June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH] alsactl: Do not run udev rule before datadir is mounted
by David Henningsson 14 Jun '14

14 Jun '14

The 90-alsa-restore.rules udev rule is primarily meant to handle hotplugging, but sometimes it is also run before /usr is mounted, and alsactl restore depends on /usr/share/alsa being present. If /usr/share/alsa is not present, we're so early in the boot process that alsa-restore.service (or upstart equivalent) will run later on. BugLink: https://bugs.launchpad.net/bugs/1289730 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I'm not really sure of all quirks in the boot process, and I haven't tested the patch either, hence the RFC above. Do you think it makes sense? Should we also add @sbindir@ to the things to test for? alsactl/90-alsa-restore.rules.in | 2 +- alsactl/Makefile.am | 1 + configure.ac | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index c68119d..c83c6b5 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -1,4 +1,4 @@ -ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO="alsa_restore_go" +ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", TEST=="@datadir@", GOTO="alsa_restore_go" GOTO="alsa_restore_end" LABEL="alsa_restore_go" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index b862412..afbedb3 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -44,6 +44,7 @@ endif edit = \ $(SED) -r -e 's,@sbindir\@,$(sbindir),g' \ + -e 's,@datadir\@,$(mydatadir),g' \ -e 's,@daemonswitch\@,$(ALSACTL_DAEMONSWITCH),g' \ < $< > $@ || rm $@ diff --git a/configure.ac b/configure.ac index 7e6894a..a4d2db3 100644 --- a/configure.ac +++ b/configure.ac @@ -292,6 +292,7 @@ AC_DEFINE_UNQUOTED(SOUNDSDIR, "$soundsdir", [directory containing sample data]) mydatadir="$dir/alsa" AC_DEFINE_UNQUOTED(DATADIR, "$mydatadir", [directory containing alsa configuration]) +AC_SUBST(mydatadir) AC_ARG_WITH(testsound, AS_HELP_STRING([--with-testsound=file], [give the path of test sound file for alsaconf]), -- 1.9.1

5 13

[alsa-devel] [RFC][PATCH 0/2] Fix aplay and alsa-lib write_areas() freezing with 100% cpu usage on stuck slave pcm
by Sergey 13 Jun '14

13 Jun '14

These patches originate from `aplay -vvv /dev/zero` not responding to Ctrl+C when playing to plugged "null", e.g. pcm.!default { type plug slave.pcm { type null } slave.format S32_LE } First patch fixes snd_pcm_write_areas() stuck in a loop because of transfer func() always returning 0. That caused application calling it to freeze. In the example above it have frozen aplay. If it was called by firefox, it would freeze firefox. Is it a good way to fix it? Should snd_pcm_read_areas() be fixed like that too? Second patch fixes aplay itself, not returning from pcm_write() after SIGINT. Additionally aplay may restore original interrupt handler, to make second Ctrl+C stop it for sure. In case of other bugs in alsa-lib or third-party slave pcms it seems better than SIGkilling it manually. Comments and suggestions are welcome. Regardless of these patches' fate null plugin must be fixed on its own. Patches (2): pcm: break out of snd_pcm_write_areas() if write function returned 0 aplay: Escape from pcm_write() when in_aborting state src/pcm/pcm.c | 2 +- aplay/aplay.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH] drm/i915, HD-audio: Don't continue probing when nomodeset is given
by Takashi Iwai 13 Jun '14

13 Jun '14

When a machine is booted with nomodeset option, i915 driver skips the whole initialization. Meanwhile, HD-audio tries to bind wth i915 just by request_symbol() without knowing that the initialization was skipped, and eventually it hits WARN_ON() in i915_request_power_well() and i915_release_power_well() wrongly but still continues probing, even though it doesn't work at all. In this patch, both functions are changed to return an error in case of uninitialized state instead of WARN_ON(), so that HD-audio driver can give up HDMI controller initialization at the right time. Cc: <stable(a)vger.kernel.org> [3.15] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- The bug actually exists since 3.11, but this commit will be applicable only to recent kernels due to ALSA and i915 code changes, that's why marked with [3.15] for stable. drivers/gpu/drm/i915/intel_pm.c | 14 ++++++++------ include/drm/i915_powerwell.h | 4 ++-- sound/pci/hda/hda_i915.c | 12 ++++++------ sound/pci/hda/hda_i915.h | 4 ++-- sound/pci/hda/hda_intel.c | 7 ++++++- 5 files changed, 24 insertions(+), 17 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_pm.c b/drivers/gpu/drm/i915/intel_pm.c index d93dcf683e8c..0d6bd247009b 100644 --- a/drivers/gpu/drm/i915/intel_pm.c +++ b/drivers/gpu/drm/i915/intel_pm.c @@ -5706,30 +5706,32 @@ void intel_display_power_put(struct drm_i915_private *dev_priv, static struct i915_power_domains *hsw_pwr; /* Display audio driver power well request */ -void i915_request_power_well(void) +int i915_request_power_well(void) { struct drm_i915_private *dev_priv; - if (WARN_ON(!hsw_pwr)) - return; + if (!hsw_pwr) + return -ENODEV; dev_priv = container_of(hsw_pwr, struct drm_i915_private, power_domains); intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); + return 0; } EXPORT_SYMBOL_GPL(i915_request_power_well); /* Display audio driver power well release */ -void i915_release_power_well(void) +int i915_release_power_well(void) { struct drm_i915_private *dev_priv; - if (WARN_ON(!hsw_pwr)) - return; + if (!hsw_pwr) + return -ENODEV; dev_priv = container_of(hsw_pwr, struct drm_i915_private, power_domains); intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO); + return 0; } EXPORT_SYMBOL_GPL(i915_release_power_well); diff --git a/include/drm/i915_powerwell.h b/include/drm/i915_powerwell.h index cfdc884405b7..2baba9996094 100644 --- a/include/drm/i915_powerwell.h +++ b/include/drm/i915_powerwell.h @@ -30,7 +30,7 @@ #define _I915_POWERWELL_H_ /* For use by hda_i915 driver */ -extern void i915_request_power_well(void); -extern void i915_release_power_well(void); +extern int i915_request_power_well(void); +extern int i915_release_power_well(void); #endif /* _I915_POWERWELL_H_ */ diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 9d07e4edacdb..e9e8a4a4a9a1 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -22,20 +22,20 @@ #include <drm/i915_powerwell.h> #include "hda_i915.h" -static void (*get_power)(void); -static void (*put_power)(void); +static int (*get_power)(void); +static int (*put_power)(void); -void hda_display_power(bool enable) +int hda_display_power(bool enable) { if (!get_power || !put_power) - return; + return -ENODEV; pr_debug("HDA display power %s \n", enable ? "Enable" : "Disable"); if (enable) - get_power(); + return get_power(); else - put_power(); + return put_power(); } int hda_i915_init(void) diff --git a/sound/pci/hda/hda_i915.h b/sound/pci/hda/hda_i915.h index 5a63da2c53e5..bfd835f8f1aa 100644 --- a/sound/pci/hda/hda_i915.h +++ b/sound/pci/hda/hda_i915.h @@ -17,11 +17,11 @@ #define __SOUND_HDA_I915_H #ifdef CONFIG_SND_HDA_I915 -void hda_display_power(bool enable); +int hda_display_power(bool enable); int hda_i915_init(void); int hda_i915_exit(void); #else -static inline void hda_display_power(bool enable) {} +static inline int hda_display_power(bool enable) { return 0; } static inline int hda_i915_init(void) { return -ENODEV; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bb65a124e006..23fd6b9aecca 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1656,8 +1656,13 @@ static int azx_probe_continue(struct azx *chip) "Error request power-well from i915\n"); goto out_free; } + err = hda_display_power(true); + if (err < 0) { + dev_err(chip->card->dev, + "Cannot turn on display power on i915\n"); + goto out_free; + } #endif - hda_display_power(true); } err = azx_first_init(chip); -- 1.9.3

2 2

[alsa-devel] [PATCH v3] ASoC: fsl-ssi: fix do_div build warning in fsl_ssi_set_bclk()
by Timur Tabi 13 Jun '14

13 Jun '14

do_div() requires that the first parameter be a 64-bit integer, but clkrate was defined as an unsigned long. Instead, just use a normal division. This change fixes the following warnings: CC sound/soc/fsl/fsl_ssi.o sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk': sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *' Signed-off-by: Timur Tabi <timur(a)tabi.org> --- v3: fix grammatical errors in commit message v2: use normal division instead of do_div Note: I cannot test this code because I do not have a platform that runs the SSI in master mode. sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55..3043d57 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -590,8 +590,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - do_div(clkrate, factor); - afreq = (u32)clkrate / (i + 1); + clkrate /= factor; + afreq = clkrate / (i + 1); if (freq == afreq) sub = 0; -- 1.7.11.7

1 0

[alsa-devel] pcm: Add missing callbacks that cannot be NULL
by Alexander E. Patrakov 13 Jun '14

13 Jun '14

Some plugins don't implement some mandatory fast_ops. Thus, these fast_ops get defaulted to NULL, and programs crash when calling the corresponding snd_pcm_* functions. The following callbacks were missing: file: htimestamp multi: rewindable, forwardable null: rewindable, forwardable rate: rewindable, forwardable After this series, all required callbacks are present, but I cannot guarantee that they return meaningful values. All patches in this series are only compile-tested. -- Alexander E. Patrakov

2 9

[alsa-devel] [PATCH] ALSA: hda - Add quirk for external mic on Lifebook U904
by David Henningsson 13 Jun '14

13 Jun '14

According to the bug reporter (Данило Шеган), the external mic starts to work and has proper jack detection if only pin 0x19 is marked properly as an external headset mic. AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachme… Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1328587 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c095fa..89af62f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4246,6 +4246,7 @@ enum { ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, + ALC269_FIXUP_LIFEBOOK_EXTMIC, ALC269_FIXUP_AMIC, ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, @@ -4383,6 +4384,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_QUANTA_MUTE }, + [ALC269_FIXUP_LIFEBOOK_EXTMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1903c }, /* headset mic, with jack detect */ + { } + }, + }, [ALC269_FIXUP_AMIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4844,6 +4852,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk table
by Hui Wang 13 Jun '14

13 Jun '14

The fixup value for codec alc293 was set to ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it, the Dock mic will be overwriten by the headset mic, this will make the Dock mic can't work. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c095fa..06d6063 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5159,7 +5159,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x1e, 0x411111f0}, }, - .value = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + .value = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, }, {} }; -- 1.8.1.2

2 1

[alsa-devel] [RFC][PATCH v1] ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series
by Nicolin Chen 13 Jun '14

13 Jun '14

From: Nicolin Chen <b42378(a)freescale.com> This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_ssi.h | 2 + 2 files changed, 278 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569..b2ebaf8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -502,6 +511,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } /** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + +/** * fsl_ssi_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69..e6b6324 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- 1.8.4

4 9

[alsa-devel] [PATCH] ALSA: au1x00: Use resource_size
by Himangi Saraogi 13 Jun '14

13 Jun '14

Use the function resource_size, which reduces the chance of introducing off-by-one errors in calculating the resource size. The semantic patch that makes this change is as follows: // <smpl> @@ struct resource *res; @@ - (res->end - res->start) + 1 + resource_size(res) // </smpl> Signed-off-by: Himangi Saraogi <himangi774(a)gmail.com> Acked-by: Julia Lawall <julia.lawall(a)lip6.fr> --- Not compile tested. sound/mips/au1x00.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index d10ef76..86583c7 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -649,7 +649,7 @@ static int au1000_ac97_probe(struct platform_device *pdev) err = -EBUSY; au1000->ac97_res_port = request_mem_region(r->start, - r->end - r->start + 1, pdev->name); + resource_size(r), pdev->name); if (!au1000->ac97_res_port) { snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); goto out; -- 1.9.1

2 1

[alsa-devel] ALSA userspace release time?
by David Henningsson 13 Jun '14

13 Jun '14

Hi, It was around a year since we last had a release of ALSA's userspace component. Is it perhaps time for 1.0.28, or are we waiting for something that would be nice to get in first? We've currently got e g speaker-test improvements/fixes, 2.1 surround support, "alsactl monitor" command, and random fixes here and there. Sounds good enough for a release to me. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 6

[alsa-devel] [PATCH 4/4] Sound: tlv320aic31xx: Fixed Coding Style Issues
by Shahina Shaik 13 Jun '14

13 Jun '14

From: Shahina Shaik <sharab.shaik(a)gmail.com> Fixed coding style issues of "Missing Blank line after declaration" Signed-off-by: Shahina Shaik <sharab.shaik(a)gmail.com> --- sound/soc/codecs/tlv320aic31xx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2341910..a9c4a8b 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -329,6 +329,7 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, unsigned int bits; int counter = count; int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { usleep_range(sleep, sleep * 2); ret = regmap_read(aic31xx->regmap, reg, &bits); @@ -435,6 +436,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* change mic bias voltage to user defined */ -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/1] ASoC: mc13783: Add missing of_node_put
by Sachin Kamat 13 Jun '14

13 Jun '14

of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: fsl-ssi: fix do_div build warning in fsl_ssi_set_bclk()
by Timur Tabi 13 Jun '14

13 Jun '14

do_div() requires that the first parameter is a 64-bit integer, which but clkrate was defined as an unsigned long. This caused the following warnings: CC sound/soc/fsl/fsl_ssi.o sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk': sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *' Signed-off-by: Timur Tabi <timur(a)tabi.org> --- Note: I cannot test this code because I do not have a platform that runs the SSI in master mode. sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9bfef55..9d98340 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -555,8 +555,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; - unsigned long clkrate, baudrate, tmprate; - u64 sub, savesub = 100000; + unsigned long baudrate, tmprate; + u64 clkrate, sub, savesub = 100000; unsigned int freq; bool baudclk_is_used; -- 1.7.11.7

1 0

[alsa-devel] [PATCH] aplay: Fix VU-meter stdout/stderr mess
by Sergey 13 Jun '14

13 Jun '14

Commit 6bbeb78d forgot to change some of VU-meter output to stderr. This patch makes the output of `aplay -vvv` consistent again. --- aplay/aplay.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 33a8ed5..77e40a3 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1774,12 +1774,12 @@ static void compute_max_peak(u_char *data, size_t count) fflush(stderr); } else if(verbose==3) { - printf(_("Max peak (%li samples): 0x%08x "), (long)ocount, max_peak[0]); + fprintf(stderr, _("Max peak (%li samples): 0x%08x "), (long)ocount, max_peak[0]); for (val = 0; val < 20; val++) if (val <= perc[0] / 5) - putchar('#'); + putc('#', stderr); else - putchar(' '); + putc(' ', stderr); fprintf(stderr, " %i%%\n", perc[0]); fflush(stderr); } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/2] [RFC] ASoC: samsung: move s3c24xx over to dmaengine
by Arnd Bergmann 12 Jun '14

12 Jun '14

The s3c24xx sound support keeps giving randconfig build errors, this is a new attempt to solve it by moving the driver over to use dmaengine exclusively. The removal of the legacy DMA support and the addition of the s3c24xx_dma_filter pointer are fairly obvious here. The part I'm not completely sure about is the removal of the s3c2410_dma_ctrl(..., S3C2410_DMAOP_STARTED) and dma_data->ops->started() calls. My understanding is that these are only required for drivers that do not support cyclic transfers, which the new dma engine driver now does, so we can simply remove them. This would also fix at least one bug in the ac97 driver on newer machines, which currently gives us a NULL pointer dereference from trying to call dma_data->ops->started(). Any insights about this, or testing would be very welcome. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Kukjin Kim <kgene.kim(a)samsung.com> Cc: linux-samsung-soc(a)vger.kernel.org --- sound/soc/samsung/Kconfig | 13 -- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/ac97.c | 16 -- sound/soc/samsung/dma.c | 454 ------------------------------------ sound/soc/samsung/dmaengine.c | 3 + sound/soc/samsung/s3c24xx-i2s.c | 1 - 6 files changed, 4 insertions(+), 487 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 7745629..32bfcf1 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,25 +1,15 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to select the audio interfaces to support below. -config SND_S3C_DMA - tristate - -config SND_S3C_DMA_LEGACY - tristate - config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +17,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate @@ -83,7 +72,6 @@ config SND_SOC_SAMSUNG_SMDK_WM8994 config SND_SOC_SAMSUNG_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_SOC_SAMSUNG && MACH_SMDK2443 - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 @@ -94,7 +82,6 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX - select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC select SND_SAMSUNG_AC97 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6d0212b..f69f490 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,6 +1,5 @@ # S3c24XX Platform Support snd-soc-s3c-dma-objs := dmaengine.o -snd-soc-s3c-dma-legacy-objs := dma.o snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o @@ -10,8 +9,7 @@ snd-soc-samsung-spdif-objs := spdif.o snd-soc-pcm-objs := pcm.o snd-soc-i2s-objs := i2s.o -obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o -obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o +obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 68d9303..a674827 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -225,9 +225,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -253,11 +250,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } @@ -265,9 +257,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -287,11 +276,6 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (!dma_data->ops) - dma_data->ops = samsung_dma_get_ops(); - - dma_data->ops->started(dma_data->channel); - return 0; } diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c deleted file mode 100644 index d9dc7bc..0000000 --- a/sound/soc/samsung/dma.c +++ /dev/null @@ -1,454 +0,0 @@ -/* - * dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * 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. - */ - -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/module.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "dma.h" - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -static const struct snd_pcm_hardware dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -static void audio_buffdone(void *data); - -/* dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. - */ -static void dma_enqueue(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - struct samsung_dma_prep dma_info; - - pr_debug("Entered %s\n", __func__); - - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE); - dma_info.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - dma_info.fp = audio_buffdone; - dma_info.fp_param = substream; - dma_info.period = prtd->dma_period; - dma_info.len = prtd->dma_period*limit; - - if (dma_info.cap == DMA_CYCLIC) { - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - prtd->dma_loaded += limit; - return; - } - - while (prtd->dma_loaded < limit) { - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + dma_info.period) > prtd->dma_end) { - dma_info.period = prtd->dma_end - pos; - pr_debug("%s: corrected dma len %ld\n", - __func__, dma_info.period); - } - - dma_info.buf = pos; - prtd->params->ops->prepare(prtd->params->ch, &dma_info); - - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } - - prtd->dma_pos = pos; -} - -static void audio_buffdone(void *data) -{ - struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (prtd->state & ST_RUNNING) { - prtd->dma_pos += prtd->dma_period; - if (prtd->dma_pos >= prtd->dma_end) - prtd->dma_pos = prtd->dma_start; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (!samsung_dma_has_circular()) { - prtd->dma_loaded--; - dma_enqueue(substream); - } - spin_unlock(&prtd->lock); - } -} - -static int dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - struct samsung_dma_req req; - struct samsung_dma_config config; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - prtd->params->ops = samsung_dma_get_ops(); - - req.cap = (samsung_dma_has_circular() ? - DMA_CYCLIC : DMA_SLAVE); - req.client = prtd->params->client; - config.direction = - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - config.width = prtd->params->dma_size; - config.fifo = prtd->params->dma_addr; - prtd->params->ch = prtd->params->ops->request( - prtd->params->channel, &req, rtd->cpu_dai->dev, - prtd->params->ch_name); - if (!prtd->params->ch) { - pr_err("Failed to allocate DMA channel\n"); - return -ENXIO; - } - prtd->params->ops->config(prtd->params->ch, &config); - } - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int dma_hw_free(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - prtd->params->ops->flush(prtd->params->ch); - prtd->params->ops->release(prtd->params->ch, - prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int dma_prepare(struct snd_pcm_substream *substream) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* flush the DMA channel */ - prtd->params->ops->flush(prtd->params->ch); - - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - dma_enqueue(substream); - - return ret; -} - -static int dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->state |= ST_RUNNING; - prtd->params->ops->trigger(prtd->params->ch); - break; - - case SNDRV_PCM_TRIGGER_STOP: - prtd->state &= ~ST_RUNNING; - prtd->params->ops->stop(prtd->params->ch); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - unsigned long res; - - pr_debug("Entered %s\n", __func__); - - res = prtd->dma_pos - prtd->dma_start; - - pr_debug("Pointer offset: %lu\n", res); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * called... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &dma_hardware); - - prtd = kzalloc(sizeof(struct runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops dma_ops = { - .open = dma_open, - .close = dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dma_hw_params, - .hw_free = dma_hw_free, - .prepare = dma_prepare, - .trigger = dma_trigger, - .pointer = dma_pointer, - .mmap = dma_mmap, -}; - -static int preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static int dma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - pr_debug("Entered %s\n", __func__); - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } -out: - return ret; -} - -static struct snd_soc_platform_driver samsung_asoc_platform = { - .ops = &dma_ops, - .pcm_new = dma_new, - .pcm_free = dma_free_dma_buffers, -}; - -void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, - struct s3c_dma_params *playback, - struct s3c_dma_params *capture) -{ - snd_soc_dai_init_dma_data(dai, playback, capture); -} -EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); - -int samsung_asoc_dma_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &samsung_asoc_platform); -} -EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index a0e4e79..506f5bf 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 9aba9fb..890d241 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -290,7 +290,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND:

3 5

[alsa-devel] [PATCH] ALSA: hda - verify pin:converter connection on unsol event for HSW and VLV
by mengdong.lin＠intel.com 12 Jun '14

12 Jun '14

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch will verify the pin's coverter selection for an active stream when an unsol event reports this pin becomes available again after a display mode change or hot-plug event. For Haswell+ and Valleyview: display mode change or hot-plug can change the transcoder:port connection and make all the involved audio pins share the 1st converter. So the stream using 1st convertor will flow to multiple pins but active streams using other converters will fail. This workaround is to assure the pin selects the right conveter and an assigned converter is not shared by other unused pins. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index be0a9ee..3e4417b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1594,10 +1594,18 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * Re-setup pin and infoframe. This is needed e.g. when * - sink is first plugged-in (infoframe is not set up if !monitor_present) * - transcoder can change during stream playback on Haswell + * and this can make HW reset converter selection on a pin. */ - if (eld->eld_valid && !old_eld_valid && per_pin->setup) + if (eld->eld_valid && !old_eld_valid && per_pin->setup) { + if (is_haswell_plus(codec) || is_valleyview(codec)) { + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, pin_nid, + per_pin->mux_idx); + } + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + } } if (eld_changed) -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: wm5102: Convert snd_kcontrol_chip to snd_soc_kcontrol_codec
by Charles Keepax 12 Jun '14

12 Jun '14

Controls for shaping the ultrasonic frequency response were introduced in this commit: commit 720630c002ffc7b0fa2ed5b3f4bfb36fd8f87ca6 ASoC: wm5102: Add controls to allow shaping of ultrasonic response However, they mistakenly used snd_kcontrol_chip instead of snd_soc_kcontrol_codec, which has replaced it now the framework is moving to componentisation. This patch fixes this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8613504..fa24d55 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -615,7 +615,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); uint16_t data; @@ -630,7 +630,7 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); @@ -645,7 +645,7 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); @@ -658,7 +658,7 @@ static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); mutex_lock(&codec->mutex); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: fsl: Fix build problem
by Guenter Roeck 12 Jun '14

12 Jun '14

Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi. Unfortunately, the structure is still used. This causes mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with sound/soc/fsl/fsl_dma.c:926:50: error: invalid use of undefined type 'struct ccsr_ssi' dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); ound/soc/fsl/fsl_dma.c:927:50: error: invalid use of undefined type 'struct ccsr_ssi' dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); Fix by using constants, similar to original commit. Cc: Markus Pargmann <mpa(a)pengutronix.de> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/fsl/fsl_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 6bb0ea5..a609aaf 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -923,8 +923,8 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) dma->dai.pcm_free = fsl_dma_free_dma_buffers; /* Store the SSI-specific information that we need */ - dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); - dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); + dma->ssi_stx_phys = res.start + CCSR_SSI_STX0; + dma->ssi_srx_phys = res.start + CCSR_SSI_SRX0; iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); if (iprop) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup index of src/dst mod when capture
by Kuninori Morimoto 12 Jun '14

12 Jun '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Index of dma name should use -1, not +1 when capture case. Thank you Dan. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 2d0f98f..5a4d7e0 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -289,7 +289,7 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma, dst_mod = mod[index]; } else { src_mod = mod[index]; - dst_mod = mod[index + 1]; + dst_mod = mod[index - 1]; } index = 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] OSS/DMASOUND: add __init to dmasound_init
by Fabian Frederick 11 Jun '14

11 Jun '14

dmasound_init is only called by __init functions: __init dmasound_q40_init __init amiga_audio_probe __init dmasound_atari_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/oss/dmasound/dmasound.h | 2 +- sound/oss/dmasound/dmasound_core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index 01019f0..a181ed2 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -86,7 +86,7 @@ static inline int ioctl_return(int __user *addr, int value) * Initialization */ -extern int dmasound_init(void); +extern int __init dmasound_init(void); #ifdef MODULE extern void dmasound_deinit(void); #else diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index f4ee85a..1f0a42c 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1394,7 +1394,7 @@ static int state_init(void) * This function is called by _one_ chipset-specific driver */ -int dmasound_init(void) +int __init dmasound_init(void) { int res ; #ifdef MODULE -- 1.8.4.5

1 1

Re: [alsa-devel] ASoC: rsnd: care DMA slave channel name for DT
by Dan Carpenter 11 Jun '14

11 Jun '14

Hello Kuninori Morimoto, The patch 199e7688bdf7: "ASoC: rsnd: care DMA slave channel name for DT" from May 22, 2014, leads to the following static checker warning: sound/soc/sh/rcar/core.c:318 rsnd_dma_of_name() error: buffer overflow 'mod' 4 <= 4 sound/soc/sh/rcar/core.c 278 struct rsnd_mod *mod[MOD_MAX]; 279 struct rsnd_mod *src_mod, *dst_mod; 280 int i, index; 281 282 283 for (i = 0; i < MOD_MAX; i++) 284 mod[i] = NULL; 285 286 /* 287 * in play case... 288 * 289 * src -> dst 290 * 291 * mem -> SSI 292 * mem -> SRC -> SSI 293 * mem -> SRC -> DVC -> SSI I don't understand what this comment means. 294 */ 295 mod[0] = NULL; /* for "mem" */ We memset everything to NULL in the loop above so this isn't needed. 296 index = 1; 297 for (i = 1; i < MOD_MAX; i++) { 298 if (!src) { 299 mod[i] = ssi; 300 break; 301 } else if (!dvc) { 302 mod[i] = src; 303 src = NULL; 304 } else { 305 mod[i] = dvc; 306 dvc = NULL; 307 } 308 309 if (mod[i] == this) 310 index = i; In theory, index can be MOD_MAX - 1. 311 } 312 313 if (is_play) { 314 src_mod = mod[index - 1]; 315 dst_mod = mod[index]; 316 } else { 317 src_mod = mod[index]; 318 dst_mod = mod[index + 1]; So then it complains that mod[] has only MOD_MAX elements so we're one space past the end of the array. Probably the way this is called, there is something to prevent it? 319 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/1] SND/PCI/HDA: add __init to snd_hda_add_codec_preset
by Fabian Frederick 10 Jun '14

10 Jun '14

snd_hda_add_codec_preset is only called by __init functions: __init patch_via_init __init patch_sigmatel_init __init patch_si3054_init __init patch_realtek_init __init patch_hdmi_init __init patch_conexant_init __init patch_cmedia_init __init patch_cirrus_init __init patch_ca0132_init __init patch_ca0110_init __init patch_analog_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_codec.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4c20277..5770c5c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -77,7 +77,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { static DEFINE_MUTEX(preset_mutex); static LIST_HEAD(hda_preset_tables); -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) +int __init snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) { mutex_lock(&preset_mutex); list_add_tail(&preset->list, &hda_preset_tables); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5825aa1..02a9123 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -182,7 +182,7 @@ struct hda_codec_preset_list { }; /* initial hook */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); +int __init snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); /* ops set by the preset patch */ -- 1.8.4.5

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[alsa-devel] [PATCH 1/2] misc: atmel-ssc: support to read ssc version
by Bo Shen 10 Jun '14

10 Jun '14

Read SSC IP version to check whether it supports frame sync length extension feature. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- drivers/misc/atmel-ssc.c | 7 +++++++ include/linux/atmel-ssc.h | 3 +++ 2 files changed, 10 insertions(+) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 22de137..9f49bcb 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -136,6 +136,7 @@ static int ssc_probe(struct platform_device *pdev) struct resource *regs; struct ssc_device *ssc; const struct atmel_ssc_platform_data *plat_dat; + u32 version; ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { @@ -173,8 +174,14 @@ static int ssc_probe(struct platform_device *pdev) clk_prepare_enable(ssc->clk); ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); + version = ssc_readl(ssc->regs, VERSION); clk_disable_unprepare(ssc->clk); + if (version >= 0x300) + ssc->has_fslen_ext = 1; + else + ssc->has_fslen_ext = 0; + ssc->irq = platform_get_irq(pdev, 0); if (!ssc->irq) { dev_dbg(&pdev->dev, "could not get irq\n"); diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index 571a12e..9ee1b68 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -19,6 +19,7 @@ struct ssc_device { int user; int irq; bool clk_from_rk_pin; + int has_fslen_ext; }; struct ssc_device * __must_check ssc_request(unsigned int ssc_num); @@ -258,6 +259,8 @@ void ssc_free(struct ssc_device *ssc); #define SSC_IMR_TXSYN_SIZE 1 #define SSC_IMR_TXSYN_OFFSET 10 +#define SSC_VERSION 0x000000fc + /* SSC PDC Receive Pointer Register */ #define SSC_PDC_RPR 0x00000100 -- 1.8.5.2

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[alsa-devel] [PATCH] ALSA: hda - Add quirk for ABit AA8XE
by David Henningsson 10 Jun '14

10 Jun '14

Bios does not set up the pin config default correctly (everything is set to zero). Reporter claims that 6stack-dig and 6stack-automute solve the problem. Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05 BugLink: https://bugs.launchpad.net/bugs/1319291 Reported-by: Stefano Statuti <stefano.statuti(a)hotmail.it> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- 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 d77745a..9c095fa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1427,6 +1427,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS W5A", ALC880_FIXUP_ASUS_W5A), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), + SND_PCI_QUIRK(0x147b, 0x1045, "ABit AA8XE", ALC880_FIXUP_6ST_AUTOMUTE), SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), -- 1.9.1

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[alsa-devel] Fwd: Re: Patch for power on headphone amp on Realtek 883
by asd 10 Jun '14

10 Jun '14

Hi Here is content; #uname -a Linux marian1000.go.ro 3.14-1-amd64 #1 SMP Debian 3.14.4-1 (2014-05-13) x86_64 GNU/Linux #cat /proc/asound/Intel/codec#0 Codec: Realtek ALC883 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10ec0883 Subsystem Id: 0x1043829f Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC883 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=2 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=4 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=6 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC883 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC883 Analog", type="Audio", device=0 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC883 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=3, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Side Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Front 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: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=05, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Surround 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: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=06, enabled=1 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Line Out CLFE 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: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=07, enabled=1 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Side Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Side 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: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=08, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic 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 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19840: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic 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 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line 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 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x0181304f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone 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: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4015e601: [N/A] Speaker at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x01441130: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x25 0x0b

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