June 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Headset mic support for two more machines
by David Henningsson 10 Jun '13

10 Jun '13

They need these quirks to have headset mic support. BugLink: https://bugs.launchpad.net/bugs/1189363 Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 02e22b4..cbd443d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3483,6 +3483,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3494,6 +3495,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), -- 1.7.9.5

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[alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices
by o-takashi＠sakamocchi.jp 08 Jun '13

08 Jun '13

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> This driver module is originally developed by Clemens Ladish 2010. But it includes duplicate codes which current snd-firewire-lib has and lack of automatical sampling rate/the number of channels arrangement. http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=refs/heads/fi… This driver module is based on snd-firewire-lib with my previous patches. [PATCH v2 0/3] snd-firewire-lib: add handling CMP output connection http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061607.html [PATCH v2 0/4] snd-firewire-lib: add handling AMDTP receive stream http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061611.html [PATCH 0/2] snd-firewire-lib: add MIDI stream support http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062610.html Currently this driver module support just PCM/MIDI kernel streaming via ALSA interfaces. So there are some issues about this driver. 1.whether adding each PCM devices for analog and digital interface 2.where the codes for device control like volume, routing and etc are 3.card definition files for ALSA applications like PulseAudio I hope this driver module promote developers to discuss about how to implement Firewire devices which has many channels and options. Takashi Sakamoto (8): add main file of driver module add device specific command add control interfaces add handling AMDTP stream add MIDI interface add PCM interface add proc interface modify Makefile and Kconfig to build this module sound/firewire/Kconfig | 16 + sound/firewire/Makefile | 2 + sound/firewire/fireworks/Makefile | 2 + sound/firewire/fireworks/fireworks.c | 488 +++++++++++++++++++++++ sound/firewire/fireworks/fireworks.h | 241 ++++++++++++ sound/firewire/fireworks/fireworks_command.c | 535 +++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_control.c | 537 ++++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_midi.c | 236 +++++++++++ sound/firewire/fireworks/fireworks_pcm.c | 530 +++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 183 +++++++++ sound/firewire/fireworks/fireworks_stream.c | 107 +++++ 11 files changed, 2877 insertions(+) create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_control.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: sgtl5000: Make device cache only when powered off
by Mark Brown 08 Jun '13

08 Jun '13

When the regulators have been disabled mark the device as cache only so that we don't try to interact with the hardware. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..85a32e6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -911,10 +911,25 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); + + regcache_cache_only(sgtl5000->regmap, false); + + ret = regcache_sync(sgtl5000->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + + regcache_cache_only(sgtl5000->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + return ret; + } } break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); break; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: codecs: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x()
by Andreas Irestal 07 Jun '13

07 Jun '13

From: Andreas Irestål <Andreas.Irestal(a)axis.com> When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is prevented. This problem was introduced in Linux 3.5 Signed-off-by: Andreas Irestål <Andreas.Irestal(a)axis.com> --- sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d6..1514bf8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, break; } - - if (found) - snd_soc_dapm_sync(widget->dapm); } - ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); - mutex_unlock(&widget->codec->mutex); + + if (found) + snd_soc_dapm_sync(widget->dapm); + + ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); return ret; }

2 1

[alsa-devel] [PATCH] ASoC: WM8962: Set CLKREG_OVD if GP5_FN is not its reset value
by Nicolin Chen 07 Jun '13

07 Jun '13

According to WM8962 datasheet, the GP5_FN bit of R516 (GPIO 5) depends on the CLKREG_OVD bit of R8 (Clocking2): "Note that GPIO5 functions are only supported when CLKREG_OVD=1. When CLKREG_OVD=0, the contents of Register R516 must not be changed from the default value." So set CLKREG_OVD bit before regcache_sync() updates the GP5_FN bit to a non-reset value. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 26219ea..9c8fd11 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3739,6 +3739,23 @@ static int wm8962_i2c_remove(struct i2c_client *client) } #ifdef CONFIG_PM_RUNTIME +static void wm8962_check_gp5_fn(struct snd_soc_codec *codec) +{ + int changed; + + /* + * If GP5_FN is not going to be the reset value 0x0, + * need to set CLKREG_OVD bit of Register CLOCKING2 + * before regcache_sync() updates GP5_FN bit to a + * non-reset value. The reset value should be 0x0. + */ + changed = snd_soc_test_bits(codec, WM8962_GPIO_5, + WM8962_GP5_FN_MASK, 0x0); + if (changed) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD_MASK, WM8962_CLKREG_OVD); +} + static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); @@ -3756,6 +3773,8 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + wm8962_check_gp5_fn(wm8962->codec); + regcache_sync(wm8962->regmap); return 0; -- 1.7.1

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[alsa-devel] [PATCH] ALSA: get rid of CONFIG_SND_VERBOSE_PRINTK
by Alan Stern 07 Jun '13

07 Jun '13

According to Takashi Iwai, CONFIG_SND_VERBOSE_PRINTK is "useless". All it does is add filenames and line numbers to certain log messages. This patch gets rid of it. Signed-off-by: Alan Stern <stern(a)rowland.harvard.edu> CC: Joe Perches <joe(a)perches.com> --- Updating all the snd_printk() and snd_printd() calls will be a huge job. Getting rid of the verbose logging is a lot easier, so that's where I'm starting. Alan Stern [as1688] Documentation/DocBook/writing-an-alsa-driver.tmpl | 7 +--- arch/arm/configs/eseries_pxa_defconfig | 1 arch/arm/configs/omap2plus_defconfig | 1 arch/arm/configs/s3c2410_defconfig | 1 arch/arm/configs/trizeps4_defconfig | 1 arch/ia64/configs/generic_defconfig | 1 arch/ia64/configs/gensparse_defconfig | 1 arch/mips/configs/pnx8335_stb225_defconfig | 1 arch/powerpc/configs/ppc6xx_defconfig | 1 arch/sh/configs/edosk7760_defconfig | 1 arch/sh/configs/sh7785lcr_32bit_defconfig | 1 include/sound/core.h | 4 +- sound/core/Kconfig | 9 ------ sound/core/misc.c | 33 ---------------------- sound/pci/hda/hda_intel.c | 4 -- 15 files changed, 5 insertions(+), 62 deletions(-) Index: usb-3.10/Documentation/DocBook/writing-an-alsa-driver.tmpl =================================================================== --- usb-3.10.orig/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ usb-3.10/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -6087,11 +6087,8 @@ struct _snd_pcm_runtime { <title><function>snd_printk()</function> and friends</title> <para> ALSA provides a verbose version of the - <function>printk()</function> function. If a kernel config - <constant>CONFIG_SND_VERBOSE_PRINTK</constant> is set, this - function prints the given message together with the file name - and the line of the caller. The <constant>KERN_XXX</constant> - prefix is processed as + <function>printk()</function> function. + The <constant>KERN_XXX</constant> prefix is processed as well as the original <function>printk()</function> does, so it's recommended to add this prefix, e.g. Index: usb-3.10/arch/arm/configs/eseries_pxa_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/eseries_pxa_defconfig +++ usb-3.10/arch/arm/configs/eseries_pxa_defconfig @@ -90,7 +90,6 @@ CONFIG_SND=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_DYNAMIC_MINORS=y -CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_PCMCIA is not set CONFIG_SND_SOC=m CONFIG_SND_PXA2XX_SOC=m Index: usb-3.10/arch/arm/configs/omap2plus_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/omap2plus_defconfig +++ usb-3.10/arch/arm/configs/omap2plus_defconfig @@ -192,7 +192,6 @@ CONFIG_SOUND=m CONFIG_SND=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_USB_AUDIO=m CONFIG_SND_SOC=m Index: usb-3.10/arch/arm/configs/s3c2410_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/s3c2410_defconfig +++ usb-3.10/arch/arm/configs/s3c2410_defconfig @@ -315,7 +315,6 @@ CONFIG_SND_SEQUENCER=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_DRIVERS is not set # CONFIG_SND_ARM is not set # CONFIG_SND_SPI is not set Index: usb-3.10/arch/arm/configs/trizeps4_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/trizeps4_defconfig +++ usb-3.10/arch/arm/configs/trizeps4_defconfig @@ -159,7 +159,6 @@ CONFIG_SND=y CONFIG_SND_SEQUENCER=y CONFIG_SND_MIXER_OSS=y CONFIG_SND_PCM_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_PXA2XX_AC97=y CONFIG_SND_USB_AUDIO=m Index: usb-3.10/arch/ia64/configs/generic_defconfig =================================================================== --- usb-3.10.orig/arch/ia64/configs/generic_defconfig +++ usb-3.10/arch/ia64/configs/generic_defconfig @@ -127,7 +127,6 @@ CONFIG_SND_SEQ_DUMMY=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m Index: usb-3.10/arch/ia64/configs/gensparse_defconfig =================================================================== --- usb-3.10.orig/arch/ia64/configs/gensparse_defconfig +++ usb-3.10/arch/ia64/configs/gensparse_defconfig @@ -115,7 +115,6 @@ CONFIG_SND_SEQ_DUMMY=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m Index: usb-3.10/arch/mips/configs/pnx8335_stb225_defconfig =================================================================== --- usb-3.10.orig/arch/mips/configs/pnx8335_stb225_defconfig +++ usb-3.10/arch/mips/configs/pnx8335_stb225_defconfig @@ -73,7 +73,6 @@ CONFIG_SND_SEQUENCER=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_EXT2_FS=m # CONFIG_DNOTIFY is not set Index: usb-3.10/arch/powerpc/configs/ppc6xx_defconfig =================================================================== --- usb-3.10.orig/arch/powerpc/configs/ppc6xx_defconfig +++ usb-3.10/arch/powerpc/configs/ppc6xx_defconfig @@ -870,7 +870,6 @@ CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y CONFIG_SND_DYNAMIC_MINORS=y # CONFIG_SND_SUPPORT_OLD_API is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_PCM_XRUN_DEBUG=y Index: usb-3.10/arch/sh/configs/edosk7760_defconfig =================================================================== --- usb-3.10.orig/arch/sh/configs/edosk7760_defconfig +++ usb-3.10/arch/sh/configs/edosk7760_defconfig @@ -90,7 +90,6 @@ CONFIG_SOUND=y CONFIG_SND=y # CONFIG_SND_SUPPORT_OLD_API is not set # CONFIG_SND_VERBOSE_PROCFS is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_SOC=y # CONFIG_HID_SUPPORT is not set # CONFIG_USB_SUPPORT is not set Index: usb-3.10/arch/sh/configs/sh7785lcr_32bit_defconfig =================================================================== --- usb-3.10.orig/arch/sh/configs/sh7785lcr_32bit_defconfig +++ usb-3.10/arch/sh/configs/sh7785lcr_32bit_defconfig @@ -105,7 +105,6 @@ CONFIG_SND_HRTIMER=y CONFIG_SND_DYNAMIC_MINORS=y # CONFIG_SND_SUPPORT_OLD_API is not set # CONFIG_SND_VERBOSE_PROCFS is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y # CONFIG_SND_DRIVERS is not set Index: usb-3.10/include/sound/core.h =================================================================== --- usb-3.10.orig/include/sound/core.h +++ usb-3.10/include/sound/core.h @@ -335,7 +335,7 @@ enum { SND_PR_VERBOSE, }; -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +#if defined(CONFIG_SND_DEBUG) __printf(4, 5) void __snd_printk(unsigned int level, const char *file, int line, const char *format, ...); @@ -349,7 +349,7 @@ void __snd_printk(unsigned int level, co * @fmt: format string * * Works like printk() but prints the file and the line of the caller - * when configured with CONFIG_SND_VERBOSE_PRINTK. + * when configured with CONFIG_SND_DEBUG. */ #define snd_printk(fmt, args...) \ __snd_printk(0, __FILE__, __LINE__, fmt, ##args) Index: usb-3.10/sound/core/Kconfig =================================================================== --- usb-3.10.orig/sound/core/Kconfig +++ usb-3.10/sound/core/Kconfig @@ -173,15 +173,6 @@ config SND_VERBOSE_PROCFS useful information to developers when a problem occurs). On the other side, it makes the ALSA subsystem larger. -config SND_VERBOSE_PRINTK - bool "Verbose printk" - help - Say Y here to enable verbose log messages. These messages - will help to identify source file and position containing - printed messages. - - You don't need this unless you're debugging ALSA. - config SND_DEBUG bool "Debug" help Index: usb-3.10/sound/core/misc.c =================================================================== --- usb-3.10.orig/sound/core/misc.c +++ usb-3.10/sound/core/misc.c @@ -51,27 +51,11 @@ void release_and_free_resource(struct re EXPORT_SYMBOL(release_and_free_resource); -#ifdef CONFIG_SND_VERBOSE_PRINTK -/* strip the leading path if the given path is absolute */ -static const char *sanity_file_name(const char *path) -{ - if (*path == '/') - return strrchr(path, '/') + 1; - else - return path; -} -#endif - -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +#if defined(CONFIG_SND_DEBUG) void __snd_printk(unsigned int level, const char *path, int line, const char *format, ...) { va_list args; -#ifdef CONFIG_SND_VERBOSE_PRINTK - int kern_level; - struct va_format vaf; - char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; -#endif #ifdef CONFIG_SND_DEBUG if (debug < level) @@ -79,22 +63,7 @@ void __snd_printk(unsigned int level, co #endif va_start(args, format); -#ifdef CONFIG_SND_VERBOSE_PRINTK - vaf.fmt = format; - vaf.va = &args; - - kern_level = printk_get_level(format); - if (kern_level) { - const char *end_of_header = printk_skip_level(format); - memcpy(verbose_fmt, format, end_of_header - format); - vaf.fmt = end_of_header; - } else if (level) - memcpy(verbose_fmt, KERN_DEBUG, sizeof(KERN_DEBUG) - 1); - printk(verbose_fmt, sanity_file_name(path), line, &vaf); - -#else vprintk(format, args); -#endif va_end(args); } EXPORT_SYMBOL_GPL(__snd_printk); Index: usb-3.10/sound/pci/hda/hda_intel.c =================================================================== --- usb-3.10.orig/sound/pci/hda/hda_intel.c +++ usb-3.10/sound/pci/hda/hda_intel.c @@ -189,11 +189,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}, "{ULI, M5461}}"); MODULE_DESCRIPTION("Intel HDA driver"); -#ifdef CONFIG_SND_VERBOSE_PRINTK -#define SFX /* nop */ -#else #define SFX "hda-intel " -#endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) #ifdef CONFIG_SND_HDA_CODEC_HDMI

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[alsa-devel] [ANNOUNCE] tinycompress: release 0.2.0
by Vinod Koul 07 Jun '13

07 Jun '13

Hello folks, Time to cut another release for the library i am working on. tinycompress library - the user mode library for using the ALSA compressed APIs has received following new features and now is v0.2.0 - support for non blocking I/O - support for recording compressed audio content - no poll when writing last file buffer - get_alsa_rate helper for rate conversion - bunch of other fixes - now all releases will be tagged accordingly For details & contributions, pls see the below changelog. Lakshmi N Vinnakota (1): fix error reporting in tinycompress Richard Fitzgerald (6): tinycompress: cplay: Correct a couple of small message typos compress: Add support for capture streams compress: fix hpointer error when no sample rate compress: copy final version of config struct compress: do not poll if enough space to write remaining data compress: Add non-blocking I/O Vinod Koul (4): compress: check for config first add utils file with rate conversion helper cplay: use get_alsa_rate to convert sample rate Tinycompress version 0.2.0 Android.mk | 2 +- compress.c | 171 +++++++++++++++++++++++++++-------- cplay.c | 52 ++--------- include/tinycompress/tinycompress.h | 27 +++++- include/tinycompress/version.h | 4 +- makefile.linux | 2 +- utils.c | 96 ++++++++++++++++++++ 7 files changed, 263 insertions(+), 91 deletions(-) create mode 100644 utils.c Thanks ~Vinod --

1 0

[alsa-devel] [PATCH V4] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; v3->v4: * Revised binding doc to describe two gpio pins more clearly. .../devicetree/bindings/sound/imx-audio-wm8962.txt | 55 ++++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 336 ++++++++++++++++++++ 4 files changed, 405 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..a34b94b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,55 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Optional properties: +- hp-det-gpios : The gpio pin to detect plug in/out event that happens to + Headphone jack. + +- mic-det-gpios: The gpio pin to detect plug in/out event that happens to + Micphone jack. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..0b2eb65 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,336 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "could not get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

3 4

[alsa-devel] Question on wait_for_avail function
by gsantosh＠codeaurora.org 07 Jun '13

07 Jun '13

Hi Liam, Consider an use case 1) Write is blocked as there is no space to write the data, after the return from snd_pcm_playback_avail()function in wait_for_avail 2) HAL issues pause in parallel to the write I see that the following function snd_pcm_post_pause unblock the write wake_up(&runtime->sleep); wake_up(&runtime->tsleep); But i do not see any case handling for the pause state in the wait_for_avail function, it is not coming out of the for loop as there will not be any change in the buffer states. Please explain what is the intention behind the wake in pause callback. Want to understand why there is no handling of pause in wait_for_avail function. and also is there any restriction from ALSA framework that we should not issue pause while write is blocked or going on. Regards, Santosh M G.

4 5

[alsa-devel] [PATCH V3] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; .../devicetree/bindings/sound/imx-audio-wm8962.txt | 52 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 336 ++++++++++++++++++++ 4 files changed, 402 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..bb76097 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,52 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Optional properties: +- hp-det-gpios : The gpio port of Headphone jack detection. +- mic-det-gpios: The gpio port of Micphone jack detection. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..0b2eb65 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,336 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "could not get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

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