December 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: omap-twl4030: Update the header file to support more boards
by Peter Ujfalusi 07 Dec '12

07 Dec '12

The common machine driver will be able to support new boards where the voice port is also in use. At the same time add possibility to fine tune the connections from twl4030 on the board: jack detection GPIO, input/output selection Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam Tony, I just finalized a series which I intend to send for 3.9: It is going to update the omap-twl4030 ASoC machine driver to support more boards, thus allowing me to remove the driver for zoom2 and sdp3440. The issue is that in order to do it I need to make modification in arch/arm/mach-omap2/ and sound/soc/omap/ as well. One patch which changes the header file is the only compile time requirement for the changes. As a reference this is how my tree looks like regarding to this change (need to sort them still), git log --oneline : 1702213 ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3403 41ff1a6 ARM: OMAP: sdp3430: Audio support via the common omap-twl4030 machine driver f51363ed ARM: OMAP: zoom: Audio support via the common omap-twl4030 machine driver 159e443 ASoC: omap-twl4030: Add support for routing, voice port and jack detect 77bda45 ARM: OMAP2+: twl-common: Allow boards to customize the twl4030 audio setup 0db92ac ARM: OMAP2+: twl-common: Add default twl4030 audio configuration 9005d5b ARM: OMAP: zoom: Zoom2 does not have extmute functionality c7ffd18 ASoC: sdp3430: No need to configure pin mux for extmute 43dc9d9 ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use db9207c ARM: OMAP: 3430sdp: Enable extmute functionality for audio e6aea6f ASoC: twl4030: Convert MICBIAS to SUPPLY widget 8eadb74 ASoC: sdp3430: No need to configure the Voice port anymore f11c1be ASoC: zoom2: No need to configure the Voice port anymore 63da45d ASoC: twl4030: Correct the support for Voice port a98e934 ASoC: omap-twl4030: Update the header file to support more boards I know it is quite late already, but Mark would you be able the squeeze in this patch for 3.8? If this is in, I can separate the arch/arm/mach-omap2/ and sound/soc/omap/ patches and queue them via the appropriate tree without creating cross tree merge issues (yes, again cross tree issue). All of these will come together for 3.9 and in linux-next. This header change does not affect the current driver since the new parameters are ignored by the current driver Mark, would you be able to do this? Thank you, Peter include/linux/platform_data/omap-twl4030.h | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/include/linux/platform_data/omap-twl4030.h b/include/linux/platform_data/omap-twl4030.h index c7bef78..ee60ef7 100644 --- a/include/linux/platform_data/omap-twl4030.h +++ b/include/linux/platform_data/omap-twl4030.h @@ -25,8 +25,34 @@ #ifndef _OMAP_TWL4030_H_ #define _OMAP_TWL4030_H_ +/* To select if only one channel is connected in a stereo port */ +#define OMAP_TWL4030_LEFT (1 << 0) +#define OMAP_TWL4030_RIGHT (1 << 1) + struct omap_tw4030_pdata { const char *card_name; + /* Voice port is connected to McBSP3 */ + bool voice_connected; + + /* The driver will parse the connection flags if this flag is set */ + bool custom_routing; + /* Flags to indicate connected audio ports. */ + u8 has_hs; + u8 has_hf; + u8 has_predriv; + u8 has_carkit; + bool has_ear; + + bool has_mainmic; + bool has_submic; + bool has_hsmic; + bool has_carkitmic; + bool has_digimic0; + bool has_digimic1; + u8 has_linein; + + /* Jack detect GPIO or <= 0 if it is not implemented */ + int jack_detect; }; #endif /* _OMAP_TWL4030_H_ */ -- 1.8.0

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[alsa-devel] [PATCH] ASoC: arizona: Log the clock we're setting the DAI to use
by Mark Brown 07 Dec '12

07 Dec '12

Useful for diagnostics. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a7eb553..d5385dc 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -811,6 +811,9 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return -EBUSY; } + dev_dbg(codec->dev, "Setting AIF%d to %s\n", dai->id + 1, + arizona_dai_clk_str(clk_id)); + memset(&routes, 0, sizeof(routes)); routes[0].sink = dai->driver->capture.stream_name; routes[1].sink = dai->driver->playback.stream_name; -- 1.7.10.4

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Re: [alsa-devel] I would like to resubmit an ALSA device driver for the Maxim 98090 Codec...
by Mark Brown 07 Dec '12

07 Dec '12

On Thu, Dec 06, 2012 at 05:14:50PM -0800, Jerry Wong wrote: > Here is the resubmission of the sound/soc/max98090.c driver with the changes you suggested including regmap I/O and some stylistic updates. Please fix your mailer to word wrap within paragraphs. Please also follow the submission procedure in SubmittingPatches, as previously mentioned this is essential especially the Signed-off-by bit. I really can't apply or even properly review something that doesn't follow that. Looking at the code you want to check the register defaults - defaults for volatile registers don't make a huge amount of sense. For the ranges based on bases use ... or if (else) rather than listing the individual registers by hand. For the custom get and sets I can't see why you're not able to use generic controls, if you aren't there's enough of these that you should add a generic type. Gains should always be volume controls with TLV, enable/disable stuff should be Switch and controls called Gain Volume don't make a lot of sense (just Volume). Prints that just log the build time aren't helpful, you'll notice no other mainline drivers do that... there's already a kernel level build time. Similarly other stuff like interrupt numbers that are getting logged are a bit noisy. Build time options for things like performance need to be made runtime or platform data based. Are you sure you want the bandgap reference running the entire time the system is powered up?

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Re: [alsa-devel] [PATCH 05/20] ALSA: firewire-speakers: remove __dev* attributes
by Clemens Ladisch 06 Dec '12

06 Dec '12

Bill Pemberton wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Clemens Ladisch <clemens(a)ladisch.de> Acked-by: Clemens Ladisch <clemens(a)ladisch.de> for all these (5,10,11): > sound/firewire/speakers.c | 8 ++++---- > sound/pci/bt87x.c | 22 +++++++++++----------- > sound/pci/oxygen/oxygen.c | 10 +++++----- > sound/pci/oxygen/virtuoso.c | 10 +++++----- > sound/pci/oxygen/xonar_cs43xx.c | 4 ++-- > sound/pci/oxygen/xonar_pcm179x.c | 4 ++-- > sound/pci/oxygen/xonar_wm87x6.c | 4 ++--

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[alsa-devel] How to mix two PCM channels with ALSA
by itxEpsilon 06 Dec '12

06 Dec '12

Dear Alsa developers: I'm trying to write a simple application for mixing two PCM sound channels in C++. Could anyone provide me a quick example about how to implement this operation? Thank you in advance. Itx

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Re: [alsa-devel] [PATCH 00/20] ALSA: remove __dev* attributes for retirement of HOTPLUG
by Takashi Iwai 06 Dec '12

06 Dec '12

At Thu, 6 Dec 2012 12:35:09 -0500, Bill Pemberton wrote: > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit so that HOTPLUG can be retired. > > Bill Pemberton (20): > ALSA: pci: remove __dev* attributes > ALSA: AACI: remove __dev* attributes > ALSA: pxa2xx: remove __dev* attributes > ALSA: atmel: remove __dev* attributes > ALSA: firewire-speakers: remove __dev* attributes > ALSA: mips: remove __dev* attributes > ALSA: parisc/harmony: remove __dev* attributes > ALSA: oss: remove __dev* attributes > ALSA: ad1889: remove __dev* attributes > ALSA: bt87X: remove __dev* attributes > ALSA: oxygen: remove __dev* attributes Any reason the three above to be separate patches? > ALSA: isa: remove __dev* attributes > ALSA: sound/ps3: remove __dev* attributes > ALSA: ppc: remove __dev* attributes > ALSA: sh: remove __dev* attributes > ALSA: sparc: remove __dev* attributes > ALSA: at73c213: remove __dev* attributes > ALSA: drivers: remove __dev* attributes > ALSA: snd-usb-6fire: remove __dev* attributes > ALSA: snd-usb-caiaq: remove __dev* attributes Also, these can be a single clean up for sound/usb/*... It's no big matter, but I just wonder since I don't see a logic... Takashi > sound/arm/aaci.c | 18 +++--- > sound/arm/pxa2xx-ac97-lib.c | 2 +- > sound/arm/pxa2xx-ac97.c | 6 +- > sound/atmel/abdac.c | 8 +-- > sound/atmel/ac97c.c | 10 ++-- > sound/drivers/aloop.c | 16 +++--- > sound/drivers/dummy.c | 14 ++--- > sound/drivers/ml403-ac97cr.c | 10 ++-- > sound/drivers/mpu401/mpu401.c | 18 +++--- > sound/drivers/mtpav.c | 14 ++--- > sound/drivers/mts64.c | 40 +++++++------- > sound/drivers/pcsp/pcsp.c | 14 ++--- > sound/drivers/pcsp/pcsp_input.c | 2 +- > sound/drivers/pcsp/pcsp_input.h | 2 +- > sound/drivers/pcsp/pcsp_lib.c | 2 +- > sound/drivers/pcsp/pcsp_mixer.c | 10 ++-- > sound/drivers/portman2x4.c | 18 +++--- > sound/drivers/serial-u16550.c | 32 +++++------ > sound/drivers/virmidi.c | 6 +- > sound/firewire/speakers.c | 8 +-- > sound/isa/ad1816a/ad1816a.c | 18 +++--- > sound/isa/ad1816a/ad1816a_lib.c | 19 ++++--- > sound/isa/ad1848/ad1848.c | 8 +-- > sound/isa/adlib.c | 8 +-- > sound/isa/als100.c | 22 ++++---- > sound/isa/azt2320.c | 26 ++++----- > sound/isa/cmi8328.c | 8 +-- > sound/isa/cmi8330.c | 42 +++++++------- > sound/isa/cs423x/cs4231.c | 8 +-- > sound/isa/cs423x/cs4236.c | 50 ++++++++--------- > sound/isa/es1688/es1688.c | 28 +++++----- > sound/isa/es18xx.c | 80 +++++++++++++-------------- > sound/isa/galaxy/galaxy.c | 26 ++++----- > sound/isa/gus/gusclassic.c | 15 ++--- > sound/isa/gus/gusextreme.c | 24 ++++---- > sound/isa/gus/gusmax.c | 16 +++--- > sound/isa/gus/interwave.c | 54 +++++++++--------- > sound/isa/msnd/msnd.h | 2 +- > sound/isa/msnd/msnd_pinnacle.c | 44 +++++++-------- > sound/isa/msnd/msnd_pinnacle_mixer.c | 2 +- > sound/isa/opl3sa2.c | 40 +++++++------- > sound/isa/opti9xx/miro.c | 68 +++++++++++------------ > sound/isa/opti9xx/opti92x-ad1848.c | 42 +++++++------- > sound/isa/sb/emu8000.c | 22 ++++---- > sound/isa/sb/jazz16.c | 18 +++--- > sound/isa/sb/sb16.c | 26 ++++----- > sound/isa/sb/sb8.c | 8 +-- > sound/isa/sc6000.c | 38 ++++++------- > sound/isa/sscape.c | 32 +++++------ > sound/isa/wavefront/wavefront.c | 53 +++++++++--------- > sound/isa/wavefront/wavefront_fx.c | 2 +- > sound/isa/wavefront/wavefront_midi.c | 2 +- > sound/isa/wavefront/wavefront_synth.c | 14 ++--- > sound/mips/au1x00.c | 4 +- > sound/mips/hal2.c | 14 ++--- > sound/mips/sgio2audio.c | 28 +++++----- > sound/oss/ad1848.c | 2 +- > sound/oss/kahlua.c | 10 ++-- > sound/parisc/harmony.c | 12 ++-- > sound/pci/ad1889.c | 18 +++--- > sound/pci/ak4531_codec.c | 10 ++-- > sound/pci/ali5451/ali5451.c | 32 +++++------ > sound/pci/als300.c | 14 ++--- > sound/pci/als4000.c | 12 ++-- > sound/pci/asihpi/asihpi.c | 55 +++++++++---------- > sound/pci/asihpi/hpioctl.c | 6 +- > sound/pci/asihpi/hpioctl.h | 6 +- > sound/pci/atiixp.c | 32 +++++------ > sound/pci/atiixp_modem.c | 20 +++---- > sound/pci/au88x0/au88x0.c | 10 ++-- > sound/pci/au88x0/au88x0_a3d.c | 6 +- > sound/pci/au88x0/au88x0_core.c | 2 +- > sound/pci/au88x0/au88x0_eq.c | 10 ++-- > sound/pci/au88x0/au88x0_game.c | 2 +- > sound/pci/au88x0/au88x0_mixer.c | 2 +- > sound/pci/au88x0/au88x0_mpu401.c | 2 +- > sound/pci/au88x0/au88x0_pcm.c | 6 +- > sound/pci/aw2/aw2-alsa.c | 28 +++++----- > sound/pci/azt3328.c | 22 ++++---- > sound/pci/bt87x.c | 22 ++++---- > sound/pci/ca0106/ca0106_main.c | 12 ++-- > sound/pci/ca0106/ca0106_mixer.c | 26 ++++----- > sound/pci/ca0106/ca0106_proc.c | 2 +- > sound/pci/ca0106/ca_midi.c | 2 +- > sound/pci/cmipci.c | 48 ++++++++-------- > sound/pci/cs4281.c | 30 +++++----- > sound/pci/cs46xx/cs46xx.c | 8 +-- > sound/pci/cs46xx/cs46xx_lib.c | 35 ++++++------ > sound/pci/cs5530.c | 16 +++--- > sound/pci/cs5535audio/cs5535audio.c | 18 +++--- > sound/pci/cs5535audio/cs5535audio.h | 10 ++-- > sound/pci/cs5535audio/cs5535audio_olpc.c | 10 ++-- > sound/pci/cs5535audio/cs5535audio_pcm.c | 2 +- > sound/pci/ctxfi/ctatc.c | 20 +++---- > sound/pci/ctxfi/ctatc.h | 8 +-- > sound/pci/ctxfi/cthardware.c | 4 +- > sound/pci/ctxfi/cthw20k1.c | 4 +- > sound/pci/ctxfi/cthw20k2.c | 4 +- > sound/pci/ctxfi/xfi.c | 6 +- > sound/pci/echoaudio/echoaudio.c | 46 ++++++++-------- > sound/pci/echoaudio/echoaudio.h | 4 +- > sound/pci/echoaudio/midi.c | 4 +- > sound/pci/emu10k1/emu10k1.c | 8 +-- > sound/pci/emu10k1/emu10k1_main.c | 4 +- > sound/pci/emu10k1/emu10k1x.c | 29 +++++----- > sound/pci/emu10k1/emufx.c | 25 +++++---- > sound/pci/emu10k1/emumixer.c | 22 ++++---- > sound/pci/emu10k1/emumpu401.c | 6 +- > sound/pci/emu10k1/emupcm.c | 11 ++-- > sound/pci/emu10k1/emuproc.c | 2 +- > sound/pci/emu10k1/p16v.c | 8 +-- > sound/pci/emu10k1/timer.c | 2 +- > sound/pci/ens1370.c | 52 +++++++++--------- > sound/pci/es1938.c | 20 +++---- > sound/pci/es1968.c | 42 +++++++------- > sound/pci/fm801.c | 26 ++++----- > sound/pci/hda/hda_intel.c | 48 ++++++++-------- > sound/pci/ice1712/amp.c | 6 +- > sound/pci/ice1712/aureon.c | 24 ++++---- > sound/pci/ice1712/delta.c | 42 +++++++------- > sound/pci/ice1712/ews.c | 28 +++++----- > sound/pci/ice1712/hoontech.c | 26 ++++----- > sound/pci/ice1712/ice1712.c | 94 ++++++++++++++++---------------- > sound/pci/ice1712/ice1724.c | 64 +++++++++++----------- > sound/pci/ice1712/juli.c | 26 ++++----- > sound/pci/ice1712/maya44.c | 16 +++--- > sound/pci/ice1712/phase.c | 22 ++++---- > sound/pci/ice1712/pontis.c | 10 ++-- > sound/pci/ice1712/prodigy192.c | 12 ++-- > sound/pci/ice1712/prodigy_hifi.c | 20 +++---- > sound/pci/ice1712/psc724.c | 8 +-- > sound/pci/ice1712/quartet.c | 24 ++++---- > sound/pci/ice1712/revo.c | 28 +++++----- > sound/pci/ice1712/se.c | 20 +++---- > sound/pci/ice1712/vt1720_mobo.c | 10 ++-- > sound/pci/ice1712/wtm.c | 10 ++-- > sound/pci/intel8x0.c | 56 +++++++++---------- > sound/pci/intel8x0m.c | 30 +++++----- > sound/pci/korg1212/korg1212.c | 12 ++-- > sound/pci/lola/lola.c | 14 ++--- > sound/pci/lola/lola_clock.c | 2 +- > sound/pci/lola/lola_mixer.c | 32 +++++------ > sound/pci/lola/lola_pcm.c | 4 +- > sound/pci/lola/lola_proc.c | 2 +- > sound/pci/lx6464es/lx6464es.c | 30 +++++----- > sound/pci/lx6464es/lx_core.c | 2 +- > sound/pci/lx6464es/lx_core.h | 2 +- > sound/pci/maestro3.c | 24 ++++---- > sound/pci/mixart/mixart.c | 12 ++-- > sound/pci/nm256/nm256.c | 18 +++--- > sound/pci/oxygen/oxygen.c | 10 ++-- > sound/pci/oxygen/virtuoso.c | 10 ++-- > sound/pci/oxygen/xonar_cs43xx.c | 4 +- > sound/pci/oxygen/xonar_pcm179x.c | 4 +- > sound/pci/oxygen/xonar_wm87x6.c | 4 +- > sound/pci/pcxhr/pcxhr.c | 14 ++--- > sound/pci/riptide/riptide.c | 20 +++---- > sound/pci/rme32.c | 10 ++-- > sound/pci/rme96.c | 14 ++--- > sound/pci/rme9652/hdsp.c | 14 ++--- > sound/pci/rme9652/hdspm.c | 41 +++++++------- > sound/pci/rme9652/rme9652.c | 22 ++++---- > sound/pci/sis7019.c | 16 +++--- > sound/pci/sonicvibes.c | 37 +++++++------ > sound/pci/trident/trident.c | 8 +-- > sound/pci/trident/trident_main.c | 46 ++++++++-------- > sound/pci/via82xx.c | 50 ++++++++--------- > sound/pci/via82xx_modem.c | 26 ++++----- > sound/pci/vx222/vx222.c | 14 ++--- > sound/pci/ymfpci/ymfpci.c | 12 ++-- > sound/pci/ymfpci/ymfpci_main.c | 44 ++++++++------- > sound/ppc/awacs.c | 54 +++++++++--------- > sound/ppc/beep.c | 2 +- > sound/ppc/burgundy.c | 22 ++++---- > sound/ppc/daca.c | 2 +- > sound/ppc/keywest.c | 4 +- > sound/ppc/pmac.c | 12 ++-- > sound/ppc/powermac.c | 6 +- > sound/ppc/snd_ps3.c | 12 ++-- > sound/ppc/tumbler.c | 16 +++--- > sound/sh/aica.c | 13 ++--- > sound/sh/sh_dac_audio.c | 10 ++-- > sound/sparc/amd7930.c | 16 +++--- > sound/sparc/cs4231.c | 38 ++++++------- > sound/sparc/dbri.c | 28 +++++----- > sound/spi/at73c213.c | 20 +++---- > sound/usb/6fire/chip.c | 4 +- > sound/usb/6fire/comm.c | 2 +- > sound/usb/6fire/comm.h | 2 +- > sound/usb/6fire/control.c | 8 +-- > sound/usb/6fire/control.h | 2 +- > sound/usb/6fire/firmware.h | 2 +- > sound/usb/6fire/midi.c | 2 +- > sound/usb/6fire/midi.h | 2 +- > sound/usb/6fire/pcm.c | 8 +-- > sound/usb/6fire/pcm.h | 2 +- > sound/usb/caiaq/control.c | 8 +-- > sound/usb/caiaq/device.c | 6 +- > 198 files changed, 1794 insertions(+), 1785 deletions(-) > > -- > 1.8.0.1 >

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Re: [alsa-devel] [PATCH 20/20] ALSA: snd-usb-caiaq: remove __dev* attributes
by Daniel Mack 06 Dec '12

06 Dec '12

On 06.12.2012 18:35, Bill Pemberton wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Daniel Mack <zonque(a)gmail.com> Acked-by: Daniel Mack <zonque(a)gmail.com> Thanks! Daniel > --- > sound/usb/caiaq/control.c | 8 ++++---- > sound/usb/caiaq/device.c | 6 +++--- > 2 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c > index 00e5d0a..adb8d03 100644 > --- a/sound/usb/caiaq/control.c > +++ b/sound/usb/caiaq/control.c > @@ -137,7 +137,7 @@ static int control_put(struct snd_kcontrol *kcontrol, > return 1; > } > > -static struct snd_kcontrol_new kcontrol_template __devinitdata = { > +static struct snd_kcontrol_new kcontrol_template = { > .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, > .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, > .index = 0, > @@ -489,8 +489,8 @@ static struct caiaq_controller kontrols4_controller[] = { > { "LED: FX2: Mode", 133 | CNT_INTVAL }, > }; > > -static int __devinit add_controls(struct caiaq_controller *c, int num, > - struct snd_usb_caiaqdev *dev) > +static int add_controls(struct caiaq_controller *c, int num, > + struct snd_usb_caiaqdev *dev) > { > int i, ret; > struct snd_kcontrol *kc; > @@ -507,7 +507,7 @@ static int __devinit add_controls(struct caiaq_controller *c, int num, > return 0; > } > > -int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) > +int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) > { > int ret = 0; > > diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c > index 7da0d0a..c828f81 100644 > --- a/sound/usb/caiaq/device.c > +++ b/sound/usb/caiaq/device.c > @@ -289,7 +289,7 @@ int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, > tmp, sizeof(tmp)); > } > > -static void __devinit setup_card(struct snd_usb_caiaqdev *dev) > +static void setup_card(struct snd_usb_caiaqdev *dev) > { > int ret; > char val[4]; > @@ -407,7 +407,7 @@ static int create_card(struct usb_device *usb_dev, > return 0; > } > > -static int __devinit init_card(struct snd_usb_caiaqdev *dev) > +static int init_card(struct snd_usb_caiaqdev *dev) > { > char *c, usbpath[32]; > struct usb_device *usb_dev = dev->chip.dev; > @@ -481,7 +481,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) > return 0; > } > > -static int __devinit snd_probe(struct usb_interface *intf, > +static int snd_probe(struct usb_interface *intf, > const struct usb_device_id *id) > { > int ret; >

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[alsa-devel] [PATCH] ASoC: lm49453: Fix analog and digital volume ranges
by MR.Swami.Reddy＠ti.com 06 Dec '12

06 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix analog and digital volume ranges Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..afbefa2 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -495,7 +495,8 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), /* TLV Declarations */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(analog_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ @@ -516,9 +517,9 @@ SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),

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[alsa-devel] [PATCH 1/3] ALSA: ASoC: McASP: remove unused variables
by Daniel Mack 06 Dec '12

06 Dec '12

codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/linux/platform_data/davinci_asp.h | 1 - sound/soc/davinci/davinci-mcasp.c | 1 - sound/soc/davinci/davinci-mcasp.h | 2 -- 3 files changed, 4 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 46350b8..023a70d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1174,7 +1174,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 06 Dec '12

06 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix adc, mic and sidetone volume ranges Signed-off-by: MR Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..379fcdf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -538,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv),

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