devel] [PATCH 0/3] ASoC: Mediatek: Add support for MT8173 SoC
by Koro Chen 15 Jun '15

15 Jun '15

This adds basic support for the Mediatek AFE (Audio Front End) unit for MT8173. This patch is based on Linux 4.1-rc1 and Sascha's SCPSYS power patch [1]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). [1] https://lkml.org/lkml/2015/6/9/172 changes since RFC: - Use DPCM to model AFE AFE memory interfaces such as DL1, VUL are modeled as frontend DAIs, and AFE IOs such as I2S, HDMI are backend DAIs. Which memif and IO to be used can be set in the machine driver. There is no need to indicate this in dts anymore so include/dt-bindings/sound/mtk-afe.h is removed. AFE connections are represented by DAPM widgets and routes, and set connection APIs/tables are no longer required. (mtk-afe-connection.c and mtk-afe-connection.h are removed) Instead, mixer controls are used to do connection/disconnection.

3 13

Re: [alsa-devel] M-Audio Midisport 2x2 Anniversary edition, latency problem
by Clemens Ladisch 14 Jun '15

14 Jun '15

Ilya Dikariev wrote: >>>>> I bought a MIDI-interface von M-Audio (see subject), but there are >>>>> strange things with it on my desktop (Arch, latest kernel). The >>>>> latency is about 5 seconds. >>>> >>>> How are you measuring the latency? >>> >>> I measured it only with my feeling of time. >> >> What are the specific events between which there is an interval of >> 5 seconds? > > Between I press any key on a MIDI-keyboard and I hear the sound from > the timidity-daemon (or any other MIDI-reciever in a computer). There are lots of components involved besides the USB/MIDI interface: Timidity, any sound server (if you're using one), and the sound card's drivers. It is extremely unlikely that the USB/MIDI interface could introduce latency, because the only way to do that would be to go to the extra effort of storing the MIDI events in some buffer, and then transmitting them later. > the MIDI events from the Pitch-wheel on the keyboard arrives the MIDI- > reciever immedietly (I saw it, when i tried to check out the output > from the /dev/midi). This proves that another component introduces the latency. (Note: you could also use amidi or aseqdump to do these tests.) How did you configure the audio output of Timidity? Regards, Clemens

2 2

[alsa-devel] 2 speakers are assigned to the same DAC, this can't support 4.0/2.1 channles
by hwang4 14 Jun '15

14 Jun '15

I am trying to enable the subwoofer speaker on a HP laptop, on this machine, there are two speakers and one headphone, but the BIOS verb only enabled one speaker(nid 0xd) and one headphone(nid 0xb), I need to use quirk in the kernel driver to configure the second speaker (subwoofer speaker, nid 0x10). Under current alsa driver, the headphone will be assigned a dac (nid 0x13) and the 2 speakers will be assigned a dac (nid 0x14), this assignment is not good since 2 speakers share the same dac, this means 2 speakers can't work at the same time to support 4.0/2.1 channels. On another Dell machine with realtek codec, there are also 2 speakers, 1 headphone and 2 dacs, on this machine, 1 speaker and 1 headphone are assigned 1 dac, and the other speaker is assigned another dac, so there is no problem for this machine to support 4.0/2.1 channels. Through debugging, I found on Dell machine, the speaker nid only has one connection to dac (hardwired), so when driver assign dac to it, the map_single() can successfully assign the each dac to the 2 speakers respectively. But on that HP machine, the speaker has multiple connections for dac, the map_single() can't work for this machine. The alsa-info.txt for that HP machine is at http://pastebin.ubuntu.com/11667947/ The alsa-info.txt for that Dell machine is at http://pastebin.ubuntu.com/11667982/ To fix this problem, I can use the quirk in the driver to make the speaker have only one connection for dac or do some change in the hda_generic.c like below, in your opinion, which one is better or probably you have a different idea to fix this problem? diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac0db16..8194ff1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1363,6 +1363,7 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, dac = try_dac(codec, get_primary_out(codec, i), pin); if (!dac) dac = try_dac(codec, dacs[0], pin); if (!dac) dac = try_dac(codec, get_primary_out(codec, i), pin); if (dac) { @@ -1762,6 +1763,12 @@ static int fill_and_eval_dacs(struct hda_codec *codec, if (err < 0) return err; badness += err; + /* if there are 2 speakers and both of them are assigned to the same dac, + we need to increase the badness for this situation, because in this situation + the 2 speakers can't work together to support 4.0/2.1 channels */ + if (cfg->speaker_outs == 2 && err < spec->extra_out_badness->no_dac && + spec->multiout.extra_out_nid[1] == 0) + badness += spec->extra_out_badness->no_dac; } if (!spec->no_multi_io && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bdcda6a..717259c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2764,6 +2764,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e, "HP ENVY TS", STAC_HP_ENVY_TS_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1967, + "HP ENVY TS", STAC_HP_ENVY_TS_BASS), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940, "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941,

5 27

Re: [alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Raymond Yau 13 Jun '15

13 Jun '15

> > Do you mean hdajackretask cannot shutdown pulseaudio or you still have other > > applicaion using the sound card which prevent dynamic reconfiguration of > > snd-hda-intel ? > > > I was testing using a browser on a youtube video. When I apply the > changes, I lose the sound (pulseaudio gets killed), that message > appears and I have to refresh the page to get sound again - but > nothing changes. > > > > > Can you post the output of alsa-info.sh after you patched the line out jacks > > ? > http://www.alsa-project.org/db/?f=82a057b76a781edb53217b2cc096bce35459ab0a > > > Which node is the headphone spdif combo jack ? > > > Using hdajacksensetest I found out the mapping: > SPDIF port (jack 1): Pin 0x16 (Green Line Out, Rear side) > jack 2: Pin 0x18 (Green Line Out, Rear side) > jack 3: Pin 0x15 (Green Line Out, Rear side) > > > Do both headphone jacks have headphone , headset or line out icon ? > You can see the icons here [0] at page 20 > > > Do you mean switch to 5.1 profile in sound preference is not enough ? > By listening on Youtube I can't hear the center and subwoofer without > modifying the pulseaudio conf > > [0] http://dlcdnet.asus.com/pub/ASUS/nb/G751JM/0409_E9481.pdf?_ga=1.206896403.1… If your specifcaion and user guide does not mention the support of surround 5.1 You need to find out the internal subwoofer use both channel or just left or right speaker-test -c4 -t wav -D hw:0,0 If you hear high frequency from subwoofer, this mean you need software low pass filtter Two headphone jacks, internal speaker and subwoofer There are only three playback volume controls http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093668.html Your case are A. 3 DACs, 2 HP, 2 speakers Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Bass Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Surround Phantom Jack", index=0, device=0 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 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90170130: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x0c 0x0d* 0x0e

2 3

[alsa-devel] ASoC: How to declare volume TLV dB scale?
by Sergej Sawazki 13 Jun '15

13 Jun '15

Dear all, could someone please help me to understand the mixer controls TLV scale? What would be the DECLARE_TLV_DB_SCALE values for the following example? reg. value: 0 .. 0x3ff phys. value: 0dB .. -128dB volume: 100% .. 0%(mute) What does TLV actually stand for? Many thanks in advance, Sergej Sawazki

3 7

[alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Nicolae Rosia 13 Jun '15

13 Jun '15

Hello, I can't get 4.0/5.1 sound working on my laptop [0] [1] with Fedora 22. With Linux 4.0 there is no output on jacks but with 4.1-rc6 I'm getting sound only on one output. Switching between pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-stereo and pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-surround-40 only boosts the volume on the first output. Attached is dmesg [2], lsmod [3], result of lspci -vv [4] and pacmd list-cards and list-sinks [5] and output of various information from /proc/asound while runnig 4.1-rc6. Let me know if you need more information or how I can help. Best regards, Nicolae Rosia [rosia@localhost ~]$ cat /proc/asound/devices 1: : sequencer 2: [ 0] : control 3: [ 0- 3]: digital audio playback 4: [ 0- 7]: digital audio playback 5: [ 0- 8]: digital audio playback 6: [ 0- 0]: hardware dependent 7: [ 1] : control 8: [ 1- 0]: digital audio playback 9: [ 1- 0]: digital audio capture 10: [ 1- 1]: digital audio playback 11: [ 1- 0]: hardware dependent 33: : timer [rosia@localhost ~]$ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xeda14000 irq 33 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xeda10000 irq 32 [rosia@localhost ~]$ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel [rosia@localhost ~]$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.1.0-0.rc6.git2.1.fc23.x86_64. [0] http://www.emag.ro/laptop-gaming-asus-cu-procesor-intelr-coretm-i7-4710hq-2… [1] https://www.asus.com/Notebooks_Ultrabooks/ASUS_ROG_G751JM/specifications/ [2] http://pastebin.com/rm3dPPYm [3] http://pastebin.com/gHZWjgBx [4] http://pastebin.com/UUf91Q31 [5] http://pastebin.com/ca5dcxy8

3 12

Re: [alsa-devel] [PATCH] hda-beep: Update authors dead email address
by Takashi Iwai 13 Jun '15

13 Jun '15

At Fri, 12 Jun 2015 21:47:46 -0700, Matt Ranostay wrote: > > Signed-off-by: Matt Ranostay <mranostay(a)gmail.com> Applied, thanks. Takashi > --- > sound/pci/hda/hda_beep.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c > index 3364dc0..c397e7d 100644 > --- a/sound/pci/hda/hda_beep.c > +++ b/sound/pci/hda/hda_beep.c > @@ -1,7 +1,7 @@ > /* > * Digital Beep Input Interface for HD-audio codec > * > - * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> > + * Author: Matt Ranostay <mranostay(a)gmail.com> > * Copyright (c) 2008 Embedded Alley Solutions Inc > * > * This driver is free software; you can redistribute it and/or modify > -- > 1.9.1 >

1 0

[alsa-devel] [PATCH] hda-beep: Update authors dead email address
by Matt Ranostay 13 Jun '15

13 Jun '15

Signed-off-by: Matt Ranostay <mranostay(a)gmail.com> --- sound/pci/hda/hda_beep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 3364dc0..c397e7d 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -1,7 +1,7 @@ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> + * Author: Matt Ranostay <mranostay(a)gmail.com> * Copyright (c) 2008 Embedded Alley Solutions Inc * * This driver is free software; you can redistribute it and/or modify -- 1.9.1

1 0

[alsa-devel] [PATCH 1/1 linux-next] ALSA: gus: use swap() in snd_ics_put_double()
by Fabian Frederick 12 Jun '15

12 Jun '15

Use kernel.h macro definition. Thanks to Julia Lawall for Coccinelle scripting support. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/isa/gus/gus_mixer.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/isa/gus/gus_mixer.c b/sound/isa/gus/gus_mixer.c index 0dd4341..3b5d9a7 100644 --- a/sound/isa/gus/gus_mixer.c +++ b/sound/isa/gus/gus_mixer.c @@ -109,7 +109,7 @@ static int snd_ics_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem unsigned long flags; int addr = kcontrol->private_value & 0xff; int change; - unsigned char val1, val2, oval1, oval2, tmp; + unsigned char val1, val2, oval1, oval2; val1 = ucontrol->value.integer.value[0] & 127; val2 = ucontrol->value.integer.value[1] & 127; @@ -120,11 +120,8 @@ static int snd_ics_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem gus->gf1.ics_regs[addr][0] = val1; gus->gf1.ics_regs[addr][1] = val2; if (gus->ics_flag && gus->ics_flipped && - (addr == SNDRV_ICS_GF1_DEV || addr == SNDRV_ICS_MASTER_DEV)) { - tmp = val1; - val1 = val2; - val2 = tmp; - } + (addr == SNDRV_ICS_GF1_DEV || addr == SNDRV_ICS_MASTER_DEV)) + swap(val1, val2); addr <<= 3; outb(addr | 0, GUSP(gus, MIXCNTRLPORT)); outb(1, GUSP(gus, MIXDATAPORT)); -- 2.4.2

2 1

[alsa-devel] [PATCH] ALSA: aoa: convert bus code to use dev_groups
by Quentin Lambert 12 Jun '15

12 Jun '15

You should use dev_groups instead of the dev_attrs field of struct bus_type. This converts the MDIO bus code to use the correct field. These modifications were made using Coccinelle. Signed-off-by: Quentin Lambert <lambert.quentin(a)gmail.com> --- sound/aoa/soundbus/core.c | 3 ++- sound/aoa/soundbus/soundbus.h | 2 +- sound/aoa/soundbus/sysfs.c | 13 ++++++++----- 3 files changed, 11 insertions(+), 7 deletions(-) --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -150,6 +150,7 @@ static int soundbus_device_resume(struct #endif /* CONFIG_PM */ +ATTRIBUTE_GROUPS(soundbus_dev); static struct bus_type soundbus_bus_type = { .name = "aoa-soundbus", .probe = soundbus_probe, @@ -160,7 +161,7 @@ static struct bus_type soundbus_bus_type .suspend = soundbus_device_suspend, .resume = soundbus_device_resume, #endif - .dev_attrs = soundbus_dev_attrs, + .dev_groups = soundbus_dev_groups, }; int soundbus_add_one(struct soundbus_dev *dev) --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -199,6 +199,6 @@ struct soundbus_driver { extern int soundbus_register_driver(struct soundbus_driver *drv); extern void soundbus_unregister_driver(struct soundbus_driver *drv); -extern struct device_attribute soundbus_dev_attrs[]; +extern struct attribute *soundbus_dev_attrs[]; #endif /* __SOUNDBUS_H */ --- a/sound/aoa/soundbus/sysfs.c +++ b/sound/aoa/soundbus/sysfs.c @@ -30,13 +30,16 @@ static ssize_t modalias_show(struct devi return length; } +static DEVICE_ATTR_RO(modalias); soundbus_config_of_attr (name, "%s\n"); +static DEVICE_ATTR_RO(name); soundbus_config_of_attr (type, "%s\n"); +static DEVICE_ATTR_RO(type); -struct device_attribute soundbus_dev_attrs[] = { - __ATTR_RO(name), - __ATTR_RO(type), - __ATTR_RO(modalias), - __ATTR_NULL +struct attribute *soundbus_dev_attrs[] = { + &dev_attr_name.attr, + &dev_attr_type.attr, + &dev_attr_modalias.attr, + NULL, };

4 19

[alsa-devel] [PATCH 1/1 linux-next] sound: oss/sb_audio: use swap() in sb_audio_close()
by Fabian Frederick 12 Jun '15

12 Jun '15

Use kernel.h macro definition. Thanks to Julia Lawall for Coccinelle scripting support. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/oss/sb_audio.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 048439a..dc91072 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -102,12 +102,8 @@ void sb_audio_close(int dev) if(devc->duplex && !devc->fullduplex && (devc->opened & OPEN_READ) && (devc->opened & OPEN_WRITE)) - { - struct dma_buffparms *dmap_temp; - dmap_temp = audio_devs[dev]->dmap_out; - audio_devs[dev]->dmap_out = audio_devs[dev]->dmap_in; - audio_devs[dev]->dmap_in = dmap_temp; - } + swap(audio_devs[dev]->dmap_out, audio_devs[dev]->dmap_in); + audio_devs[dev]->dmap_out->dma = devc->dma8; audio_devs[dev]->dmap_in->dma = ( devc->duplex ) ? devc->dma16 : devc->dma8; -- 2.4.2

1 0

[alsa-devel] [PATCH] ASoC: wm8523: Set bclk ratio for master mode
by Julian Scheel 12 Jun '15

12 Jun '15

When running in master mode the bclk divider must be configured to generate a sane bitclock. Pick the smallest fs multiplicator, which can hold all transmitted bits. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/soc/codecs/wm8523.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index ec1f574..9c60058 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -113,6 +113,15 @@ static struct { { 7, 1152 }, }; +static struct { + int value; + int ratio; +} bclk_ratios[WM8523_NUM_RATES] = { + { 2, 32 }, + { 3, 64 }, + { 4, 128 }, +}; + static int wm8523_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,6 +171,23 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, aifctrl2 &= ~WM8523_SR_MASK; aifctrl2 |= lrclk_ratios[i].value; + if (aifctrl1 & WM8523_AIF_MSTR) { + /* Find a fs->bclk ratio */ + for (i = 0; i < ARRAY_SIZE(bclk_ratios); i++) + if (params_width(params) * 2 <= bclk_ratios[i].ratio) + break; + + if (i == ARRAY_SIZE(bclk_ratios)) { + dev_err(codec->dev, + "No matching BCLK/fs ratio for word length %d\n", + params_width(params)); + return -EINVAL; + } + + aifctrl2 &= ~WM8523_BCLKDIV_MASK; + aifctrl2 |= bclk_ratios[i].value << WM8523_BCLKDIV_SHIFT; + } + aifctrl1 &= ~WM8523_WL_MASK; switch (params_width(params)) { case 16: -- 2.4.0

3 2

[alsa-devel] [PATCH - seq 1/1] ALSA seq: expose the card number of ALSA seq clients
by Adam Goode 12 Jun '15

12 Jun '15

Expose the card number of seq clients. This allows interested userspace programs to discover the hardware device that backs a particular seq client. Before this change, the only way to get information about the hardware for a client was by using brittle heuristics. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 5a5fa49..c2a659be 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -25,7 +25,7 @@ #include <sound/asound.h> /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -357,7 +357,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card_number; /* RO: card number, or -1 if no card. Added in protocol version 1.0.2 */ + char reserved[60]; /* for future use */ }; diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index edbdab8..fcae784 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -216,7 +216,7 @@ int __init client_init_data(void) } -static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) +static struct snd_seq_client *seq_create_client1(int client_index, int poolsize, int card_number) { unsigned long flags; int c; @@ -232,6 +232,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) return NULL; } client->type = NO_CLIENT; + client->card_number = card_number; snd_use_lock_init(&client->use_lock); rwlock_init(&client->ports_lock); mutex_init(&client->ports_mutex); @@ -327,7 +328,7 @@ static int snd_seq_open(struct inode *inode, struct file *file) if (mutex_lock_interruptible(&register_mutex)) return -ERESTARTSYS; - client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS); + client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS, -1); if (client == NULL) { mutex_unlock(&register_mutex); return -ENOMEM; /* failure code */ @@ -1194,6 +1195,7 @@ static void get_client_info(struct snd_seq_client *cptr, info->event_lost = cptr->event_lost; memcpy(info->event_filter, cptr->event_filter, 32); info->num_ports = cptr->num_ports; + info->card_number = cptr->card_number; memset(info->reserved, 0, sizeof(info->reserved)); } @@ -2239,6 +2241,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, { struct snd_seq_client *client; va_list args; + int card_number = -1; if (snd_BUG_ON(in_interrupt())) return -EBUSY; @@ -2252,6 +2255,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, return -ERESTARTSYS; if (card) { + card_number = card->number; client_index += SNDRV_SEQ_GLOBAL_CLIENTS + card->number * SNDRV_SEQ_CLIENTS_PER_CARD; if (client_index >= SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN) @@ -2259,7 +2263,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, } /* empty write queue as default */ - client = seq_create_client1(client_index, 0); + client = seq_create_client1(client_index, 0, card_number); if (client == NULL) { mutex_unlock(&register_mutex); return -EBUSY; /* failure code */ @@ -2533,9 +2537,13 @@ void snd_seq_info_clients_read(struct snd_info_entry *entry, continue; } - snd_iprintf(buffer, "Client %3d : \"%s\" [%s]\n", + snd_iprintf(buffer, "Client %3d : \"%s\" [%s]", c, client->name, client->type == USER_CLIENT ? "User" : "Kernel"); + if (client->card_number != -1) { + snd_iprintf(buffer, ", card %d", client->card_number); + } + snd_iprintf(buffer, "\n"); snd_seq_info_dump_ports(buffer, client); if (snd_seq_write_pool_allocated(client)) { snd_iprintf(buffer, " Output pool :\n"); diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 20f0a72..627be9d 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -50,6 +50,7 @@ struct snd_seq_client { accept_output: 1; char name[64]; /* client name */ int number; /* client number */ + int card_number; /* card number, or -1 if no card */ unsigned int filter; /* filter flags */ DECLARE_BITMAP(event_filter, 256); snd_use_lock_t use_lock; -- 2.2.0.rc0.207.ga3a616c

2 4

[alsa-devel] [PATCH 1/2] ALSA: hda - Continue probing even if i915 binding fails
by Takashi Iwai 12 Jun '15

12 Jun '15

Currently snd-hda-intel driver aborts the probing of Intel HD-audio controller with i915 power well management when binding with i915 driver via hda_i915_init() fails. This is no big problem for Haswell and Broadwell where the HD-audio controllers are dedicated to HDMI/DP, thus i915 link is mandatory. However, Skylake, Baytrail and Braswell have only one controller and both HDMI/DP and analog codecs share the same bus. Thus, even if HDMI/DP isn't usable, we should keep the controller working for other codecs. For fixing this, this patch simply allows continuing the probing even if hda_i915_init() call fails. This may leave stale sound components for HDMI/DP devices that are unbound with graphics. We could abort the probing selectively, but from the code simplicity POV, it's better to continue in all cases. Reported-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is for 4.1. Appying to 4.2 may result in trivial conflicts. sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fea198c58196..8a0af6770e1d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1855,7 +1855,7 @@ static int azx_probe_continue(struct azx *chip) #ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(hda); if (err < 0) - goto out_free; + goto skip_i915; err = hda_display_power(hda, true); if (err < 0) { dev_err(chip->card->dev, @@ -1865,6 +1865,7 @@ static int azx_probe_continue(struct azx *chip) #endif } + skip_i915: err = azx_first_init(chip); if (err < 0) goto out_free; -- 2.4.3

3 11

[alsa-devel] [PATCH 4/4] ASoC: wm8960: the enum of "DAC Polarity" should be wm8960_enum[1]
by Zidan Wang 12 Jun '15

12 Jun '15

the enum of "DAC Polarity" should be wm8960_enum[1]. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 62246f1..d482c0f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -254,7 +254,7 @@ SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), SOC_ENUM("ADC Polarity", wm8960_enum[0]), SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), -SOC_ENUM("DAC Polarity", wm8960_enum[2]), +SOC_ENUM("DAC Polarity", wm8960_enum[1]), SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0, wm8960_get_deemph, wm8960_put_deemph), -- 1.9.1

3 2

[alsa-devel] [PATCH 3/4] ASoC: wm8960: add kcontrol to select ADC data output
by Zidan Wang 12 Jun '15

12 Jun '15

add kcontrol to select ADC data output. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index c1be15c..62246f1 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -141,6 +141,12 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adc_data_sel[] = { + "ldata = left ADC; rdata = right ADC", + "ldata = left ADC; rdata = left ADC", + "ldata = right ADC; rdata = right ADC", + "ldata = right ADC; rdata = left ADC", +}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -149,6 +155,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), + SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_sel), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -279,6 +286,8 @@ SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", WM8960_BYPASS2, 4, 7, 1, bypass_tlv), SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), + +SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: core: Pass kcontrol pointer to bytes tlv callbacks
by Ben Zhang 12 Jun '15

12 Jun '15

The get/put callbacks need the kcontrol pointer to get context information like snd_soc_codec and drvdata. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- include/sound/soc.h | 6 ++++-- sound/soc/soc-ops.c | 4 ++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index fcb312b..404265d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1169,8 +1169,10 @@ struct soc_bytes { struct soc_bytes_ext { int max; /* used for TLV byte control */ - int (*get)(unsigned int __user *bytes, unsigned int size); - int (*put)(const unsigned int __user *bytes, unsigned int size); + int (*get)(struct snd_kcontrol *kcontrol, + unsigned int __user *bytes, unsigned int size); + int (*put)(struct snd_kcontrol *kcontrol, + const unsigned int __user *bytes, unsigned int size); }; /* multi register control */ diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 100d92b..7f53da9 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -751,11 +751,11 @@ int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, switch (op_flag) { case SNDRV_CTL_TLV_OP_READ: if (params->get) - ret = params->get(tlv, count); + ret = params->get(kcontrol, tlv, count); break; case SNDRV_CTL_TLV_OP_WRITE: if (params->put) - ret = params->put(tlv, count); + ret = params->put(kcontrol, tlv, count); break; } return ret; -- 2.2.0.rc0.207.ga3a616c

3 4

[alsa-devel] [PATCH] ASoC: Intel: Added I2C dependency to two BSW machine drivers
by yang.a.fang＠intel.com 12 Jun '15

12 Jun '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> Fixed below error/warnings sound/built-in.o: In function `rt5645_i2c_probe': >> rt5645.c:(.text+0xe38f5): undefined reference to >> `devm_regmap_init_i2c' sound/built-in.o: In function `rt5645_i2c_driver_exit': >> rt5645.c:(.exit.text+0x60e): undefined reference to `i2c_del_driver' sound/built-in.o: In function `rt5645_i2c_driver_init': >> rt5645.c:(.init.text+0x1a90): undefined reference to >> `i2c_register_driver' Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 791953f..f3060a4 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -112,7 +112,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS + depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -123,7 +123,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec" - depends on X86_INTEL_LPSS + depends on X86_INTEL_LPSS && I2C select SND_SOC_MAX98090 select SND_SOC_TS3A227E select SND_SST_MFLD_PLATFORM -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Use devm_gpiod_get_optional for hp-detect
by Axel Lin 12 Jun '15

12 Jun '15

Since hp-detect is optional, use devm_gpiod_get_optional instead. In additional, it should return error if devm_gpiod_get_optional fails. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5645.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7d04b1a..04d9ff8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3255,11 +3255,12 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } - rt5645->gpiod_hp_det = devm_gpiod_get(&i2c->dev, "hp-detect", GPIOD_IN); + rt5645->gpiod_hp_det = devm_gpiod_get_optional(&i2c->dev, "hp-detect", + GPIOD_IN); if (IS_ERR(rt5645->gpiod_hp_det)) { - rt5645->gpiod_hp_det = NULL; dev_err(&i2c->dev, "failed to initialize gpiod\n"); + return PTR_ERR(rt5645->gpiod_hp_det); } rt5645->regmap = devm_regmap_init_i2c(i2c, &rt5645_regmap); -- 2.1.0

2 1

[alsa-devel] [PATCH V2] PCM: snd_pcm_xxxx_drain() maybe blocked after suspend and resume
by Shengjiu Wang 12 Jun '15

12 Jun '15

After suspend and resume, the alsa driver is stopped. But if alsa-lib run into snd_pcm_xxxx_drain(), it need to wait avail >= pcm->stop_threshold, otherwise, it will not exit the loop, so finally it is blocked at poll() of snd_pcm_wait_nocheck(pcm, -1). This patch is to add state check after snd_pcm_wait_nocheck(pcm, -1), if the state is SND_PCM_STATE_SUSPENDED, then return error. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- changes in v2: - Add state check in beginning of drain() - Add fix for pcm_dshare() - Mov the state check to the place that is after snd_pcm_direct_clear_timer_queue(dmix). That the suspend may happen just before calling snd_pcm_direct_clear_timer_queue(dmix), which will force the next poll to wait. So move state check to new place may safty. src/pcm/pcm_dmix.c | 14 ++++++++++++++ src/pcm/pcm_dshare.c | 14 ++++++++++++++ 2 files changed, 28 insertions(+) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index babde6a..91148b9 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -616,6 +616,13 @@ static int snd_pcm_dmix_drain(snd_pcm_t *pcm) snd_pcm_uframes_t stop_threshold; int err; + switch (snd_pcm_state(dmix->spcm)) { + case SND_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + default: + break; + } + if (dmix->state == SND_PCM_STATE_OPEN) return -EBADFD; if (pcm->mode & SND_PCM_NONBLOCK) @@ -648,6 +655,13 @@ static int snd_pcm_dmix_drain(snd_pcm_t *pcm) snd_pcm_dmix_sync_area(pcm); snd_pcm_wait_nocheck(pcm, -1); snd_pcm_direct_clear_timer_queue(dmix); /* force poll to wait */ + + switch (snd_pcm_state(dmix->spcm)) { + case SND_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + default: + break; + } } } while (dmix->state == SND_PCM_STATE_DRAINING); pcm->stop_threshold = stop_threshold; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 020e6f7..24975d4 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -367,6 +367,13 @@ static int snd_pcm_dshare_drain(snd_pcm_t *pcm) snd_pcm_uframes_t stop_threshold; int err; + switch (snd_pcm_state(dshare->spcm)) { + case SND_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + default: + break; + } + if (dshare->state == SND_PCM_STATE_OPEN) return -EBADFD; if (pcm->mode & SND_PCM_NONBLOCK) @@ -399,6 +406,13 @@ static int snd_pcm_dshare_drain(snd_pcm_t *pcm) snd_pcm_dshare_sync_area(pcm); snd_pcm_wait_nocheck(pcm, -1); snd_pcm_direct_clear_timer_queue(dshare); /* force poll to wait */ + + switch (snd_pcm_state(dshare->spcm)) { + case SND_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + default: + break; + } } } while (dshare->state == SND_PCM_STATE_DRAINING); pcm->stop_threshold = stop_threshold; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: sst: use ; instead of , at the of a C statement
by Sebastian Andrzej Siewior 12 Jun '15

12 Jun '15

This was spotted by Fernando Lopez-Lezcano <nando(a)ccrma.Stanford.EDU> while he tried to compile a -RT kernel with this driver enabled. "make C=2" would also warn about this. This is is based on his patch. Reported-by: Fernando Lopez-Lezcano <nando(a)ccrma.Stanford.EDU> Signed-off-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> --- sound/soc/intel/atom/sst/sst.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 96c2e420cce6..b0c3e0e56302 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -368,8 +368,8 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx, * initialize by FW or driver when firmware is loaded */ spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); - sst_shim_write64(shim, SST_IMRX, shim_regs->imrx), - sst_shim_write64(shim, SST_CSR, shim_regs->csr), + sst_shim_write64(shim, SST_IMRX, shim_regs->imrx): + sst_shim_write64(shim, SST_CSR, shim_regs->csr); spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } -- 2.1.4

4 4

[alsa-devel] [PATCH 1/2] Revert "ASoC: atmel: compile pcm driver in snd-soc-atmel_ssc_dai"
by Alexandre Belloni 12 Jun '15

12 Jun '15

After discussing with the Kconfig maintainer, we found a better fiw allowing to keep each driver as modules. This reverts commit 0ef9dc139db2fca304ce4eadb5b8fb40d3dedb5e. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Kconfig | 4 ++-- sound/soc/atmel/Makefile | 8 +++++--- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index c3152072d682..93abe4e6d596 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,10 +9,10 @@ config SND_ATMEL_SOC if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC - bool + tristate config SND_ATMEL_SOC_DMA - bool + tristate select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 4fa7ac91f972..b327e5cc8de3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,8 +1,10 @@ # AT91 Platform Support -snd-soc-atmel-pcm-$(CONFIG_SND_ATMEL_SOC_PDC) := atmel-pcm-pdc.o -snd-soc-atmel-pcm-$(CONFIG_SND_ATMEL_SOC_DMA) += atmel-pcm-dma.o -snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o $(snd-soc-atmel-pcm-y) +snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o +snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o +obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o +obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support -- 2.1.4

2 2

[alsa-devel] [PATCH v2] ASoC: rt5645: Add the device tree parser
by Oder Chiou 12 Jun '15

12 Jun '15

Modify the RT5645 driver to parse platform data from device tree. This is missing from previous patch in sound/soc/codecs/rt5645.c, that was present in v3. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7d04b1a..1884264 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3231,6 +3231,20 @@ static struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) +{ + rt5645->pdata.in2_diff = device_property_read_bool(dev, + "realtek,in2-differential"); + device_property_read_u32(dev, + "realtek,dmic1-data-pin", &rt5645->pdata.dmic1_data_pin); + device_property_read_u32(dev, + "realtek,dmic2-data-pin", &rt5645->pdata.dmic2_data_pin); + device_property_read_u32(dev, + "realtek,jd-mode", &rt5645->pdata.jd_mode); + + return 0; +} + static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3247,13 +3261,12 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, rt5645->i2c = i2c; i2c_set_clientdata(i2c, rt5645); - if (pdata) { + if (pdata) rt5645->pdata = *pdata; - } else { - if (dmi_check_system(dmi_platform_intel_braswell)) { - rt5645->pdata = *rt5645_pdata; - } - } + else if (dmi_check_system(dmi_platform_intel_braswell)) + rt5645->pdata = *rt5645_pdata; + else + rt5645_parse_dt(rt5645, &i2c->dev); rt5645->gpiod_hp_det = devm_gpiod_get(&i2c->dev, "hp-detect", GPIOD_IN); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fixed incorrect sst pcm_delay
by yang.a.fang＠intel.com 12 Jun '15

12 Jun '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This patch corrected pcm_delay calculation in BSW sst driver Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 7b50a9d..620da1d 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -533,7 +533,7 @@ static inline int sst_calc_tstamp(struct intel_sst_drv *ctx, info->buffer_ptr = pointer_samples / substream->runtime->channels; - info->pcm_delay = delay_frames / substream->runtime->channels; + info->pcm_delay = delay_frames; dev_dbg(ctx->dev, "buffer ptr %llu pcm_delay rep: %llu\n", info->buffer_ptr, info->pcm_delay); return 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rt286: Add RL6347A class device shared support for RT286
by Oder Chiou 12 Jun '15

12 Jun '15

The patch separates the IO function from the rt286. It is prepared to share for new chips that support the same IO function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 5 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rl6347a.c | 128 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rl6347a.h | 32 ++++++++++++ sound/soc/codecs/rt286.c | 97 +++------------------------------- 5 files changed, 173 insertions(+), 91 deletions(-) create mode 100644 sound/soc/codecs/rl6347a.c create mode 100644 sound/soc/codecs/rl6347a.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9b36011..efaafce 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -509,6 +509,11 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m +config SND_SOC_RL6347A + tristate + default y if SND_SOC_RT286=y + default m if SND_SOC_RT286=m + config SND_SOC_RT286 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3dcf5ac..cf160d9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -77,6 +77,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -263,6 +264,7 @@ 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_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rl6347a.c b/sound/soc/codecs/rl6347a.c new file mode 100644 index 0000000..91d5166 --- /dev/null +++ b/sound/soc/codecs/rl6347a.c @@ -0,0 +1,128 @@ +/* + * rl6347a.c - RL6347A class device shared support + * + * Copyright 2015 Realtek Semiconductor Corp. + * + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/dmi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/hda_verbs.h> + +#include "rl6347a.h" + +int rl6347a_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rl6347a_priv *rl6347a = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /* handle index registers */ + if (reg <= 0xff) { + rl6347a_hw_write(client, RL6347A_COEF_INDEX, reg); + for (i = 0; i < rl6347a->index_cache_size; i++) { + if (reg == rl6347a->index_cache[i].reg) { + rl6347a->index_cache[i].def = value; + break; + } + + } + reg = RL6347A_PROC_COEF; + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} +EXPORT_SYMBOL_GPL(rl6347a_hw_write); + +int rl6347a_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /* handle index registers */ + if (reg <= 0xff) { + rl6347a_hw_write(client, RL6347A_COEF_INDEX, reg); + reg = RL6347A_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rl6347a_hw_read); + +MODULE_DESCRIPTION("RL6347A class device shared support"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rl6347a.h b/sound/soc/codecs/rl6347a.h new file mode 100644 index 0000000..1cb56e5 --- /dev/null +++ b/sound/soc/codecs/rl6347a.h @@ -0,0 +1,32 @@ +/* + * rl6347a.h - RL6347A class device shared support + * + * Copyright 2015 Realtek Semiconductor Corp. + * + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#ifndef __RL6347A_H__ +#define __RL6347A_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RL6347A_VENDOR_REGISTERS 0x20 + +#define RL6347A_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RL6347A_VENDOR_REGISTERS, 0) +#define RL6347A_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RL6347A_VENDOR_REGISTERS, 0) + +struct rl6347a_priv { + struct reg_default *index_cache; + int index_cache_size; +}; + +int rl6347a_hw_write(void *context, unsigned int reg, unsigned int value); +int rl6347a_hw_read(void *context, unsigned int reg, unsigned int *value); + +#endif /* __RL6347A_H__ */ diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index c6cca06..5c43e26 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -31,12 +31,15 @@ #include <sound/rt286.h> #include <sound/hda_verbs.h> +#include "rl6347a.h" #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { + struct reg_default *index_cache; + int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; struct rt286_platform_data pdata; @@ -45,7 +48,6 @@ struct rt286_priv { struct delayed_work jack_detect_work; int sys_clk; int clk_id; - struct reg_default *index_cache; }; static struct reg_default rt286_index_def[] = { @@ -185,94 +187,6 @@ static bool rt286_readable_register(struct device *dev, unsigned int reg) } } -static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) -{ - struct i2c_client *client = context; - struct rt286_priv *rt286 = i2c_get_clientdata(client); - u8 data[4]; - int ret, i; - - /* handle index registers */ - if (reg <= 0xff) { - rt286_hw_write(client, RT286_COEF_INDEX, reg); - for (i = 0; i < INDEX_CACHE_SIZE; i++) { - if (reg == rt286->index_cache[i].reg) { - rt286->index_cache[i].def = value; - break; - } - - } - reg = RT286_PROC_COEF; - } - - data[0] = (reg >> 24) & 0xff; - data[1] = (reg >> 16) & 0xff; - /* - * 4 bit VID: reg should be 0 - * 12 bit VID: value should be 0 - * So we use an OR operator to handle it rather than use if condition. - */ - data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); - data[3] = value & 0xff; - - ret = i2c_master_send(client, data, 4); - - if (ret == 4) - return 0; - else - pr_err("ret=%d\n", ret); - if (ret < 0) - return ret; - else - return -EIO; -} - -static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) -{ - struct i2c_client *client = context; - struct i2c_msg xfer[2]; - int ret; - __be32 be_reg; - unsigned int index, vid, buf = 0x0; - - /* handle index registers */ - if (reg <= 0xff) { - rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; - } - - reg = reg | 0x80000; - vid = (reg >> 8) & 0xfff; - - if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { - index = (reg >> 8) & 0xf; - reg = (reg & ~0xf0f) | index; - } - be_reg = cpu_to_be32(reg); - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 4; - xfer[0].buf = (u8 *)&be_reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 4; - xfer[1].buf = (u8 *)&buf; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret < 0) - return ret; - else if (ret != 2) - return -EIO; - - *value = be32_to_cpu(buf); - - return 0; -} - #ifdef CONFIG_PM static void rt286_index_sync(struct snd_soc_codec *codec) { @@ -1174,8 +1088,8 @@ static const struct regmap_config rt286_regmap = { .max_register = 0x02370100, .volatile_reg = rt286_volatile_register, .readable_reg = rt286_readable_register, - .reg_write = rt286_hw_write, - .reg_read = rt286_hw_read, + .reg_write = rl6347a_hw_write, + .reg_read = rl6347a_hw_read, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt286_reg, .num_reg_defaults = ARRAY_SIZE(rt286_reg), @@ -1248,6 +1162,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } rt286->index_cache = rt286_index_def; + rt286->index_cache_size = INDEX_CACHE_SIZE; rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoc: rt5645: move RT5650 muxes to rt5650_specific_dapm_widgets
by Michele Curti 12 Jun '15

12 Jun '15

Developing a driver for an Asus X205TA laptop I get these dmesg errors: rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC1 Swap Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC2 Swap Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC3 Swap Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 L Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 R Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 L Mux has no paths rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 R Mux has no paths so, move these muxes to the rt5650_specific_dapm_widgets[] list. Signed-off-by: Michele Curti <michele.curti(a)gmail.com> --- sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++----------------- 1 file changed, 18 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 46dc795..e92e734 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1706,15 +1706,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if1_adc_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC1 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc1_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC2 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc2_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC3 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc3_in_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 ADC Mux", SND_SOC_NOPM, - 0, 0, &rt5650_if1_adc_in_mux), - SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if2_adc_in_mux), @@ -1733,14 +1724,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { &rt5645_if1_dac2_tdm_sel_mux), SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, &rt5645_if1_dac3_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac0_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac1_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac2_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, - &rt5650_if1_dac3_tdm_sel_mux), SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1882,6 +1865,24 @@ static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { 0, 0, &rt5650_a_dac2_l_mux), SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, 0, 0, &rt5650_a_dac2_r_mux), + + SND_SOC_DAPM_MUX("RT5650 IF1 ADC1 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc1_in_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 ADC2 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 ADC3 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc3_in_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5650_if1_adc_in_mux), + + SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5650 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5650_if1_dac3_tdm_sel_mux), }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { -- 2.4.2

1 0

[alsa-devel] Small pop on microphone capture on SGTL5000
by Jean-Michel Hautbois 12 Jun '15

12 Jun '15

Hi, We are using a SGTL5000 on a custom board, and right now, it is a 4.0.5 kernel. When we record on the MIC in a pop is heard at the very beginning. Is it a known issue ? Thanks, JM

1 0

[alsa-devel] [PATCH 0/2] genalloc: rename dev_get_gen_pool() and of_get_named_gen_pool()
by Vladimir Zapolskiy 12 Jun '15

12 Jun '15

Trivial nonfunctional change initially based on discussion https://lkml.org/lkml/2015/6/8/588 Worth to mention that instead of the assumed new name dev_gen_pool_get(), this change attempts to be more close to other in-kernel interfaces and new function name is just gen_pool_get(). The change is based and tested on linux-next. Vladimir Zapolskiy (2): genalloc: rename dev_get_gen_pool() to gen_pool_get() genalloc: rename of_get_named_gen_pool() to of_gen_pool_get() arch/arm/mach-at91/pm.c | 2 +- arch/arm/mach-imx/pm-imx5.c | 2 +- arch/arm/mach-imx/pm-imx6.c | 2 +- drivers/dma/mmp_tdma.c | 2 +- drivers/media/platform/coda/coda-common.c | 4 ++-- include/linux/genalloc.h | 6 +++--- lib/genalloc.c | 14 +++++++------- sound/core/memalloc.c | 2 +- 8 files changed, 17 insertions(+), 17 deletions(-) -- 2.1.4

2 2

[alsa-devel] [RFC PATCH] ALSA: split params refinement to seperated file
by Jie Yang 12 Jun '15

12 Jun '15

ALSA parameter refinement takes up about 13kB of memory. The refinement is useful on the desktop but is not usually required for some bespoke applications typically found on small inexpensive devices with reduced memory. These bespoke applications will typically configure the audio configuration with the exact requirements that will not require any further refinement from the kernel. The change moves refinement functions from original pcm_native.c and pcm_lib.c, to pcm_refine.c(similarly the declaration from pcm.h to pcm_refine.h), so it is not so big as it looks like. Please be noticed that this change may block alsa-lib usage, user app such as aplay/arecord doesn't work when CONFIG_SND_PARAMS_REFINEMENT is unselected. But it may be still worthy, some small player(e.g. tinyplay) can work with fixed params passed in(some small changes for the player may be needed), at this case, the change here will save about 13KB footprint on x86_64 platform for those small inexpensive devices(e.g. IoT ones). Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/pcm.h | 123 +--- include/sound/pcm_refine.h | 307 +++++++++ sound/core/Kconfig | 9 + sound/core/Makefile | 1 + sound/core/pcm_lib.c | 982 ---------------------------- sound/core/pcm_native.c | 505 --------------- sound/core/pcm_refine.c | 1519 ++++++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1837 insertions(+), 1609 deletions(-) create mode 100644 include/sound/pcm_refine.h create mode 100644 sound/core/pcm_refine.c diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee..f2d6425 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -26,6 +26,7 @@ #include <sound/asound.h> #include <sound/memalloc.h> #include <sound/minors.h> +#include <sound/pcm_refine.h> #include <linux/poll.h> #include <linux/mm.h> #include <linux/bitops.h> @@ -213,34 +214,6 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE #endif -struct snd_pcm_file { - struct snd_pcm_substream *substream; - int no_compat_mmap; -}; - -struct snd_pcm_hw_rule; -typedef int (*snd_pcm_hw_rule_func_t)(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule); - -struct snd_pcm_hw_rule { - unsigned int cond; - int var; - int deps[4]; - - snd_pcm_hw_rule_func_t func; - void *private; -}; - -struct snd_pcm_hw_constraints { - struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; - struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; - unsigned int rules_num; - unsigned int rules_all; - struct snd_pcm_hw_rule *rules; -}; - static inline struct snd_mask *constrs_mask(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { @@ -253,38 +226,6 @@ static inline struct snd_interval *constrs_interval(struct snd_pcm_hw_constraint return &constrs->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } -struct snd_ratnum { - unsigned int num; - unsigned int den_min, den_max, den_step; -}; - -struct snd_ratden { - unsigned int num_min, num_max, num_step; - unsigned int den; -}; - -struct snd_pcm_hw_constraint_ratnums { - int nrats; - struct snd_ratnum *rats; -}; - -struct snd_pcm_hw_constraint_ratdens { - int nrats; - struct snd_ratden *rats; -}; - -struct snd_pcm_hw_constraint_list { - const unsigned int *list; - unsigned int count; - unsigned int mask; -}; - -struct snd_pcm_hw_constraint_ranges { - unsigned int count; - const struct snd_interval *ranges; - unsigned int mask; -}; - struct snd_pcm_hwptr_log; /* @@ -968,72 +909,10 @@ static inline unsigned int params_buffer_bytes(const struct snd_pcm_hw_params *p return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min; } -int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); -void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c); -void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c); -void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b, - unsigned int k, struct snd_interval *c); -void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, - const struct snd_interval *b, struct snd_interval *c); -int snd_interval_list(struct snd_interval *i, unsigned int count, - const unsigned int *list, unsigned int mask); -int snd_interval_ranges(struct snd_interval *i, unsigned int count, - const struct snd_interval *list, unsigned int mask); -int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, - unsigned int *nump, unsigned int *denp); - void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params); void _snd_pcm_hw_param_setempty(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var); int snd_pcm_hw_params_choose(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); -int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); - -int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream); -int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream); - -int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, - u_int32_t mask); -int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, - u_int64_t mask); -int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, - unsigned int min, unsigned int max); -int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var); -int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - const struct snd_pcm_hw_constraint_list *l); -int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - const struct snd_pcm_hw_constraint_ranges *r); -int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r); -int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r); -int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, - unsigned int cond, - unsigned int width, - unsigned int msbits); -int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - unsigned long step); -int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var); -int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, - unsigned int base_rate); -int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, - unsigned int cond, - int var, - snd_pcm_hw_rule_func_t func, void *private, - int dep, ...); - int snd_pcm_format_signed(snd_pcm_format_t format); int snd_pcm_format_unsigned(snd_pcm_format_t format); int snd_pcm_format_linear(snd_pcm_format_t format); diff --git a/include/sound/pcm_refine.h b/include/sound/pcm_refine.h new file mode 100644 index 0000000..2e47620 --- /dev/null +++ b/include/sound/pcm_refine.h @@ -0,0 +1,307 @@ +#ifndef __SOUND_PCM_REFINE_H +#define __SOUND_PCM_REFINE_H + +/* + * Digital Audio (PCM) HW/SW params refinement layer + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Abramo Bagnara <abramo(a)alsa-project.org> + * Jie Yang <yang.jie(a)intel.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +struct snd_pcm_file { + struct snd_pcm_substream *substream; + int no_compat_mmap; +}; + +struct snd_pcm_hw_rule; +typedef int (*snd_pcm_hw_rule_func_t)(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); + +struct snd_pcm_hw_rule { + unsigned int cond; + int var; + int deps[4]; + + snd_pcm_hw_rule_func_t func; + void *private; +}; + +struct snd_pcm_hw_constraints { + struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - + SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; + struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - + SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; + unsigned int rules_num; + unsigned int rules_all; + struct snd_pcm_hw_rule *rules; +}; + +struct snd_ratnum { + unsigned int num; + unsigned int den_min, den_max, den_step; +}; + +struct snd_ratden { + unsigned int num_min, num_max, num_step; + unsigned int den; +}; + +struct snd_pcm_hw_constraint_ratnums { + int nrats; + struct snd_ratnum *rats; +}; + +struct snd_pcm_hw_constraint_ratdens { + int nrats; + struct snd_ratden *rats; +}; + +struct snd_pcm_hw_constraint_list { + const unsigned int *list; + unsigned int count; + unsigned int mask; +}; + +struct snd_pcm_hw_constraint_ranges { + unsigned int count; + const struct snd_interval *ranges; + unsigned int mask; +}; + +struct snd_pcm_runtime; + +#ifdef CONFIG_SND_PARAMS_REFINEMENT +int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); +void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c); +void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c); +void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b, + unsigned int k, struct snd_interval *c); +void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, + const struct snd_interval *b, struct snd_interval *c); +int snd_interval_list(struct snd_interval *i, unsigned int count, + const unsigned int *list, unsigned int mask); +int snd_interval_ranges(struct snd_interval *i, unsigned int count, + const struct snd_interval *list, unsigned int mask); +int snd_interval_ratnum(struct snd_interval *i, + unsigned int rats_count, struct snd_ratnum *rats, + unsigned int *nump, unsigned int *denp); + + +int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); + +int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream); +int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream); + +int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + u_int32_t mask); +int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + u_int64_t mask); +int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + unsigned int min, unsigned int max); +int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var); +int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_list *l); +int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_ranges *r); +int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + struct snd_pcm_hw_constraint_ratnums *r); +int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + struct snd_pcm_hw_constraint_ratdens *r); +int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, + unsigned int cond, + unsigned int width, + unsigned int msbits); +int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + unsigned long step); +int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var); +int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate); +int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, + unsigned int cond, + int var, + snd_pcm_hw_rule_func_t func, void *private, + int dep, ...); + +#else +static inline int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v) +{ + return 0; +} + +static inline void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) +{ +} + +static inline void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) +{ +} + +static inline void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b, + unsigned int k, struct snd_interval *c) +{ +} + +static inline void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, + const struct snd_interval *b, struct snd_interval *c) +{ +} + +static inline int snd_interval_list(struct snd_interval *i, unsigned int count, + const unsigned int *list, unsigned int mask) +{ + return 0; +} + +static inline int snd_interval_ranges(struct snd_interval *i, unsigned int count, + const struct snd_interval *list, unsigned int mask) +{ + return 0; +} + +static inline int snd_interval_ratnum(struct snd_interval *i, + unsigned int rats_count, struct snd_ratnum *rats, + unsigned int *nump, unsigned int *denp) +{ + return 0; +} + +static inline int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + return 0; +} + +static inline int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) +{ + return 0; +} + +static inline int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + u_int32_t mask) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + u_int64_t mask) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + unsigned int min, unsigned int max) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_list *l) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_ranges *r) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + struct snd_pcm_hw_constraint_ratnums *r) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + struct snd_pcm_hw_constraint_ratdens *r) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, + unsigned int cond, + unsigned int width, + unsigned int msbits) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + unsigned long step) +{ + return 0; +} + +static inline int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var) +{ + return 0; +} + +static inline int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate) +{ + return 0; +} + +static inline int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, + unsigned int cond, + int var, + snd_pcm_hw_rule_func_t func, void *private, + int dep, ...) +{ + return 0; +} + +#endif + +#endif /* __SOUND_PCM_REFINE_H */ + diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6c96fee..644e837 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -191,6 +191,15 @@ config SND_PROC_FS 9KB on x86_64 platform. If unsure say Y. +config SND_PARAMS_REFINEMENT + bool "HW/SW Params Refinement Support" if EXPERT + default y + help + Say 'N' to disable the HW/SW Params Refinement feature, which may + save about 13KB code size on x86_64 object target. + Please be noted that enabling this may break alsa on the desktop, + only use it with tinyalsa/tinyplay for device such as IoT. + config SND_VERBOSE_PROCFS bool "Verbose procfs contents" depends on SND_PROC_FS diff --git a/sound/core/Makefile b/sound/core/Makefile index 3354f91..c615404 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -18,6 +18,7 @@ snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o +snd-pcm-$(CONFIG_SND_PARAMS_REFINEMENT) += pcm_refine.o # for trace-points CFLAGS_pcm_lib.o := -I$(src) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d45645..0bcc3ec 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -532,119 +532,6 @@ EXPORT_SYMBOL(snd_pcm_set_sync); * Standard ioctl routine */ -static inline unsigned int div32(unsigned int a, unsigned int b, - unsigned int *r) -{ - if (b == 0) { - *r = 0; - return UINT_MAX; - } - *r = a % b; - return a / b; -} - -static inline unsigned int div_down(unsigned int a, unsigned int b) -{ - if (b == 0) - return UINT_MAX; - return a / b; -} - -static inline unsigned int div_up(unsigned int a, unsigned int b) -{ - unsigned int r; - unsigned int q; - if (b == 0) - return UINT_MAX; - q = div32(a, b, &r); - if (r) - ++q; - return q; -} - -static inline unsigned int mul(unsigned int a, unsigned int b) -{ - if (a == 0) - return 0; - if (div_down(UINT_MAX, a) < b) - return UINT_MAX; - return a * b; -} - -static inline unsigned int muldiv32(unsigned int a, unsigned int b, - unsigned int c, unsigned int *r) -{ - u_int64_t n = (u_int64_t) a * b; - if (c == 0) { - snd_BUG_ON(!n); - *r = 0; - return UINT_MAX; - } - n = div_u64_rem(n, c, r); - if (n >= UINT_MAX) { - *r = 0; - return UINT_MAX; - } - return n; -} - -/** - * snd_interval_refine - refine the interval value of configurator - * @i: the interval value to refine - * @v: the interval value to refer to - * - * Refines the interval value with the reference value. - * The interval is changed to the range satisfying both intervals. - * The interval status (min, max, integer, etc.) are evaluated. - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v) -{ - int changed = 0; - if (snd_BUG_ON(snd_interval_empty(i))) - return -EINVAL; - if (i->min < v->min) { - i->min = v->min; - i->openmin = v->openmin; - changed = 1; - } else if (i->min == v->min && !i->openmin && v->openmin) { - i->openmin = 1; - changed = 1; - } - if (i->max > v->max) { - i->max = v->max; - i->openmax = v->openmax; - changed = 1; - } else if (i->max == v->max && !i->openmax && v->openmax) { - i->openmax = 1; - changed = 1; - } - if (!i->integer && v->integer) { - i->integer = 1; - changed = 1; - } - if (i->integer) { - if (i->openmin) { - i->min++; - i->openmin = 0; - } - if (i->openmax) { - i->max--; - i->openmax = 0; - } - } else if (!i->openmin && !i->openmax && i->min == i->max) - i->integer = 1; - if (snd_interval_checkempty(i)) { - snd_interval_none(i); - return -EINVAL; - } - return changed; -} - -EXPORT_SYMBOL(snd_interval_refine); - static int snd_interval_refine_first(struct snd_interval *i) { if (snd_BUG_ON(snd_interval_empty(i))) @@ -671,875 +558,6 @@ static int snd_interval_refine_last(struct snd_interval *i) return 1; } -void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) -{ - if (a->empty || b->empty) { - snd_interval_none(c); - return; - } - c->empty = 0; - c->min = mul(a->min, b->min); - c->openmin = (a->openmin || b->openmin); - c->max = mul(a->max, b->max); - c->openmax = (a->openmax || b->openmax); - c->integer = (a->integer && b->integer); -} - -/** - * snd_interval_div - refine the interval value with division - * @a: dividend - * @b: divisor - * @c: quotient - * - * c = a / b - * - * Returns non-zero if the value is changed, zero if not changed. - */ -void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) -{ - unsigned int r; - if (a->empty || b->empty) { - snd_interval_none(c); - return; - } - c->empty = 0; - c->min = div32(a->min, b->max, &r); - c->openmin = (r || a->openmin || b->openmax); - if (b->min > 0) { - c->max = div32(a->max, b->min, &r); - if (r) { - c->max++; - c->openmax = 1; - } else - c->openmax = (a->openmax || b->openmin); - } else { - c->max = UINT_MAX; - c->openmax = 0; - } - c->integer = 0; -} - -/** - * snd_interval_muldivk - refine the interval value - * @a: dividend 1 - * @b: dividend 2 - * @k: divisor (as integer) - * @c: result - * - * c = a * b / k - * - * Returns non-zero if the value is changed, zero if not changed. - */ -void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b, - unsigned int k, struct snd_interval *c) -{ - unsigned int r; - if (a->empty || b->empty) { - snd_interval_none(c); - return; - } - c->empty = 0; - c->min = muldiv32(a->min, b->min, k, &r); - c->openmin = (r || a->openmin || b->openmin); - c->max = muldiv32(a->max, b->max, k, &r); - if (r) { - c->max++; - c->openmax = 1; - } else - c->openmax = (a->openmax || b->openmax); - c->integer = 0; -} - -/** - * snd_interval_mulkdiv - refine the interval value - * @a: dividend 1 - * @k: dividend 2 (as integer) - * @b: divisor - * @c: result - * - * c = a * k / b - * - * Returns non-zero if the value is changed, zero if not changed. - */ -void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, - const struct snd_interval *b, struct snd_interval *c) -{ - unsigned int r; - if (a->empty || b->empty) { - snd_interval_none(c); - return; - } - c->empty = 0; - c->min = muldiv32(a->min, k, b->max, &r); - c->openmin = (r || a->openmin || b->openmax); - if (b->min > 0) { - c->max = muldiv32(a->max, k, b->min, &r); - if (r) { - c->max++; - c->openmax = 1; - } else - c->openmax = (a->openmax || b->openmin); - } else { - c->max = UINT_MAX; - c->openmax = 0; - } - c->integer = 0; -} - -/* ---- */ - - -/** - * snd_interval_ratnum - refine the interval value - * @i: interval to refine - * @rats_count: number of ratnum_t - * @rats: ratnum_t array - * @nump: pointer to store the resultant numerator - * @denp: pointer to store the resultant denominator - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, - unsigned int *nump, unsigned int *denp) -{ - unsigned int best_num, best_den; - int best_diff; - unsigned int k; - struct snd_interval t; - int err; - unsigned int result_num, result_den; - int result_diff; - - best_num = best_den = best_diff = 0; - for (k = 0; k < rats_count; ++k) { - unsigned int num = rats[k].num; - unsigned int den; - unsigned int q = i->min; - int diff; - if (q == 0) - q = 1; - den = div_up(num, q); - if (den < rats[k].den_min) - continue; - if (den > rats[k].den_max) - den = rats[k].den_max; - else { - unsigned int r; - r = (den - rats[k].den_min) % rats[k].den_step; - if (r != 0) - den -= r; - } - diff = num - q * den; - if (diff < 0) - diff = -diff; - if (best_num == 0 || - diff * best_den < best_diff * den) { - best_diff = diff; - best_den = den; - best_num = num; - } - } - if (best_den == 0) { - i->empty = 1; - return -EINVAL; - } - t.min = div_down(best_num, best_den); - t.openmin = !!(best_num % best_den); - - result_num = best_num; - result_diff = best_diff; - result_den = best_den; - best_num = best_den = best_diff = 0; - for (k = 0; k < rats_count; ++k) { - unsigned int num = rats[k].num; - unsigned int den; - unsigned int q = i->max; - int diff; - if (q == 0) { - i->empty = 1; - return -EINVAL; - } - den = div_down(num, q); - if (den > rats[k].den_max) - continue; - if (den < rats[k].den_min) - den = rats[k].den_min; - else { - unsigned int r; - r = (den - rats[k].den_min) % rats[k].den_step; - if (r != 0) - den += rats[k].den_step - r; - } - diff = q * den - num; - if (diff < 0) - diff = -diff; - if (best_num == 0 || - diff * best_den < best_diff * den) { - best_diff = diff; - best_den = den; - best_num = num; - } - } - if (best_den == 0) { - i->empty = 1; - return -EINVAL; - } - t.max = div_up(best_num, best_den); - t.openmax = !!(best_num % best_den); - t.integer = 0; - err = snd_interval_refine(i, &t); - if (err < 0) - return err; - - if (snd_interval_single(i)) { - if (best_diff * result_den < result_diff * best_den) { - result_num = best_num; - result_den = best_den; - } - if (nump) - *nump = result_num; - if (denp) - *denp = result_den; - } - return err; -} - -EXPORT_SYMBOL(snd_interval_ratnum); - -/** - * snd_interval_ratden - refine the interval value - * @i: interval to refine - * @rats_count: number of struct ratden - * @rats: struct ratden array - * @nump: pointer to store the resultant numerator - * @denp: pointer to store the resultant denominator - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -static int snd_interval_ratden(struct snd_interval *i, - unsigned int rats_count, struct snd_ratden *rats, - unsigned int *nump, unsigned int *denp) -{ - unsigned int best_num, best_diff, best_den; - unsigned int k; - struct snd_interval t; - int err; - - best_num = best_den = best_diff = 0; - for (k = 0; k < rats_count; ++k) { - unsigned int num; - unsigned int den = rats[k].den; - unsigned int q = i->min; - int diff; - num = mul(q, den); - if (num > rats[k].num_max) - continue; - if (num < rats[k].num_min) - num = rats[k].num_max; - else { - unsigned int r; - r = (num - rats[k].num_min) % rats[k].num_step; - if (r != 0) - num += rats[k].num_step - r; - } - diff = num - q * den; - if (best_num == 0 || - diff * best_den < best_diff * den) { - best_diff = diff; - best_den = den; - best_num = num; - } - } - if (best_den == 0) { - i->empty = 1; - return -EINVAL; - } - t.min = div_down(best_num, best_den); - t.openmin = !!(best_num % best_den); - - best_num = best_den = best_diff = 0; - for (k = 0; k < rats_count; ++k) { - unsigned int num; - unsigned int den = rats[k].den; - unsigned int q = i->max; - int diff; - num = mul(q, den); - if (num < rats[k].num_min) - continue; - if (num > rats[k].num_max) - num = rats[k].num_max; - else { - unsigned int r; - r = (num - rats[k].num_min) % rats[k].num_step; - if (r != 0) - num -= r; - } - diff = q * den - num; - if (best_num == 0 || - diff * best_den < best_diff * den) { - best_diff = diff; - best_den = den; - best_num = num; - } - } - if (best_den == 0) { - i->empty = 1; - return -EINVAL; - } - t.max = div_up(best_num, best_den); - t.openmax = !!(best_num % best_den); - t.integer = 0; - err = snd_interval_refine(i, &t); - if (err < 0) - return err; - - if (snd_interval_single(i)) { - if (nump) - *nump = best_num; - if (denp) - *denp = best_den; - } - return err; -} - -/** - * snd_interval_list - refine the interval value from the list - * @i: the interval value to refine - * @count: the number of elements in the list - * @list: the value list - * @mask: the bit-mask to evaluate - * - * Refines the interval value from the list. - * When mask is non-zero, only the elements corresponding to bit 1 are - * evaluated. - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -int snd_interval_list(struct snd_interval *i, unsigned int count, - const unsigned int *list, unsigned int mask) -{ - unsigned int k; - struct snd_interval list_range; - - if (!count) { - i->empty = 1; - return -EINVAL; - } - snd_interval_any(&list_range); - list_range.min = UINT_MAX; - list_range.max = 0; - for (k = 0; k < count; k++) { - if (mask && !(mask & (1 << k))) - continue; - if (!snd_interval_test(i, list[k])) - continue; - list_range.min = min(list_range.min, list[k]); - list_range.max = max(list_range.max, list[k]); - } - return snd_interval_refine(i, &list_range); -} - -EXPORT_SYMBOL(snd_interval_list); - -/** - * snd_interval_ranges - refine the interval value from the list of ranges - * @i: the interval value to refine - * @count: the number of elements in the list of ranges - * @ranges: the ranges list - * @mask: the bit-mask to evaluate - * - * Refines the interval value from the list of ranges. - * When mask is non-zero, only the elements corresponding to bit 1 are - * evaluated. - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -int snd_interval_ranges(struct snd_interval *i, unsigned int count, - const struct snd_interval *ranges, unsigned int mask) -{ - unsigned int k; - struct snd_interval range_union; - struct snd_interval range; - - if (!count) { - snd_interval_none(i); - return -EINVAL; - } - snd_interval_any(&range_union); - range_union.min = UINT_MAX; - range_union.max = 0; - for (k = 0; k < count; k++) { - if (mask && !(mask & (1 << k))) - continue; - snd_interval_copy(&range, &ranges[k]); - if (snd_interval_refine(&range, i) < 0) - continue; - if (snd_interval_empty(&range)) - continue; - - if (range.min < range_union.min) { - range_union.min = range.min; - range_union.openmin = 1; - } - if (range.min == range_union.min && !range.openmin) - range_union.openmin = 0; - if (range.max > range_union.max) { - range_union.max = range.max; - range_union.openmax = 1; - } - if (range.max == range_union.max && !range.openmax) - range_union.openmax = 0; - } - return snd_interval_refine(i, &range_union); -} -EXPORT_SYMBOL(snd_interval_ranges); - -static int snd_interval_step(struct snd_interval *i, unsigned int step) -{ - unsigned int n; - int changed = 0; - n = i->min % step; - if (n != 0 || i->openmin) { - i->min += step - n; - i->openmin = 0; - changed = 1; - } - n = i->max % step; - if (n != 0 || i->openmax) { - i->max -= n; - i->openmax = 0; - changed = 1; - } - if (snd_interval_checkempty(i)) { - i->empty = 1; - return -EINVAL; - } - return changed; -} - -/* Info constraints helpers */ - -/** - * snd_pcm_hw_rule_add - add the hw-constraint rule - * @runtime: the pcm runtime instance - * @cond: condition bits - * @var: the variable to evaluate - * @func: the evaluation function - * @private: the private data pointer passed to function - * @dep: the dependent variables - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, - int var, - snd_pcm_hw_rule_func_t func, void *private, - int dep, ...) -{ - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; - struct snd_pcm_hw_rule *c; - unsigned int k; - va_list args; - va_start(args, dep); - if (constrs->rules_num >= constrs->rules_all) { - struct snd_pcm_hw_rule *new; - unsigned int new_rules = constrs->rules_all + 16; - new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL); - if (!new) { - va_end(args); - return -ENOMEM; - } - if (constrs->rules) { - memcpy(new, constrs->rules, - constrs->rules_num * sizeof(*c)); - kfree(constrs->rules); - } - constrs->rules = new; - constrs->rules_all = new_rules; - } - c = &constrs->rules[constrs->rules_num]; - c->cond = cond; - c->func = func; - c->var = var; - c->private = private; - k = 0; - while (1) { - if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps))) { - va_end(args); - return -EINVAL; - } - c->deps[k++] = dep; - if (dep < 0) - break; - dep = va_arg(args, int); - } - constrs->rules_num++; - va_end(args); - return 0; -} - -EXPORT_SYMBOL(snd_pcm_hw_rule_add); - -/** - * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint - * @runtime: PCM runtime instance - * @var: hw_params variable to apply the mask - * @mask: the bitmap mask - * - * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, - u_int32_t mask) -{ - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; - struct snd_mask *maskp = constrs_mask(constrs, var); - *maskp->bits &= mask; - memset(maskp->bits + 1, 0, (SNDRV_MASK_MAX-32) / 8); /* clear rest */ - if (*maskp->bits == 0) - return -EINVAL; - return 0; -} - -/** - * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint - * @runtime: PCM runtime instance - * @var: hw_params variable to apply the mask - * @mask: the 64bit bitmap mask - * - * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, - u_int64_t mask) -{ - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; - struct snd_mask *maskp = constrs_mask(constrs, var); - maskp->bits[0] &= (u_int32_t)mask; - maskp->bits[1] &= (u_int32_t)(mask >> 32); - memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */ - if (! maskp->bits[0] && ! maskp->bits[1]) - return -EINVAL; - return 0; -} -EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); - -/** - * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval - * @runtime: PCM runtime instance - * @var: hw_params variable to apply the integer constraint - * - * Apply the constraint of integer to an interval parameter. - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var) -{ - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; - return snd_interval_setinteger(constrs_interval(constrs, var)); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); - -/** - * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval - * @runtime: PCM runtime instance - * @var: hw_params variable to apply the range - * @min: the minimal value - * @max: the maximal value - * - * Apply the min/max range constraint to an interval parameter. - * - * Return: Positive if the value is changed, zero if it's not changed, or a - * negative error code. - */ -int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, - unsigned int min, unsigned int max) -{ - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; - struct snd_interval t; - t.min = min; - t.max = max; - t.openmin = t.openmax = 0; - t.integer = 0; - return snd_interval_refine(constrs_interval(constrs, var), &t); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax); - -static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_hw_constraint_list *list = rule->private; - return snd_interval_list(hw_param_interval(params, rule->var), list->count, list->list, list->mask); -} - - -/** - * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter - * @runtime: PCM runtime instance - * @cond: condition bits - * @var: hw_params variable to apply the list constraint - * @l: list - * - * Apply the list of constraints to an interval parameter. - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - const struct snd_pcm_hw_constraint_list *l) -{ - return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_list, (void *)l, - var, -1); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_list); - -static int snd_pcm_hw_rule_ranges(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_hw_constraint_ranges *r = rule->private; - return snd_interval_ranges(hw_param_interval(params, rule->var), - r->count, r->ranges, r->mask); -} - - -/** - * snd_pcm_hw_constraint_ranges - apply list of range constraints to a parameter - * @runtime: PCM runtime instance - * @cond: condition bits - * @var: hw_params variable to apply the list of range constraints - * @r: ranges - * - * Apply the list of range constraints to an interval parameter. - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - const struct snd_pcm_hw_constraint_ranges *r) -{ - return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ranges, (void *)r, - var, -1); -} -EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges); - -static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_hw_constraint_ratnums *r = rule->private; - unsigned int num = 0, den = 0; - int err; - err = snd_interval_ratnum(hw_param_interval(params, rule->var), - r->nrats, r->rats, &num, &den); - if (err >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { - params->rate_num = num; - params->rate_den = den; - } - return err; -} - -/** - * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter - * @runtime: PCM runtime instance - * @cond: condition bits - * @var: hw_params variable to apply the ratnums constraint - * @r: struct snd_ratnums constriants - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r) -{ - return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratnums, r, - var, -1); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); - -static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_hw_constraint_ratdens *r = rule->private; - unsigned int num = 0, den = 0; - int err = snd_interval_ratden(hw_param_interval(params, rule->var), - r->nrats, r->rats, &num, &den); - if (err >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { - params->rate_num = num; - params->rate_den = den; - } - return err; -} - -/** - * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter - * @runtime: PCM runtime instance - * @cond: condition bits - * @var: hw_params variable to apply the ratdens constraint - * @r: struct snd_ratdens constriants - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r) -{ - return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratdens, r, - var, -1); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_ratdens); - -static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - unsigned int l = (unsigned long) rule->private; - int width = l & 0xffff; - unsigned int msbits = l >> 16; - struct snd_interval *i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - - if (!snd_interval_single(i)) - return 0; - - if ((snd_interval_value(i) == width) || - (width == 0 && snd_interval_value(i) > msbits)) - params->msbits = min_not_zero(params->msbits, msbits); - - return 0; -} - -/** - * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule - * @runtime: PCM runtime instance - * @cond: condition bits - * @width: sample bits width - * @msbits: msbits width - * - * This constraint will set the number of most significant bits (msbits) if a - * sample format with the specified width has been select. If width is set to 0 - * the msbits will be set for any sample format with a width larger than the - * specified msbits. - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, - unsigned int cond, - unsigned int width, - unsigned int msbits) -{ - unsigned long l = (msbits << 16) | width; - return snd_pcm_hw_rule_add(runtime, cond, -1, - snd_pcm_hw_rule_msbits, - (void*) l, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_msbits); - -static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - unsigned long step = (unsigned long) rule->private; - return snd_interval_step(hw_param_interval(params, rule->var), step); -} - -/** - * snd_pcm_hw_constraint_step - add a hw constraint step rule - * @runtime: PCM runtime instance - * @cond: condition bits - * @var: hw_params variable to apply the step constraint - * @step: step size - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var, - unsigned long step) -{ - return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_step, (void *) step, - var, -1); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_step); - -static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) -{ - static unsigned int pow2_sizes[] = { - 1<<0, 1<<1, 1<<2, 1<<3, 1<<4, 1<<5, 1<<6, 1<<7, - 1<<8, 1<<9, 1<<10, 1<<11, 1<<12, 1<<13, 1<<14, 1<<15, - 1<<16, 1<<17, 1<<18, 1<<19, 1<<20, 1<<21, 1<<22, 1<<23, - 1<<24, 1<<25, 1<<26, 1<<27, 1<<28, 1<<29, 1<<30 - }; - return snd_interval_list(hw_param_interval(params, rule->var), - ARRAY_SIZE(pow2_sizes), pow2_sizes, 0); -} - -/** - * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule - * @runtime: PCM runtime instance - * @cond: condition bits - * @var: hw_params variable to apply the power-of-2 constraint - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, - unsigned int cond, - snd_pcm_hw_param_t var) -{ - return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_pow2, NULL, - var, -1); -} - -EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); - -static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - unsigned int base_rate = (unsigned int)(uintptr_t)rule->private; - struct snd_interval *rate; - - rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - return snd_interval_list(rate, 1, &base_rate, 0); -} - -/** - * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling - * @runtime: PCM runtime instance - * @base_rate: the rate at which the hardware does not resample - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, - unsigned int base_rate) -{ - return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE, - SNDRV_PCM_HW_PARAM_RATE, - snd_pcm_hw_rule_noresample_func, - (void *)(uintptr_t)base_rate, - SNDRV_PCM_HW_PARAM_RATE, -1); -} -EXPORT_SYMBOL(snd_pcm_hw_rule_noresample); - static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d126c03..b4c4829 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -227,21 +227,6 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, return err; } -static bool hw_support_mmap(struct snd_pcm_substream *substream) -{ - if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) - return false; - /* check architectures that return -EINVAL from dma_mmap_coherent() */ - /* FIXME: this should be some global flag */ -#if defined(CONFIG_C6X) || defined(CONFIG_FRV) || defined(CONFIG_MN10300) ||\ - defined(CONFIG_PARISC) || defined(CONFIG_XTENSA) - if (!substream->ops->mmap && - substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) - return false; -#endif - return true; -} - #undef RULES_DEBUG #ifdef RULES_DEBUG @@ -265,184 +250,6 @@ static const char * const snd_pcm_hw_param_names[] = { }; #endif -int snd_pcm_hw_refine(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - unsigned int k; - struct snd_pcm_hardware *hw; - struct snd_interval *i = NULL; - struct snd_mask *m = NULL; - struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; - unsigned int rstamps[constrs->rules_num]; - unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; - unsigned int stamp = 2; - int changed, again; - - params->info = 0; - params->fifo_size = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) - params->msbits = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) { - params->rate_num = 0; - params->rate_den = 0; - } - - for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) { - m = hw_param_mask(params, k); - if (snd_mask_empty(m)) - return -EINVAL; - if (!(params->rmask & (1 << k))) - continue; -#ifdef RULES_DEBUG - pr_debug("%s = ", snd_pcm_hw_param_names[k]); - pr_cont("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); -#endif - changed = snd_mask_refine(m, constrs_mask(constrs, k)); -#ifdef RULES_DEBUG - pr_cont("%04x%04x%04x%04x\n", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); -#endif - if (changed) - params->cmask |= 1 << k; - if (changed < 0) - return changed; - } - - for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) { - i = hw_param_interval(params, k); - if (snd_interval_empty(i)) - return -EINVAL; - if (!(params->rmask & (1 << k))) - continue; -#ifdef RULES_DEBUG - pr_debug("%s = ", snd_pcm_hw_param_names[k]); - if (i->empty) - pr_cont("empty"); - else - pr_cont("%c%u %u%c", - i->openmin ? '(' : '[', i->min, - i->max, i->openmax ? ')' : ']'); - pr_cont(" -> "); -#endif - changed = snd_interval_refine(i, constrs_interval(constrs, k)); -#ifdef RULES_DEBUG - if (i->empty) - pr_cont("empty\n"); - else - pr_cont("%c%u %u%c\n", - i->openmin ? '(' : '[', i->min, - i->max, i->openmax ? ')' : ']'); -#endif - if (changed) - params->cmask |= 1 << k; - if (changed < 0) - return changed; - } - - for (k = 0; k < constrs->rules_num; k++) - rstamps[k] = 0; - for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) - vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; - do { - again = 0; - for (k = 0; k < constrs->rules_num; k++) { - struct snd_pcm_hw_rule *r = &constrs->rules[k]; - unsigned int d; - int doit = 0; - if (r->cond && !(r->cond & params->flags)) - continue; - for (d = 0; r->deps[d] >= 0; d++) { - if (vstamps[r->deps[d]] > rstamps[k]) { - doit = 1; - break; - } - } - if (!doit) - continue; -#ifdef RULES_DEBUG - pr_debug("Rule %d [%p]: ", k, r->func); - if (r->var >= 0) { - pr_cont("%s = ", snd_pcm_hw_param_names[r->var]); - if (hw_is_mask(r->var)) { - m = hw_param_mask(params, r->var); - pr_cont("%x", *m->bits); - } else { - i = hw_param_interval(params, r->var); - if (i->empty) - pr_cont("empty"); - else - pr_cont("%c%u %u%c", - i->openmin ? '(' : '[', i->min, - i->max, i->openmax ? ')' : ']'); - } - } -#endif - changed = r->func(params, r); -#ifdef RULES_DEBUG - if (r->var >= 0) { - pr_cont(" -> "); - if (hw_is_mask(r->var)) - pr_cont("%x", *m->bits); - else { - if (i->empty) - pr_cont("empty"); - else - pr_cont("%c%u %u%c", - i->openmin ? '(' : '[', i->min, - i->max, i->openmax ? ')' : ']'); - } - } - pr_cont("\n"); -#endif - rstamps[k] = stamp; - if (changed && r->var >= 0) { - params->cmask |= (1 << r->var); - vstamps[r->var] = stamp; - again = 1; - } - if (changed < 0) - return changed; - stamp++; - } - } while (again); - if (!params->msbits) { - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - if (snd_interval_single(i)) - params->msbits = snd_interval_value(i); - } - - if (!params->rate_den) { - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - if (snd_interval_single(i)) { - params->rate_num = snd_interval_value(i); - params->rate_den = 1; - } - } - - hw = &substream->runtime->hw; - if (!params->info) { - params->info = hw->info & ~(SNDRV_PCM_INFO_FIFO_IN_FRAMES | - SNDRV_PCM_INFO_DRAIN_TRIGGER); - if (!hw_support_mmap(substream)) - params->info &= ~(SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID); - } - if (!params->fifo_size) { - m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - if (snd_mask_min(m) == snd_mask_max(m) && - snd_interval_min(i) == snd_interval_max(i)) { - changed = substream->ops->ioctl(substream, - SNDRV_PCM_IOCTL1_FIFO_SIZE, params); - if (changed < 0) - return changed; - } - } - params->rmask = 0; - return 0; -} - -EXPORT_SYMBOL(snd_pcm_hw_refine); - static int snd_pcm_hw_refine_user(struct snd_pcm_substream *substream, struct snd_pcm_hw_params __user * _params) { @@ -1888,89 +1695,6 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) /* * hw configurator */ -static int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval t; - snd_interval_mul(hw_param_interval_c(params, rule->deps[0]), - hw_param_interval_c(params, rule->deps[1]), &t); - return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} - -static int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval t; - snd_interval_div(hw_param_interval_c(params, rule->deps[0]), - hw_param_interval_c(params, rule->deps[1]), &t); - return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} - -static int snd_pcm_hw_rule_muldivk(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval t; - snd_interval_muldivk(hw_param_interval_c(params, rule->deps[0]), - hw_param_interval_c(params, rule->deps[1]), - (unsigned long) rule->private, &t); - return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} - -static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval t; - snd_interval_mulkdiv(hw_param_interval_c(params, rule->deps[0]), - (unsigned long) rule->private, - hw_param_interval_c(params, rule->deps[1]), &t); - return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} - -static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - unsigned int k; - struct snd_interval *i = hw_param_interval(params, rule->deps[0]); - struct snd_mask m; - struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - snd_mask_any(&m); - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { - int bits; - if (! snd_mask_test(mask, k)) - continue; - bits = snd_pcm_format_physical_width(k); - if (bits <= 0) - continue; /* ignore invalid formats */ - if ((unsigned)bits < i->min || (unsigned)bits > i->max) - snd_mask_reset(&m, k); - } - return snd_mask_refine(mask, &m); -} - -static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval t; - unsigned int k; - t.min = UINT_MAX; - t.max = 0; - t.openmin = 0; - t.openmax = 0; - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { - int bits; - if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) - continue; - bits = snd_pcm_format_physical_width(k); - if (bits <= 0) - continue; /* ignore invalid formats */ - if (t.min > (unsigned)bits) - t.min = bits; - if (t.max < (unsigned)bits) - t.max = bits; - } - t.integer = 1; - return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} #if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 #error "Change this table" @@ -1984,235 +1708,6 @@ const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .list = rates, }; -static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_hardware *hw = rule->private; - return snd_interval_list(hw_param_interval(params, rule->var), - snd_pcm_known_rates.count, - snd_pcm_known_rates.list, hw->rates); -} - -static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval t; - struct snd_pcm_substream *substream = rule->private; - t.min = 0; - t.max = substream->buffer_bytes_max; - t.openmin = 0; - t.openmax = 0; - t.integer = 1; - return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} - -int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; - int k, err; - - for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) { - snd_mask_any(constrs_mask(constrs, k)); - } - - for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) { - snd_interval_any(constrs_interval(constrs, k)); - } - - snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_CHANNELS)); - snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_SIZE)); - snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)); - snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)); - snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS)); - - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - snd_pcm_hw_rule_format, NULL, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - snd_pcm_hw_rule_sample_bits, NULL, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, - snd_pcm_hw_rule_mul, NULL, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, - snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - snd_pcm_hw_rule_div, NULL, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - snd_pcm_hw_rule_muldivk, (void*) 1000000, - SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - snd_pcm_hw_rule_mul, NULL, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - snd_pcm_hw_rule_mulkdiv, (void*) 8, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - snd_pcm_hw_rule_muldivk, (void*) 1000000, - SNDRV_PCM_HW_PARAM_BUFFER_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - snd_pcm_hw_rule_muldivk, (void*) 8, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - snd_pcm_hw_rule_muldivk, (void*) 8, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, - snd_pcm_hw_rule_mulkdiv, (void*) 1000000, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - return 0; -} - -int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_pcm_hardware *hw = &runtime->hw; - int err; - unsigned int mask = 0; - - if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED; - if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED; - if (hw_support_mmap(substream)) { - if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED; - if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED; - if (hw->info & SNDRV_PCM_INFO_COMPLEX) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX; - } - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, hw->formats); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - hw->channels_min, hw->channels_max); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, - hw->rate_min, hw->rate_max); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - hw->period_bytes_min, hw->period_bytes_max); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, - hw->periods_min, hw->periods_max); - if (err < 0) - return err; - - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - hw->period_bytes_min, hw->buffer_bytes_max); - if (err < 0) - return err; - - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - snd_pcm_hw_rule_buffer_bytes_max, substream, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); - if (err < 0) - return err; - - /* FIXME: remove */ - if (runtime->dma_bytes) { - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, runtime->dma_bytes); - if (err < 0) - return err; - } - - if (!(hw->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))) { - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_pcm_hw_rule_rate, hw, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - } - - /* FIXME: this belong to lowlevel */ - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); - - return 0; -} - static void pcm_release_private(struct snd_pcm_substream *substream) { snd_pcm_unlink(substream); diff --git a/sound/core/pcm_refine.c b/sound/core/pcm_refine.c new file mode 100644 index 0000000..40ae80e --- /dev/null +++ b/sound/core/pcm_refine.c @@ -0,0 +1,1519 @@ +/* + * Digital Audio (PCM) abstract layer + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Abramo Bagnara <abramo(a)alsa-project.org> + * Jie Yang <yang.jie(a)intel.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + + + +static bool hw_support_mmap(struct snd_pcm_substream *substream) +{ + if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) + return false; + /* check architectures that return -EINVAL from dma_mmap_coherent() */ + /* FIXME: this should be some global flag */ +#if defined(CONFIG_C6X) || defined(CONFIG_FRV) || defined(CONFIG_MN10300) ||\ + defined(CONFIG_PARISC) || defined(CONFIG_XTENSA) + if (!substream->ops->mmap && + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return false; +#endif + return true; +} + +int snd_pcm_hw_refine(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + unsigned int k; + struct snd_pcm_hardware *hw; + struct snd_interval *i = NULL; + struct snd_mask *m = NULL; + struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; + unsigned int rstamps[constrs->rules_num]; + unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; + unsigned int stamp = 2; + int changed, again; + + params->info = 0; + params->fifo_size = 0; + if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) + params->msbits = 0; + if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) { + params->rate_num = 0; + params->rate_den = 0; + } + + for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) { + m = hw_param_mask(params, k); + if (snd_mask_empty(m)) + return -EINVAL; + if (!(params->rmask & (1 << k))) + continue; +#ifdef RULES_DEBUG + pr_debug("%s = ", snd_pcm_hw_param_names[k]); + pr_cont("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); +#endif + changed = snd_mask_refine(m, constrs_mask(constrs, k)); +#ifdef RULES_DEBUG + pr_cont("%04x%04x%04x%04x\n", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); +#endif + if (changed) + params->cmask |= 1 << k; + if (changed < 0) + return changed; + } + + for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) { + i = hw_param_interval(params, k); + if (snd_interval_empty(i)) + return -EINVAL; + if (!(params->rmask & (1 << k))) + continue; +#ifdef RULES_DEBUG + pr_debug("%s = ", snd_pcm_hw_param_names[k]); + if (i->empty) + pr_cont("empty"); + else + pr_cont("%c%u %u%c", + i->openmin ? '(' : '[', i->min, + i->max, i->openmax ? ')' : ']'); + pr_cont(" -> "); +#endif + changed = snd_interval_refine(i, constrs_interval(constrs, k)); +#ifdef RULES_DEBUG + if (i->empty) + pr_cont("empty\n"); + else + pr_cont("%c%u %u%c\n", + i->openmin ? '(' : '[', i->min, + i->max, i->openmax ? ')' : ']'); +#endif + if (changed) + params->cmask |= 1 << k; + if (changed < 0) + return changed; + } + + for (k = 0; k < constrs->rules_num; k++) + rstamps[k] = 0; + for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) + vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; + do { + again = 0; + for (k = 0; k < constrs->rules_num; k++) { + struct snd_pcm_hw_rule *r = &constrs->rules[k]; + unsigned int d; + int doit = 0; + if (r->cond && !(r->cond & params->flags)) + continue; + for (d = 0; r->deps[d] >= 0; d++) { + if (vstamps[r->deps[d]] > rstamps[k]) { + doit = 1; + break; + } + } + if (!doit) + continue; +#ifdef RULES_DEBUG + pr_debug("Rule %d [%p]: ", k, r->func); + if (r->var >= 0) { + pr_cont("%s = ", snd_pcm_hw_param_names[r->var]); + if (hw_is_mask(r->var)) { + m = hw_param_mask(params, r->var); + pr_cont("%x", *m->bits); + } else { + i = hw_param_interval(params, r->var); + if (i->empty) + pr_cont("empty"); + else + pr_cont("%c%u %u%c", + i->openmin ? '(' : '[', i->min, + i->max, i->openmax ? ')' : ']'); + } + } +#endif + changed = r->func(params, r); +#ifdef RULES_DEBUG + if (r->var >= 0) { + pr_cont(" -> "); + if (hw_is_mask(r->var)) + pr_cont("%x", *m->bits); + else { + if (i->empty) + pr_cont("empty"); + else + pr_cont("%c%u %u%c", + i->openmin ? '(' : '[', i->min, + i->max, i->openmax ? ')' : ']'); + } + } + pr_cont("\n"); +#endif + rstamps[k] = stamp; + if (changed && r->var >= 0) { + params->cmask |= (1 << r->var); + vstamps[r->var] = stamp; + again = 1; + } + if (changed < 0) + return changed; + stamp++; + } + } while (again); + if (!params->msbits) { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + if (snd_interval_single(i)) + params->msbits = snd_interval_value(i); + } + + if (!params->rate_den) { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + if (snd_interval_single(i)) { + params->rate_num = snd_interval_value(i); + params->rate_den = 1; + } + } + + hw = &substream->runtime->hw; + if (!params->info) { + params->info = hw->info & ~(SNDRV_PCM_INFO_FIFO_IN_FRAMES | + SNDRV_PCM_INFO_DRAIN_TRIGGER); + if (!hw_support_mmap(substream)) + params->info &= ~(SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID); + } + if (!params->fifo_size) { + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + if (snd_mask_min(m) == snd_mask_max(m) && + snd_interval_min(i) == snd_interval_max(i)) { + changed = substream->ops->ioctl(substream, + SNDRV_PCM_IOCTL1_FIFO_SIZE, params); + if (changed < 0) + return changed; + } + } + params->rmask = 0; + return 0; +} + +EXPORT_SYMBOL(snd_pcm_hw_refine); + +static int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval t; + snd_interval_mul(hw_param_interval_c(params, rule->deps[0]), + hw_param_interval_c(params, rule->deps[1]), &t); + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +static int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval t; + snd_interval_div(hw_param_interval_c(params, rule->deps[0]), + hw_param_interval_c(params, rule->deps[1]), &t); + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +static int snd_pcm_hw_rule_muldivk(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval t; + snd_interval_muldivk(hw_param_interval_c(params, rule->deps[0]), + hw_param_interval_c(params, rule->deps[1]), + (unsigned long) rule->private, &t); + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval t; + snd_interval_mulkdiv(hw_param_interval_c(params, rule->deps[0]), + (unsigned long) rule->private, + hw_param_interval_c(params, rule->deps[1]), &t); + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned int k; + struct snd_interval *i = hw_param_interval(params, rule->deps[0]); + struct snd_mask m; + struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_any(&m); + for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + int bits; + if (! snd_mask_test(mask, k)) + continue; + bits = snd_pcm_format_physical_width(k); + if (bits <= 0) + continue; /* ignore invalid formats */ + if ((unsigned)bits < i->min || (unsigned)bits > i->max) + snd_mask_reset(&m, k); + } + return snd_mask_refine(mask, &m); +} + +static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval t; + unsigned int k; + t.min = UINT_MAX; + t.max = 0; + t.openmin = 0; + t.openmax = 0; + for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + int bits; + if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) + continue; + bits = snd_pcm_format_physical_width(k); + if (bits <= 0) + continue; /* ignore invalid formats */ + if (t.min > (unsigned)bits) + t.min = bits; + if (t.max < (unsigned)bits) + t.max = bits; + } + t.integer = 1; + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hardware *hw = rule->private; + return snd_interval_list(hw_param_interval(params, rule->var), + snd_pcm_known_rates.count, + snd_pcm_known_rates.list, hw->rates); +} + +static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval t; + struct snd_pcm_substream *substream = rule->private; + t.min = 0; + t.max = substream->buffer_bytes_max; + t.openmin = 0; + t.openmax = 0; + t.integer = 1; + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + int k, err; + + for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) { + snd_mask_any(constrs_mask(constrs, k)); + } + + for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) { + snd_interval_any(constrs_interval(constrs, k)); + } + + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_CHANNELS)); + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_SIZE)); + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)); + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)); + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS)); + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + snd_pcm_hw_rule_format, NULL, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + snd_pcm_hw_rule_sample_bits, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + snd_pcm_hw_rule_div, NULL, + SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + snd_pcm_hw_rule_mul, NULL, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + snd_pcm_hw_rule_mulkdiv, (void*) 8, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + snd_pcm_hw_rule_mulkdiv, (void*) 8, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_div, NULL, + SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_mulkdiv, (void*) 1000000, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_mulkdiv, (void*) 1000000, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + snd_pcm_hw_rule_div, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + snd_pcm_hw_rule_div, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + snd_pcm_hw_rule_mulkdiv, (void*) 8, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + snd_pcm_hw_rule_muldivk, (void*) 1000000, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + snd_pcm_hw_rule_mul, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + snd_pcm_hw_rule_mulkdiv, (void*) 8, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + snd_pcm_hw_rule_muldivk, (void*) 1000000, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_rule_muldivk, (void*) 8, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_rule_muldivk, (void*) 8, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, + snd_pcm_hw_rule_mulkdiv, (void*) 1000000, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, + snd_pcm_hw_rule_mulkdiv, (void*) 1000000, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + return 0; +} + +int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; + int err; + unsigned int mask = 0; + + if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) + mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED; + if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) + mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED; + if (hw_support_mmap(substream)) { + if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) + mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED; + if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) + mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED; + if (hw->info & SNDRV_PCM_INFO_COMPLEX) + mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX; + } + err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, hw->formats); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, + hw->channels_min, hw->channels_max); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, + hw->rate_min, hw->rate_max); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + hw->period_bytes_min, hw->period_bytes_max); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, + hw->periods_min, hw->periods_max); + if (err < 0) + return err; + + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + hw->period_bytes_min, hw->buffer_bytes_max); + if (err < 0) + return err; + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_rule_buffer_bytes_max, substream, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); + if (err < 0) + return err; + + /* FIXME: remove */ + if (runtime->dma_bytes) { + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, runtime->dma_bytes); + if (err < 0) + return err; + } + + if (!(hw->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))) { + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_rate, hw, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + } + + /* FIXME: this belong to lowlevel */ + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + + return 0; +} + +static inline unsigned int div32(unsigned int a, unsigned int b, + unsigned int *r) +{ + if (b == 0) { + *r = 0; + return UINT_MAX; + } + *r = a % b; + return a / b; +} + +static inline unsigned int div_down(unsigned int a, unsigned int b) +{ + if (b == 0) + return UINT_MAX; + return a / b; +} + +static inline unsigned int div_up(unsigned int a, unsigned int b) +{ + unsigned int r; + unsigned int q; + if (b == 0) + return UINT_MAX; + q = div32(a, b, &r); + if (r) + ++q; + return q; +} + +static inline unsigned int mul(unsigned int a, unsigned int b) +{ + if (a == 0) + return 0; + if (div_down(UINT_MAX, a) < b) + return UINT_MAX; + return a * b; +} + +static inline unsigned int muldiv32(unsigned int a, unsigned int b, + unsigned int c, unsigned int *r) +{ + u_int64_t n = (u_int64_t) a * b; + if (c == 0) { + snd_BUG_ON(!n); + *r = 0; + return UINT_MAX; + } + n = div_u64_rem(n, c, r); + if (n >= UINT_MAX) { + *r = 0; + return UINT_MAX; + } + return n; +} + +/** + * snd_interval_refine - refine the interval value of configurator + * @i: the interval value to refine + * @v: the interval value to refer to + * + * Refines the interval value with the reference value. + * The interval is changed to the range satisfying both intervals. + * The interval status (min, max, integer, etc.) are evaluated. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v) +{ + int changed = 0; + if (snd_BUG_ON(snd_interval_empty(i))) + return -EINVAL; + if (i->min < v->min) { + i->min = v->min; + i->openmin = v->openmin; + changed = 1; + } else if (i->min == v->min && !i->openmin && v->openmin) { + i->openmin = 1; + changed = 1; + } + if (i->max > v->max) { + i->max = v->max; + i->openmax = v->openmax; + changed = 1; + } else if (i->max == v->max && !i->openmax && v->openmax) { + i->openmax = 1; + changed = 1; + } + if (!i->integer && v->integer) { + i->integer = 1; + changed = 1; + } + if (i->integer) { + if (i->openmin) { + i->min++; + i->openmin = 0; + } + if (i->openmax) { + i->max--; + i->openmax = 0; + } + } else if (!i->openmin && !i->openmax && i->min == i->max) + i->integer = 1; + if (snd_interval_checkempty(i)) { + snd_interval_none(i); + return -EINVAL; + } + return changed; +} + +EXPORT_SYMBOL(snd_interval_refine); + +void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) +{ + if (a->empty || b->empty) { + snd_interval_none(c); + return; + } + c->empty = 0; + c->min = mul(a->min, b->min); + c->openmin = (a->openmin || b->openmin); + c->max = mul(a->max, b->max); + c->openmax = (a->openmax || b->openmax); + c->integer = (a->integer && b->integer); +} + +/** + * snd_interval_div - refine the interval value with division + * @a: dividend + * @b: divisor + * @c: quotient + * + * c = a / b + * + * Returns non-zero if the value is changed, zero if not changed. + */ +void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c) +{ + unsigned int r; + if (a->empty || b->empty) { + snd_interval_none(c); + return; + } + c->empty = 0; + c->min = div32(a->min, b->max, &r); + c->openmin = (r || a->openmin || b->openmax); + if (b->min > 0) { + c->max = div32(a->max, b->min, &r); + if (r) { + c->max++; + c->openmax = 1; + } else + c->openmax = (a->openmax || b->openmin); + } else { + c->max = UINT_MAX; + c->openmax = 0; + } + c->integer = 0; +} + +/** + * snd_interval_muldivk - refine the interval value + * @a: dividend 1 + * @b: dividend 2 + * @k: divisor (as integer) + * @c: result + * + * c = a * b / k + * + * Returns non-zero if the value is changed, zero if not changed. + */ +void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b, + unsigned int k, struct snd_interval *c) +{ + unsigned int r; + if (a->empty || b->empty) { + snd_interval_none(c); + return; + } + c->empty = 0; + c->min = muldiv32(a->min, b->min, k, &r); + c->openmin = (r || a->openmin || b->openmin); + c->max = muldiv32(a->max, b->max, k, &r); + if (r) { + c->max++; + c->openmax = 1; + } else + c->openmax = (a->openmax || b->openmax); + c->integer = 0; +} + +/** + * snd_interval_mulkdiv - refine the interval value + * @a: dividend 1 + * @k: dividend 2 (as integer) + * @b: divisor + * @c: result + * + * c = a * k / b + * + * Returns non-zero if the value is changed, zero if not changed. + */ +void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, + const struct snd_interval *b, struct snd_interval *c) +{ + unsigned int r; + if (a->empty || b->empty) { + snd_interval_none(c); + return; + } + c->empty = 0; + c->min = muldiv32(a->min, k, b->max, &r); + c->openmin = (r || a->openmin || b->openmax); + if (b->min > 0) { + c->max = muldiv32(a->max, k, b->min, &r); + if (r) { + c->max++; + c->openmax = 1; + } else + c->openmax = (a->openmax || b->openmin); + } else { + c->max = UINT_MAX; + c->openmax = 0; + } + c->integer = 0; +} + +/* ---- */ + + +/** + * snd_interval_ratnum - refine the interval value + * @i: interval to refine + * @rats_count: number of ratnum_t + * @rats: ratnum_t array + * @nump: pointer to store the resultant numerator + * @denp: pointer to store the resultant denominator + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_interval_ratnum(struct snd_interval *i, + unsigned int rats_count, struct snd_ratnum *rats, + unsigned int *nump, unsigned int *denp) +{ + unsigned int best_num, best_den; + int best_diff; + unsigned int k; + struct snd_interval t; + int err; + unsigned int result_num, result_den; + int result_diff; + + best_num = best_den = best_diff = 0; + for (k = 0; k < rats_count; ++k) { + unsigned int num = rats[k].num; + unsigned int den; + unsigned int q = i->min; + int diff; + if (q == 0) + q = 1; + den = div_up(num, q); + if (den < rats[k].den_min) + continue; + if (den > rats[k].den_max) + den = rats[k].den_max; + else { + unsigned int r; + r = (den - rats[k].den_min) % rats[k].den_step; + if (r != 0) + den -= r; + } + diff = num - q * den; + if (diff < 0) + diff = -diff; + if (best_num == 0 || + diff * best_den < best_diff * den) { + best_diff = diff; + best_den = den; + best_num = num; + } + } + if (best_den == 0) { + i->empty = 1; + return -EINVAL; + } + t.min = div_down(best_num, best_den); + t.openmin = !!(best_num % best_den); + + result_num = best_num; + result_diff = best_diff; + result_den = best_den; + best_num = best_den = best_diff = 0; + for (k = 0; k < rats_count; ++k) { + unsigned int num = rats[k].num; + unsigned int den; + unsigned int q = i->max; + int diff; + if (q == 0) { + i->empty = 1; + return -EINVAL; + } + den = div_down(num, q); + if (den > rats[k].den_max) + continue; + if (den < rats[k].den_min) + den = rats[k].den_min; + else { + unsigned int r; + r = (den - rats[k].den_min) % rats[k].den_step; + if (r != 0) + den += rats[k].den_step - r; + } + diff = q * den - num; + if (diff < 0) + diff = -diff; + if (best_num == 0 || + diff * best_den < best_diff * den) { + best_diff = diff; + best_den = den; + best_num = num; + } + } + if (best_den == 0) { + i->empty = 1; + return -EINVAL; + } + t.max = div_up(best_num, best_den); + t.openmax = !!(best_num % best_den); + t.integer = 0; + err = snd_interval_refine(i, &t); + if (err < 0) + return err; + + if (snd_interval_single(i)) { + if (best_diff * result_den < result_diff * best_den) { + result_num = best_num; + result_den = best_den; + } + if (nump) + *nump = result_num; + if (denp) + *denp = result_den; + } + return err; +} + +EXPORT_SYMBOL(snd_interval_ratnum); + +/** + * snd_interval_ratden - refine the interval value + * @i: interval to refine + * @rats_count: number of struct ratden + * @rats: struct ratden array + * @nump: pointer to store the resultant numerator + * @denp: pointer to store the resultant denominator + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +static int snd_interval_ratden(struct snd_interval *i, + unsigned int rats_count, struct snd_ratden *rats, + unsigned int *nump, unsigned int *denp) +{ + unsigned int best_num, best_diff, best_den; + unsigned int k; + struct snd_interval t; + int err; + + best_num = best_den = best_diff = 0; + for (k = 0; k < rats_count; ++k) { + unsigned int num; + unsigned int den = rats[k].den; + unsigned int q = i->min; + int diff; + num = mul(q, den); + if (num > rats[k].num_max) + continue; + if (num < rats[k].num_min) + num = rats[k].num_max; + else { + unsigned int r; + r = (num - rats[k].num_min) % rats[k].num_step; + if (r != 0) + num += rats[k].num_step - r; + } + diff = num - q * den; + if (best_num == 0 || + diff * best_den < best_diff * den) { + best_diff = diff; + best_den = den; + best_num = num; + } + } + if (best_den == 0) { + i->empty = 1; + return -EINVAL; + } + t.min = div_down(best_num, best_den); + t.openmin = !!(best_num % best_den); + + best_num = best_den = best_diff = 0; + for (k = 0; k < rats_count; ++k) { + unsigned int num; + unsigned int den = rats[k].den; + unsigned int q = i->max; + int diff; + num = mul(q, den); + if (num < rats[k].num_min) + continue; + if (num > rats[k].num_max) + num = rats[k].num_max; + else { + unsigned int r; + r = (num - rats[k].num_min) % rats[k].num_step; + if (r != 0) + num -= r; + } + diff = q * den - num; + if (best_num == 0 || + diff * best_den < best_diff * den) { + best_diff = diff; + best_den = den; + best_num = num; + } + } + if (best_den == 0) { + i->empty = 1; + return -EINVAL; + } + t.max = div_up(best_num, best_den); + t.openmax = !!(best_num % best_den); + t.integer = 0; + err = snd_interval_refine(i, &t); + if (err < 0) + return err; + + if (snd_interval_single(i)) { + if (nump) + *nump = best_num; + if (denp) + *denp = best_den; + } + return err; +} + +/** + * snd_interval_list - refine the interval value from the list + * @i: the interval value to refine + * @count: the number of elements in the list + * @list: the value list + * @mask: the bit-mask to evaluate + * + * Refines the interval value from the list. + * When mask is non-zero, only the elements corresponding to bit 1 are + * evaluated. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_interval_list(struct snd_interval *i, unsigned int count, + const unsigned int *list, unsigned int mask) +{ + unsigned int k; + struct snd_interval list_range; + + if (!count) { + i->empty = 1; + return -EINVAL; + } + snd_interval_any(&list_range); + list_range.min = UINT_MAX; + list_range.max = 0; + for (k = 0; k < count; k++) { + if (mask && !(mask & (1 << k))) + continue; + if (!snd_interval_test(i, list[k])) + continue; + list_range.min = min(list_range.min, list[k]); + list_range.max = max(list_range.max, list[k]); + } + return snd_interval_refine(i, &list_range); +} + +EXPORT_SYMBOL(snd_interval_list); + +/** + * snd_interval_ranges - refine the interval value from the list of ranges + * @i: the interval value to refine + * @count: the number of elements in the list of ranges + * @ranges: the ranges list + * @mask: the bit-mask to evaluate + * + * Refines the interval value from the list of ranges. + * When mask is non-zero, only the elements corresponding to bit 1 are + * evaluated. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_interval_ranges(struct snd_interval *i, unsigned int count, + const struct snd_interval *ranges, unsigned int mask) +{ + unsigned int k; + struct snd_interval range_union; + struct snd_interval range; + + if (!count) { + snd_interval_none(i); + return -EINVAL; + } + snd_interval_any(&range_union); + range_union.min = UINT_MAX; + range_union.max = 0; + for (k = 0; k < count; k++) { + if (mask && !(mask & (1 << k))) + continue; + snd_interval_copy(&range, &ranges[k]); + if (snd_interval_refine(&range, i) < 0) + continue; + if (snd_interval_empty(&range)) + continue; + + if (range.min < range_union.min) { + range_union.min = range.min; + range_union.openmin = 1; + } + if (range.min == range_union.min && !range.openmin) + range_union.openmin = 0; + if (range.max > range_union.max) { + range_union.max = range.max; + range_union.openmax = 1; + } + if (range.max == range_union.max && !range.openmax) + range_union.openmax = 0; + } + return snd_interval_refine(i, &range_union); +} +EXPORT_SYMBOL(snd_interval_ranges); + +static int snd_interval_step(struct snd_interval *i, unsigned int step) +{ + unsigned int n; + int changed = 0; + n = i->min % step; + if (n != 0 || i->openmin) { + i->min += step - n; + i->openmin = 0; + changed = 1; + } + n = i->max % step; + if (n != 0 || i->openmax) { + i->max -= n; + i->openmax = 0; + changed = 1; + } + if (snd_interval_checkempty(i)) { + i->empty = 1; + return -EINVAL; + } + return changed; +} + +/* Info constraints helpers */ + +/** + * snd_pcm_hw_rule_add - add the hw-constraint rule + * @runtime: the pcm runtime instance + * @cond: condition bits + * @var: the variable to evaluate + * @func: the evaluation function + * @private: the private data pointer passed to function + * @dep: the dependent variables + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, + int var, + snd_pcm_hw_rule_func_t func, void *private, + int dep, ...) +{ + struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + struct snd_pcm_hw_rule *c; + unsigned int k; + va_list args; + va_start(args, dep); + if (constrs->rules_num >= constrs->rules_all) { + struct snd_pcm_hw_rule *new; + unsigned int new_rules = constrs->rules_all + 16; + new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL); + if (!new) { + va_end(args); + return -ENOMEM; + } + if (constrs->rules) { + memcpy(new, constrs->rules, + constrs->rules_num * sizeof(*c)); + kfree(constrs->rules); + } + constrs->rules = new; + constrs->rules_all = new_rules; + } + c = &constrs->rules[constrs->rules_num]; + c->cond = cond; + c->func = func; + c->var = var; + c->private = private; + k = 0; + while (1) { + if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps))) { + va_end(args); + return -EINVAL; + } + c->deps[k++] = dep; + if (dep < 0) + break; + dep = va_arg(args, int); + } + constrs->rules_num++; + va_end(args); + return 0; +} + +EXPORT_SYMBOL(snd_pcm_hw_rule_add); + +/** + * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint + * @runtime: PCM runtime instance + * @var: hw_params variable to apply the mask + * @mask: the bitmap mask + * + * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + u_int32_t mask) +{ + struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + struct snd_mask *maskp = constrs_mask(constrs, var); + *maskp->bits &= mask; + memset(maskp->bits + 1, 0, (SNDRV_MASK_MAX-32) / 8); /* clear rest */ + if (*maskp->bits == 0) + return -EINVAL; + return 0; +} + +/** + * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint + * @runtime: PCM runtime instance + * @var: hw_params variable to apply the mask + * @mask: the 64bit bitmap mask + * + * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + u_int64_t mask) +{ + struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + struct snd_mask *maskp = constrs_mask(constrs, var); + maskp->bits[0] &= (u_int32_t)mask; + maskp->bits[1] &= (u_int32_t)(mask >> 32); + memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */ + if (! maskp->bits[0] && ! maskp->bits[1]) + return -EINVAL; + return 0; +} +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); + +/** + * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval + * @runtime: PCM runtime instance + * @var: hw_params variable to apply the integer constraint + * + * Apply the constraint of integer to an interval parameter. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var) +{ + struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + return snd_interval_setinteger(constrs_interval(constrs, var)); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); + +/** + * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval + * @runtime: PCM runtime instance + * @var: hw_params variable to apply the range + * @min: the minimal value + * @max: the maximal value + * + * Apply the min/max range constraint to an interval parameter. + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + unsigned int min, unsigned int max) +{ + struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + struct snd_interval t; + t.min = min; + t.max = max; + t.openmin = t.openmax = 0; + t.integer = 0; + return snd_interval_refine(constrs_interval(constrs, var), &t); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax); + +static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hw_constraint_list *list = rule->private; + return snd_interval_list(hw_param_interval(params, rule->var), list->count, list->list, list->mask); +} + + +/** + * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the list constraint + * @l: list + * + * Apply the list of constraints to an interval parameter. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_list *l) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_list, (void *)l, + var, -1); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_list); + +static int snd_pcm_hw_rule_ranges(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hw_constraint_ranges *r = rule->private; + return snd_interval_ranges(hw_param_interval(params, rule->var), + r->count, r->ranges, r->mask); +} + + +/** + * snd_pcm_hw_constraint_ranges - apply list of range constraints to a parameter + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the list of range constraints + * @r: ranges + * + * Apply the list of range constraints to an interval parameter. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + const struct snd_pcm_hw_constraint_ranges *r) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_ranges, (void *)r, + var, -1); +} +EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges); + +static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hw_constraint_ratnums *r = rule->private; + unsigned int num = 0, den = 0; + int err; + err = snd_interval_ratnum(hw_param_interval(params, rule->var), + r->nrats, r->rats, &num, &den); + if (err >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + return err; +} + +/** + * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the ratnums constraint + * @r: struct snd_ratnums constriants + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + struct snd_pcm_hw_constraint_ratnums *r) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_ratnums, r, + var, -1); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); + +static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hw_constraint_ratdens *r = rule->private; + unsigned int num = 0, den = 0; + int err = snd_interval_ratden(hw_param_interval(params, rule->var), + r->nrats, r->rats, &num, &den); + if (err >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + return err; +} + +/** + * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the ratdens constraint + * @r: struct snd_ratdens constriants + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + struct snd_pcm_hw_constraint_ratdens *r) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_ratdens, r, + var, -1); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_ratdens); + +static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned int l = (unsigned long) rule->private; + int width = l & 0xffff; + unsigned int msbits = l >> 16; + struct snd_interval *i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + + if (!snd_interval_single(i)) + return 0; + + if ((snd_interval_value(i) == width) || + (width == 0 && snd_interval_value(i) > msbits)) + params->msbits = min_not_zero(params->msbits, msbits); + + return 0; +} + +/** + * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule + * @runtime: PCM runtime instance + * @cond: condition bits + * @width: sample bits width + * @msbits: msbits width + * + * This constraint will set the number of most significant bits (msbits) if a + * sample format with the specified width has been select. If width is set to 0 + * the msbits will be set for any sample format with a width larger than the + * specified msbits. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, + unsigned int cond, + unsigned int width, + unsigned int msbits) +{ + unsigned long l = (msbits << 16) | width; + return snd_pcm_hw_rule_add(runtime, cond, -1, + snd_pcm_hw_rule_msbits, + (void*) l, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_msbits); + +static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned long step = (unsigned long) rule->private; + return snd_interval_step(hw_param_interval(params, rule->var), step); +} + +/** + * snd_pcm_hw_constraint_step - add a hw constraint step rule + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the step constraint + * @step: step size + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var, + unsigned long step) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_step, (void *) step, + var, -1); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_step); + +static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) +{ + static unsigned int pow2_sizes[] = { + 1<<0, 1<<1, 1<<2, 1<<3, 1<<4, 1<<5, 1<<6, 1<<7, + 1<<8, 1<<9, 1<<10, 1<<11, 1<<12, 1<<13, 1<<14, 1<<15, + 1<<16, 1<<17, 1<<18, 1<<19, 1<<20, 1<<21, 1<<22, 1<<23, + 1<<24, 1<<25, 1<<26, 1<<27, 1<<28, 1<<29, 1<<30 + }; + return snd_interval_list(hw_param_interval(params, rule->var), + ARRAY_SIZE(pow2_sizes), pow2_sizes, 0); +} + +/** + * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule + * @runtime: PCM runtime instance + * @cond: condition bits + * @var: hw_params variable to apply the power-of-2 constraint + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, + unsigned int cond, + snd_pcm_hw_param_t var) +{ + return snd_pcm_hw_rule_add(runtime, cond, var, + snd_pcm_hw_rule_pow2, NULL, + var, -1); +} + +EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2); + +static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + unsigned int base_rate = (unsigned int)(uintptr_t)rule->private; + struct snd_interval *rate; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + return snd_interval_list(rate, 1, &base_rate, 0); +} + +/** + * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling + * @runtime: PCM runtime instance + * @base_rate: the rate at which the hardware does not resample + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, + unsigned int base_rate) +{ + return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE, + SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_noresample_func, + (void *)(uintptr_t)base_rate, + SNDRV_PCM_HW_PARAM_RATE, -1); +} +EXPORT_SYMBOL(snd_pcm_hw_rule_noresample); + + -- 1.9.1

4 9

[alsa-devel] [PATCH] ASoC: fsl: add imx-wm8960 machine driver
by Zidan Wang 12 Jun '15

12 Jun '15

This imx-wm8960 device-tree-only machine driver works with sai driver and have below feature. * support codec master and slave mode * support headphone jack detect * support headphone and micphone jack event * support asrc<->sai<->wm8960 mode Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8960.txt | 68 ++ sound/soc/fsl/Kconfig | 13 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8960.c | 711 +++++++++++++++++++++ 4 files changed, 794 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8960.txt create mode 100644 sound/soc/fsl/imx-wm8960.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8960.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8960.txt new file mode 100644 index 0000000..300d027 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8960.txt @@ -0,0 +1,68 @@ +Freescale i.MX audio complex with WM8960 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8960" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8960 audio codec + + - hp-det : ADCLRC/GPIO1, LINPUT3/JD2 and RINPUT3/JD3 pins can + be selected as headphone jack detect inputs to + automatically disable the speaker output and enable + the headphone output. + hp-det = <hp-det-pin hp-det-polarity>; + hp-det-pin = 1: ADCLRC/GPIO1 used as detect pin + hp-det-pin = 2: LINPUT3/JD2 used as detect pin + hp-det-pin = 3: RINPUT3/JD3 used as detect pin + hp-det-polarity = 0: hp detect high for headphone + hp-det-polarity = 1: hp detect high for speaker + + - codec-master : If "codec-master" present, codec works as master. + Otherwise, codec works as slave. + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, WM8962 + pins, and the jacks on the board: + + Power supplies: + * MICB + + Board connectors: + * Hp MIC + * Main MIC + * Headset Jack + * Ext Spk + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8960", + "fsl,imx-audio-wm8960"; + model = "wm8960-audio"; + cpu-dai = <&sai2>; + audio-codec = <&codec>; + asrc-controller = <&asrc>; + codec-master; + hp-det = <3 0>; + audio-routing = + "Headset Jack", "HP_L", + "Headset Jack", "HP_R", + "Ext Spk", "SPK_LP", + "Ext Spk", "SPK_LN", + "Ext Spk", "SPK_RP", + "Ext Spk", "SPK_RN", + "LINPUT2", "Hp MIC", + "LINPUT3", "Hp MIC", + "RINPUT1", "Main MIC", + "RINPUT2", "Main MIC", + "Hp MIC", "MICB", + "Main MIC", "MICB", + "CPU-Playback", "ASRC-Playback", + "Playback", "CPU-Playback", + "ASRC-Capture", "CPU-Capture", + "CPU-Capture", "Capture"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19c302b..0d9aa56 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,19 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8960 + tristate "SoC Audio support for i.MX boards with wm8960" + depends on OF && I2C + select SND_SOC_WM8960 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8960 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8960 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..84a9a10 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8960-objs := imx-wm8960.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8960) += snd-soc-imx-wm8960.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8960.c b/sound/soc/fsl/imx-wm8960.c new file mode 100644 index 0000000..ddd17cb --- /dev/null +++ b/sound/soc/fsl/imx-wm8960.c @@ -0,0 +1,711 @@ +/* + * Copyright (C) 2015 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> +#include <linux/mfd/syscon.h> +#include "../codecs/wm8960.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8960_data { + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; + bool is_codec_master; + bool is_stream_in_use[2]; + struct regmap *gpr; + unsigned int hp_det[2]; + u32 asrc_rate; + u32 asrc_format; +}; + +struct imx_priv { + int hp_set_gpio; + int hp_active_low; + struct snd_kcontrol *headset_kctl; + struct snd_soc_codec *codec; + struct platform_device *pdev; + struct platform_device *asrc_pdev; + struct snd_card *snd_card; +}; + +static struct imx_priv card_priv; + +static struct snd_soc_jack imx_hp_set; +static struct snd_soc_jack_pin imx_hp_set_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + }, +}; +static struct snd_soc_jack_gpio imx_hp_set_gpio = { + .name = "headset detect", + .report = SND_JACK_HEADSET, + .debounce_time = 250, + .invert = 1, +}; + +static int hp_set_status_check(void *data) +{ + struct imx_priv *priv = &card_priv; + struct platform_device *pdev = priv->pdev; + char *envp[3], *buf; + int hp_status, ret; + + if (!gpio_is_valid(priv->hp_set_gpio)) + return 0; + + hp_status = gpio_get_value(priv->hp_set_gpio) ? 1 : 0; + buf = kmalloc(32, GFP_ATOMIC); + if (!buf) { + dev_err(&pdev->dev, "%s kmalloc failed\n", __func__); + return -ENOMEM; + } + + if (hp_status != priv->hp_active_low) { + snprintf(buf, 32, "STATE=%d", 2); + snd_soc_dapm_disable_pin(&priv->codec->dapm, "Ext Spk"); + snd_soc_dapm_disable_pin(&priv->codec->dapm, "Main MIC"); + ret = imx_hp_set_gpio.report; + + /* + * As the hp MIC only connect the input for left channel, we + * need to route it for right channel. + */ + snd_soc_update_bits(priv->codec, WM8960_ADDCTL1, 3<<2, 1<<2); + + snd_kctl_jack_report(priv->snd_card, priv->headset_kctl, 1); + } else { + snprintf(buf, 32, "STATE=%d", 0); + snd_soc_dapm_enable_pin(&priv->codec->dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(&priv->codec->dapm, "Main MIC"); + ret = 0; + + /* + * As the Main MIC only connect the input for right channel, + * we need to route it for left channel. + */ + snd_soc_update_bits(priv->codec, WM8960_ADDCTL1, 3<<2, 2<<2); + + snd_kctl_jack_report(priv->snd_card, priv->headset_kctl, 0); + } + + envp[0] = "NAME=headset"; + envp[1] = buf; + envp[2] = NULL; + kobject_uevent_env(&pdev->dev.kobj, KOBJ_CHANGE, envp); + kfree(buf); + + return ret; +} + +static const struct snd_soc_dapm_widget imx_wm8960_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headset Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Hp MIC", NULL), + SND_SOC_DAPM_MIC("Main MIC", NULL), +}; + +static int imx_wm8960_gpio_init(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct imx_priv *priv = &card_priv; + int ret; + + priv->codec = codec; + + if (gpio_is_valid(priv->hp_set_gpio)) { + imx_hp_set_gpio.gpio = priv->hp_set_gpio; + imx_hp_set_gpio.jack_status_check = hp_set_status_check; + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET, &imx_hp_set, + imx_hp_set_pins, ARRAY_SIZE(imx_hp_set_pins)); + if (ret) + return ret; + ret = snd_soc_jack_add_gpios(&imx_hp_set, 1, + &imx_hp_set_gpio); + if (ret) + return ret; + } + + return 0; +} + +static ssize_t show_headphone(struct device_driver *dev, char *buf) +{ + struct imx_priv *priv = &card_priv; + int hp_status; + + if (!gpio_is_valid(priv->hp_set_gpio)) { + strcpy(buf, "no detect gpio connected\n"); + return strlen(buf); + } + + /* Check if headphone is plugged in */ + hp_status = gpio_get_value(priv->hp_set_gpio) ? 1 : 0; + + if (hp_status != priv->hp_active_low) + strcpy(buf, "Headphone\n"); + else + strcpy(buf, "Speaker\n"); + + return strlen(buf); +} + +static ssize_t show_micphone(struct device_driver *dev, char *buf) +{ + struct imx_priv *priv = &card_priv; + int hp_status; + + if (!gpio_is_valid(priv->hp_set_gpio)) { + strcpy(buf, "no detect gpio connected\n"); + return strlen(buf); + } + + /* Check if headphone is plugged in */ + hp_status = gpio_get_value(priv->hp_set_gpio) ? 1 : 0; + + if (hp_status != priv->hp_active_low) + strcpy(buf, "Hp MIC\n"); + else + strcpy(buf, "Main MIC\n"); + + return strlen(buf); +} +static DRIVER_ATTR(headphone, S_IRUGO | S_IWUSR, show_headphone, NULL); +static DRIVER_ATTR(micphone, S_IRUGO | S_IWUSR, show_micphone, NULL); + +static void wm8960_init(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + + /* + * codec ADCLRC pin configured as GPIO, DACLRC pin is used as + * a frame clock for ADCs and DACs + */ + snd_soc_update_bits(codec, WM8960_IFACE2, 1<<6, 1<<6); + + /* GPIO1 used as headphone detect output */ + snd_soc_update_bits(codec, WM8960_ADDCTL4, 7<<4, 3<<4); + + /* Enable headphone jack detect */ + snd_soc_update_bits(codec, WM8960_ADDCTL2, 1<<6, 1<<6); + snd_soc_update_bits(codec, WM8960_ADDCTL2, 1<<5, data->hp_det[1]<<5); + snd_soc_update_bits(codec, WM8960_ADDCTL4, 3<<2, data->hp_det[0]<<2); + snd_soc_update_bits(codec, WM8960_ADDCTL1, 3, 3); + + /* route ADC left channel to right channel in default */ + snd_soc_update_bits(codec, WM8960_ADDCTL1, 3<<2, 1<<2); +} + +/* -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 }; + +/* Multiply 10 to eliminate decimials */ +static const int bclk_divs[] = { + 10, 15, 20, 30, 40, 55, 60, 80, 110, + 120, 160, 220, 240, 320, 320, 320 +}; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct device *dev = card->dev; + unsigned int sample_rate = params_rate(params); + unsigned int sysclk, pll_in, pll_out; + int i, j, k, ret = 0; + int bclk = snd_soc_params_to_bclk(params); + unsigned int fmt, plln; + + if (params_channels(params) == 1) + bclk *= 2; + + data->is_stream_in_use[tx] = true; + + if (data->is_stream_in_use[!tx]) + return 0; + + if (data->is_codec_master) + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + else + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret) { + dev_err(dev, "failed to set codec dai fmt: %d\n", ret); + return ret; + } + + if (!data->is_codec_master) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, + 0, 0, 2, params_width(params)); + if (ret) { + dev_err(dev, + "failed to set cpu dai tdm slot: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + return 0; + } else { + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + + /* Use MCLK to provide sysclk directly */ + sysclk = data->clk_frequency; + + 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 == sample_rate * dac_divs[j]) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; + } + if (i != ARRAY_SIZE(sysclk_divs)) { + /* Set codec sysclk */ + snd_soc_dai_set_sysclk(codec_dai, + WM8960_SYSCLK_MCLK, sysclk, 0); + snd_soc_dai_set_clkdiv(codec_dai, WM8960_SYSCLKDIV, i << 1); + return 0; + } + + /* Use PLL to provide sysclk */ + pll_in = data->clk_frequency; + + 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]; + /* + * we should make sure 5 < plln < 13, or we can't get + * such pll_out. + */ + plln = pll_out * 4 / pll_in; + if (plln < 6) + plln *= 2; + if ((plln < 6) || (plln > 12)) + continue; + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10) { + /* Set codec pll */ + ret = snd_soc_dai_set_pll(codec_dai, 0, + 0, data->clk_frequency, + pll_out); + if (ret != 0) + continue; + else + break; + } + } + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + if (j != ARRAY_SIZE(dac_divs)) + break; + } + if (i != ARRAY_SIZE(sysclk_divs)) { + /* Set codec sysclk */ + snd_soc_dai_set_sysclk(codec_dai, WM8960_SYSCLK_PLL, sysclk, 0); + snd_soc_dai_set_clkdiv(codec_dai, WM8960_SYSCLKDIV, i << 1); + return 0; + } + + dev_err(dev, "failed to configure system clock"); + return -EINVAL; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + data->is_stream_in_use[tx] = false; + + /* Power down PLL to save power*/ + if (data->is_codec_master && !data->is_stream_in_use[!tx]) + snd_soc_dai_set_pll(codec_dai, 0, 0, 0, 0); + + return 0; +} + +static u32 imx_wm8960_rates[] = { 8000, 16000, 32000, 48000 }; +static struct snd_pcm_hw_constraint_list imx_wm8960_rate_constraints = { + .count = ARRAY_SIZE(imx_wm8960_rates), + .list = imx_wm8960_rates, +}; + +static int imx_hifi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + int ret = 0; + + if (!data->is_codec_master) { + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_wm8960_rate_constraints); + if (ret) + return ret; + } + + /* codec mclk should be enabled early to avoid jack detect error */ + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(card->dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + + return ret; +} + +static void imx_hifi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + + clk_disable_unprepare(data->codec_clk); +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, + .startup = imx_hifi_startup, + .shutdown = imx_hifi_shutdown, +}; + +static int imx_wm8960_late_probe(struct snd_soc_card *card) +{ + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + + /* set SAI2_MCLK_DIR to enable codec MCLK */ + if (data->gpr) + regmap_update_bits(data->gpr, 4, 1<<20, 1<<20); + + wm8960_init(card); + return 0; +} + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_card *card = rtd->card; + struct imx_wm8960_data *data = snd_soc_card_get_drvdata(card); + struct imx_priv *priv = &card_priv; + struct snd_interval *rate; + struct snd_mask *mask; + + if (!priv->asrc_pdev) + return -EINVAL; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = rate->min = data->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, data->asrc_format); + + return 0; +} + +static struct snd_soc_dai_link imx_wm8960_dai[] = { + { + .name = "HiFi", + .stream_name = "HiFi", + .codec_dai_name = "wm8960-hifi", + .ops = &imx_hifi_ops, + }, + { + .name = "HiFi-ASRC-FE", + .stream_name = "HiFi-ASRC-FE", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "HiFi-ASRC-BE", + .stream_name = "HiFi-ASRC-BE", + .codec_dai_name = "wm8960-hifi", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &imx_hifi_ops, + .be_hw_params_fixup = be_hw_params_fixup, + }, +}; + +static int imx_wm8960_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np, *gpr_np; + struct platform_device *cpu_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8960_data *data; + struct platform_device *asrc_pdev = NULL; + struct device_node *asrc_np; + u32 width; + int ret; + + priv->pdev = pdev; + + cpu_np = of_parse_phandle(pdev->dev.of_node, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + if (of_property_read_bool(pdev->dev.of_node, "codec-master")) + data->is_codec_master = true; + + data->codec_clk = devm_clk_get(&codec_dev->dev, "mclk"); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&pdev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + gpr_np = of_parse_phandle(pdev->dev.of_node, "gpr", 0); + if (gpr_np) { + data->gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(data->gpr)) { + ret = PTR_ERR(data->gpr); + dev_err(&pdev->dev, "failed to get gpr regmap\n"); + goto fail; + } + } + + /* + * hp-det = <hp-det-pin hp-det-polarity>; + * hp-det-pin: JD1 JD2 or JD3 + * hp-det-polarity = 0: hp detect high for headphone + * hp-det-polarity = 1: hp detect high for speaker + */ + of_property_read_u32_array(pdev->dev.of_node, + "hp-det", data->hp_det, 2); + + asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0); + if (asrc_np) { + asrc_pdev = of_find_device_by_node(asrc_np); + priv->asrc_pdev = asrc_pdev; + } + + data->card.dai_link = imx_wm8960_dai; + + imx_wm8960_dai[0].codec_of_node = codec_np; + imx_wm8960_dai[0].cpu_dai_name = dev_name(&cpu_pdev->dev); + imx_wm8960_dai[0].platform_of_node = cpu_np; + + if (!asrc_pdev) { + data->card.num_links = 1; + } else { + imx_wm8960_dai[1].cpu_of_node = asrc_np; + imx_wm8960_dai[1].platform_of_node = asrc_np; + imx_wm8960_dai[2].codec_of_node = codec_np; + imx_wm8960_dai[2].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->card.num_links = 3; + + ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", + &data->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + if (width == 24) + data->asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + data->asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + data->card.dapm_widgets = imx_wm8960_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8960_dapm_widgets); + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + data->card.late_probe = imx_wm8960_late_probe; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "failed to register card (%d)\n", ret); + goto fail; + } + + priv->snd_card = data->card.snd_card; + + priv->hp_set_gpio = of_get_named_gpio_flags(pdev->dev.of_node, + "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + if (IS_ERR(ERR_PTR(priv->hp_set_gpio))) + goto fail; + + priv->headset_kctl = snd_kctl_jack_new("Headset", priv->snd_card); + ret = snd_ctl_add(data->card.snd_card, priv->headset_kctl); + if (ret) + goto fail; + + ret = imx_wm8960_gpio_init(&data->card); + + if (gpio_is_valid(priv->hp_set_gpio)) { + ret = driver_create_file(pdev->dev.driver, + &driver_attr_headphone); + if (ret) { + dev_err(&pdev->dev, + "create hp attr failed (%d)\n", ret); + goto fail; + } + ret = driver_create_file(pdev->dev.driver, + &driver_attr_micphone); + if (ret) { + dev_err(&pdev->dev, + "create mic attr failed (%d)\n", ret); + goto fail; + } + } +fail: + if (cpu_np) + of_node_put(cpu_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8960_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8960", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8960_dt_ids); + +static struct platform_driver imx_wm8960_driver = { + .driver = { + .name = "imx-wm8960", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8960_dt_ids, + }, + .probe = imx_wm8960_probe, +}; +module_platform_driver(imx_wm8960_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8960 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8960"); -- 1.9.1

3 5

Re: [alsa-devel] Can't Unload ALSA Driver Without Disconnecting the Device
by Takashi Iwai 12 Jun '15

12 Jun '15

At Fri, 12 Jun 2015 02:57:27 -0400, Azizul Hakim wrote: > > I see. So in this scenario what'll be the way to unload the driver > successfully? Is there a way to stop the mixer interface? Kill pulseaudio properly. It'll auto respawn itself usually, so for killing PA, you'd have to change the PA configuration. Takashi > > On Fri, Jun 12, 2015 at 2:55 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Fri, 12 Jun 2015 02:51:03 -0400, > > Azizul Hakim wrote: > > > > > > I tried that command. The output is as below: > > > > > > USER PID ACCESS COMMAND > > > /dev/snd/controlC0: mhaki005 2043 F.... pulseaudio > > > /dev/snd/controlC1: mhaki005 2043 F.... pulseaudio > > > /dev/snd/pcmC0D0c: mhaki005 2043 F...m pulseaudio > > > > > > > > > I think pcmC0D0c stands for "Capture" device. My device is a playback > > > device. So I don't think it is being used by anyone. > > > > /dev/snd/controlC1 is being used. It's the mixer interface. > > > > > > Takashi > > > > > > > > > > > On Fri, Jun 12, 2015 at 2:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > At Fri, 12 Jun 2015 02:28:16 -0400, > > > > Azizul Hakim wrote: > > > > > > > > > > Hi, > > > > > > > > > > I've been developing a sound card driver for a USB device using > > ALSA. My > > > > > driver is almost working but I've got one small issue. If I try to > > unload > > > > > the driver module using "rmmod" command when the device is still > > > > connected > > > > > to the USB port, it says "Module is in use" even though I'm not > > playing > > > > any > > > > > sound files or anything. > > > > > > > > > > > > > > > If I try to execute the "lsmod" command I see that it is saying the > > usage > > > > > count of my driver to be 1, but it doesn't say anything about who is > > > > using > > > > > it. Once I disconnect the device from the USB port, I can easily > > unload > > > > the > > > > > module. > > > > > > > > > > > > > > > So it seems to me it won't be possible to unload a sound card driver > > if > > > > the > > > > > card is attached in the computer. Is my assumption correct? If not, > > what > > > > > might be some key points to solve the issue? > > > > > > > > Check "fuser -v /dev/snd/*" as root. Something (e.g. a mixer > > > > application or sound backend) must be accessing the device. > > > > > > > > > > > > Takashi > > > >

2 1

[alsa-devel] Can't Unload ALSA Driver Without Disconnecting the Device
by Azizul Hakim 12 Jun '15

12 Jun '15

Hi, I've been developing a sound card driver for a USB device using ALSA. My driver is almost working but I've got one small issue. If I try to unload the driver module using "rmmod" command when the device is still connected to the USB port, it says "Module is in use" even though I'm not playing any sound files or anything. If I try to execute the "lsmod" command I see that it is saying the usage count of my driver to be 1, but it doesn't say anything about who is using it. Once I disconnect the device from the USB port, I can easily unload the module. So it seems to me it won't be possible to unload a sound card driver if the card is attached in the computer. Is my assumption correct? If not, what might be some key points to solve the issue? Thanks for any suggestion Regards Azizul Hakim

2 4

[alsa-devel] [PATCH] ASoC: rt5645: Add the device tree parser
by Oder Chiou 12 Jun '15

12 Jun '15

Modify the RT5645 driver to parse platform data from device tree. This is missing from previous patch in sound/soc/codecs/rt5645.c, that was present in v3. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7d04b1a..7bfc17e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3231,6 +3231,20 @@ static struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) +{ + rt5645->pdata.in2_diff = device_property_read_bool(dev, + "realtek,in2-differential"); + device_property_read_u32(dev, + "realtek,dmic1-data-pin", &rt5645->pdata.dmic1_data_pin); + device_property_read_u32(dev, + "realtek,dmic2-data-pin", &rt5645->pdata.dmic2_data_pin); + device_property_read_u32(dev, + "realtek,jd-mode", &rt5645->pdata.jd_mode); + + return 0; +} + static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3249,6 +3263,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) { rt5645->pdata = *pdata; + } else if (i2c->dev.of_node) { + rt5645_parse_dt(rt5645, &i2c->dev); } else { if (dmi_check_system(dmi_platform_intel_braswell)) { rt5645->pdata = *rt5645_pdata; -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH v2 - seq 1/1] ALSA seq: expose the card number of ALSA seq clients
by Adam Goode 12 Jun '15

12 Jun '15

Expose the card number of seq clients. This allows interested userspace programs to discover the hardware device that backs a particular seq client. Before this change, the only way to get information about the hardware for a client was by using brittle heuristics. SNDRV_SEQ_VERSION is not yet incremented in this patch, but should be relatively soon. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 5a5fa49..2627bee 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -357,7 +357,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card_number; /* RO: card number, or -1 if no card. Added in protocol version 1.0.2 */ + char reserved[60]; /* for future use */ }; diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index edbdab8..5f2e3bf 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -216,7 +216,7 @@ int __init client_init_data(void) } -static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) +static struct snd_seq_client *seq_create_client1(int client_index, int poolsize, int card_number) { unsigned long flags; int c; @@ -232,6 +232,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) return NULL; } client->type = NO_CLIENT; + client->card_number = card_number; snd_use_lock_init(&client->use_lock); rwlock_init(&client->ports_lock); mutex_init(&client->ports_mutex); @@ -327,7 +328,7 @@ static int snd_seq_open(struct inode *inode, struct file *file) if (mutex_lock_interruptible(&register_mutex)) return -ERESTARTSYS; - client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS); + client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS, -1); if (client == NULL) { mutex_unlock(&register_mutex); return -ENOMEM; /* failure code */ @@ -1194,6 +1195,7 @@ static void get_client_info(struct snd_seq_client *cptr, info->event_lost = cptr->event_lost; memcpy(info->event_filter, cptr->event_filter, 32); info->num_ports = cptr->num_ports; + info->card_number = cptr->card_number; memset(info->reserved, 0, sizeof(info->reserved)); } @@ -2239,6 +2241,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, { struct snd_seq_client *client; va_list args; + int card_number = -1; if (snd_BUG_ON(in_interrupt())) return -EBUSY; @@ -2252,6 +2255,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, return -ERESTARTSYS; if (card) { + card_number = card->number; client_index += SNDRV_SEQ_GLOBAL_CLIENTS + card->number * SNDRV_SEQ_CLIENTS_PER_CARD; if (client_index >= SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN) @@ -2259,7 +2263,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, } /* empty write queue as default */ - client = seq_create_client1(client_index, 0); + client = seq_create_client1(client_index, 0, card_number); if (client == NULL) { mutex_unlock(&register_mutex); return -EBUSY; /* failure code */ @@ -2533,9 +2537,12 @@ void snd_seq_info_clients_read(struct snd_info_entry *entry, continue; } - snd_iprintf(buffer, "Client %3d : \"%s\" [%s]\n", + snd_iprintf(buffer, "Client %3d : \"%s\" [%s]", c, client->name, client->type == USER_CLIENT ? "User" : "Kernel"); + if (client->card_number != -1) + snd_iprintf(buffer, ", card %d", client->card_number); + snd_iprintf(buffer, "\n"); snd_seq_info_dump_ports(buffer, client); if (snd_seq_write_pool_allocated(client)) { snd_iprintf(buffer, " Output pool :\n"); diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 20f0a72..627be9d 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -50,6 +50,7 @@ struct snd_seq_client { accept_output: 1; char name[64]; /* client name */ int number; /* client number */ + int card_number; /* card number, or -1 if no card */ unsigned int filter; /* filter flags */ DECLARE_BITMAP(event_filter, 256); snd_use_lock_t use_lock; -- 2.4.2

1 0

Re: [alsa-devel] ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller
by Dan Carpenter 11 Jun '15

11 Jun '15

Hello Jyri Sarha, The patch 9fbd58cf4ab0: "ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller" from Jun 2, 2015, leads to the following static checker warning: sound/soc/davinci/davinci-mcasp.c:1842 davinci_mcasp_probe() info: ignoring unreachable code. sound/soc/davinci/davinci-mcasp.c 1812 switch (ret) { 1813 case PCM_EDMA: 1814 #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ 1815 (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ 1816 IS_MODULE(CONFIG_SND_EDMA_SOC)) 1817 ret = edma_pcm_platform_register(&pdev->dev); 1818 #else 1819 dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); 1820 ret = -EINVAL; 1821 goto err; 1822 #endif 1823 break; 1824 case PCM_SDMA: 1825 #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ 1826 (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ 1827 IS_MODULE(CONFIG_SND_OMAP_SOC)) 1828 ret = omap_pcm_platform_register(&pdev->dev); 1829 #else 1830 dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); 1831 ret = -EINVAL; 1832 goto err; 1833 #endif 1834 break; 1835 default: 1836 dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); 1837 case -EPROBE_DEFER: 1838 goto err; 1839 break; 1840 } 1841 Depending on the ifdefs then this is reachable, I guess... 1842 if (ret) { 1843 dev_err(&pdev->dev, "register PCM failed: %d\n", ret); 1844 goto err; 1845 } 1846 1847 return 0; 1848 1849 err: 1850 pm_runtime_disable(&pdev->dev); 1851 return ret; 1852 } 1853 regards, dan carpenter

2 1

[alsa-devel] [PATCH] ALSA: HDA: remove the unused function declaration
by Vinod Koul 11 Jun '15

11 Jun '15

Commit a1b3fda6ae ALSA: hdac_ext: add hdac extended controller, erroneously added snd_hdac_ext_bus_map_codec_to_link() function declaration, so remove it Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hdaudio_ext.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 0044caa19155..202350a8eddb 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -47,7 +47,6 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *chip, bool enable, int index); int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *bus); -int snd_hdac_ext_bus_map_codec_to_link(struct hdac_ext_bus *bus, int addr); struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *bus, const char *codec_name); -- 1.9.1

2 1

[alsa-devel] [PATCH v9 0/3] HDA: add extended HDA
by Vinod Koul 11 Jun '15

11 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Changes in v9: - removed snd_hdac_ext_bus_map_codec_to_link - updated snd_hdac_ext_bus_get_link per Takahsi comment and drop the array - add index in bus instance Changes in v8: - fixed some dead code crept in - checkpatch cleanup Changes in v7: - Feedback from Takashi - removed the macros for register read/writes - add new macros for reg update - fix bus pointer macro - header cleanup on c/h files - new device name ehdaudioiDn - removed flags for ext caps, use pointer only - moved module info to patch 1 - sparse check on series - remove codec name in link Jeeja KP (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 129 ++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 117 +++++++++++ sound/hda/ext/hdac_ext_controller.c | 270 ++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 402 ++++++++++++++++++++++++++++++++++++ 7 files changed, 928 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

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[alsa-devel] [PATCH spi-pxa2xx.c 1/1] : fix spi init of WM510205 codec via ACPI, the chip hangs on chipselect 1
by Christian Hartmann 11 Jun '15

11 Jun '15

Hello, I have one patch that enables the spi initialization of the WM510205 audio codec Without the patch, this message indicates that the codec chip fails with the current chipselect number in spi-pxa2xx: [ 0.296256] pxa2xx-spi 80860F0E:00: cs1 >= max 1 [ 0.296270] spi_master spi32766: failed to add SPI device WM510205:00 from ACPI So raise num_chipselect by one to enable the codec. The WM510202 is on chipselect = 1. With the patch applied I got in dmesg now: [ 6.056829] pxa2xx-spi 80860F0E:00: no DMA channels available, using PIO [ 6.067035] pxa2xx-spi 80860F0E:00: registered master spi32766 (dynamic) [ 6.067214] spi spi-WM510205:00: 8333333 Hz actual, PIO [ 6.067221] spi spi-WM510205:00: setup mode 0, 8 bits/w, 8000000 Hz max --> 0 [ 6.067287] pxa2xx-spi 80860F0E:00: registered child spi-WM510205:00 would be nice to hear some feedback or comments about this patch. Notice: Thats my first attempt to make the codec chip via ACPI working on Linux systems and some other patches are needed to let the codec work together with the intel_sst_acpi driver. Currently I am trying to write the also needed machine driver which binds to this codec chip. But I can need help from others too, cause I am not so familiar with the Intel SST ACPI stuff and the baytrail code itself, nor with the arizona stuff... cheers Chris

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix link power unbalance at device removal
by Takashi Iwai 11 Jun '15

11 Jun '15

snd_hdac_link_power() has to be called after unregistering the codec device. Otherwise the device might be already runtime-suspended, thus the refcount goes under zero, triggering a warning like: WARNING: CPU: 7 PID: 2014 at sound/hda/hdac_i915.c:63 snd_hdac_display_power+0x106/0x120 [snd_hda_core]() CPU: 7 PID: 2014 Comm: modprobe Not tainted 4.1.0-rc7-test+ #1 Call Trace: [<ffffffff81697fe3>] dump_stack+0x4c/0x6e [<ffffffff810696da>] warn_slowpath_common+0x8a/0xc0 [<ffffffff810697ca>] warn_slowpath_null+0x1a/0x20 [<ffffffffa02dd526>] snd_hdac_display_power+0x106/0x120 [snd_hda_core] [<ffffffffa030b422>] azx_intel_link_power+0x12/0x20 [snd_hda_intel] [<ffffffffa037139f>] azx_link_power+0x1f/0x30 [snd_hda_codec] [<ffffffffa02d89fe>] snd_hdac_link_power+0x2e/0x40 [snd_hda_core] [<ffffffffa0368524>] snd_hda_codec_dev_free+0x34/0x50 [snd_hda_codec] [<ffffffffa0252061>] __snd_device_free+0x51/0xa0 [snd] ..... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d70229841156..497bd0f9dcda 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -886,8 +886,8 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - snd_hdac_link_power(&codec->core, false); snd_hdac_device_unregister(&codec->core); + snd_hdac_link_power(&codec->core, false); put_device(hda_codec_dev(codec)); return 0; } -- 2.4.3

1 0

[alsa-devel] [PATCH] ASoC: tas2552: Use devm_gpiod_get_optional for enable-gpio
by Axel Lin 11 Jun '15

11 Jun '15

commit ea178d1456dc ("ASoC: tas2552: Make the enable-gpio really optional") makes enable-gpio optional. devm_gpiod_get_optional() is the better function for optional gpio, so let's switch to use it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 169a703..4f25a7d 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -703,13 +703,10 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); - if (IS_ERR(data->enable_gpio)) { - if (PTR_ERR(data->enable_gpio) == -EPROBE_DEFER) - return -EPROBE_DEFER; - - data->enable_gpio = NULL; - } + data->enable_gpio = devm_gpiod_get_optional(dev, "enable", + GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); -- 2.1.0

3 2

[alsa-devel] [PATCH v4 1/3] ASoC: wm_adsp: Add codec_probe and codec_remove stubs
by Richard Fitzgerald 11 Jun '15

11 Jun '15

Currently the only init function in wm_adsp is called by the codec driver early in its probe before the codec has been registered with SOC. This patch adds stubs for the codec_probe and codec_remove stages and calls them from WM5102 and WM5110 codec drivers. This allows us to hang anything that needs setup during the codec probe stage off these functions without further modification of the codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 12 +++++++++++- sound/soc/codecs/wm_adsp.c | 12 ++++++++++++ sound/soc/codecs/wm_adsp.h | 2 ++ 4 files changed, 31 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 36a3921..785b3cf 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = wm_adsp2_codec_probe(&priv->core.adsp[0], codec); + if (ret) + return ret; + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1897,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c83b0a9..45d109a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; priv->core.arizona->dapm = &codec->dapm; @@ -1607,6 +1607,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + for (i = 0; i < WM5110_NUM_ADSP; ++i) { + ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); + if (ret) + return ret; + } + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) return ret; @@ -1621,6 +1627,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; ++i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b62ffd0..9fad2fd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1933,6 +1933,18 @@ err: } EXPORT_SYMBOL_GPL(wm_adsp2_event); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); + +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_remove); + int wm_adsp2_init(struct wm_adsp *dsp) { int ret; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..5584e34 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -79,6 +79,8 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 3

[alsa-devel] [PATCH 1/1] usb-audio: Set correct type for some UAC2 mixer controls.
by Johan Rastén 11 Jun '15

11 Jun '15

Changed ctl type for Input Gain Control and Input Gain Pad Control to USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0 definition. Signed-off-by: Johan Rastén <johan(a)oljud.se> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 8b7e391..6b3acba 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -809,12 +809,12 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Tone Control - Treble", USB_MIXER_S8 }, { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, + { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ { "Bass Boost", USB_MIXER_BOOLEAN }, { "Loudness", USB_MIXER_BOOLEAN }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_U16 }, - { "Input Gain Pad Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16 }, + { "Input Gain Pad Control", USB_MIXER_S16 }, { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; -- 2.1.4

3 3

[alsa-devel] [PATCH] ALSA: hda - Re-add the lost fake mute support
by Takashi Iwai 11 Jun '15

11 Jun '15

Yet another regression by the transition to regmap cache; for better usability, we had the fake mute control using the zero amp value for Conexant codecs, and this was forgotten in the new hda core code. Since the bits 4-7 are unused for the amp registers (as we follow the syntax of AMP_GET verb), the bit 4 is now used to indicate the fake mute. For setting this flag, snd_hda_codec_amp_update() becomes a function from a simple macro. The bonus is that it gained a proper function description. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 2 ++ sound/hda/hdac_regmap.c | 5 +++++ sound/pci/hda/hda_codec.c | 23 +++++++++++++++++++++++ sound/pci/hda/hda_local.h | 4 ++-- 4 files changed, 32 insertions(+), 2 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 53a18b3635e2..df705908480a 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -9,6 +9,8 @@ #include <sound/core.h> #include <sound/hdaudio.h> +#define AC_AMP_FAKE_MUTE 0x10 /* fake mute bit set to amp verbs */ + int snd_hdac_regmap_init(struct hdac_device *codec); void snd_hdac_regmap_exit(struct hdac_device *codec); int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index c4f5e61d4404..1eabcdf69457 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -246,6 +246,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) return hda_reg_read_stereo_amp(codec, reg, val); if (verb == AC_VERB_GET_PROC_COEF) return hda_reg_read_coef(codec, reg, val); + if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE) + reg &= ~AC_AMP_FAKE_MUTE; + err = snd_hdac_exec_verb(codec, reg, 0, val); if (err < 0) return err; @@ -283,6 +286,8 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: + if ((reg & AC_AMP_FAKE_MUTE) && (val & AC_AMP_MUTE)) + val = 0; verb = AC_VERB_SET_AMP_GAIN_MUTE; if (reg & AC_AMP_GET_LEFT) verb |= AC_AMP_SET_LEFT >> 8; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7782212dd64..3a3670168f12 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1376,6 +1376,29 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); /** + * snd_hda_codec_amp_update - update the AMP mono value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel to update (0 or 1) + * @dir: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP values for the given channel, direction and index. + */ +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, int mask, int val) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx); + + if (query_amp_caps(codec, nid, dir) & AC_AMPCAP_MIN_MUTE) + cmd |= AC_AMP_FAKE_MUTE; + return snd_hdac_regmap_update_raw(&codec->core, cmd, mask, val); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); + +/** * snd_hda_codec_amp_stereo - update the AMP stereo values * @codec: HD-audio codec * @nid: NID to read the AMP value diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3b567f42296b..bed66c314431 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -129,8 +129,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, /* lowlevel accessor with caching; use carefully */ #define snd_hda_codec_amp_read(codec, nid, ch, dir, idx) \ snd_hdac_regmap_get_amp(&(codec)->core, nid, ch, dir, idx) -#define snd_hda_codec_amp_update(codec, nid, ch, dir, idx, mask, val) \ - snd_hdac_regmap_update_amp(&(codec)->core, nid, ch, dir, idx, mask, val) +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, int mask, int val); int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, -- 2.4.3

1 0

[alsa-devel] [PATCH v8 0/3] HDA: add extended HDA
by Vinod Koul 11 Jun '15

11 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Changes in v8: - fixed some dead codec crept in - checkpatch cleanup Changes in v7: - Feedback from Takashi - removed the macros for register read/writes - add new macros for reg update - fix bus pointer macro - header cleanup on c/h files - new device name ehdaudioiDn - removed flags for ext caps, use pointer only - moved module info to patch 1 - sparse check on series - remove codec name in link Jeeja KP (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 129 ++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 116 +++++++++++ sound/hda/ext/hdac_ext_controller.c | 290 ++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 402 ++++++++++++++++++++++++++++++++++++ 7 files changed, 947 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

2 6

[alsa-devel] [PATCH v4 1/3] SoC: rt5645: make RT5650_TDM_CTRL_4 readable
by Oder Chiou 11 Jun '15

11 Jun '15

Register RT5650_TDM_CTRL_4(0x7A) is readable and used for mixer setting. It should be added in rt5645_readable_register function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d5f0f56..b86474a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -349,6 +349,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: case RT5645_TDM_CTRL_3: + case RT5650_TDM_CTRL_4: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] [patch] ASoC: fsl: add an error check
by Dan Carpenter 10 Jun '15

10 Jun '15

My static checker complains that: sound/soc/fsl/imx-wm8962.c:196 imx_wm8962_probe() warn: we tested 'ret' before and it was 'false' The intent was that we use "ret" to check imx_audmux_v2_configure_port(). Fixes: 8de2ae2a7f1f ('ASoC: fsl: add imx-wm8962 machine driver') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I assume checking for an error here will not change anything but I have not tested this patch. diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index cd146d4..b38b98c 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -190,7 +190,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) {

3 3

[alsa-devel] [PATCH v3 1/3] ASoC: wm_adsp: Add code_probe and codec_remove stubs
by Richard Fitzgerald 10 Jun '15

10 Jun '15

Currently the only init function in wm_adsp is called by the codec driver early in its probe before the codec has been registered with SOC. This patch adds stubs for the codec_probe and codec_remove stages and calls them from WM5102 and WM5110 codec drivers. This allows us to hang anything that needs setup during the codec probe stage off these functions without further modification of the codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 12 +++++++++++- sound/soc/codecs/wm_adsp.c | 12 ++++++++++++ sound/soc/codecs/wm_adsp.h | 2 ++ 4 files changed, 31 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 11eba0e..341d96e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = wm_adsp2_codec_probe(&priv->core.adsp[0], codec); + if (ret) + return ret; + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1897,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d65364e..6e15d9c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; priv->core.arizona->dapm = &codec->dapm; @@ -1607,6 +1607,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + for (i = 0; i < WM5110_NUM_ADSP; ++i) { + ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); + if (ret) + return ret; + } + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) return ret; @@ -1621,6 +1627,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; ++i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6e358b5..490edb3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,6 +1906,18 @@ err: } EXPORT_SYMBOL_GPL(wm_adsp2_event); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); + +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_remove); + int wm_adsp2_init(struct wm_adsp *dsp) { int ret; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..5584e34 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -79,6 +79,8 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 3

[alsa-devel] [PATCH 1/3 RESEND] ASoC: rt5645: make RT5650_TDM_CTRL_4 readable
by Oder Chiou 10 Jun '15

10 Jun '15

Register RT5650_TDM_CTRL_4(0x7A) is readable and used for mixer setting. It should be added in rt5645_readable_register function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d5f0f56..b86474a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -349,6 +349,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: case RT5645_TDM_CTRL_3: + case RT5650_TDM_CTRL_4: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Lock mutex in rt5645_enable_push_button_irq
by Nicolas Boichat 10 Jun '15

10 Jun '15

rt5645_enable_push_button_irq uses snd_soc_dapm_*_unlocked functions, so it needs to lock the required dapm mutex. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d5f0f56..025cf14 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2761,6 +2761,7 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); if (enable) { + snd_soc_dapm_mutex_lock(&codec->dapm); snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, "ADC L power"); snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, @@ -2770,6 +2771,8 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_dapm_mutex_unlock(&codec->dapm); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x8); snd_soc_update_bits(codec, @@ -2780,6 +2783,8 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, } else { snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + + snd_soc_dapm_mutex_lock(&codec->dapm); snd_soc_dapm_disable_pin_unlocked(&codec->dapm, "ADC L power"); snd_soc_dapm_disable_pin_unlocked(&codec->dapm, @@ -2790,6 +2795,7 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, snd_soc_dapm_disable_pin_unlocked(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_dapm_mutex_unlock(&codec->dapm); } } -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH v6 0/2] ASoC: qcom: add support to apq8016 sbc machine driver
by Srinivas Kandagatla 10 Jun '15

10 Jun '15

This patchset adds sound card support to APQ8016 SBC board aka DB410c. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing and various board specifics are controlled by 2 mux registers. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC, msm8916-mtp boards. Changes since v5: - converted if-else to switch case as suggested by Mark. - moved usage of startup() to dai link init() as suggested by Mark. - replaced "external" boolean property with "link-name" property as suggested by Mark. Changes since v4 (https://lkml.org/lkml/2015/5/22/628) - moved all the dt parsing to single function, spotted by Mark. - removed unnecessary setting of card->dev = NULL spotted by Mark. - renamed the dai links to the correct codec names. - removed the special case for EPROBE_DEFER in probe, spotted by Mark. - Dropped all the Changes since comments as most of the patches in last series are merged to topic/qcom branch. --srini Srinivas Kandagatla (2): ASoC: qcom: document apq8016 sbc machine driver bindings ASoC: qcom: add apq8016 sound card support .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 60 +++++++ sound/soc/qcom/Kconfig | 9 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/apq8016_sbc.c | 198 +++++++++++++++++++++ 4 files changed, 269 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt create mode 100644 sound/soc/qcom/apq8016_sbc.c -- 1.9.1

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[alsa-devel] [PATCH - alsa-lib 1/1] Introduce snd_seq_client_info_get_card_number, for getting the card number of a seq client on recent kernels
by Adam Goode 10 Jun '15

10 Jun '15

Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/configure.ac b/configure.ac index 9621d4e..01bb0fb 100644 --- a/configure.ac +++ b/configure.ac @@ -14,7 +14,7 @@ dnl remove API = c+1:0:0 dnl ************************************************* AC_CANONICAL_HOST AM_INIT_AUTOMAKE -eval LIBTOOL_VERSION_INFO="2:0:0" +eval LIBTOOL_VERSION_INFO="3:0:1" dnl ************************************************* AM_CONDITIONAL([INSTALL_M4], [test -n "${ACLOCAL}"]) diff --git a/include/seq.h b/include/seq.h index 9576822..e0f9b78 100644 --- a/include/seq.h +++ b/include/seq.h @@ -146,6 +146,7 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card_number(const snd_seq_client_info_t *info); void snd_seq_client_info_set_client(snd_seq_client_info_t *info, int client); void snd_seq_client_info_set_name(snd_seq_client_info_t *info, const char *name); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..c2a659b 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -22,9 +22,10 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include <sound/asound.h> /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +357,8 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card_number; /* RO: card number, or -1 if no card. Added in protocol version 1.0.2 */ + char reserved[60]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..bc39046 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1651,6 +1651,18 @@ int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info) } /** + * \brief Get the card number of a client_info container + * \param info client_info container + * \return the card number, -1 if no card associated with this client, or -ENOSYS if the + * kernel does not support reporting this field + */ +int snd_seq_client_info_get_card_number(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card_number; +} + +/** * \brief Set the client id of a client_info container * \param info client_info container * \param client client id diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..8502d63 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + /* The card_number field was added in protocol 1.0.2, so set to -ENOSYS if older. */ + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card_number = -ENOSYS; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + /* The card_number field was added in protocol 1.0.2, so set to -ENOSYS if older. */ + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card_number = -ENOSYS; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->version = ver; if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { diff --git a/test/seq.c b/test/seq.c index 34b000f..a4e5f1f 100644 --- a/test/seq.c +++ b/test/seq.c @@ -111,7 +111,7 @@ void show_port_info(snd_seq_t *handle, int client, int port) void show_client_info(snd_seq_t *handle, int client) { - int err, idx, min, max; + int err, idx, min, max, card_number; snd_seq_client_info_t *info; snd_seq_client_info_alloca(&info); @@ -124,11 +124,14 @@ void show_client_info(snd_seq_t *handle, int client) fprintf(stderr, "Client %i info error: %s\n", idx, snd_strerror(err)); exit(0); } + card_number = snd_seq_client_info_get_card_number(info); printf("Client %i info\n", idx); if (verbose) printf(" Client : %i\n", snd_seq_client_info_get_client(info)); printf(" Type : %s\n", snd_seq_client_info_get_type(info) == SND_SEQ_KERNEL_CLIENT ? "kernel" : "user"); printf(" Name : %s\n", snd_seq_client_info_get_name(info)); + if (card_number != -ENOSYS) + printf(" Card Number : %i\n", card_number); } } -- 2.2.0.rc0.207.ga3a616c

3 3

[alsa-devel] [PATCH v3 0/2] Update ALSA driver to use media controller API
by Shuah Khan 10 Jun '15

10 Jun '15

This patch series updates ALSA driver to use media controller API to share tuner with DVB and V4L2 drivers that control AU0828 media device. Two new interfaces are added to media controller API to enable creating media device as a device resource. This allows creating media device as a device resource on the main struct device that is common to all drivers that control a single media hardware and share resources on it. Drivers then can find the common media device to register entities and manage links, and pipelines. Tested with and without CONFIG_MEDIA_CONTROLLER enabled. Tested tuner entity doesn't exist case as au0828 v4l2 driver is the one that will create the tuner when it gets updated to use media controller API. Please note that au0828 updates media controller are necessary before the resource sharing can work across ALSA and au0828 dvb and v4l2 drivers. This work is in progress and another developer is working on it. Changes since v1: - Moved media specific code into new header and source file - Created a new structure for media controller specific fields - Added a new define USE_MEDIA_CONTROLLER to add media specific code. This new define is defined only when Media Controller support is enabled. New media specific code is active only when this define is defined. - Moved media_deviced_delete() check under !was_shutdown block - Removed return check for media_device_init() - Changed new Media Controller interface names for clarity. Now they are named media_device_get_devres() and media_device_find_devres(). - Calling media_entity_pipeline_stop() on a pipe that isn't active appears to be safe. This needs further testing once AU0828 media controller patches are ready. Changes since v2: - Moved checks to define USE_MEDIA_CONTROLLER to Makefile - Moved stubs for media controller specific functions to header file. - Added new register_notify operation to media_entity_operations structure. This new hook is called when a new entity is registered to allow other entities associated with the media device to create link as needed. - Added a new field to struct media_entity for entity owners to save their private data. - Is it safe for ALSA to call media_entity_remove_links()? Laurent Pinchart, please comment. - Do start/stop pipeline functions have to be concered about V4l2? Are they safe the way they are? Hans, please comment. Shuah Khan (2): media: media controller entity framework enhancements for ALSA sound/usb: Update ALSA driver to use media controller API drivers/media/media-device.c | 7 ++ include/media/media-entity.h | 4 + sound/usb/Makefile | 15 +++- sound/usb/card.c | 5 ++ sound/usb/card.h | 1 + sound/usb/media.c | 185 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 47 +++++++++++ sound/usb/pcm.c | 10 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 ++- sound/usb/stream.c | 3 + sound/usb/usbaudio.h | 1 + 12 files changed, 285 insertions(+), 3 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

1 2

Re: [alsa-devel] [PATCH v7 0/3] HDA: add extended HDA
by Vinod Koul 10 Jun '15

10 Jun '15

On Wed, Jun 10, 2015 at 04:56:28PM +0530, Subhransu S. Prusty wrote: > There is an error in this series. Vinod will resend shortly. > Yes some dead code crept in and checkpatch wasnt too happy with few lines, so I have updated this and resent the fixed series. Sorry about the noise -- ~Vinod > On Wed, Jun 10, 2015 at 05:03:33PM +0530, Vinod Koul wrote: > > New HDA controllers from Intel have extended HDA capabilities like multilink, > > pipe processing, SPIB, GTS etc In order to use them we create an extended > > HDA bus, controller and stream which uses the extended configuration > > > > Changes in v7: > > - Feedback from Takashi > > - removed the macros for register read/writes > > - add new macros for reg update > > - fix bus pointer macro > > - header cleanup on c/h files > > - new device name ehdaudioiDn > > - removed flags for ext caps, use pointer only > > - moved module info to patch 1 > > - sparse check on series > > - remove codec name in link > > > > Jeeja KP (3): > > ALSA: hdac_ext: add extended HDA bus > > ALSA: hdac_ext: add hdac extended controller > > ALSA: hdac_ext: add extended stream capabilities > > > > include/sound/hdaudio_ext.h | 129 ++++++++++++ > > sound/hda/Kconfig | 4 + > > sound/hda/Makefile | 3 + > > sound/hda/ext/Makefile | 3 + > > sound/hda/ext/hdac_ext_bus.c | 116 +++++++++++ > > sound/hda/ext/hdac_ext_controller.c | 297 ++++++++++++++++++++++++++ > > sound/hda/ext/hdac_ext_stream.c | 402 ++++++++++++++++++++++++++++++++++++ > > 7 files changed, 954 insertions(+) > > create mode 100644 include/sound/hdaudio_ext.h > > create mode 100644 sound/hda/ext/Makefile > > create mode 100644 sound/hda/ext/hdac_ext_bus.c > > create mode 100644 sound/hda/ext/hdac_ext_controller.c > > create mode 100644 sound/hda/ext/hdac_ext_stream.c > > > > -- > > 1.9.1 > > > > -- --

1 0

[alsa-devel] [PATCH v7 0/3] HDA: add extended HDA
by Vinod Koul 10 Jun '15

10 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Changes in v7: - Feedback from Takashi - removed the macros for register read/writes - add new macros for reg update - fix bus pointer macro - header cleanup on c/h files - new device name ehdaudioiDn - removed flags for ext caps, use pointer only - moved module info to patch 1 - sparse check on series - remove codec name in link Jeeja KP (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 129 ++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 116 +++++++++++ sound/hda/ext/hdac_ext_controller.c | 297 ++++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 402 ++++++++++++++++++++++++++++++++++++ 7 files changed, 954 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

2 4

[alsa-devel] [PATCH] ALSA: hda - PCH HDA controller not controlled by i915 power
by libin.yang＠intel.com 10 Jun '15

10 Jun '15

From: Libin Yang <libin.yang(a)intel.com> On some Intel platforms, the HDMI codec is connected to PCH HDA controller. In this case, AZX_DCAPS_I915_POWERWELL is set and the i915 power well failure should not block the hda controller initialization. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/hda/hdac_i915.c | 6 +++--- sound/pci/hda/hda_intel.c | 11 +++++++++-- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index cb78c25..4f4036a 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -27,7 +27,7 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { struct i915_audio_component *acomp = bus->audio_component; - if (!acomp->ops) + if (!acomp || !acomp->ops) return -ENODEV; if (!acomp->ops->codec_wake_override) { @@ -49,7 +49,7 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { struct i915_audio_component *acomp = bus->audio_component; - if (!acomp->ops) + if (!acomp || !acomp->ops) return -ENODEV; dev_dbg(bus->dev, "display power %s\n", @@ -72,7 +72,7 @@ int snd_hdac_get_display_clk(struct hdac_bus *bus) { struct i915_audio_component *acomp = bus->audio_component; - if (!acomp->ops) + if (!acomp || !acomp->ops) return -ENODEV; return acomp->ops->get_cdclk_freq(acomp->dev); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7933fb0..775e020 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1986,14 +1986,21 @@ static int azx_probe_continue(struct azx *chip) hda->need_i915_power = 1; err = snd_hdac_i915_init(bus); + /* if controller is located in powerwell, + * the initialization should be stopped. + * if only codec is located in powerwell, + * the hd audio is still ok for analog audio + */ if (err < 0) - goto i915_power_fail; + if (hda->need_i915_power) + goto i915_power_fail; err = snd_hdac_display_power(bus, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); - goto i915_power_fail; + if (hda->need_i915_power) + goto i915_power_fail; } } -- 1.9.1

4 9

[alsa-devel] [PATCH] control: Allow cset'ing specific values in the multi-value case
by arun＠accosted.net 10 Jun '15

10 Jun '15

From: Arun Raghavan <git(a)arunraghavan.net> This works with syntax like ... amixer cset 'IIR1 Band1' ,,200 ... to set the third value of the control to 200. --- src/control/ctlparse.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 8d6c385..64c6480 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -48,9 +48,9 @@ static inline long int convert_prange1(long val, long min, long max) #define check_range(val, min, max) \ ((val < min) ? (min) : ((val > max) ? (max) : (val))) -static long get_integer(const char **ptr, long min, long max) +static long get_integer(const char **ptr, long def, long min, long max) { - long val = min; + long val = def; char *p = (char *)*ptr, *s; if (*p == ':') @@ -351,6 +351,8 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, break; case SND_CTL_ELEM_TYPE_INTEGER: tmp = get_integer(&ptr, + snd_ctl_elem_value_get_integer(dst, + idx), snd_ctl_elem_info_get_min(info), snd_ctl_elem_info_get_max(info)); snd_ctl_elem_value_set_integer(dst, idx, tmp); @@ -364,12 +366,12 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, case SND_CTL_ELEM_TYPE_ENUMERATED: tmp = get_ctl_enum_item_index(handle, info, &ptr); if (tmp < 0) - tmp = get_integer(&ptr, 0, + tmp = get_integer(&ptr, 0, 0, snd_ctl_elem_info_get_items(info) - 1); snd_ctl_elem_value_set_enumerated(dst, idx, tmp); break; case SND_CTL_ELEM_TYPE_BYTES: - tmp = get_integer(&ptr, 0, 255); + tmp = get_integer(&ptr, 0, 0, 255); snd_ctl_elem_value_set_byte(dst, idx, tmp); break; default: -- 2.4.2

3 3

[alsa-devel] [PATCH] ALSA: hda - Don't actually write registers for caps overwrites
by Takashi Iwai 10 Jun '15

10 Jun '15

Along with the transition to regmap for managing the cached parameter reads, the caps overwrite was also moved to regmap cache. The cache change itself works, but it still tries to write the non-existing verb (the HDA parameter is read-only) wrongly. It's harmless in most cases, but some chips are picky and may result in the codec communication stall. This patch avoids it just by adding the missing flag check in reg_write ops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 7371e0c3926f..c4f5e61d4404 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -265,6 +265,9 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; + if (codec->caps_overwriting) + return 0; + reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); verb = get_verb(reg); -- 2.4.3

1 0

[alsa-devel] OPL3 on cmipci doesn't survive suspend/resume
by Adam Goode 10 Jun '15

10 Jun '15

Hi, I have this card: 05:00.0 Multimedia audio controller: C-Media Electronics Inc CMI8738/CMI8768 PCI Audio (rev 10) Subsystem: C-Media Electronics Inc CMI8738/C3DX PCI Audio Device Flags: bus master, stepping, medium devsel, latency 32, IRQ 19 I/O ports at d000 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: snd_cmipci Kernel modules: snd_cmipci When I suspend/resume, the sound from the OPL3 is permanently distorted. It sounds like not all of the operators are active. I can fix this by doing a full PCI remove/rescan cycle: the driver correctly initializes the card at this time. Any ideas before I peek into the code? Thanks, Adam

3 6

Re: [alsa-devel] [pulseaudio-discuss] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-5)
by Raymond Yau 10 Jun '15

10 Jun '15

>> >> Some error messages can be reproduced by set default-sample-format to s24le >> >> But cannot reproduce SNDRV_PCM_IOCTL_HW_PARAMS >> >> Any error meesage in system log , it seem snd_usb_pcm_hw_params return error with your usb sound car >> > The only thing related to sound card found in log, last 3 lines are multiple times repeated afterwards: > > [ 1306.010420] usb 1-6: new full-speed USB device number 23 using xhci_hcd > [ 1306.177219] usb 1-6: New USB device found, idVendor=041e, idProduct=322c > [ 1306.177222] usb 1-6: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 1306.177223] usb 1-6: Product: SB Omni Surround 5.1 > [ 1306.177225] usb 1-6: Manufacturer: Creative Technology Ltd > [ 1306.177226] usb 1-6: SerialNumber: 000000Q6 > [ 1306.177381] usb 1-6: ep 0x83 - rounding interval to 64 microframes, ep desc says 80 microframes > [ 1306.208636] input: Creative Technology Ltd SB Omni Surround 5.1 as /devices/pci0000:00/0000:00:14.0/usb1/1-6/1-6:1.4/0003:041E:322C.0013/input/input42 > [ 1306.259885] hid-generic 0003:041E:322C.0013: input,hidraw0: USB HID v1.11 Keyboard [Creative Technology Ltd SB Omni Surround 5.1] on usb-0000:00:14.0-6/input4 > [ 1306.292463] usb 1-6: Not enough bandwidth for new device state. > [ 1306.292470] usb 1-6: Not enough bandwidth for altsetting 2 > [ 1306.292472] usb 1-6: 3:2: usb_set_interface failed (-28) > > I have no idea why is that the case, device is USB 2.0, even disabling all other external USB devices and pluggin it into USB 3.0 doesn't help. > Nothing more in log. > Can "not enough bandwidth" or "usb_set_interface failed (-28)" message repoduced by using aplay or arecord with s24le/s16le at different rate instead of using pulseaudio ?

1 0

[alsa-devel] wm8731 - redundant register initialization, stream name vs DAPM widgets, snd_soc_io to regmap_io conversion
by Sergey Kiselev 09 Jun '15

09 Jun '15

Hi, I have a few questions regarding wm8731. Some might be applicable to other codecs as well. 1. According to the datasheet bit #8 of all the registers below is set to '0' by the hardware reset. Is there a good reason to set them anyway? (Perhaps an errata in previous chip version?) /* Latch the update bits */ snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); 2. In wm8731's snd_soc_dai_driver structure the steam names are 'Playback' and 'Capture', while DAPM widgets have 'HiFi' prepended to the names: SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1), Should the name match? DAPM.txt says so, but the DAPM code actually looks for a substring match (DAI name as a substring in widget name). 3. Do I understand correctly the direction is to move from snd_soc_${io} to more generic regmap_${io}? If so, I'll submit a patch. Thanks -- Sergey

4 3

[alsa-devel] [PATCH v5 0/2] ASoC: qcom: add support to apq8016 sbc machine driver
by Srinivas Kandagatla 09 Jun '15

09 Jun '15

This patchset adds sound card support to APQ8016 SBC board aka DB410c. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing and various board specifics are controlled by 2 mux registers. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC, msm8916-mtp boards. Changes since v4 (https://lkml.org/lkml/2015/5/22/628) - moved all the dt parsing to single function, spotted by Mark. - removed unnecessary setting of card->dev = NULL spotted by Mark. - renamed the dai links to the correct codec names. - removed the special case for EPROBE_DEFER in probe, spotted by Mark. - Dropped all the Changes since comments as most of the patches in last series are merged to topic/qcom branch. --srini Srinivas Kandagatla (2): ASoC: qcom: document apq8016 sbc machine driver bindings ASoC: qcom: add apq8016 sound card support .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 61 ++++++ sound/soc/qcom/Kconfig | 9 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/apq8016_sbc.c | 208 +++++++++++++++++++++ 4 files changed, 280 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt create mode 100644 sound/soc/qcom/apq8016_sbc.c -- 1.9.1

2 10

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller
by Jyri Sarha 09 Jun '15

09 Jun '15

Find the configured DMA controller by asking for a DMA channel in the probe phase and releasing it right after. The controller device can be found via the dma_chan struct and the controller is recognized from the compatible property of its device node. The patch assumes EDMA if there is no device node. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Changes since v2: - Print only one error message if DMA "probing" fails and none in case of -EPROBE_DEFER Changes since first patch version: - Change enum names from MCASP_?DMA to PCM_?DMA - Return PCM_EDMA if !mcasp->dev->of_node - Do not ignore possible error code returned by davinci_mcasp_get_dma_type() sound/soc/davinci/davinci-mcasp.c | 70 +++++++++++++++++++++++++++++++++------ 1 file changed, 60 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d793494..a73cd57 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1565,6 +1565,49 @@ static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp) return ret; } +enum { + PCM_EDMA, + PCM_SDMA, +}; +static const char *sdma_prefix = "ti,omap"; + +static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) +{ + struct dma_chan *chan; + const char *tmp; + int ret = PCM_EDMA; + + if (!mcasp->dev->of_node) + return PCM_EDMA; + + tmp = mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data; + chan = dma_request_slave_channel_reason(mcasp->dev, tmp); + if (IS_ERR(chan)) { + if (PTR_ERR(chan) != -EPROBE_DEFER) + dev_err(mcasp->dev, + "Can't verify DMA configuration (%ld)\n", + PTR_ERR(chan)); + return PTR_ERR(chan); + } + BUG_ON(!chan->device || !chan->device->dev); + + if (chan->device->dev->of_node) + ret = of_property_read_string(chan->device->dev->of_node, + "compatible", &tmp); + else + dev_dbg(mcasp->dev, "DMA controller has no of-node\n"); + + dma_release_channel(chan); + if (ret) + return ret; + + dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp); + if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix))) + return PCM_SDMA; + + return PCM_EDMA; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1763,27 +1806,34 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err; - switch (mcasp->version) { + ret = davinci_mcasp_get_dma_type(mcasp); + switch (ret) { + case PCM_EDMA: #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); + ret = -EINVAL; + goto err; #endif + break; + case PCM_SDMA: #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_OMAP_SOC)) - case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); + ret = -EINVAL; + goto err; #endif + break; default: - dev_err(&pdev->dev, "Invalid McASP version: %d\n", - mcasp->version); - ret = -EINVAL; + dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + case -EPROBE_DEFER: + goto err; break; } -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rx51: Automatically disconnect non-connected pins
by Lars-Peter Clausen 09 Jun '15

09 Jun '15

According to the schematics the external speaker is differential, so the HPLCOM and HPRCOM pins of the CODEC should be connected to it. Otherwise the routing looks complete, so add the missing routes and set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/rx51.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index fded993..3bebfb1 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -245,6 +245,8 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Ext Spk", NULL, "HPLCOM"}, + {"Ext Spk", NULL, "HPRCOM"}, {"Headphone Jack", NULL, "LLOUT"}, {"Headphone Jack", NULL, "RLOUT"}, {"FM Transmitter", NULL, "LLOUT"}, @@ -288,15 +290,8 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card); - - struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Set up NC codec pins */ - snd_soc_dapm_nc_pin(dapm, "MIC3L"); - snd_soc_dapm_nc_pin(dapm, "MIC3R"); - snd_soc_dapm_nc_pin(dapm, "LINE1R"); - err = tpa6130a2_add_controls(codec); if (err < 0) { dev_err(card->dev, "Failed to add TPA6130A2 controls\n"); @@ -383,6 +378,7 @@ static struct snd_soc_card rx51_sound_card = { .num_aux_devs = ARRAY_SIZE(rx51_aux_dev), .codec_conf = rx51_codec_conf, .num_configs = ARRAY_SIZE(rx51_codec_conf), + .fully_routed = true, .controls = aic34_rx51_controls, .num_controls = ARRAY_SIZE(aic34_rx51_controls), -- 2.1.4

4 3

[alsa-devel] [PATCH v2 1/2] ASoC: wm_adsp: Add code_probe and codec_remove stubs
by Richard Fitzgerald 09 Jun '15

09 Jun '15

Currently the only init function in wm_adsp is called by the codec driver early in its probe before the codec has been registered with SOC. This patch adds stubs for the codec_probe and codec_remove stages and calls them from WM5102 and WM5110 codec drivers. This allows us to hang anything that needs setup during the codec probe stage off these functions without further modification of the codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 12 +++++++++++- sound/soc/codecs/wm_adsp.c | 12 ++++++++++++ sound/soc/codecs/wm_adsp.h | 2 ++ 4 files changed, 31 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 11eba0e..341d96e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = wm_adsp2_codec_probe(&priv->core.adsp[0], codec); + if (ret) + return ret; + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1897,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d65364e..6e15d9c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; priv->core.arizona->dapm = &codec->dapm; @@ -1607,6 +1607,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + for (i = 0; i < WM5110_NUM_ADSP; ++i) { + ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); + if (ret) + return ret; + } + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) return ret; @@ -1621,6 +1627,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; ++i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6e358b5..490edb3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,6 +1906,18 @@ err: } EXPORT_SYMBOL_GPL(wm_adsp2_event); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); + +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_remove); + int wm_adsp2_init(struct wm_adsp *dsp) { int ret; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..5584e34 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -79,6 +79,8 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 6

[alsa-devel] [PATCH 1/2] ASoC: qcom: storm: allocate snd_soc_card struct dynamically.
by Srinivas Kandagatla 09 Jun '15

09 Jun '15

This patch moves static allocation of snd_soc_card to dynamic allocation, the reason to do this is to avoid holding up any dangling pointers in this static structures. And I see no use for having this struct as static given that the card->name is also populated dynamically from dt. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/storm.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index b8bd296..ee06613 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -69,11 +69,6 @@ static struct snd_soc_dai_link storm_dai_link = { .ops = &storm_soc_ops, }; -static struct snd_soc_card storm_soc_card = { - .name = "ipq806x-storm", - .dev = NULL, -}; - static int storm_parse_of(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link = card->dai_link; @@ -99,14 +94,13 @@ static int storm_parse_of(struct snd_soc_card *card) static int storm_platform_probe(struct platform_device *pdev) { - struct snd_soc_card *card = &storm_soc_card; + struct snd_soc_card *card; int ret; - if (card->dev) { - dev_err(&pdev->dev, "%s() error, existing soundcard\n", - __func__); - return -ENODEV; - } + card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); @@ -129,7 +123,6 @@ static int storm_platform_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret == -EPROBE_DEFER) { - card->dev = NULL; return ret; } else if (ret) { dev_err(&pdev->dev, "%s() error registering soundcard: %d\n", -- 1.9.1

2 2

[alsa-devel] Softvol only after device has opened
by Ricard Wanderlof 09 Jun '15

09 Jun '15

I'm working on implementing support for a MEMS microphone with an I2S connection. Since the microphone has no controls but simply outputs I2S data from the word go, this means that any gain must be applied later. (Microphones of this type are generally specified at outputting 0 dB FS at some rediculously high SPL, 120 dB SPL being typical). Since we use 16 bit audio in our (SOC) system, I'm trying to apply gain using the softvol ALSA plugin, before converting to 16 bit audio (the microphone outputs 24 bit audio). One thing though, the 'softvol' plugin very correctly creates a mixer control and it all works fine, but only after the device has been opened. Before that the mixer control doesn't seem to exist, which means that we cannot adjust the gain until the device is opened by some application. Is this intentional, a bug, or something that is misconfigured on my part? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 10

[alsa-devel] [PATCH] ASoC: wm_adsp: Add basic debugfs entries
by Richard Fitzgerald 09 Jun '15

09 Jun '15

This patch adds some debugfs nodes to get information about the currently running firmware. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 4 + sound/soc/codecs/wm5110.c | 9 ++- sound/soc/codecs/wm_adsp.c | 182 ++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm_adsp.h | 25 ++++++- 4 files changed, 212 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 11eba0e..22fc9e9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,8 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + wm_adsp_init_debugfs(&priv->core.adsp[0], codec); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1895,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp_cleanup_debugfs(&priv->core.adsp[0]); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d65364e..f9ca1a9 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,10 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; + + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp_init_debugfs(&priv->core.adsp[i], codec); priv->core.arizona->dapm = &codec->dapm; @@ -1621,6 +1624,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp_cleanup_debugfs(&priv->core.adsp[i]); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6e358b5..c8def3ea 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -23,6 +23,7 @@ #include <linux/slab.h> #include <linux/vmalloc.h> #include <linux/workqueue.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -243,6 +244,26 @@ struct wm_coeff_ctl { unsigned int flags; }; +#ifdef CONFIG_DEBUG_FS +static void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, const char *s); +static void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, const char *s); +static void wm_adsp_debugfs_clear(struct wm_adsp *dsp); +#else +static inline void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, + const char *s) +{ +} + +static inline void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, + const char *s) +{ +} + +static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) +{ +} +#endif + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1104,6 +1125,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); + wm_adsp_debugfs_save_wmfwname(dsp, file); + out_fw: regmap_async_complete(regmap); wm_adsp_buf_free(&buf_list); @@ -1218,11 +1241,12 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) n_algs = be32_to_cpu(adsp1_id.n_algs); dsp->fw_id = be32_to_cpu(adsp1_id.fw.id); + dsp->fw_id_version = be32_to_cpu(adsp1_id.fw.ver); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", dsp->fw_id, - (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, - be32_to_cpu(adsp1_id.fw.ver) & 0xff, + (dsp->fw_id_version & 0xff0000) >> 16, + (dsp->fw_id_version & 0xff00) >> 8, + dsp->fw_id_version & 0xff, n_algs); alg_region = wm_adsp_create_region(dsp, WMFW_ADSP1_ZM, @@ -1321,11 +1345,12 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) n_algs = be32_to_cpu(adsp2_id.n_algs); dsp->fw_id = be32_to_cpu(adsp2_id.fw.id); + dsp->fw_id_version = be32_to_cpu(adsp2_id.fw.ver); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", dsp->fw_id, - (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, - be32_to_cpu(adsp2_id.fw.ver) & 0xff, + (dsp->fw_id_version & 0xff0000) >> 16, + (dsp->fw_id_version & 0xff00) >> 8, + dsp->fw_id_version & 0xff, n_algs); alg_region = wm_adsp_create_region(dsp, WMFW_ADSP2_XM, @@ -1601,6 +1626,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); + wm_adsp_debugfs_save_binname(dsp, file); + out_fw: regmap_async_complete(regmap); release_firmware(firmware); @@ -1869,6 +1896,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + wm_adsp_debugfs_clear(dsp); + + dsp->fw_id = 0; + dsp->fw_id_version = 0; dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, @@ -1929,4 +1960,143 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +#ifdef CONFIG_DEBUG_FS +static void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, const char *s) +{ + kfree(dsp->wmfw_file_name); + dsp->wmfw_file_name = kstrdup(s, GFP_KERNEL); +} + +static void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, const char *s) +{ + kfree(dsp->bin_file_name); + dsp->bin_file_name = kstrdup(s, GFP_KERNEL); +} + +static void wm_adsp_debugfs_clear(struct wm_adsp *dsp) +{ + kfree(dsp->wmfw_file_name); + kfree(dsp->bin_file_name); + dsp->wmfw_file_name = NULL; + dsp->bin_file_name = NULL; +} + +static ssize_t wm_adsp_debugfs_string_read(struct wm_adsp *dsp, + char __user *user_buf, + size_t count, loff_t *ppos, + const char *string) +{ + char *temp; + int len; + ssize_t ret; + + if (!string || !dsp->running) + return 0; + + temp = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!temp) + return -ENOMEM; + + len = snprintf(temp, PAGE_SIZE, "%s\n", string); + ret = simple_read_from_buffer(user_buf, count, ppos, temp, len); + kfree(temp); + return ret; +} + +static ssize_t wm_adsp_debugfs_wmfw_read(struct file *file, + char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct wm_adsp *dsp = file->private_data; + + return wm_adsp_debugfs_string_read(dsp, user_buf, count, ppos, + dsp->wmfw_file_name); +} + +static ssize_t wm_adsp_debugfs_bin_read(struct file *file, + char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct wm_adsp *dsp = file->private_data; + + return wm_adsp_debugfs_string_read(dsp, user_buf, count, ppos, + dsp->bin_file_name); +} + +static const struct { + const char *name; + const struct file_operations fops; +} wm_adsp_debugfs_fops[] = { + { + .name = "wmfw_file", + .fops = { + .open = simple_open, + .read = wm_adsp_debugfs_wmfw_read, + }, + }, + { + .name = "bin_file", + .fops = { + .open = simple_open, + .read = wm_adsp_debugfs_bin_read, + }, + }, +}; + +void wm_adsp_init_debugfs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + struct dentry *root = NULL; + char *root_name; + int i; + + if (!codec->component.debugfs_root) { + adsp_err(dsp, "No codec debugfs root\n"); + goto err; + } + + root_name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!root_name) + goto err; + + snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num); + root = debugfs_create_dir(root_name, codec->component.debugfs_root); + kfree(root_name); + + if (!root) + goto err; + + if (!debugfs_create_bool("running", S_IRUGO, root, &dsp->running)) + goto err; + + if (!debugfs_create_x32("fw_id", S_IRUGO, root, &dsp->fw_id)) + goto err; + + if (!debugfs_create_x32("fw_version", S_IRUGO, root, + &dsp->fw_id_version)) + goto err; + + for (i = 0; i < ARRAY_SIZE(wm_adsp_debugfs_fops); ++i) { + if (!debugfs_create_file(wm_adsp_debugfs_fops[i].name, + S_IRUGO, root, dsp, + &wm_adsp_debugfs_fops[i].fops)) + goto err; + } + + dsp->debugfs_root = root; + return; + +err: + debugfs_remove_recursive(root); + adsp_err(dsp, "Failed to create debugfs\n"); +} +EXPORT_SYMBOL_GPL(wm_adsp_init_debugfs); + + +void wm_adsp_cleanup_debugfs(struct wm_adsp *dsp) +{ + debugfs_remove_recursive(dsp->debugfs_root); +} +EXPORT_SYMBOL_GPL(wm_adsp_cleanup_debugfs); +#endif + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..1d563fe 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -46,17 +46,25 @@ struct wm_adsp { struct list_head alg_regions; int fw_id; + int fw_id_version; const struct wm_adsp_region *mem; int num_mems; int fw; int fw_ver; - bool running; + u32 running; struct list_head ctl_list; struct work_struct boot_work; + +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_root; + char *wmfw_file_name; + char *bin_file_name; +#endif + }; #define WM_ADSP1(wname, num) \ @@ -79,6 +87,21 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); + +#ifdef CONFIG_DEBUG_FS +void wm_adsp_init_debugfs(struct wm_adsp *dsp, struct snd_soc_codec *codec); +void wm_adsp_cleanup_debugfs(struct wm_adsp *dsp); +#else +static inline void wm_adsp_init_debugfs(struct wm_adsp *dsp, + struct snd_soc_codec *codec) +{ +} + +void wm_adsp_cleanup_debugfs(struct wm_adsp *dsp) +{ +} +#endif + int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 4

[alsa-devel] [PATCH/linux-next] Doc: sound:oss: Fix typo in sound/oss
by Masanari Iida 09 Jun '15

09 Jun '15

This patch fix some spelling typo found in Documentations/sound/oss. Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- Documentation/sound/oss/PSS-updates | 2 +- Documentation/sound/oss/README.OSS | 2 +- Documentation/sound/oss/btaudio | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/oss/PSS-updates b/Documentation/sound/oss/PSS-updates index c84dd75..11914a1 100644 --- a/Documentation/sound/oss/PSS-updates +++ b/Documentation/sound/oss/PSS-updates @@ -41,7 +41,7 @@ pss_no_sound This module parameter is a flag that can be used to tell the driver to just configure non-sound components. 0 configures all components, a non-0 -value will only attept to configure the CDROM and joystick ports. This +value will only attempt to configure the CDROM and joystick ports. This parameter can be used by a user who only wished to use the builtin joystick and/or CDROM port(s) of his PSS sound card. If this driver is loaded with this parameter and with the parameter below set to true then a user can safely unload diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS index 4be2594..a085ea3 100644 --- a/Documentation/sound/oss/README.OSS +++ b/Documentation/sound/oss/README.OSS @@ -1346,7 +1346,7 @@ implement nice real-time signal processing audio effect software and network telephones. The ACI mixer has to be switched into the "solo" mode for duplex operation in order to avoid feedback caused by the mixer (input hears output signal). You can de-/activate this mode -through toggleing the record button for the wave controller with an +through toggling the record button for the wave controller with an OSS-mixer. The PCM20 contains a radio tuner, which is also controlled by diff --git a/Documentation/sound/oss/btaudio b/Documentation/sound/oss/btaudio index 1a693e6..effdb9a 100644 --- a/Documentation/sound/oss/btaudio +++ b/Documentation/sound/oss/btaudio @@ -29,7 +29,7 @@ Driver Status Still somewhat experimental. The driver should work stable, i.e. it should'nt crash your box. It might not work as expected, have bugs, -not being fully OSS API compilant, ... +not being fully OSS API compliant, ... Latest versions are available from http://bytesex.org/bttv/, the driver is in the bttv tarball. Kernel patches might be available too, -- 2.4.2.387.gf86f31a

2 1

[alsa-devel] [PATCH] mfd: wm8994: Fix NULL pointer exception on missing pdata
by Inha Song 09 Jun '15

09 Jun '15

If we use the dev_get_platdata, pdata can be NULL in DT environments. Actually, NULL pointer exception is occurs with error messages like this: [ 1.226583] Unable to handle kernel NULL pointer dereference at virtual address 00000044 [ 1.467910] [<c02f1e38>] (wm8994_irq_init) from [<c02f1bd4>] (wm8994_i2c_probe+0x684/0x858) [ 1.476241] [<c02f1bd4>] (wm8994_i2c_probe) from [<c0371c78>] (i2c_device_probe+0x11c/0x17c) In wm8994-core, we already checked wheter platform data is NULL and set the wm8994->pdata to platform data if platform data is not NULL. So, we can use wm8994->pdata instead of platform data. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-irq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/mfd/wm8994-irq.c b/drivers/mfd/wm8994-irq.c index a14407e..ec2c891 100644 --- a/drivers/mfd/wm8994-irq.c +++ b/drivers/mfd/wm8994-irq.c @@ -193,7 +193,7 @@ int wm8994_irq_init(struct wm8994 *wm8994) { int ret; unsigned long irqflags; - struct wm8994_pdata *pdata = dev_get_platdata(wm8994->dev); + struct wm8994_pdata *pdata = &wm8994->pdata; if (!wm8994->irq) { dev_warn(wm8994->dev, -- 2.0.0.390.gcb682f8

2 1

[alsa-devel] [PATCH v6 0/3] ALSA: HDA: add extended HDA
by Vinod Koul 09 Jun '15

09 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Vinod Koul (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 385 +++++++++++++++++++++++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 115 ++++++++++ sound/hda/ext/hdac_ext_controller.c | 295 +++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 414 ++++++++++++++++++++++++++++++++++++ 7 files changed, 1219 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

2 18

[alsa-devel] [PATCH-v2 1/1] HDA: Intel: enable automatic runtime pm for HDMI codecs by default
by han.lu＠intel.com 09 Jun '15

09 Jun '15

From: "Lu, Han" <han.lu(a)intel.com> Enable runtime PM of the HDMI audio codec on the latest Intel platforms. So the HD-A controller or HDMI codec can suspend when idle timeout by default and release the GFX power well. The patch influences HSW/BDW/BYT/BSW/SKL. Eariler platforms and third party analog codecs will not be influenced. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 54380ed..e9f274b6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3495,6 +3495,9 @@ static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); + if (delay == 0 && codec->auto_runtime_pm) + delay = 3000; + if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c57258..12837ab 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -251,6 +251,7 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ unsigned int power_save_node:1; /* advanced PM for each widget */ + unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 407978b..f852734 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2356,6 +2356,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->dp_mst = true; } + /* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */ + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) + codec->auto_runtime_pm = 1; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] HDA: Intel: enable automatic runtime pm for HDMI codecs by default
by han.lu＠intel.com 09 Jun '15

09 Jun '15

From: "Lu, Han" <han.lu(a)intel.com> Enable runtime PM of the HDMI audio codec on the latest Intel platforms. So the HD-A controller or HDMI codec can suspend when idle timeout by default and release the GFX power well. The patch influences HSW/BDW/BYT/BSW/SKL. Eariler platforms and third party analog codecs will not be influenced. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 54380ed..804ec24 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3491,10 +3491,13 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_PM -static void codec_set_power_save(struct hda_codec *codec, int delay) +static void codec_set_power_save(struct hda_codec *codec, int delay, bool force) { struct device *dev = hda_codec_dev(codec); + if (force && delay == 0 && codec->auto_runtime_pm) + delay = 3000; + if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); @@ -3511,15 +3514,16 @@ static void codec_set_power_save(struct hda_codec *codec, int delay) * snd_hda_set_power_save - reprogram autosuspend for the given delay * @bus: HD-audio bus * @delay: autosuspend delay in msec, 0 = off + * @force: force to enable autosuspend * * Synchronize the runtime PM autosuspend state from the power_save option. */ -void snd_hda_set_power_save(struct hda_bus *bus, int delay) +void snd_hda_set_power_save(struct hda_bus *bus, int delay, bool force) { struct hda_codec *c; list_for_each_codec(c, bus) - codec_set_power_save(c, delay); + codec_set_power_save(c, delay, force); } EXPORT_SYMBOL_GPL(snd_hda_set_power_save); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c57258..9f525ba 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -251,6 +251,7 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ unsigned int power_save_node:1; /* advanced PM for each widget */ + unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; @@ -482,10 +483,11 @@ const char *snd_hda_get_jack_location(u32 cfg); #define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core) #define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core) #ifdef CONFIG_PM -void snd_hda_set_power_save(struct hda_bus *bus, int delay); +void snd_hda_set_power_save(struct hda_bus *bus, int delay, bool force); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} +static inline void +snd_hda_set_power_save(struct hda_bus *bus, int delay, bool force) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7933fb0..11d52e5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -797,7 +797,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) chip = &hda->chip; if (!hda->probe_continued || chip->disabled) continue; - snd_hda_set_power_save(&chip->bus, power_save * 1000); + snd_hda_set_power_save(&chip->bus, power_save * 1000, false); } mutex_unlock(&card_list_lock); return 0; @@ -2034,7 +2034,7 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); - snd_hda_set_power_save(&chip->bus, power_save * 1000); + snd_hda_set_power_save(&chip->bus, power_save * 1000, true); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 477742c..bec3a37 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -518,7 +518,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - snd_hda_set_power_save(&chip->bus, power_save * 1000); + snd_hda_set_power_save(&chip->bus, power_save * 1000, false); return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 407978b..7941d81 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2356,6 +2356,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->dp_mst = true; } + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) + codec->auto_runtime_pm = 1; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.9.1

3 2

Re: [alsa-devel] [pulseaudio-discuss] snd_pcm_avail no reliable
by Raymond Yau 09 Jun '15

09 Jun '15

> > I tried using a smaller period_size (about 1/16 of hwbuf_size), snd_pcm_avail returned a better value. Period bytes = 2Kbytes which is still larger than the default rewind_safeguard 512 bytes > But cpu usage moved from 1% to 4%, since we are using it in a automotive, we don`t care about power saving, I think it is acceptable. > > But I`m not sure if there is any other down side about "time based scheduling" ? > >

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[alsa-devel] [PATCH] alsa: hda_codec: fix number of devices query on hotplug
by airlied＠gmail.com 09 Jun '15

09 Jun '15

From: Dave Airlie <airlied(a)redhat.com> The new regmap code seems to cache this, which isn't helpful for the hotplug dock situation where this gets updated. Use the uncached query for this. Signed-off-by: Dave Airlie <airlied(a)redhat.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b49feff..b778221 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -436,7 +436,7 @@ static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) get_wcaps_type(wcaps) != AC_WID_PIN) return 0; - parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + parm = snd_hdac_read_parm_uncached(&codec->core, nid, AC_PAR_DEVLIST_LEN); if (parm == -1 && codec->bus->rirb_error) parm = 0; return parm & AC_DEV_LIST_LEN_MASK; -- 2.4.1

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[alsa-devel] [PATCH] PCM: snd_pcm_dmix_drain() maybe blocked after suspend and resume
by Shengjiu Wang 09 Jun '15

09 Jun '15

After suspend and resume, the alsa driver is stopped. But if alsa-lib run into snd_pcm_dmix_drain(), it need to wait avail >= pcm->stop_threshold, otherwise, it will not exit the loop, so finally it is blocked at poll() of snd_pcm_wait_nocheck(pcm, -1). This patch is to add state check after snd_pcm_wait_nocheck(pcm, -1), if the state is SND_PCM_STATE_SUSPENDED, then return error. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_dmix.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index babde6a..0cc318e 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -647,6 +647,12 @@ static int snd_pcm_dmix_drain(snd_pcm_t *pcm) if (dmix->state == SND_PCM_STATE_DRAINING) { snd_pcm_dmix_sync_area(pcm); snd_pcm_wait_nocheck(pcm, -1); + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + default: + break; + } snd_pcm_direct_clear_timer_queue(dmix); /* force poll to wait */ } } while (dmix->state == SND_PCM_STATE_DRAINING); -- 1.7.9.5

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Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 09 Jun '15

09 Jun '15

> > > > > > No change. > > > http://www.alsa-project.org/db/?f=30b3f0087374b914a20dbe20a618fb892a5d6fd5 > > > > Are there any offical specificaion ? > > > > So far most review only mention stereo speakers and subwoofer, the service > > guide only show how to replace two internal speakers > > Service manual mentions > "Dolby®-certified surround sound system with two built-in stereo speakers and > one subwoofer supporting low-frequency effects" > > so all additional channels what I'm hearing in Windows is emulated. I hope > that it is clear now that my laptop has two internal speakers and one > subwoofer. > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… You need to define your 2.1 chmap if the subwoofer only use left channel Compiling and running PulseAudio from git http://colin.guthr.ie/2010/09/compiling-and-running-pulseaudio-from-git/

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[alsa-devel] regression in device count on hotplug
by Dave Airlie 09 Jun '15

09 Jun '15

Hi, I've been looking at mst support a bit more, and noticed a regression in the number of devices probing between 4.0 and 4.1 On 4.0 if I boot with no dock, and cat /proc/asound/HDMI/codec#0 Node 0x06 has Devices: 0 if I dock I get Devices: 3, and the info about the 3 devices. On 4.1-rc6 if I boot with no dock, and cat I get Devices:0, however when I dock, I still get Devices: 0 I haven't had time to bisect yet, but was hoping someone might be able to spot it from the symptoms. Dave.

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Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 09 Jun '15

09 Jun '15

> > > No change. > > > http://www.alsa-project.org/db/?f=30b3f0087374b914a20dbe20a618fb892a5d6fd5 > > > > Are there any offical specificaion ? > > > > So far most review only mention stereo speakers and subwoofer, the service > > guide only show how to replace two internal speakers > > Service manual mentions > "Dolby®-certified surround sound system with two built-in stereo speakers and > one subwoofer supporting low-frequency effects" > > so all additional channels what I'm hearing in Windows is emulated. I hope > that it is clear now that my laptop has two internal speakers and one > subwoofer. > You have to determine which node 0x16 or 0x17 is your subwoofer and remove the redundant speaker pin fixup >Thanks. I know how to help myself, but I expected it by default. I wonder how > many other users will manage to get their sound working good as before > regression. > And wasn't "model=acer-aspire-4930g" some sort of early patching which was > thrown out of official code? After remove the redundant internal speaker, 5.1 is only available after you change the channel mode Can alsactl restore the channel mode before pulseaudio probe the sound cards ? Auto mic select is not enabled when the notebook has mic jack and line in jack , internl mic As jack retasking is specific to snd-hda-intel and some multi cjannel ac97 codecs It is not easy to force pulseaudio to stop all client connection, reprobe the profiles on board audio after jack retasking hdajackretask also perform the same function stop pulseaudio and using dynamic reconfiguration/early patching For those notebook with headphone spdif combo jack, it need a customised iec958.conf and you can use headphone jack to make iec958 unavailable http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone] state.plugged = no state.unplugged = unknown

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[alsa-devel] [GIT PULL] ASoC updates for v4.2
by Mark Brown 08 Jun '15

08 Jun '15

The following changes since commit c65b99f046843d2455aa231747b5a07a999a9f3d: Linux 4.1-rc6 (2015-05-31 19:01:07 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.2 for you to fetch changes up to 11e688862c4c8162119a4ca51c3326555966c8bb: Merge remote-tracking branches 'asoc/topic/wm8994', 'asoc/topic/wm8996' and 'asoc/topic/zx' into asoc-next (2015-06-05 18:55:07 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.2 The big thing this release has been Liam's addition of topology support to the core. We've also seen quite a bit of driver work and the continuation of Lars' refactoring for component support. - Support for loading ASoC topology maps from firmware, intended to be used to allow self-describing DSP firmware images to be built which can map controls added by the DSP to userspace without the kernel needing to know about individual DSP firmwares. - Lots of refactoring to avoid direct access to snd_soc_codec where it's not needed supporting future refactoring. - Big refactoring and cleanup serieses for the Wolfson ADSP and TI TAS2552 drivers. - Support for TI TAS571x power amplifiers. - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs. - Support for x86 systems with RT5650 and Qualcomm Storm. ---------------------------------------------------------------- Alexander Stein (2): ASoC: sgtl5000: Use specific variable for lo_vag ASoC: sgtl5000: Calculate Lineout Channel Output Level Alexandre Belloni (2): ASoC: atmel: simplify Kconfig ASoC: atmel: compile pcm driver in snd-soc-atmel_ssc_dai Anatol Pomozov (1): ASoC: rt5677: Add reset-gpio dts option Antonio Ospite (1): ASoC: adau1977: fix typo s/Substraction/Subtraction/ Arnd Bergmann (3): ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency, again ASoC: qcom: remove incorrect dependencies ASoC: qcom: fix STORM board Kconfig Axel Lin (7): ASoC: da7210: Fix dependency to allow build with SND_SOC_I2C_AND_SPI ASoC: sta32x: Use devm_gpiod_get_optional at appropriate place ASoC: sta350: Use devm_gpiod_get_optional at appropriate place ASoC: wm8737: Fixup setting VMID Impedance control register ASoC: wm8903: Fix define for WM8903_VMID_RES_250K ASoC: wm8955: Fix setting wrong register for WM8955_K_8_0_MASK bits ASoC: max98925: Fix mask for setting DAI invert mode Bard Liao (10): ASoC: rt5645: Adds push button support for rt5650 ASoC: rt5677: add i2s asrc clk src selection ASoC: rt5645: fix implicit declaration error ASoC: rt5645: fix wrong mask for button report ASoC: rt5645: remove RT5645_I2S_BCLK_MS1 control ASoC: rt5645: fix PLL source register definitions ASoC: rt5645: add TDM slot control into dapm route ASoC: rt5645: make volume TLV closer to reality ASoC: rt5645: fix add missing widget ASoC: rt5645: remove unused field in pdata Charles Keepax (22): ASoC: wm_adsp: Split out adsp1 & 2 setup algorithms ASoC: wm_adsp: Improve variable naming ASoC: wm_adsp: Remove len field from wm_adsp_alg_region ASoC: wm_adsp: Limit firmware control name to ALSA control name size ASoC: wm_adsp: Move temporary control name to the stack ASoC: wm_adsp: Clean up low level control read/write functions ASoC: wm_adsp: Factor out creation of alg_regions ASoC: wm_adsp: Remove private field from wm_coeff_ctl ASoC: wm_adsp: Group all the ALSA control functions together ASoC: wm_adsp: Add basic support for rev 1 firmware file format ASoC: wm_adsp: Add support for rev 2 firmware file format ASoC: wm_adsp: Warn that firmware file format 0 is depreciated ASoC: wm_adsp: Use __leXX for little endian data ASoC: wm_adsp: Add support for DSP control flags ASoC: dapm: Remove local OOM error message ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro ASoC: dapm: Append "Autodisable" to autodisable widget names ASoC: dapm: Add support for autodisable mux controls ASoC: arizona: Use auto disable muxes for routing ASoC: dapm: Add missing mutex unlock ASoC: dapm: Break out of widget search when source and sink are located ASoC: dapm: Add cache to speed up adding of routes Dan Carpenter (1): ASoC: wm8741: check for error returns from wm8741_set_pdata() Dylan Reid (2): ASoC: Add jack types to dt-bindings ASoC: jack - add_gpiods accepts filled descriptors Fabio Estevam (8): ASoC: fsl_ssi: No need call of_device_is_available() ASoC: fsl: fsl_dma: Use true/false for boolean init ASoC: fsl: imx-mc13783: Simplify trivial if-return sequence ASoC: wm8996: Pass the IRQF_ONESHOT flag ASoC: wm5100: Pass the IRQF_ONESHOT flag ASoC: wm8994: Pass the IRQF_ONESHOT flag ASoC: twl6040: Pass the IRQF_ONESHOT flag ASoC: max98095: Pass the IRQF_ONESHOT flag Fang, Yang A (10): ASoC: Intel: Add support rt5650 in sst driver ASoC: rt5645: set platform data base on DMI ASoC: rt5645: fixed kbuild err ASoC: Intel: Add support max98090 in sst driver ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_max98090_ti ASoC: Intel: Support rt5650 codec for Cherrytrail & Braswell ASoC: Intel: Fixed kbuild warnings ASoC: Intel: Enabled button jack for BSW platform with rt5650 codec ASoC: max98090: read micbias from device property ASoC: ts3a227e: use device property api Geert Uytterhoeven (2): ASoC: rsnd: Use generic names for device nodes ASoC: rsnd: Document r8a7778-specific binding Jie Yang (6): ASoC: Intel: remove unused function hsw_pcm_free_modules() ASoC: Intel: don't need compress offload for broadwell ASoC: intel: Revert "ASoC: Intel: remove unused function hsw_pcm_free_modules()" ASoC: intel: Revert "ASoC: Intel: fix broadwell module removing failed issue" ASoC: Intel: check and clear runtime module pointer ASoC: Intel: handle haswell pcm suspend including runtime modules freeing Jin Yao (1): ASoC: Intel: Remove invalid kfree of devm allocated data John Lin (5): ASoC: rt5645: improve headphone depop function ASoC: rt5645: remove unnecessary power in JD function ASoC: rt5645: fix IRQ error in jack detection ASoC: rt5645: fix jack type detect error ASoC: rt5645: fix kernel hang when call rt5645_set_jack_detect() Jun Nie (5): ASoC: simple-card: support platform in dts parse ASoC: zx: Add zx296702 SPDIF support ASoC: zx: Add ZTE zx296702 I2S DAI driver dt: Add documentation for the ZTE SPDIF controller dt: Add documentation for the ZTE I2S controller Jyri Sarha (4): ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule ASoC: davinci-mcasp: Channel count constraints for multi-serializer case ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule ASoC: core: If component doesn't have of_node use parent's node instead Kevin Cernekee (4): ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver ASoC: tas571x: Eliminate redundant dev->of_node NULL check Krzysztof Kozlowski (4): ASoC: au1x: Constify platform_device_id ASoC: bt-sco: Constify platform_device_id ASoC: fsl: Constify platform_device_id ASoC: samsung: Constify platform_device_id Kuninori Morimoto (13): ASoC: rsnd: add rsnd_dai_to_priv() macro ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call ASoC: rsnd: tidyup SSI parent related function/macro names ASoC: rsnd: make sure SSI parent/child uses same number of sound channel. ASoC: rsnd: care snd_kcontrol's index ASoC: soc.h: tidyup struct snd_soc_dai_link definition order ASoC: rsnd: revert lock for calls to rsnd_dai_call ASoC: rsnd: indicate unknown HW start ASoC: rsnd: add rsnd_dai_stream_quit() ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM dmaengine: rcar-dmac: fixup spinlock in rcar-dmac ASoC: rsnd: spin lock for interrupt handler Lars-Peter Clausen (114): ASoC: at91sam9g20ek: Automatically disconnect non-connected pins ASoC: dmaengine_pcm: Make FLAG_NO_RESIDUE internal ASoC: Add helper function getting CODEC's DAPM context ASoC: Add helper functions bias level management ASoC: Route all bias level updates through the core ASoC: Move bias level update to the core ASoC: samsung: wolfson: Improve compile test coverage ASoC: lowland: Use card DAPM context to access widgets ASoC: smdk_wm8994: Use card DAPM context to access widgets ASoC: smartq: Remove unnecessary snd_soc_dapm_disable_pin() ASoC: smartq: Use card DAPM context to access widgets ASoC: speyside: Use snd_soc_codec_get_dapm() ASoC: ad1836: Replace direct snd_soc_codec dapm field access ASoC: adau17x1: Replace direct snd_soc_codec dapm field access ASoC: adau1977: Replace direct snd_soc_codec dapm field access ASoC: adav80x: Replace direct snd_soc_codec dapm field access ASoC: ssm2518: Replace direct snd_soc_codec dapm field access ASoC: ssm2602: Replace direct snd_soc_codec dapm field access ASoC: ssm4567: Replace direct snd_soc_codec dapm field access ASoC: dapm: Add new widgets to the end of the widget list ASoC: dapm: Add new widgets to the end of the widget list ASoC: dapm: Add demux support ASoC: lm4857: Use DAPM demux ASoC: lm4857: Convert to component ASoC: omap-twl4030: Use card DAPM context to access widgets ASoC: brownstone: Automatically disconnect non-connected pins ASoC: poodle: Automatically disconnect non-connected pins ASoC: tosa: Automatically disconnect non-connected pins ASoC: z2: Automatically disconnect non-connected pins ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access ASoC: ab8500: Replace direct snd_soc_codec dapm field access ASoC: ak4641: Replace direct snd_soc_codec dapm field access ASoC: cx20442: Replace direct snd_soc_codec dapm field access ASoC: es8328: Replace direct snd_soc_codec dapm field access ASoC: jz4740: Replace direct snd_soc_codec dapm field access ASoC: ml26124: Replace direct snd_soc_codec dapm field access ASoC: uda134x: Replace direct snd_soc_codec dapm field access ASoC: uda1380: Replace direct snd_soc_codec dapm field access ASoC: sgtl5000: Replace direct snd_soc_codec dapm field access ASoC: sirf-audio-codec: Replace direct snd_soc_codec dapm field access ASoC: sn95031: Replace direct snd_soc_codec dapm field access ASoC: lm49453: Replace direct snd_soc_codec dapm field access ASoC: pcm512x: Replace direct snd_soc_codec dapm field access ASoC: tlv320aix31xx: Replace direct snd_soc_codec dapm field access ASoC: tlv320aic3x: Replace direct snd_soc_codec dapm field access ASoC: tlv320dac33: Replace direct snd_soc_codec dapm field access ASoC: twl4030: Replace direct snd_soc_codec dapm field access ASoC: twl6040: Replace direct snd_soc_codec dapm field access ASoC: Drop unnecessary bias level check on resume ASoC: sta32x: Replace direct snd_soc_codec dapm field access ASoC: sta350: Replace direct snd_soc_codec dapm field access ASoC: sta529: Replace direct snd_soc_codec dapm field access ASoC: da7213: Replace direct snd_soc_codec dapm field access ASoC: da732x: Replace direct snd_soc_codec dapm field access ASoC: da9055: Replace direct snd_soc_codec dapm field access ASoC: max98088: Replace direct snd_soc_codec dapm field access ASoC: max98090: Remove unnecessary snd_soc_dapm_sync() ASoC: max98090: Replace direct snd_soc_codec dapm field access ASoC: max98095: Replace direct snd_soc_codec dapm field access ASoC: max9850: Replace direct snd_soc_codec dapm field access ASoC: alc5623: Replace direct snd_soc_codec dapm field access ASoC: rt286: Replace direct snd_soc_codec dapm field access ASoC: rt5631: Replace direct snd_soc_codec dapm field access ASoC: rt5640: Replace direct snd_soc_codec dapm field access ASoC: rt5651: Replace direct snd_soc_codec dapm field access ASoC: rt5670: Replace direct snd_soc_codec dapm field access ASoC: rt5677: Replace direct snd_soc_codec dapm field access ASoC: cs42l52: Replace direct snd_soc_codec dapm field access ASoC: cs42l56: Replace direct snd_soc_codec dapm field access ASoC: cs42l73: Replace direct snd_soc_codec dapm field access ASoC: cs42xx8: Replace direct snd_soc_codec dapm field access ASoC: arizona: Replace direct snd_soc_codec dapm field access ASoC: wm0010: Replace direct snd_soc_codec dapm field access ASoC: wm5100: Replace direct snd_soc_codec dapm field access ASoC: wm5102: Replace direct snd_soc_codec dapm field access ASoC: wm5110: Replace direct snd_soc_codec dapm field access ASoC: wm8350: Replace direct snd_soc_codec dapm field access ASoC: wm8400: Replace direct snd_soc_codec dapm field access ASoC: wm8510: Replace direct snd_soc_codec dapm field access ASoC: wm8523: Replace direct snd_soc_codec dapm field access ASoC: wm8580: Replace direct snd_soc_codec dapm field access ASoC: wm8711: Replace direct snd_soc_codec dapm field access ASoC: wm8728: Replace direct snd_soc_codec dapm field access ASoC: wm8731: Replace direct snd_soc_codec dapm field access ASoC: wm8737: Replace direct snd_soc_codec dapm field access ASoC: wm8750: Replace direct snd_soc_codec dapm field access ASoC: wm8753: Replace direct snd_soc_codec dapm field access ASoC: wm8770: Replace direct snd_soc_codec dapm field access ASoC: wm8776: Replace direct snd_soc_codec dapm field access ASoC: wm8804: Replace direct snd_soc_codec dapm field access ASoC: wm8900: Replace direct snd_soc_codec dapm field access ASoC: wm8903: Replace direct snd_soc_codec dapm field access ASoC: wm8904: Replace direct snd_soc_codec dapm field access ASoC: wm8940: Replace direct snd_soc_codec dapm field access ASoC: wm8955: Replace direct snd_soc_codec dapm field access ASoC: wm8960: Replace direct snd_soc_codec dapm field access ASoC: wm8961: Replace direct snd_soc_codec dapm field access ASoC: wm8962: Replace direct snd_soc_codec dapm field access ASoC: wm8971: Replace direct snd_soc_codec dapm field access ASoC: wm8974: Replace direct snd_soc_codec dapm field access ASoC: wm8978: Replace direct snd_soc_codec dapm field access ASoC: wm8983: Replace direct snd_soc_codec dapm field access ASoC: wm8985: Replace direct snd_soc_codec dapm field access ASoC: wm8988: Replace direct snd_soc_codec dapm field access ASoC: wm8990: Replace direct snd_soc_codec dapm field access ASoC: wm8991: Replace direct snd_soc_codec dapm field access ASoC: wm8993: Replace direct snd_soc_codec dapm field access ASoC: wm8994: Replace direct snd_soc_codec dapm field access ASoC: wm8995: Replace direct snd_soc_codec dapm field access ASoC: wm8996: Replace direct snd_soc_codec dapm field access ASoC: wm8997: Replace direct snd_soc_codec dapm field access ASoC: wm9081: Replace direct snd_soc_codec dapm field access ASoC: wm9090: Replace direct snd_soc_codec dapm field access ASoC: wm_hubs: Replace direct snd_soc_codec dapm field access Liam Girdwood (6): ASoC: Intel: fix broadwell module removing failed issue ASoC: DAPM: Add APIs to create individual DAPM controls. ASoC: topology: Add topology UAPI header ASoC: topology: Add topology core ALSA: topology: Export ID types for TLV controls. ASoC: dapm: fix build errors for missing snd_soc_dapm_new_control symbol Maciej S. Szmigiero (3): ASoC: ac97: Remove rate constraints ASoC: ac97: make selectable in config ASoC: codecs: use SNDRV_PCM_FMTBIT_* for format bitmask Marek Belisko (2): ASoC: gtm601: Document GTM601 bindings ASoC: bt-sco: Add devicetree support for bt-sco codec Mark Brown (27): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-tas751x ASoC: tas751x: Factor setting of new bias level into the core Merge branch 'topic/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/dpcm' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'topic/rcar-dma' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge branch 'topic/adsp' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: zte: Fix missing dev in devm operation Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fmtbit', 'asoc/fix/max98925', 'asoc/fix/rcar' and 'asoc/fix/ux500' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8737', 'asoc/fix/wm8903' and 'asoc/fix/wm8955' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ad1836', 'asoc/topic/ada1977', 'asoc/topic/adau1701' and 'asoc/topic/adau17x1' into asoc-next Merge remote-tracking branches 'asoc/topic/adau1977', 'asoc/topic/adav80x', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/au1x', 'asoc/topic/bt-sco', 'asoc/topic/cs35l32' and 'asoc/topic/da7210' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci' and 'asoc/topic/dpcm' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-sai', 'asoc/topic/fsl-spdif', 'asoc/topic/fsl-ssi' and 'asoc/topic/gpio-jack' into asoc-next Merge remote-tracking branches 'asoc/topic/gpiod-flags', 'asoc/topic/gtm601', 'asoc/topic/intel', 'asoc/topic/lm3857' and 'asoc/topic/max98090' into asoc-next Merge remote-tracking branches 'asoc/topic/max98095', 'asoc/topic/omap', 'asoc/topic/pxa', 'asoc/topic/qcom' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5677', 'asoc/topic/samsung' and 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branches 'asoc/topic/simple', 'asoc/topic/ssm2518', 'asoc/topic/ssm2602', 'asoc/topic/ssm4567' and 'asoc/topic/sta32x' into asoc-next Merge remote-tracking branches 'asoc/topic/sta350', 'asoc/topic/tas2552', 'asoc/topic/tas3a227e' and 'asoc/topic/tas571x' into asoc-next Merge remote-tracking branches 'asoc/topic/topology', 'asoc/topic/twl6040', 'asoc/topic/wm5100', 'asoc/topic/wm8741' and 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8994', 'asoc/topic/wm8996' and 'asoc/topic/zx' into asoc-next Nicolas Boichat (2): ASoC: rt5645: Rename HP control to Headphone ASoC: rt5645: Init jack_detect_work before registering irq Nicolin Chen (1): ASoC: fsl_spdif: Don't try to round-up for clock divisor calculation Oder Chiou (1): ASoC: rt5677: Add DMIC ASRC detect function Pascal Huerst (1): ASoC: adau1701: add regulator consumer support Peter Ujfalusi (18): ASoC: tas2552: Make the enable-gpio really optional ASoC: tas2552: Fix kernel crash when the codec is loaded but not part of a card ASoC: tas2552: Fix kernel crash caused by wrong kcontrol entry ASoC: tas2552: Correct PDM configuration register bit definitions ASoC: tas2552: Correct CFG1 register bit definitions ASoC: tas2552: Simplify the tas2552_mute function ASoC: tas2552: Simplify and reverse the functionality of tas2552_sw_shutdown ASoC: tas2552: Rename mclk parameter to pll_clkin to match with the datasheet ASoC: tas2552: bindings header file for tas2552 codec ASoC: tas2552: Add support for pll and pdm source clock selection ASoC: tas2552: Correct dai format support ASoC: tas2552: Correct and clean up data format and BCLK/WCLK direction ASoC: tas2552: Add TDM support ASoC: tas2552: Clean up the Digital - Analog DAPM route definition ASoC: tas2552: Correct the Speaker Driver Playback Volume (PGA_GAIN) ASoC: tas2552: Implement startup/stop sequence as per TRM ASoC: tas2552: Add support for word length configuration ASoC: tas2552: Configure the WCLK frequency based on the stream Rajan Vaja (1): ASoC: mop500_ab8500: Add a NULL pointer check in mop500_ab8500_machine_init() Richard Fitzgerald (6): ASoC: arizona: Fix noise generator gain TLV ASoC: wm_adsp: Dump scratch registers on DSP shutdown ASoC: arizona: Export functions to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control regulator: arizona-ldo1: Do not control DVFS clocking from regulator Sergej Sawazki (1): ASoC: wm8741: Add differential mono mode support Srinivas Kandagatla (13): ASoC: qcom: Remove redundant error check. ASoC: qcom: remove unnecessary header files ASoC: qcom: move ipq806x specific bits out of lpass driver. ASoC: qcom: remove hardcoded i2s port number ASoC: qcom: remove hardcoded dma channel ASoC: skip legacy dai naming if dai driver has all the information ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: make osr clock optional ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Document apq8016 bindings. ASoC: qcom: Add apq8016 lpass driver support Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size Sudip Mukherjee (1): ASoC: rt5645: fixed section mismatch Takashi Iwai (1): ASoC: Simplify format_register_str() without stack usages Uwe Kleine-König (3): ASoC: sta32x: use devm_gpiod_get_optional for optional reset gpio ASoC: max98357a: use flags argument of devm_gpiod_get to set direction ASoC: rx51: use flags argument of devm_gpiod_get to set direction Vinod Koul (6): ASoC: intel - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN ASoC: Intel: add frame and data polarity to ssp config ASoC: Intel: use local values for ssp configuration ASoC: Intel: load hw_defaults in hw_params of ssp be ASoC: intel: add support for specifying PCM format ASoC: Intel: add support for configuring TDM slots for SSP Vladimir Zapolskiy (1): ASoC: dapm: fix snd_soc_dapm_new_control() implicit declaration Wei Yongjun (1): ASoC: cs35l32: Remove unused including <linux/version.h> Zidan Wang (4): ASoC: fsl_sai: add sai master mode support ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock ASoC: wm8960: add 32 bit word length support kbuild test robot (7): ASoC: adau1977: fix simple_return.cocci warnings ASoC: ssm2518: fix simple_return.cocci warnings ASoC: qcom: ipq806x_lpass_alloc_dma_channel() can be static ASoC: zx: zx_i2s_dai can be static ASoC: zx: fix platform_no_drv_owner.cocci warnings ASoC: zx: fix simple_return.cocci warnings ASoC: zx: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/adi,adau1701.txt | 4 + Documentation/devicetree/bindings/sound/bt-sco.txt | 13 + Documentation/devicetree/bindings/sound/gtm601.txt | 13 + .../devicetree/bindings/sound/max98090.txt | 6 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 3 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 + .../devicetree/bindings/sound/tas571x.txt | 41 + Documentation/devicetree/bindings/sound/wm8741.txt | 11 + .../devicetree/bindings/sound/zte,zx-i2s.txt | 44 + .../devicetree/bindings/sound/zte,zx-spdif.txt | 28 + .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 6 + drivers/dma/sh/rcar-dmac.c | 37 +- drivers/regulator/arizona-ldo1.c | 5 - include/dt-bindings/sound/apq8016-lpass.h | 9 + include/dt-bindings/sound/audio-jack-events.h | 9 + include/dt-bindings/sound/tas2552.h | 18 + include/sound/dmaengine_pcm.h | 5 - include/sound/rt5645.h | 3 - include/sound/soc-dapm.h | 49 + include/sound/soc-topology.h | 168 ++ include/sound/soc.h | 118 +- include/sound/tlv.h | 15 +- include/uapi/sound/asoc.h | 388 +++++ include/uapi/sound/tlv.h | 31 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 + sound/soc/atmel/Kconfig | 20 +- sound/soc/atmel/Makefile | 8 +- sound/soc/atmel/atmel-pcm-dma.c | 3 +- sound/soc/atmel/sam9g20_wm8731.c | 10 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/cirrus/ep93xx-pcm.c | 1 - sound/soc/codecs/88pm860x-codec.c | 19 +- sound/soc/codecs/Kconfig | 12 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ab8500-codec.c | 20 +- sound/soc/codecs/ac97.c | 8 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 126 +- sound/soc/codecs/adau1761.c | 27 +- sound/soc/codecs/adau1781.c | 10 +- sound/soc/codecs/adau17x1.c | 20 +- sound/soc/codecs/adau1977.c | 14 +- sound/soc/codecs/adav80x.c | 11 +- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 3 +- sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/alc5623.c | 3 +- sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/arizona.c | 174 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/bt-sco.c | 11 +- sound/soc/codecs/cq93vc.c | 1 - sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs42l52.c | 5 +- sound/soc/codecs/cs42l56.c | 5 +- sound/soc/codecs/cs42l73.c | 3 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cx20442.c | 6 +- sound/soc/codecs/da7213.c | 3 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 3 +- sound/soc/codecs/es8328.c | 3 +- sound/soc/codecs/isabelle.c | 2 - sound/soc/codecs/jz4740.c | 4 +- sound/soc/codecs/lm4857.c | 114 +- sound/soc/codecs/lm49453.c | 4 +- sound/soc/codecs/max98088.c | 3 +- sound/soc/codecs/max98090.c | 34 +- sound/soc/codecs/max98095.c | 24 +- sound/soc/codecs/max98357a.c | 3 +- sound/soc/codecs/max9850.c | 3 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/ml26124.c | 3 +- sound/soc/codecs/pcm512x.c | 8 +- sound/soc/codecs/rt286.c | 33 +- sound/soc/codecs/rt5631.c | 5 +- sound/soc/codecs/rt5640.c | 16 +- sound/soc/codecs/rt5645.c | 1089 +++++++++--- sound/soc/codecs/rt5645.h | 30 +- sound/soc/codecs/rt5651.c | 5 +- sound/soc/codecs/rt5670.c | 26 +- sound/soc/codecs/rt5677.c | 157 +- sound/soc/codecs/rt5677.h | 15 + sound/soc/codecs/sgtl5000.c | 56 +- sound/soc/codecs/sirf-audio-codec.c | 2 +- sound/soc/codecs/sn95031.c | 12 +- sound/soc/codecs/ssm2518.c | 9 +- sound/soc/codecs/ssm2602.c | 5 +- sound/soc/codecs/ssm4567.c | 9 +- sound/soc/codecs/sta32x.c | 19 +- sound/soc/codecs/sta350.c | 9 +- sound/soc/codecs/sta529.c | 8 +- sound/soc/codecs/stac9766.c | 3 +- sound/soc/codecs/tas2552.c | 312 +++- sound/soc/codecs/tas2552.h | 90 +- sound/soc/codecs/tas571x.c | 514 ++++++ sound/soc/codecs/tas571x.h | 33 + sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 11 +- sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 10 +- sound/soc/codecs/tlv320dac33.c | 5 +- sound/soc/codecs/ts3a227e.c | 15 +- sound/soc/codecs/twl4030.c | 3 +- sound/soc/codecs/twl6040.c | 9 +- sound/soc/codecs/uda134x.c | 4 +- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm1250-ev1.c | 2 - sound/soc/codecs/wm5100.c | 12 +- sound/soc/codecs/wm5102.c | 62 +- sound/soc/codecs/wm5110.c | 11 +- sound/soc/codecs/wm8350.c | 3 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 3 +- sound/soc/codecs/wm8523.c | 3 +- sound/soc/codecs/wm8580.c | 3 +- sound/soc/codecs/wm8711.c | 3 +- sound/soc/codecs/wm8728.c | 3 +- sound/soc/codecs/wm8731.c | 8 +- sound/soc/codecs/wm8737.c | 11 +- sound/soc/codecs/wm8741.c | 129 +- sound/soc/codecs/wm8741.h | 10 + sound/soc/codecs/wm8750.c | 3 +- sound/soc/codecs/wm8753.c | 3 +- sound/soc/codecs/wm8770.c | 3 +- sound/soc/codecs/wm8776.c | 3 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 13 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8903.h | 2 +- sound/soc/codecs/wm8904.c | 5 +- sound/soc/codecs/wm8940.c | 6 +- sound/soc/codecs/wm8955.c | 7 +- sound/soc/codecs/wm8960.c | 125 +- sound/soc/codecs/wm8961.c | 6 +- sound/soc/codecs/wm8962.c | 21 +- sound/soc/codecs/wm8971.c | 3 +- sound/soc/codecs/wm8974.c | 3 +- sound/soc/codecs/wm8978.c | 7 +- sound/soc/codecs/wm8983.c | 3 +- sound/soc/codecs/wm8985.c | 3 +- sound/soc/codecs/wm8988.c | 3 +- sound/soc/codecs/wm8990.c | 5 +- sound/soc/codecs/wm8991.c | 3 +- sound/soc/codecs/wm8993.c | 12 +- sound/soc/codecs/wm8994.c | 68 +- sound/soc/codecs/wm8995.c | 6 +- sound/soc/codecs/wm8996.c | 23 +- sound/soc/codecs/wm8997.c | 18 +- sound/soc/codecs/wm9081.c | 4 +- sound/soc/codecs/wm9090.c | 6 +- sound/soc/codecs/wm9712.c | 3 +- sound/soc/codecs/wm9713.c | 7 +- sound/soc/codecs/wm_adsp.c | 1210 +++++++------ sound/soc/codecs/wm_adsp.h | 19 +- sound/soc/codecs/wm_hubs.c | 4 +- sound/soc/codecs/wmfw.h | 44 +- sound/soc/davinci/davinci-mcasp.c | 167 +- sound/soc/fsl/fsl_dma.c | 4 +- sound/soc/fsl/fsl_sai.c | 144 +- sound/soc/fsl/fsl_sai.h | 9 +- sound/soc/fsl/fsl_spdif.c | 10 +- sound/soc/fsl/fsl_ssi.c | 7 - sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-mc13783.c | 6 +- sound/soc/generic/simple-card.c | 16 +- sound/soc/intel/Kconfig | 17 +- sound/soc/intel/atom/sst-atom-controls.c | 167 +- sound/soc/intel/atom/sst-atom-controls.h | 9 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 47 +- sound/soc/intel/atom/sst-mfld-platform.h | 2 + sound/soc/intel/atom/sst/sst_acpi.c | 4 + sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 318 ++++ sound/soc/intel/boards/cht_bsw_rt5645.c | 118 +- sound/soc/intel/common/sst-ipc.c | 34 +- sound/soc/intel/common/sst-ipc.h | 7 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 + sound/soc/intel/haswell/sst-haswell-pcm.c | 31 +- sound/soc/omap/Kconfig | 5 +- sound/soc/omap/omap-twl4030.c | 3 +- sound/soc/omap/rx51.c | 30 +- sound/soc/pxa/brownstone.c | 25 +- sound/soc/pxa/poodle.c | 19 +- sound/soc/pxa/tosa.c | 13 +- sound/soc/pxa/z2.c | 9 +- sound/soc/qcom/Kconfig | 19 +- sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-apq8016.c | 242 +++ sound/soc/qcom/lpass-cpu.c | 240 +-- sound/soc/qcom/lpass-ipq806x.c | 109 ++ .../{lpass-lpaif-ipq806x.h => lpass-lpaif-reg.h} | 92 +- sound/soc/qcom/lpass-platform.c | 202 ++- sound/soc/qcom/lpass.h | 51 +- sound/soc/samsung/Kconfig | 15 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 6 +- sound/soc/samsung/smdk_wm8994.c | 3 +- sound/soc/samsung/speyside.c | 2 +- sound/soc/sh/rcar/core.c | 32 +- sound/soc/sh/rcar/rsnd.h | 3 +- sound/soc/sh/rcar/rsrc-card.c | 1 + sound/soc/sh/rcar/src.c | 11 +- sound/soc/sh/rcar/ssi.c | 70 +- sound/soc/soc-core.c | 67 +- sound/soc/soc-dapm.c | 349 +++- sound/soc/soc-generic-dmaengine-pcm.c | 25 +- sound/soc/soc-jack.c | 9 +- sound/soc/soc-pcm.c | 47 +- sound/soc/soc-topology.c | 1826 ++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.c | 4 + sound/soc/ux500/ux500_pcm.c | 1 - sound/soc/zte/Kconfig | 17 + sound/soc/zte/Makefile | 2 + sound/soc/zte/zx296702-i2s.c | 436 +++++ sound/soc/zte/zx296702-spdif.c | 365 ++++ 225 files changed, 9460 insertions(+), 2305 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 include/dt-bindings/sound/apq8016-lpass.h create mode 100644 include/dt-bindings/sound/audio-jack-events.h create mode 100644 include/dt-bindings/sound/tas2552.h create mode 100644 include/sound/soc-topology.h create mode 100644 include/uapi/sound/asoc.h create mode 100644 include/uapi/sound/tlv.h create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h create mode 100644 sound/soc/intel/boards/cht_bsw_max98090_ti.c create mode 100644 sound/soc/qcom/lpass-apq8016.c create mode 100644 sound/soc/qcom/lpass-ipq806x.c rename sound/soc/qcom/{lpass-lpaif-ipq806x.h => lpass-lpaif-reg.h} (64%) create mode 100644 sound/soc/soc-topology.c create mode 100644 sound/soc/zte/Kconfig create mode 100644 sound/soc/zte/Makefile create mode 100644 sound/soc/zte/zx296702-i2s.c create mode 100644 sound/soc/zte/zx296702-spdif.c

2 1

[alsa-devel] [PATCH] alsactl: Try to create state file directory
by Dan Nicholson 08 Jun '15

08 Jun '15

Try to create the directory for the state file when saving so we don't depend on it being created ahead of time. This only checks for failures on existing directories and doesn't try to create the leading directories or workaround any other errors. This should catch the common case where /var/lib exists, but /var/lib/alsa doesn't. Signed-off-by: Dan Nicholson <nicholson(a)endlessm.com> --- alsactl/state.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/alsactl/state.c b/alsactl/state.c index 3908ec4..8ca3d6e 100644 --- a/alsactl/state.c +++ b/alsactl/state.c @@ -27,6 +27,9 @@ #include <stdio.h> #include <assert.h> #include <errno.h> +#include <string.h> +#include <sys/stat.h> +#include <sys/types.h> #include <alsa/asoundlib.h> #include "alsactl.h" @@ -1544,6 +1547,7 @@ int save_state(const char *file, const char *cardname) snd_output_t *out; int stdio; char *nfile = NULL; + char *filedir = NULL; int lock_fd = -EINVAL; err = snd_config_top(&config); @@ -1553,6 +1557,9 @@ int save_state(const char *file, const char *cardname) } stdio = !strcmp(file, "-"); if (!stdio) { + char *tmp; + mode_t mode = S_IRWXU | S_IRWXG | S_IRWXO; + nfile = malloc(strlen(file) + 5); if (nfile == NULL) { error("No enough memory..."); @@ -1561,6 +1568,21 @@ int save_state(const char *file, const char *cardname) } strcpy(nfile, file); strcat(nfile, ".new"); + filedir = strdup(file); + if (filedir == NULL) { + error("Not enough memory..."); + err = -ENOMEM; + goto out; + } + tmp = strrchr(filedir, '/'); + if (tmp && tmp != filedir) { + *tmp = '\0'; + if (mkdir(filedir, mode) != 0 && errno != EEXIST) { + error("Could not create directory %s: %s", + filedir, strerror(errno)); + goto out; + } + } lock_fd = state_lock(file, 10); if (lock_fd < 0) { err = lock_fd; @@ -1640,6 +1662,7 @@ out: if (!stdio && lock_fd >= 0) state_unlock(lock_fd, file); free(nfile); + free(filedir); snd_config_delete(config); snd_config_update_free_global(); return err; -- 1.8.1.2

2 7

[alsa-devel] [asoc:topic/rt5645 22/22] sound/soc/codecs/rt5645.c:3230:27: warning: passing argument 1 of 'device_property_read_bool' from incompatible pointer type
by kbuild test robot 08 Jun '15

08 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5645 head: d35b41b504096eba03a3b2d58e27ba2a2806f9de commit: d35b41b504096eba03a3b2d58e27ba2a2806f9de [22/22] ASoC: rt5645: add device tree support config: xtensa-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d35b41b504096eba03a3b2d58e27ba2a2806f9de # save the attached .config to linux build tree make.cross ARCH=xtensa All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_parse_dt': >> sound/soc/codecs/rt5645.c:3230:27: warning: passing argument 1 of 'device_property_read_bool' from incompatible pointer type rt5645->pdata.in2_diff = device_property_read_bool(np, ^ In file included from include/linux/of.h:26:0, from include/linux/i2c.h:33, from sound/soc/codecs/rt5645.c:17: include/linux/property.h:74:20: note: expected 'struct device *' but argument is of type 'struct device_node *' static inline bool device_property_read_bool(struct device *dev, ^ >> sound/soc/codecs/rt5645.c:3232:2: warning: passing argument 1 of 'device_property_read_u32' from incompatible pointer type device_property_read_u32(np, ^ In file included from include/linux/of.h:26:0, from include/linux/i2c.h:33, from sound/soc/codecs/rt5645.c:17: include/linux/property.h:92:19: note: expected 'struct device *' but argument is of type 'struct device_node *' static inline int device_property_read_u32(struct device *dev, ^ sound/soc/codecs/rt5645.c:3234:2: warning: passing argument 1 of 'device_property_read_u32' from incompatible pointer type device_property_read_u32(np, ^ In file included from include/linux/of.h:26:0, from include/linux/i2c.h:33, from sound/soc/codecs/rt5645.c:17: include/linux/property.h:92:19: note: expected 'struct device *' but argument is of type 'struct device_node *' static inline int device_property_read_u32(struct device *dev, ^ sound/soc/codecs/rt5645.c:3236:2: warning: passing argument 1 of 'device_property_read_u32' from incompatible pointer type device_property_read_u32(np, ^ In file included from include/linux/of.h:26:0, from include/linux/i2c.h:33, from sound/soc/codecs/rt5645.c:17: include/linux/property.h:92:19: note: expected 'struct device *' but argument is of type 'struct device_node *' static inline int device_property_read_u32(struct device *dev, ^ vim +/device_property_read_bool +3230 sound/soc/codecs/rt5645.c 3224 }, 3225 { } 3226 }; 3227 3228 static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device_node *np) 3229 { > 3230 rt5645->pdata.in2_diff = device_property_read_bool(np, 3231 "realtek,in2-differential"); > 3232 device_property_read_u32(np, 3233 "realtek,dmic1-data-pin", &rt5645->pdata.dmic1_data_pin); 3234 device_property_read_u32(np, 3235 "realtek,dmic2-data-pin", &rt5645->pdata.dmic2_data_pin); --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

1 0

[alsa-devel] [PATCH v3 1/3] ASoC: rt5645: make RT5650_TDM_CTRL_4 readable
by Bard Liao 08 Jun '15

08 Jun '15

Register RT5650_TDM_CTRL_4(0x7A) is readable and used for mixer setting. It should be added in rt5645_readable_register function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index aaede45..d12da40 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -349,6 +349,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: case RT5645_TDM_CTRL_3: + case RT5650_TDM_CTRL_4: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH v2] ASoC: wm8741: Allow master clock switching
by Sergej Sawazki 08 Jun '15

08 Jun '15

The set of supported sample rates depends on the master clock supplied to the codec. Allow the machine driver to set the required master clock in hw_params(). Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Charles, Mark, thanks for your comments. This patch has been tested on a board with two master clocks, 22.5792MHz (for 44.1kHz, 88.2kHz, ...) and 24.576MHz (for 48kHz, 96kHz, ...). Changes since v1 (Suggested by Charles Keepax): - Do not entirely delete the startup() method, just don't set any constraints if there is no SYSCLK configured. - Clear the SYSCLK in set_dai_sysclk() if freq is 0. Many thanks for your time and effort, Sergej sound/soc/codecs/wm8741.c | 61 ++++++++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 09ff01f..b346237 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -125,18 +125,6 @@ static const struct snd_soc_dapm_route wm8741_dapm_routes[] = { { "VOUTRN", NULL, "DACR" }, }; -static struct { - int value; - int ratio; -} lrclk_ratios[WM8741_NUM_RATES] = { - { 1, 128 }, - { 2, 192 }, - { 3, 256 }, - { 4, 384 }, - { 5, 512 }, - { 6, 768 }, -}; - static const unsigned int rates_11289[] = { 44100, 88200, }; @@ -209,25 +197,16 @@ static const struct snd_pcm_hw_constraint_list constraints_36864 = { .list = rates_36864, }; - static int wm8741_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC - enforce this. - */ - if (!wm8741->sysclk) { - dev_err(codec->dev, - "No MCLK configured, call set_sysclk() on init\n"); - return -EINVAL; - } - - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - wm8741->sysclk_constraints); + if (wm8741->sysclk) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8741->sysclk_constraints); return 0; } @@ -241,17 +220,24 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; - /* Find a supported LRCLK ratio */ - for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { - if (wm8741->sysclk / params_rate(params) == - lrclk_ratios[i].ratio) + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8741->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init or in hw_params\n"); + return -EINVAL; + } + + /* Find a supported LRCLK rate */ + for (i = 0; i < wm8741->sysclk_constraints->count; i++) { + if (wm8741->sysclk_constraints->list[i] == params_rate(params)) break; } - /* Should never happen, should be handled by constraints */ - if (i == ARRAY_SIZE(lrclk_ratios)) { - dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", - wm8741->sysclk / params_rate(params)); + if (i == wm8741->sysclk_constraints->count) { + dev_err(codec->dev, "LRCLK %d unsupported with MCLK %d\n", + params_rate(params), wm8741->sysclk); return -EINVAL; } @@ -274,8 +260,8 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", - params_width(params)); + dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", + params_width(params), params_rate(params)); snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); return 0; @@ -290,6 +276,11 @@ static int wm8741_set_dai_sysclk(struct snd_soc_dai *codec_dai, dev_dbg(codec->dev, "wm8741_set_dai_sysclk info: freq=%dHz\n", freq); switch (freq) { + case 0: + wm8741->sysclk_constraints = NULL; + wm8741->sysclk = freq; + return 0; + case 11289600: wm8741->sysclk_constraints = &constraints_11289; wm8741->sysclk = freq; -- 1.9.1

3 2

[alsa-devel] [PATCH v3 0/9] ASoC: tas2552: Fixes, cleanups and improvements
by Peter Ujfalusi 08 Jun '15

08 Jun '15

Hi, Changes since v02: - Dropped already applied patches from v02 - Fix bit shift in the PLL configuration patch (1/9) and add macro for it for readability - New patches since v02: - fix module removing and loading issue by calling pm_runtime_disable when the module is removed - Remove extra semicolon - Consitent naming for 'struct tas2552_data': tas_data -> tas2552 - Binding document update regarding to the added header file for clock selection Changes since v01: - missing patch found. It was merged with the previous patch while fixing checkpatch errors (patch 19 got squashed to patch 18) - Sorry for this. I have used the TAS2552-EVM to verify that my setup is in working condition for a work on other audio product. During the bringup of the EVM I have faced some issues and limitation with the driver. The result is this series against the tas2552 driver. Notable changes: - Fixes for kernel crashes - The enable-gpio was not optional as it was documented in the DTS - Register bit definitions were not correct in several places - Adding recommended startup/stop sequence - Handling of sample formats the driver claims to support - PLL configuration fixes and reference clock selection - TDM support - Gain control fix and control to select DIN source Regards, Peter --- Peter Ujfalusi (9): ASoC: tas2552: Correct the PLL configuration ASoC: tas2552: Add control for selecting DIN source ASoC: tas2552: Correct Output Data register usage ASoC: tas2552: Correct Boost Auto-Pass Through Control register usage ASoC: tas2552: Code, define alignment changes for uniformity ASoC: tas2552: Call pm_runtime_disable when the module is removed ASoC: tas2552: Remove unneeded semicolon ASoC: tas2552: Use consistent name for 'struct tas2552_data' ASoC: tas2552: Update DT binding document regarding clkock configuration .../devicetree/bindings/sound/tas2552.txt | 6 + sound/soc/codecs/tas2552.c | 173 ++++++++++++++------- sound/soc/codecs/tas2552.h | 61 ++++---- 3 files changed, 155 insertions(+), 85 deletions(-) -- 2.4.2

2 10

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Logic low for inactive output slots
by Peter Ujfalusi 08 Jun '15

08 Jun '15

From: Misael Lopez Cruz <misael.lopez(a)ti.com> The default state when serializers are in inactive slots is Hi-Z. In some cases, there are no additional components driving the data lines to a safe state so they might have noise. While in inactive slots, the McASP AXR pins configured as outputs can be driven low through the serializer pin drive mode setting (DISMOD) to prevent such noise. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 ++ sound/soc/davinci/davinci-mcasp.h | 5 ++++- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d79349434a9a..26cbc213b8cf 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -686,6 +686,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, if (mcasp->serial_dir[i] == TX_MODE && tx_ser < max_active_serializers) { mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + DISMOD_LOW, DISMOD_MASK); tx_ser++; } else if (mcasp->serial_dir[i] == RX_MODE && rx_ser < max_active_serializers) { diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 79dc511180bf..a3be108a8c17 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -215,7 +215,10 @@ * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits */ #define MODE(val) (val) -#define DISMOD (val)(val<<2) +#define DISMOD_3STATE (0x0) +#define DISMOD_LOW (0x2 << 2) +#define DISMOD_HIGH (0x3 << 2) +#define DISMOD_MASK DISMOD_HIGH #define TXSTATE BIT(4) #define RXSTATE BIT(5) #define SRMOD_MASK 3 -- 2.4.2

2 1

Re: [alsa-devel] [PATCH v4 00/18] new MFD LoCoMo driver set
by Jingoo Han 08 Jun '15

08 Jun '15

On Thu, 28 May 2015 22:07:40 +0300, Dmitry Eremin-Solenikov wrote: >2015-05-28 14:58 GMT+03:00 Lee Jones wrote: >> On Wed, 27 May 2015, Dmitry Eremin-Solenikov wrote: >> >>> 2015-05-27 13:27 GMT+03:00 Lee Jones <lee.jones@xxxxxxxxxx>: >>> > On Mon, 25 May 2015, Dmitry Eremin-Solenikov wrote: >>> > >>> >> LoCoMo is a GA used on Sharp Zaurus SL-5x00. Current driver does has >>> >> several design issues (special bus instead of platform bus, doesn't use >>> >> mfd-core, etc). >>> >> >>> >> Implement 'core' parts of locomo support as an mfd driver. >>> >> >>> >> Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@xxxxxxxxx> >>> >> --- >>> > >>> > These things are much easier to re-review if you keep a change-log >>> > here. >>> > >>> >> drivers/mfd/Kconfig | 10 ++ >>> >> drivers/mfd/Makefile | 1 + >>> >> drivers/mfd/locomo.c | 336 +++++++++++++++++++++++++++++++++++++++++++++ >>> >> include/linux/mfd/locomo.h | 169 +++++++++++++++++++++++ >>> >> 4 files changed, 516 insertions(+) >>> >> create mode 100644 drivers/mfd/locomo.c >>> >> create mode 100644 include/linux/mfd/locomo.h >>> > >>> > Looks good now though: >>> > Acked-by: Lee Jones <lee.jones@xxxxxxxxxx> >>> > >>> > How do you plan for this to be merged? >>> >>> I'd like for this to be merged either through mfd tree or through the >>> arm-soc (or rmk's ARM) tree. >>> Some of the respective maintainers already gave their blessing for >>> this patchset to be merged >>> not through their subsystem trees. >> >> I can take it through MFD. Once you have all the required Acks, >> resubmit the set with them all applied with a note in the cover-letter >> to that affect and I'll apply. > >Thank you. This is what I still miss: > >* Russell, could you please review sa1100 parts? >* Jingoo, could you please review the backlight part? For the following two patches, [PATCH v4 06/18] video: backlight: add new locomo backlight driver [PATCH v4 07/18] video: lcd: add LoCoMo LCD driver Acked-by: Jingoo Han <jingoohan1(a)gmail.com> Pleased send the patches to 'jingoohan1(a)gmail.com', instead of 'jg1.han(a)samsung.com'. Best regards, Jingoo Han >* Mark, same question for the updated SPI driver. >* Linus, can I assume that yours 'Reviewed-by' mean > 'Acked-by' for the GPIO-related parts?

1 0

[alsa-devel] [RFC][PATCH 00/11] improve support for DM1500/BeBoBv3
by Takashi Sakamoto 08 Jun '15

08 Jun '15

Hi, This patchset is to improve support for DM1500 and BeBoBv3; DM1500 is an ASIC which ArchWave produced. BeBoB is an interface protocol for DM series firmware. Currently, Some devices are reported not to work with current ALSA BeBoB driver. [FFADO-user] Support Behringer UFX1604 http://sourceforge.net/p/ffado/mailman/message/32846563/ [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/090967.html They commonly install BeBoB version 3 firmware. As long as seeing these devices, they may have below quirks comparing to version 1 and 2. Furthermore, DM1500 seems to have its own quirks. The first half of this patchset is for improvement of clock detection, the rest is for quirks. The latter is tested by Kim Tore Jensen (Thanks!), while the first half is not widely tested. I hope someone to test it till opening next merge window (but just one week rest!). I push these patches into my own repository. It's DKMS available. https://github.com/takaswie/snd-firewire-improve/tree/bebob3 Regards Takashi Sakamoto (11): ALSA: bebob: improve signal mode detection for clock source ALSA: bebob: apply new enumerator to normalize the type of clock source ALSA: bebob: preparation for replacing string literals by normalized representation for model-dependent structures ALSA: bebob: use normalized representation for the type of clock source ALSA: bebob: obsolete string literal expression for clock source ALSA: bebob: add SYT-Match support ALSA: bebob: add 'version' member for BeBoB protocol version ALSA: bebob: expand timeout for DM1500 quirk ALSA: bebob: loosen up severity of checking continuity for BeBoB v3 quirk ALSA: bebob: keep duplex streams always to keep internal multiplexer properly ALSA: bebob: add support for Behringer FCA 610/1616 sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 14 ++- sound/firewire/bebob/bebob.h | 17 ++-- sound/firewire/bebob/bebob_focusrite.c | 33 +++++--- sound/firewire/bebob/bebob_maudio.c | 23 +++-- sound/firewire/bebob/bebob_midi.c | 8 +- sound/firewire/bebob/bebob_pcm.c | 14 +-- sound/firewire/bebob/bebob_proc.c | 22 ++--- sound/firewire/bebob/bebob_stream.c | 150 ++++++++++++++++++++++++--------- sound/firewire/bebob/bebob_terratec.c | 30 +++++-- sound/firewire/bebob/bebob_yamaha.c | 20 ++++- 11 files changed, 233 insertions(+), 99 deletions(-) -- 2.1.4

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[alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Nicolae Rosia 08 Jun '15

08 Jun '15

Hello, I can't get 4.0/5.1 sound working on my laptop [0] [1] with Fedora 22. With Linux 4.0 there is no output on jacks but with 4.1-rc6 I'm getting sound only on one output. Switching between pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-stereo and pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-surround-40 only boosts the volume on the first output. Attached is dmesg, lsmod, result of lspci -vv and pacmd list-cards and list-sinks and output of various information from /proc/asound while runnig 4.1-rc6. Let me know if you need more information or how I can help. Best regards, Nicolae Rosia [rosia@localhost ~]$ cat /proc/asound/devices 1: : sequencer 2: [ 0] : control 3: [ 0- 3]: digital audio playback 4: [ 0- 7]: digital audio playback 5: [ 0- 8]: digital audio playback 6: [ 0- 0]: hardware dependent 7: [ 1] : control 8: [ 1- 0]: digital audio playback 9: [ 1- 0]: digital audio capture 10: [ 1- 1]: digital audio playback 11: [ 1- 0]: hardware dependent 33: : timer [rosia@localhost ~]$ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xeda14000 irq 33 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xeda10000 irq 32 [rosia@localhost ~]$ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel [rosia@localhost ~]$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.1.0-0.rc6.git2.1.fc23.x86_64. [0] http://www.emag.ro/laptop-gaming-asus-cu-procesor-intelr-coretm-i7-4710hq-2… [1] https://www.asus.com/Notebooks_Ultrabooks/ASUS_ROG_G751JM/specifications/

1 1

[alsa-devel] [PATCH v2 00/23] ASoC: tas2552: Fixes, cleanups and improvements
by Peter Ujfalusi 08 Jun '15

08 Jun '15

Hi, Changes since v01: - missing patch found. It was merged with the previous patch while fixing checkpatch errors (patch 19 got squashed to patch 18) - Sorry for this. I have used the TAS2552-EVM to verify that my setup is in working condition for a work on other audio product. During the bringup of the EVM I have faced some issues and limitation with the driver. The result is this series against the tas2552 driver. Notable changes: - Fixes for kernel crashes - The enable-gpio was not optional as it was documented in the DTS - Register bit definitions were not correct in several places - Adding recommended startup/stop sequence - Handling of sample formats the driver claims to support - PLL configuration fixes and reference clock selection - TDM support - Gain control fix and control to select DIN source Regards, Peter --- Peter Ujfalusi (23): ASoC: tas2552: Make the enable-gpio really optional ASoC: tas2552: Fix kernel crash when the codec is loaded but not part of a card ASoC: tas2552: Fix kernel crash caused by wrong kcontrol entry ASoC: tas2552: Correct PDM configuration register bit definitions ASoC: tas2552: Correct CFG1 register bit definitions ASoC: tas2552: Simplify the tas2552_mute function ASoC: tas2552: Simplify and reverse the functionality of tas2552_sw_shutdown ASoC: tas2552: Rename mclk parameter to pll_clkin to match with the datasheet DT/sound: bindings header file for tas2552 codec ASoC: tas2552: Add support for pll and pdm source clock selection ASoC: tas2552: Correct dai format support ASoC: tas2552: Correct and clean up data format and BCLK/WCLK direction ASoC: tas2552: Add TDM support ASoC: tas2552: Clean up the Digital - Analog DAPM route definition ASoC: tas2552: Correct the Speaker Driver Playback Volume (PGA_GAIN) ASoC: tas2552: Implement startup/stop sequence as per TRM ASoC: tas2552: Add support for word length configuration ASoC: tas2552: Configure the WCLK frequency based on the stream ASoC: tas2552: Correct the PLL configuration ASoC: tas2552: Add control for selecting DIN source ASoC: tas2552: Correct Output Data register usage ASoC: tas2552: Correct Boost Auto-Pass Through Control register usage ASoC: tas2552: Code, define alignment changes for uniformity include/dt-bindings/sound/tas2552.h | 18 ++ sound/soc/codecs/tas2552.c | 435 +++++++++++++++++++++++++++--------- sound/soc/codecs/tas2552.h | 149 ++++++------ 3 files changed, 431 insertions(+), 171 deletions(-) create mode 100644 include/dt-bindings/sound/tas2552.h -- 2.4.2

2 33

[alsa-devel] dshare plugin issue in alsa-lib.1.0.27
by Srinivasan S 08 Jun '15

08 Jun '15

Dear Clemens, Thanks a lot for all your suggestions w.r.t the thread [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Could you please provide some suggestions or guidance w.r.t the below issue Now am using linux kernel version 3.14 with alsa-lib.1.0.27 I feel dshare plugin is not working now with this linux kernel & alsa-lib version , When I try to run theaplay -D VOUTR NeroSoundTrax_test8_PCM_Mono_VBR_16SS_48000Hz.wav I don't get any errors, but it seems that audio is playing but I don't hear any audio But when I run the same in linux kernel version 3.12 with alsa-lib.1.0.26, I hear the audio output wav @am335x-evm:/# aplay -D VOUTR NeroSoundTrax_test8_PCM_Mono_VBR_16SS_48000Hz.w [ 876.513092] DAVINCIIIIIIIIII UDA134XXXXX SYSCLK=12288000 [ 876.518771] DAVINCIIIIIIIIII UDA134XXXXX BCLK FREQQ=1536000 [ 876.524777] DAVINCIIIIIIIIII UDA134XXXXX SYSCLK/BCLK_FREQ =8 [ 876.530813] uda134x_hw_params CLOCKS uda134x_hw_params uda134x->sysclk: 12288000, params_rate(params):48000 [ 876.541191] uda134x_hw_params FORMATS uda134x_hw_params dai_fmt: 16385, params_format:2 [ 876.549720] uda134x_hw_params FORMATS uda134x_hw_params uda134x->sysclk / params_rate(params) 256 [ 876.559165] UDA1345TSSSSSSSSSSSS SYSCLK / fs ratio is 256 [ 876.564929] uda134x_hw_params dai_fmt: 16385, params_format:2 [ 876.571029] UDA1345TSSSSSSSSSSSS FORMAT SND_SOC_DAIFMT_I2S [ 876.577006] ENTERED davinci_config_channel_size davinci_config_channel_size: tx_rotate = 4 [ 876.585812] ENTERED davinci_config_channel_size davinci_config_channel_size: MASK= 65535 [ 876.598147] uda134x_unnnnnnnnmuteeeeeeeeee uda134x_mute mute: 0 [ 876.605173] davinci_mcasp_starttttttttttttttttttttt SNDRV_PCM_STREAM_PLAYBACK Playing WAVE 'NeroSoundTrax_test8_PCM_Mono_VBR_16SS_48000Hz.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono I have tired the following: Firstly, I have tried with basic arecord & aplay in linux kernel version 3.14 that works, without the above asound.conf I have even tried with combination's of linux kernel version 3.14 & alsa-lib-1.0.26 & linux kernel verison 3.12 with alsa-lib-1.0.27 that didn't work, with the above asound.conf Now I feel, it seems to be there is an dependency issue with alsa-lib.1.0.27 So, could you please try from your end with linux kernel version 3.14 with the above asound.conf with above alsalib version & please let me know whether it works or not? If not, Could you please let me know how to get this thing working with linux kernel 3.14 (as all my development activities are being done in linux 3.14) Kindly do the needful as early as possible, as am stuck with this from many day's Many Many Thanks in advance,

1 0

[alsa-devel] [PATCH] usb-audio: add native DSD support for JLsounds I2SoverUSB
by Jurgen Kramer 08 Jun '15

08 Jun '15

This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2917534..04c285c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1266,8 +1266,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; - /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ - case USB_ID(0x20b1, 0x2009): + + case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ + case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.1.0

2 1

[alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Łukasz Wojniłowicz 07 Jun '15

07 Jun '15

Hi all, My problem is that when I set channel output to "Analog output surround 5.1" (or 2.1 or 4.1), I hear no sound in front speakers, but in case of 5.1 or 4.1 I can hear sound in rear speakers. I use kernel 3.19.5 and it didn't happen in earlier kernels that is 3.18 and in 3.17 for sure. My configuration is: OS: Fedora 21 64 bit SoundCard: ALC888 Model name: acer-aspire-4930g Laptop: Acer Aspire 5930g Speakers: external 5.1 with 3 jacks Sound channel configuration module is standard configuration module from KDE. Additional info: 1) When channel output is 5.1 and I unplug green jack and leave two others jacks connected then I can hear sound in laptop internal speakers. 2) Earlier (kernel 3.18 and 3.17) I didn't have possibility to choose 2.1 configuration and additionally to 4.1 configuration I could also choose 4.0 configuration. Thanks for looking into problem. Cheers LW

3 3

[alsa-devel] [PATCH v2 0/2] Update ALSA driver to use media controller API
by Shuah Khan 06 Jun '15

06 Jun '15

This patch series updates ALSA driver to use media controller API to share tuner with DVB and V4L2 drivers that control AU0828 media device. Two new interfaces are added to media controller API to enable creating media device as a device resource. This allows creating media device as a device resource on the main struct device that is common to all drivers that control a single media hardware and share resources on it. Drivers then can find the common media device to register entities and manage links, and pipelines. Tested with and without CONFIG_MEDIA_CONTROLLER enabled. Tested tuner entity doesn't exist case as au0828 v4l2 driver is the one that will create the tuner when it gets updated to use media controller API. Please note that au0828 updates media controller are necessary before the resource sharing can work across ALSA and au0828 dvb and v4l2 drivers. This work is in progress and another developer is working on it. Changes since v1: - Moved media specific code into new header and source file - Created a new structure for media controller specific fields - Added a new define USE_MEDIA_CONTROLLER to add media specific code. This new define is defined only when Media Controller support is enabled. New media specific code is active only when this define is defined. - Moved media_deviced_delete() check under !was_shutdown block - Removed return check for media_device_init() - Changed new Media Controller interface names for clarity. Now they are named media_device_get_devres() and media_device_find_devres(). - Calling media_entity_pipeline_stop() on a pipe that isn't active appears to be safe. This needs further testing once AU0828 media controller patches are ready. Shuah Khan (2): media: new media controller API for device resource support sound/usb: Update ALSA driver to use media controller API drivers/media/media-device.c | 33 ++++++++ include/media/media-device.h | 2 + sound/usb/Makefile | 3 +- sound/usb/card.c | 5 ++ sound/usb/card.h | 1 + sound/usb/media.c | 181 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 40 ++++++++++ sound/usb/pcm.c | 10 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 ++- sound/usb/stream.c | 3 + sound/usb/usbaudio.h | 1 + 12 files changed, 286 insertions(+), 3 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

2 7

[alsa-devel] [RFC PATCH] ASoC: wm8741: Allow master clock switching
by Sergej Sawazki 05 Jun '15

05 Jun '15

The set of supported sample rates depends on the master clock supplied to the codec. Allow the machine driver to set the required master clock in hw_params(). Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- This patch has been tested on a board with two master clocks, 22.5792MHz (for 44.1kHz, 88.2kHz, ...) and 24.576MHz (for 48kHz, 96kHz, ...). sound/soc/codecs/wm8741.c | 63 +++++++++++++---------------------------------- 1 file changed, 17 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 09ff01f..47c5577 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -125,18 +125,6 @@ static const struct snd_soc_dapm_route wm8741_dapm_routes[] = { { "VOUTRN", NULL, "DACR" }, }; -static struct { - int value; - int ratio; -} lrclk_ratios[WM8741_NUM_RATES] = { - { 1, 128 }, - { 2, 192 }, - { 3, 256 }, - { 4, 384 }, - { 5, 512 }, - { 6, 768 }, -}; - static const unsigned int rates_11289[] = { 44100, 88200, }; @@ -209,29 +197,6 @@ static const struct snd_pcm_hw_constraint_list constraints_36864 = { .list = rates_36864, }; - -static int wm8741_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC - enforce this. - */ - if (!wm8741->sysclk) { - dev_err(codec->dev, - "No MCLK configured, call set_sysclk() on init\n"); - return -EINVAL; - } - - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - wm8741->sysclk_constraints); - - return 0; -} - static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -241,17 +206,24 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; - /* Find a supported LRCLK ratio */ - for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { - if (wm8741->sysclk / params_rate(params) == - lrclk_ratios[i].ratio) + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8741->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on hw_params\n"); + return -EINVAL; + } + + /* Find a supported LRCLK rate */ + for (i = 0; i < wm8741->sysclk_constraints->count; i++) { + if (wm8741->sysclk_constraints->list[i] == params_rate(params)) break; } - /* Should never happen, should be handled by constraints */ - if (i == ARRAY_SIZE(lrclk_ratios)) { - dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", - wm8741->sysclk / params_rate(params)); + if (i == wm8741->sysclk_constraints->count) { + dev_err(codec->dev, "LRCLK %d unsupported with MCLK %d\n", + params_rate(params), wm8741->sysclk); return -EINVAL; } @@ -274,8 +246,8 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d", - params_width(params)); + dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", + params_width(params), params_rate(params)); snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); return 0; @@ -403,7 +375,6 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8741_dai_ops = { - .startup = wm8741_startup, .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, .set_fmt = wm8741_set_dai_fmt, -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: wm8731: initialize the hardware when loading the codec driver
by Sergey Kiselev 05 Jun '15

05 Jun '15

From: Sergey Kiselev <sergey.kiselev(a)intel.com> This patch moves the requesting supplies, hardware reset and initialization from wm8731_probe to wm8731_i2c_probe and wm8731_spi_probe. So that the codec hardware is initialized when loading the codec driver, and not when loading the machine driver. This avoids unnecesary hardware resets and re-initializations when re-loading the machine driver. Signed-off-by: Sergey Kiselev <sergey.kiselev(a)intel.com> --- sound/soc/codecs/wm8731.c | 80 ++++++++++++++++++++++++++++++----------------- 1 file changed, 52 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 915ea11..28401f6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -84,7 +84,7 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg) return reg <= WM8731_RESET; } -#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) +#define wm8731_reset(m) regmap_write(m, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -571,44 +571,54 @@ static struct snd_soc_dai_driver wm8731_dai = { .symmetric_rates = 1, }; -static int wm8731_probe(struct snd_soc_codec *codec) +static int wm8731_request_supplies(struct device *dev, + struct wm8731_priv *wm8731) { - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + dev_err(dev, "Failed to request supplies: %d\n", ret); return ret; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + dev_err(dev, "Failed to enable supplies: %d\n", ret); return ret; } - ret = wm8731_reset(codec); + return 0; +} + +static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) +{ + int ret = 0; + + ret = wm8731_reset(wm8731->regmap); if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + dev_err(dev, "Failed to issue reset: %d\n", ret); goto err_regulator_enable; } - snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Clear POWEROFF, keep everything else disabled */ + regmap_write(wm8731->regmap, WM8731_PWR, 0x7f); /* Latch the update bits */ - snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); - snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LOUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_ROUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); + regmap_update_bits(wm8731->regmap, WM8731_APANA, 0x8, 0); + + regcache_mark_dirty(wm8731->regmap); /* Regulators will have been enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -621,19 +631,7 @@ err_regulator_enable: return ret; } -/* power down chip */ -static int wm8731_remove(struct snd_soc_codec *codec) -{ - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { - .probe = wm8731_probe, - .remove = wm8731_remove, .set_bias_level = wm8731_set_bias_level, .suspend_bias_off = true, @@ -690,6 +688,12 @@ static int wm8731_spi_probe(struct spi_device *spi) mutex_init(&wm8731->lock); + spi_set_drvdata(spi, wm8731); + + ret = wm8731_request_supplies(&spi->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -698,7 +702,9 @@ static int wm8731_spi_probe(struct spi_device *spi) return ret; } - spi_set_drvdata(spi, wm8731); + ret = wm8731_hw_init(&spi->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); @@ -712,7 +718,12 @@ static int wm8731_spi_probe(struct spi_device *spi) static int wm8731_spi_remove(struct spi_device *spi) { + struct wm8731_priv *wm8731 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + return 0; } @@ -754,6 +765,12 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, mutex_init(&wm8731->lock); + i2c_set_clientdata(i2c, wm8731); + + ret = wm8731_request_supplies(&i2c->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -762,7 +779,9 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, return ret; } - i2c_set_clientdata(i2c, wm8731); + ret = wm8731_hw_init(&i2c->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); @@ -776,7 +795,12 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, static int wm8731_i2c_remove(struct i2c_client *client) { + struct wm8731_priv *wm8731 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + return 0; } -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: rt5645: Init jack_detect_work before registering irq
by Nicolas Boichat 05 Jun '15

05 Jun '15

Prevents frequent panic on boot, if the irq handler rt5645_irq gets called before the workqueue rt5645_jack_detect_work is initialized. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Applies on top of broonie/sound.git/for-next (dac48b88824392de29a437607b7dbf844bb66f74) sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e9544c6..d5f0f56 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3416,6 +3416,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING @@ -3434,8 +3436,6 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n"); } - INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, rt5645_dai, ARRAY_SIZE(rt5645_dai)); } -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Avoid recursive locking in the DEADLOCK detector
by Oder Chiou 05 Jun '15

05 Jun '15

The patch is for to avoid recursive locking in the DEADLOCK detector. In the driver, it encountered the warnning message of the recursive locking in the function "regmap_lock_mutex". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 31d969a..ee5b570 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4998,6 +4998,22 @@ static const struct regmap_config rt5677_regmap_physical = { .num_ranges = ARRAY_SIZE(rt5677_ranges), }; +static void rt5677_regmap_lock_mutex(void *__lock) +{ + struct mutex *lock = __lock; + + mutex_lock(lock); +} + +static void rt5677_regmap_unlock_mutex(void *__lock) +{ + struct mutex *lock = __lock; + + mutex_unlock(lock); +} + +static struct mutex rt5677_regmap_lock; + static const struct regmap_config rt5677_regmap = { .reg_bits = 8, .val_bits = 16, @@ -5009,6 +5025,9 @@ static const struct regmap_config rt5677_regmap = { .readable_reg = rt5677_readable_register, .reg_read = rt5677_read, .reg_write = rt5677_write, + .lock = rt5677_regmap_lock_mutex, + .unlock = rt5677_regmap_unlock_mutex, + .lock_arg = &rt5677_regmap_lock, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5677_reg, @@ -5206,6 +5225,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return ret; } + mutex_init(&rt5677_regmap_lock); + regmap_read(rt5677->regmap, RT5677_VENDOR_ID2, &val); if (val != RT5677_DEVICE_ID) { dev_err(&i2c->dev, -- 1.9.1

4 5

[alsa-devel] [PATCH v2] ASoC: dapm: fix prefixed stream name matching in snd_soc_dapm_link_dai_widgets
by Arnaud Mouiche 05 Jun '15

05 Jun '15

commit "ASoC: dapm: Modify widget stream name according to prefix" fixed a part of the problem when dai_w->name and w->sname are equal before being prefixed. Remains the case where w->sname is only a sub string of dai_w->name. For example, TLV320AIC3x codec defines the widget SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0) In snd_soc_dapm_link_dai_widgets, we will failed to match "[prefix] Left Playback" with "[prefix] Playback". Since dai_w and w necessarily belongs to the same dapm when strstr() is called, they necessarily have the same prefix which allow us to compare 'w->sname + prefix_len' with 'dai_w->name + prefix_len' directly without length check. Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/soc-dapm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e19a676..fc4408b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3392,6 +3392,9 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { + const char *prefix = soc_dapm_prefix(dai_w->dapm); + size_t prefix_len = prefix ? strlen(prefix)+1 : 0; + switch (dai_w->id) { case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: @@ -3415,7 +3418,8 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || + !strstr(w->sname + prefix_len, dai_w->name + prefix_len)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: dapm: fix prefixed stream name matching in snd_soc_dapm_link_dai_widgets
by Arnaud Mouiche 05 Jun '15

05 Jun '15

commit "ASoC: dapm: Modify widget stream name according to prefix" fixed a part of the problem when dai_w->name and w->sname are equal before being prefixed. Remains the case where w->sname is only a sub string of dai_w->name. For example, TLV320AIC3x codec defines the widget SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0) In snd_soc_dapm_link_dai_widgets, we will failed to match "[prefix] Left Playback" with "[prefix] Playback". Since dai_w and w necessarily belongs to the same dapm when strstr() is called, they necessarily have the same prefix which allow us to compare 'w->sname + prefix_len' with 'dai_w->name + prefix_len' directly without length check. Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/soc-dapm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e19a676..7194df4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3392,6 +3392,9 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { + const char *prefix = soc_dapm_prefix(dai_w->dapm); + size_t prefix_len = prefix ? strlen(prefix)+1 : 0; + switch (dai_w->id) { case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: @@ -3416,6 +3419,8 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) } if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || + !strstr(w->sname + prefix_len, dai_w->name + prefix_len)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- 1.9.1

2 1

Re: [alsa-devel] (alsa-lib)pcm_hw.c: Invalid PCM state for chmap_get: OPEN
by Raymond Yau 05 Jun '15

05 Jun '15

>> >> Have you tested surround21? > > Yes, front speakers work fine, however LFE doesn't seem to work. Instead when I try to test LFE (from sound UI) I have weird noise in one of rear speakers. There are no such problems with any other mode though (I see that officially only 5.1 mode is supported, however 4.1, 5.0 and 5.1 all work fine), I decided to leave 2.1 mode for now, not sure whose fault is that. http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=3c4a22ea49a881cd… It is strange that pcm state is open ( 0.089| 0.001) D: [pulseaudio] alsa-mixer.c: Looking at profile output:analog-surround-21 ( 0.089| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for playback on Analog Surround 2.1 (analog-surround-21) ( 0.089| 0.000) D: [pulseaudio] alsa-util.c: Trying surround21:1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.089| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: Invalid PCM state for chmap_get: OPEN ( 0.089| 0.000) D: [pulseaudio] alsa-util.c: Managed to open surround21:1 ( 0.089| 0.000) I: [pulseaudio] alsa-util.c: Disabling tsched mode since BATCH flag is set ( 0.089| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 1213 ms ( 0.090| 0.000) D: [pulseaudio] alsa-util.c: Set buffer size first (to 4797 samples), period size second (to 1199 samples). ( 0.090| 0.000) I: [pulseaudio] alsa-util.c: Device surround21:1 doesn't support 44100 Hz, changed to 48000 Hz. ( 0.090| 0.000) D: [pulseaudio] alsa-mixer.c: Profile output:analog-surround-21 supported.

1 0

[alsa-devel] [asoc:topic/tas2552 1/8] sound/soc/codecs/tas2552.c:493:27-28: Unneeded semicolon
by kbuild test robot 05 Jun '15

05 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/tas2552 head: 16bd395259cf3e9966d40478891e0e610da109d4 commit: ea178d1456dcf88875d5edd148f2df8ea0de1794 [1/8] ASoC: tas2552: Make the enable-gpio really optional coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2552.c:493:27-28: Unneeded semicolon vim +493 sound/soc/codecs/tas2552.c 477 { 478 struct device *dev; 479 struct tas2552_data *data; 480 int ret; 481 int i; 482 483 dev = &client->dev; 484 data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); 485 if (data == NULL) 486 return -ENOMEM; 487 488 data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); 489 if (IS_ERR(data->enable_gpio)) { 490 if (PTR_ERR(data->enable_gpio) == -EPROBE_DEFER) 491 return -EPROBE_DEFER; 492 > 493 data->enable_gpio = NULL;; 494 } 495 496 data->tas2552_client = client; 497 data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); 498 if (IS_ERR(data->regmap)) { 499 ret = PTR_ERR(data->regmap); 500 dev_err(&client->dev, "Failed to allocate register map: %d\n", 501 ret); --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] [asoc:topic/zx 2/8] sound/soc/zte/zx296702-i2s.c:428:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 05 Jun '15

05 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/zx head: 69ccc50231eca9a57dc06a5514103c0f17ef402e commit: e5d4cd87800ce12c356e7eb571e565e839ab3a90 [2/8] ASoC: zx: Add ZTE zx296702 I2S DAI driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/zte/zx296702-i2s.c:428:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] Reflections on implementing Web MIDI on top of ALSA seq
by Adam Goode 05 Jun '15

05 Jun '15

Hi, Chrome 43 is now in beta, with Web MIDI support. (http://blog.chromium.org/2015/04/chrome-43-beta-web-midi-and-upgrading.html) The Linux (and Chrome OS) version of Chrome uses ALSA seq to implement Web MIDI. I wanted to reflect a bit on some of the issues I discovered while implementing this new API on top of seq. This is a little bit long, so the tl;dr is: * ALSA seq worked pretty well for implementing Web MIDI, but I need a way to get the card number for a given client, before my assumptions about the kernel break. Here is a quick overview of Web MIDI (http://www.w3.org/TR/webmidi/): - provides hotplug events for port connect/disconnect - incoming events are timestamped - outgoing events can be timestamped for future delivery by the system - timestamps are high precision, and use a monotonic clock - the port abstraction contains "name", "manufacturer", and "version" fields - open ports auto-reconnect if the port disappears and reappears, for both send and receive - input and output ports are distinct: there are no duplex ports - uses MIDI bytestreams (not an event structure), but requires full MIDI messages to be sent The initial prototype of Chrome's Web MIDI support was built on top of rawmidi. I rewrote it to use seq instead (to get multiple client and user client access). Here are my observations: - Getting the "manufacturer" is very hard from seq. I needed to join against udev data, which relies on the current implementation of ALSA in the kernel. These udev attributes are also used to match for transparent reconnect. - I could not use the kernel seq timestamp, either for send or receive, since I could not get it to just use a CLOCK_MONOTONIC offset. Instead, timestamping is done in userspace. - Hotplug was tricky, since I needed to listen to both seq events and udev events. I would prefer just to use udev, since I have to anyway. I definitely want to fix the kernel so that I can stop relying on a few implementation details. Here are the current assumptions I am making: - Card based clients start at 16. (I need to know which clients actually have cards) - Clients are in the same order as cards. - There is a 1:1 correspondence between clients and OPL3+rawmidi devices. One way to fix these assumptions would be to fully expose all seq events through sysfs. Right now, there is a seq device in sysfs but it has no information. Here are some ways that the kernel might be extended to simplify the code and/or fix bad assumptions: 1. Expose all seq clients+ports as kobjects, with a symlink to the card if it's a hardware device 2. Expose all subscriptions through sysfs 3. Add a simplified timestamp mechanism that just uses a configurable system clock, with at least CLOCK_MONOTONIC available (something like what sound/core/timer.c does with tread?) The first change would allow me to eliminate all the hardcoded assumptions about which cards correspond to which clients. I believe it would also allow me to drive the hotplug logic directly from udev, instead of both udev and seq. The second change is not strictly necessary, but would remove more need to react to seq events for some uses. The third change would allow me to use the kernel timestamp reliably, since right now I timestamp events with CLOCK_MONOTONIC in userspace. If I had to start today and create a new MIDI kernel interface for ALSA, it would look like this: not-quite-raw-midi: - hotplug events completely driven by sysfs - ability to interoperate with existing seq clients (for kernel and user clients) - non-exclusive access - reads and writes use a simple structured format, with timestamp and a midi message as bytes - no or minimal alsa-lib needed - one device file per port (no subdevices) - some mechanism for creating userspace clients, preferably with some number of key-value attributes (so I can report "manufacturer" and other things if the user client sets it) Thanks for reading these reflections. Comments are welcome. Adam

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[alsa-devel] [PATCH] ASoC: zte: Fix missing dev in devm operation
by Mark Brown 04 Jun '15

04 Jun '15

Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx296702-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c index 4a93bca232ff..11a0e46a1156 100644 --- a/sound/soc/zte/zx296702-spdif.c +++ b/sound/soc/zte/zx296702-spdif.c @@ -309,7 +309,7 @@ static int zx_spdif_probe(struct platform_device *pdev) struct zx_spdif_info *zx_spdif; int ret; - zx_spdif = devm_kzalloc(sizeof(*zx_spdif), GFP_KERNEL); + zx_spdif = devm_kzalloc(&pdev->dev, sizeof(*zx_spdif), GFP_KERNEL); if (!zx_spdif) return -ENOMEM; -- 2.1.4

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[alsa-devel] [PATCH v5 0/4] Support ZTE zx296702 SPDIF/I2S
by Jun Nie 04 Jun '15

04 Jun '15

Changes vs v4: - Fix devm_zalloc argument issue in SPDIF driver and static issue in i2s. Changes vs v3: - remove HDMI stuff and will add it in video side later - Polish SPDIF driver with devm_* Changes vs v2: - remove pcm dma driver and merge the related call into DAI driver - Add zx HDMI codec with direct HDMI function call - Other typo and minor fix Jun Nie (4): dt: Add documentation for the ZTE SPDIF controller dt: Add documentation for the ZTE I2S controller ASoC: zx: Add zx296702 SPDIF support ASoC: zx: Add ZTE zx296702 I2S DAI driver .../devicetree/bindings/sound/zte,zx-i2s.txt | 44 +++ .../devicetree/bindings/sound/zte,zx-spdif.txt | 28 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/zte/Kconfig | 17 + sound/soc/zte/Makefile | 2 + sound/soc/zte/zx296702-i2s.c | 437 +++++++++++++++++++++ sound/soc/zte/zx296702-spdif.c | 370 +++++++++++++++++ 8 files changed, 900 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt create mode 100644 sound/soc/zte/Kconfig create mode 100644 sound/soc/zte/Makefile create mode 100644 sound/soc/zte/zx296702-i2s.c create mode 100644 sound/soc/zte/zx296702-spdif.c -- 1.9.1

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[alsa-devel] Different codecs for playback and capture?
by Ricard Wanderlof 04 Jun '15

04 Jun '15

Hi, I'm developing a machine driver for a system where we have two different codecs for input and output, connected to the same I2S interface. Normally a machine driver specifies which codec the I2S interface is connected to, but in this case there should really be two different ones, one for record and one for playback. I'm working on how to put this together, and just thought I'd try and get some input before going too far down the wrong track. The options I see are: 1. Specify two different codecs in the machine driver. Since one can specify several dai_links this would seem to be doable, however, I'm not sure ALSA can handle two codecs sharing a single CPU DAI in this way? 2. Use two separate machine drivers, one for each codec? Again, sharing a single CPU DAI would seem to be an issue here? 3. Write a special codec driver that is a merge between the two codecs, one for input and one for output. Seems like a bit of a hack, but should be possible. Any spontaneous thoughts? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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