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

[alsa-devel] Fake Guru Hamid Arabnia and bogus WORLDCOMP and CSCI
by jermy23＠hushmail.com 14 Jun '14

14 Jun '14

Fake Guru Hamid Arabnia and bogus WORLDCOMP and CSCI If you have any thought of attending the world’s biggest fake or bogus conference in computer science, WORLDCOMP http://www.world-academy-of-science.org you must visit any websites below https://sites.google.com/site/worlddump1 or https://sites.google.com/site/dumpconf https://sites.google.com/site/moneycomp1 https://sites.google.com/site/worlddump4 The organizer of this conference is Hamid Arabnia http://www.cs.uga.edu/~hra a professor from University of Georgia, USA. The primary goal of this conference is to collect registration fee by accepting all submitted papers. Several fake papers were published in the conference proceedings. He already earned millions of dollars from the registration fee and is using part of the money for anti-Christmas greetings campaign. http://worldcomp-fake-bogus.blogspot.com has additional information on this campaign. He recently started another new conference CSCI due to his hunger for money http://www.americancse.org Hamid Arabnia failed to reveal the reviews and reviewers' information for all the papers he received, despite repeated requests and challenges. The reason for his failure is there were no reviews and reviewers and he just cheated the research community for more than a decade by announcing that each draft paper is reviewed by two experts. We challenge him to publish these details at the conference website. Where are these experts? Where are these reviews? He pays tens of thousands of dollars and brings keynote speakers and project his conference as a great one. He lists the names of current and former keynote speakers on the front page of the conference and use those names as baits to attract paper submissions from innocent researchers. He always conducts the conferences in the “sin city” to attract people who prefer “other” activities. DBLP stopped indexing these conferences since 2011 and displayed an explicit message; The DBLP Advisory Board decided to discontinue indexing of this conference series. Visit http://www.informatik.uni-trier.de/~ley/db/conf/biocomp/index.html as a sample. He was forced to remove his name, the university of Georgia name, and university of Georgia email address from the conference’s contact page because the University has banned him from doing that. He is running the conference behind the scenes and there is no coordinator or general chair listed for this conference. Almost every week (until the conference date is so close), Hamid Arabnia updates the conference website with things like this: You are invited to submit a "Late Breaking Paper", "Position Paper", "Abstract/Poster Paper" or “Abstract of Thesis or Dissertation” for consideration, etc. He wants more papers because each paper generates around 500 US dollars in to his pocket. Additional information on this conference bogus conference is available by searching internet using the keywords worldcomp fake or Hamid Arabnia fake Apologies for posting to multiple mailing lists. Spreading the word is the only way to stop this conference from cheating innocent researchers.

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[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

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[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

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[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

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[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

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