May 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] ALSA: fix audio on HP TX25xx series notebooks
by Chuck Ebbert 22 May '09

22 May '09

ALSA: fix audio on HP TX25xx series notebooks Fixes https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4121 Taken from https://bugzilla.redhat.com/show_bug.cgi?id=498060 Signed-off-by: Adam Williamson <awilliam(a)redhat.com> --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),

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[alsa-devel] [PATCH 0/2]ASoC: TWL4030 and SDP3430 HS ramp configuration
by Peter Ujfalusi 22 May '09

22 May '09

Hello, The following series adds support for platform dependent Headset pop removal configuration for the twl4030 codec. Platform drivers can fill the twl4030_setup_data structure and pass it with snd_soc_device->codec_data to the codec driver. As an example - based on the feedback for the ramp_delay value variations - the sdp3430.c board is updated to pass the correct ramp settings. If you know for the other platforms using the twl4030 codec the correct HS ramp configuration, please update those files... Mark: This series needs the DAPM restructuring and HS pop removal related patches for twl4030 codec. --- Peter Ujfalusi (2): ASoC: TWL4030: Add support for platform dependent configuration ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal sound/soc/codecs/twl4030.c | 22 +++++++++++++++++++--- sound/soc/codecs/twl4030.h | 5 +++++ sound/soc/omap/sdp3430.c | 7 +++++++ 3 files changed, 31 insertions(+), 3 deletions(-)

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[alsa-devel] Program blocks with alsa pulse plugin
by Cristian Morales Vega 21 May '09

21 May '09

I'm not sure where exactly the problem is, probably in the alsa pulse plugin? And not sure where alsa pulse plugin should be reported, ALSA or PA? Using http://byuu.org/bsnes/ through the ALSA API with PulseAudio at some point provokes the game to block with 100% CPU usage. Stopping it I get this backtrace: Program received signal SIGTSTP, Stopped (user). 0x00007ffff2b8a386 in poll () from /lib64/libc.so.6 (gdb) bt #0 0x00007ffff2b8a386 in poll () from /lib64/libc.so.6 #1 0x00007ffff72ac94c in snd1_pcm_wait_nocheck () from /usr/lib64/libasound.so.2 #2 0x00007ffff72accb7 in snd1_pcm_write_areas () from /usr/lib64/libasound.so.2 #3 0x00007ffff72ebf3a in snd_pcm_ioplug_writei () from /usr/lib64/libasound.so.2 #4 0x000000000043e1b8 in ruby::AudioALSA::sample (this=<value optimized out>, left=<value optimized out>, right=<value optimized out>) at lib/ruby/audio/alsa.cpp:101 #5 0x000000000043ed8b in ruby::AudioInterface::sample (this=0x74c520, left=<value optimized out>, right=<value optimized out>) at lib/ruby/ruby_audio.cpp:110 #6 0x00000000004a5b82 in SNES::sDSP::enter (this=0x8528c0) at dsp/sdsp/sdsp.cpp:199 #7 0x00000000004a309a in SNES::sSMP::op_read (this=0x851860, addr=4532) at smp/ssmp/timing/timing.cpp:8 #8 0x00000000004a2652 in SNES::sSMP::enter (this=0x851860) at smp/core/memory.hpp:2 #9 0x000000000043bec0 in ?? () at lib/libco/x86-64.c:31 #10 0x0000000000000000 in ?? () There are lots of snd_pcm_writei() (lib/ruby/audio/alsa.cpp:101) calls before bsnes blocks, and the only errors are some underruns that are "fixed" with snd_pcm_recover(). Just that at some point (easier to trigger with 150% or 200% emu speed) blocks. It's using the "safe ALSA" everywhere, so... This is with openSUSE 11.1. But also with openSUSE 11.1 + PA from multimedia:libs (0.9.14) and a recent GIT snapshot of ALSA from multimedia:audio:snapshot and multimedia:audio:KMP. There is a package for Fedora in rpmfusion, and they have a readme that states the problem also happens there. To avoid this bsnes dev created a native PA backend. I inform just in case this is triggering a bug, perhaps affecting other software, you are interested in fix... PS. ...obviously bsnes never blocks when using ALSA directly.

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[alsa-devel] [PATCH] beyond ARRAY_SIZE of s3c64xx_i2s
by Roel Kluin 21 May '09

21 May '09

Do not go beyond ARRAY_SIZE of s3c64xx_i2s Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- In this case you are Maintainer, correct? diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 33c5de7..f2d3021 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -126,7 +126,7 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev, dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); - if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { + if (pdev->id < 0 || pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) { dev_err(dev, "id %d out of range\n", pdev->id); return -EINVAL; }

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[alsa-devel] About my issue with the sound on a Dell Dimension 4550.
by Kristoffer Grundström 21 May '09

21 May '09

What is the problem? I really want to know if it's about to be fixed. I'm getting tired of having no sound. https://qa.mandriva.com/show_bug.cgi?id=49048 /Kristoffer

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[alsa-devel] [PATCH 0/2] ASoC: TWL4030: DAPM restructuring and Headset pop-attenuation fix
by Peter Ujfalusi 21 May '09

21 May '09

Hello, The following series aims to fix the Headset power on/off pop noise problems. Since the twl4030 codec internal implementation have this order in HW: DAC -> Analog PGA -> Mixer While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA I needed to move the APGA control from SND_SOC_DAPM_PGA to be handled with _MIXER. Than adding a PGA to the headset outputs results correct power sequences. This patch also separates the analog and digital loopback paths, so when the analog loopback is enabled none of the DACs will be powered on, while in case of digital loopback the corresponding DACs will be powered. These are internal changes, not visible for the user. I have tested it with my custom board with the headset soldered to HeadsetL, the pop noise is really small if any. The user must also set the correct ramp delay for the given board. On my board: MCLK = 38.4MHz RAMP_DELAY = 2 (109/81/55 ms) works best (meaning 55ms ramp delay), if it is shorter, than it clicks, if it is longer, than it does not click, but the beep tone can be observed. I hope that I have not missed anything with the patches, since these are forward ported from my working tree to sound-2.6:topic/asoc. I have applied the patch from Misael on top of the current topic/asoc, so the head was: ASoC: TWL4030: Enable/disable voice digital filters I believe that this should be able to minimize the pop noise on the headset outputs. --- Peter Ujfalusi (2): ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event sound/soc/codecs/twl4030.c | 257 ++++++++++++++++++++++++++++---------------- 1 files changed, 162 insertions(+), 95 deletions(-)

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[alsa-devel] [GIT branch] Fix envy24control on glibc-2.10
by Diego E. “Flameeyes” Pettenò 21 May '09

21 May '09

I've pushed to git://git.flameeyes.eu/others/alsa/alsa-tools.git a single commit to fix building of envy24control on glibc-2.10. HTH, -- Diego Elio Pettenò — “Flameeyes” http://blog.flameeyes.eu/ If you found a .asc file in this mail and know not what it is, it's a GnuPG digital signature: http://www.gnupg.org/

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Noel Bourke 21 May '09

21 May '09

Hi Takashi, I set this up with an Auzentech X-Fi Prelude. Driver works OK but sound seemed a bit off. According to speaker-test I seem to only have one audio channel! Any thoughts? Regards, Noel

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Marcel Mulders 20 May '09

20 May '09

Hi Takashi, I tried the snd-ctxfi driver on Ubuntu Jaunty x64. Seems to work just fine for me. Audacious and Totem have no problems as far as I can see. Great work, your efforts are very much appreciated! I was wondering, is there any chance to get the headphones-output on the frontpanel working? I don't really need all the other jacks to work, just plain old analog stereo.. ;-) In one of the closed-source drivers from Creative from a year ago or so, version 1.18, that worked fine. In the opensource driver from last February, the headphones were broken again, just like with the latest snapshot alsa drivers. Can you comment on this maybe? Thanks in advance. Marcel

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Re: [alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by Takashi Iwai 20 May '09

20 May '09

[Please don't drop Cc from alsa-devel ML so that other people can follow.] At Wed, 20 May 2009 12:51:17 -0400, wirechief wrote: > > Takashi > I got brave. i did compile this: > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… > I used this: > sudo apt-get install build-essential linux-headers-$(uname -r) > ./configure --with-cards=all > make > sudo make install Note that you may have already the driver modules in the update directory in /lib/modules/*, depending on distro. As default, alsa-driver external build tries to replace the existing /lib/module/*/kernel/ module files, but the update directory always wins. Takashi > then i rebooted and re-ran the alsa script : > http://www.alsa-project.org/db/?f=249eea933a054742dc27c86822edecc0c731e1b9 > > Sorry if but i am new to this and wonder if i should of compiled more > than just this > "snapshot" the report does show the new driver however, i am including a .png of > alsamixer if that helps. > > > > On Wed, May 20, 2009 at 10:34 AM, wirechief <silvermachineman(a)gmail.com> wrote: > > thank you very much, I am trying to contact dtchen perhaps he can advise > > perhaps there is a backport for jaunty 9.04 remix that i can get a .deb > > > > On Wed, May 20, 2009 at 10:26 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> At Wed, 20 May 2009 10:23:30 -0400, > >> wirechief wrote: > >>> > >>> My skills at doing compiling is limited and not sure this netbook can > >>> accomplish > >>> this. I will look for 1.0.20 in a deb as my first option, perhaps that > >>> will work.. > >> > >> There are lots of changes since 1.0.20, so you should try anyway the > >> later version. > >> > >> Cc'ed ML, so that anyone who knows well Debian could help... > >> > >> > >> thanks, > >> > >> Takashi > >> > > > > > > > > -- > > Reach out and share life, care for others, > > > > > > -- > Reach out and share life, care for others, > [2 alsamixer.png <image/png (base64)>] >

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Re: [alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by Takashi Iwai 20 May '09

20 May '09

At Wed, 20 May 2009 10:23:30 -0400, wirechief wrote: > > My skills at doing compiling is limited and not sure this netbook can > accomplish > this. I will look for 1.0.20 in a deb as my first option, perhaps that > will work.. There are lots of changes since 1.0.20, so you should try anyway the later version. Cc'ed ML, so that anyone who knows well Debian could help... thanks, Takashi

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[alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by wirechief 20 May '09

20 May '09

Host/Kernel/OS "wirechief-notebook" running Linux 2.6.30-020630rc5-generic i686 [ Jaunty 9.04 remix ] CPU Info (1) Intel Atom N270 @ 512 KB cache flags( sse3 ) clocked at [ 1333.000 MHz ] (2) Intel Atom N270 @ 512 KB cache flags( sse3 ) clocked at [ 800.000 MHz ] Videocard Intel Mobile 945GME Express Integrated Graphics Controller X.Org 1.6.0 [ 1024x600(a)60.0hz ] Network cards Attansic Corp. L1e Gigabit Ethernet Adapter, at port: 1000 Atheros AR242x 802.11abg Wireless PCI Express Adapter Processes 154 | Uptime 2:44 | Memory 357.3/993.4MB | HDD Verbatim STORE N GO,ATA TOSHIBA MK1652GS,Generic- Multi-Card Size 166GB (1%used) | Client Shell | Infobash v3.11 my alsa info report: http://www.alsa-project.org/db/?f=83ac843507b5450f5c4d3252a0082ef6f9c6f124 I am not able to see a sound indication with : arecord -f cd -vv /dev/null -- Reach out and share life, care for others,

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[alsa-devel] [PATCH] TXx9: Add ACLC support (v2)
by Atsushi Nemoto 20 May '09

20 May '09

Add platform support for ACLC of TXx9 SoCs. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- Changes since v1: * use one platform device to provide irq/mem/dma resources and another platform device for machine driver. arch/mips/include/asm/txx9/generic.h | 5 ++++ arch/mips/include/asm/txx9/tx4927.h | 2 + arch/mips/include/asm/txx9/tx4938.h | 1 + arch/mips/include/asm/txx9/tx4939.h | 1 + arch/mips/txx9/Kconfig | 3 ++ arch/mips/txx9/generic/setup.c | 36 +++++++++++++++++++++++++++ arch/mips/txx9/generic/setup_tx4927.c | 43 +++++++++++++++++++++++++++++++++ arch/mips/txx9/generic/setup_tx4938.c | 11 ++++++++ arch/mips/txx9/generic/setup_tx4939.c | 9 +++++++ arch/mips/txx9/rbtx4927/setup.c | 8 ++++- arch/mips/txx9/rbtx4938/setup.c | 2 + arch/mips/txx9/rbtx4939/setup.c | 2 + 12 files changed, 121 insertions(+), 2 deletions(-) diff --git a/arch/mips/include/asm/txx9/generic.h b/arch/mips/include/asm/txx9/generic.h index 9cde009..8169477 100644 --- a/arch/mips/include/asm/txx9/generic.h +++ b/arch/mips/include/asm/txx9/generic.h @@ -91,4 +91,9 @@ void txx9_7segled_init(unsigned int num, void (*putc)(unsigned int pos, unsigned char val)); int txx9_7segled_putc(unsigned int pos, char c); +void __init txx9_aclc_init(unsigned long baseaddr, int irq, + unsigned int dmac_id, + unsigned int dma_chan_out, + unsigned int dma_chan_in); + #endif /* __ASM_TXX9_GENERIC_H */ diff --git a/arch/mips/include/asm/txx9/tx4927.h b/arch/mips/include/asm/txx9/tx4927.h index d92ae07..18c98c5 100644 --- a/arch/mips/include/asm/txx9/tx4927.h +++ b/arch/mips/include/asm/txx9/tx4927.h @@ -50,6 +50,7 @@ #define TX4927_NR_SIO 2 #define TX4927_SIO_REG(ch) (TX4927_REG_BASE + 0xf300 + (ch) * 0x100) #define TX4927_PIO_REG (TX4927_REG_BASE + 0xf500) +#define TX4927_ACLC_REG (TX4927_REG_BASE + 0xf700) #define TX4927_IR_ECCERR 0 #define TX4927_IR_WTOERR 1 @@ -267,5 +268,6 @@ void tx4927_setup_pcierr_irq(void); void tx4927_irq_init(void); void tx4927_mtd_init(int ch); void tx4927_dmac_init(int memcpy_chan); +void tx4927_aclc_init(unsigned int dma_chan_out, unsigned int dma_chan_in); #endif /* __ASM_TXX9_TX4927_H */ diff --git a/arch/mips/include/asm/txx9/tx4938.h b/arch/mips/include/asm/txx9/tx4938.h index 0758a0c..54e4674 100644 --- a/arch/mips/include/asm/txx9/tx4938.h +++ b/arch/mips/include/asm/txx9/tx4938.h @@ -306,5 +306,6 @@ struct tx4938ide_platform_info { void tx4938_ata_init(unsigned int irq, unsigned int shift, int tune); void tx4938_dmac_init(int memcpy_chan0, int memcpy_chan1); +void tx4938_aclc_init(void); #endif diff --git a/arch/mips/include/asm/txx9/tx4939.h b/arch/mips/include/asm/txx9/tx4939.h index 1be9798..f13b708 100644 --- a/arch/mips/include/asm/txx9/tx4939.h +++ b/arch/mips/include/asm/txx9/tx4939.h @@ -545,5 +545,6 @@ void tx4939_rtc_init(void); void tx4939_ndfmc_init(unsigned int hold, unsigned int spw, unsigned char ch_mask, unsigned char wide_mask); void tx4939_dmac_init(int memcpy_chan0, int memcpy_chan1); +void tx4939_aclc_init(void); #endif /* __ASM_TXX9_TX4939_H */ diff --git a/arch/mips/txx9/Kconfig b/arch/mips/txx9/Kconfig index 0db7cf3..852ae4b 100644 --- a/arch/mips/txx9/Kconfig +++ b/arch/mips/txx9/Kconfig @@ -69,6 +69,7 @@ config SOC_TX4927 select IRQ_TXX9 select PCI_TX4927 select GPIO_TXX9 + select HAS_TXX9_ACLC config SOC_TX4938 bool @@ -78,6 +79,7 @@ config SOC_TX4938 select IRQ_TXX9 select PCI_TX4927 select GPIO_TXX9 + select HAS_TXX9_ACLC config SOC_TX4939 bool @@ -85,6 +87,7 @@ config SOC_TX4939 select HAS_TXX9_SERIAL select HW_HAS_PCI select PCI_TX4927 + select HAS_TXX9_ACLC config TXX9_7SEGLED bool diff --git a/arch/mips/txx9/generic/setup.c b/arch/mips/txx9/generic/setup.c index 369d863..7f91012 100644 --- a/arch/mips/txx9/generic/setup.c +++ b/arch/mips/txx9/generic/setup.c @@ -876,3 +876,39 @@ void __init txx9_dmac_init(int id, unsigned long baseaddr, int irq, } #endif } + +void __init txx9_aclc_init(unsigned long baseaddr, int irq, + unsigned int dmac_id, + unsigned int dma_chan_out, + unsigned int dma_chan_in) +{ +#if defined(CONFIG_SND_SOC_TXX9ACLC) || \ + defined(CONFIG_SND_SOC_TXX9ACLC_MODULE) + unsigned int dma_base = dmac_id * TXX9_DMA_MAX_NR_CHANNELS; + struct resource res[] = { + { + .start = baseaddr, + .end = baseaddr + 0x100 - 1, + .flags = IORESOURCE_MEM, + }, { + .start = irq, + .flags = IORESOURCE_IRQ, + }, { + .name = "txx9dmac-chan", + .start = dma_base + dma_chan_out, + .flags = IORESOURCE_DMA, + }, { + .name = "txx9dmac-chan", + .start = dma_base + dma_chan_in, + .flags = IORESOURCE_DMA, + } + }; + struct platform_device *pdev = + platform_device_alloc("txx9aclc-ac97", -1); + + if (!pdev || + platform_device_add_resources(pdev, res, ARRAY_SIZE(res)) || + platform_device_add(pdev)) + platform_device_put(pdev); +#endif +} diff --git a/arch/mips/txx9/generic/setup_tx4927.c b/arch/mips/txx9/generic/setup_tx4927.c index 6b681cd..3418b2a 100644 --- a/arch/mips/txx9/generic/setup_tx4927.c +++ b/arch/mips/txx9/generic/setup_tx4927.c @@ -265,6 +265,49 @@ void __init tx4927_dmac_init(int memcpy_chan) TXX9_IRQ_BASE + TX4927_IR_DMA(0), &plat_data); } +void __init tx4927_aclc_init(unsigned int dma_chan_out, + unsigned int dma_chan_in) +{ + u64 pcfg = __raw_readq(&tx4927_ccfgptr->pcfg); + __u64 dmasel_mask = 0, dmasel = 0; + unsigned long flags; + + if (!(pcfg & TX4927_PCFG_SEL2)) + return; + /* setup DMASEL (playback:ACLC ch0, capture:ACLC ch1) */ + switch (dma_chan_out) { + case 0: + dmasel_mask |= TX4927_PCFG_DMASEL0_MASK; + dmasel |= TX4927_PCFG_DMASEL0_ACL0; + break; + case 2: + dmasel_mask |= TX4927_PCFG_DMASEL2_MASK; + dmasel |= TX4927_PCFG_DMASEL2_ACL0; + break; + default: + return; + } + switch (dma_chan_in) { + case 1: + dmasel_mask |= TX4927_PCFG_DMASEL1_MASK; + dmasel |= TX4927_PCFG_DMASEL1_ACL1; + break; + case 3: + dmasel_mask |= TX4927_PCFG_DMASEL3_MASK; + dmasel |= TX4927_PCFG_DMASEL3_ACL1; + break; + default: + return; + } + local_irq_save(flags); + txx9_clear64(&tx4927_ccfgptr->pcfg, dmasel_mask); + txx9_set64(&tx4927_ccfgptr->pcfg, dmasel); + local_irq_restore(flags); + txx9_aclc_init(TX4927_ACLC_REG & 0xfffffffffULL, + TXX9_IRQ_BASE + TX4927_IR_ACLC, + 0, dma_chan_out, dma_chan_in); +} + static void __init tx4927_stop_unused_modules(void) { __u64 pcfg, rst = 0, ckd = 0; diff --git a/arch/mips/txx9/generic/setup_tx4938.c b/arch/mips/txx9/generic/setup_tx4938.c index b2b8529..4dfdb52 100644 --- a/arch/mips/txx9/generic/setup_tx4938.c +++ b/arch/mips/txx9/generic/setup_tx4938.c @@ -414,6 +414,17 @@ void __init tx4938_dmac_init(int memcpy_chan0, int memcpy_chan1) } } +void __init tx4938_aclc_init(void) +{ + u64 pcfg = __raw_readq(&tx4938_ccfgptr->pcfg); + + if ((pcfg & TX4938_PCFG_SEL2) && + !(pcfg & TX4938_PCFG_ETH0_SEL)) + txx9_aclc_init(TX4938_ACLC_REG & 0xfffffffffULL, + TXX9_IRQ_BASE + TX4938_IR_ACLC, + 1, 0, 1); +} + static void __init tx4938_stop_unused_modules(void) { __u64 pcfg, rst = 0, ckd = 0; diff --git a/arch/mips/txx9/generic/setup_tx4939.c b/arch/mips/txx9/generic/setup_tx4939.c index 98effef..7139686 100644 --- a/arch/mips/txx9/generic/setup_tx4939.c +++ b/arch/mips/txx9/generic/setup_tx4939.c @@ -485,6 +485,15 @@ void __init tx4939_dmac_init(int memcpy_chan0, int memcpy_chan1) } } +void __init tx4939_aclc_init(void) +{ + u64 pcfg = __raw_readq(&tx4939_ccfgptr->pcfg); + + if ((pcfg & TX4939_PCFG_I2SMODE_MASK) == TX4939_PCFG_I2SMODE_ACLC) + txx9_aclc_init(TX4939_ACLC_REG & 0xfffffffffULL, + TXX9_IRQ_BASE + TX4939_IR_ACLC, 1, 0, 1); +} + static void __init tx4939_stop_unused_modules(void) { __u64 pcfg, rst = 0, ckd = 0; diff --git a/arch/mips/txx9/rbtx4927/setup.c b/arch/mips/txx9/rbtx4927/setup.c index 332cdbc..ee468ea 100644 --- a/arch/mips/txx9/rbtx4927/setup.c +++ b/arch/mips/txx9/rbtx4927/setup.c @@ -337,10 +337,14 @@ static void __init rbtx4927_device_init(void) rbtx4927_ne_init(); tx4927_wdt_init(); rbtx4927_mtd_init(); - if (TX4927_REV_PCODE() == 0x4927) + if (TX4927_REV_PCODE() == 0x4927) { tx4927_dmac_init(2); - else + tx4927_aclc_init(0, 1); + } else { tx4938_dmac_init(0, 2); + tx4938_aclc_init(); + } + platform_device_register_simple("txx9aclc-generic", -1, NULL, 0); txx9_iocled_init(RBTX4927_LED_ADDR - IO_BASE, -1, 3, 1, "green", NULL); rbtx4927_gpioled_init(); } diff --git a/arch/mips/txx9/rbtx4938/setup.c b/arch/mips/txx9/rbtx4938/setup.c index 37c5e3d..8da66e9 100644 --- a/arch/mips/txx9/rbtx4938/setup.c +++ b/arch/mips/txx9/rbtx4938/setup.c @@ -356,6 +356,8 @@ static void __init rbtx4938_device_init(void) tx4938_ndfmc_init(10, 35); tx4938_ata_init(RBTX4938_IRQ_IOC_ATA, 0, 1); tx4938_dmac_init(0, 2); + tx4938_aclc_init(); + platform_device_register_simple("txx9aclc-generic", -1, NULL, 0); txx9_iocled_init(RBTX4938_LED_ADDR - IO_BASE, -1, 8, 1, "green", NULL); } diff --git a/arch/mips/txx9/rbtx4939/setup.c b/arch/mips/txx9/rbtx4939/setup.c index 91f2ec8..d5ad5ab 100644 --- a/arch/mips/txx9/rbtx4939/setup.c +++ b/arch/mips/txx9/rbtx4939/setup.c @@ -499,6 +499,8 @@ static void __init rbtx4939_device_init(void) tx4939_ata_init(); tx4939_rtc_init(); tx4939_dmac_init(0, 2); + tx4939_aclc_init(); + platform_device_register_simple("txx9aclc-generic", -1, NULL, 0); } static void __init rbtx4939_setup(void) -- 1.5.6.5

3 2

[alsa-devel] [PATCH 1/1] Add AIC3106 support.
by Ambrose, Martin 19 May '09

19 May '09

Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Martin Ambrose <martin(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 185 ++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic3x.h | 11 ++- 2 files changed, 190 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f4..5720cad 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -57,13 +57,17 @@ struct aic3x_priv { int master; }; +static u8 pg0_end = AIC3X_GEN_PG0_END; +static u8 pg1_end = AIC3X_GEN_PG1_END; +static enum aic3x_codec_variant codec_variant; + /* * AIC3X register cache * We can't read the AIC3X register space when we are * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { +static const u8 aic3x_reg[] = { 0x00, 0x00, 0x00, 0x10, /* 0 */ 0x04, 0x00, 0x00, 0x00, /* 4 */ 0x00, 0x00, 0x00, 0x01, /* 8 */ @@ -90,6 +94,35 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ 0x00, 0x00, 0x02, /* 100 */ + + 0x00, 0x00, 0x00, 0x00, /* 103-127 unused */ + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, + 0x00, + + 0x00, 0x6b, 0xe3, 0x96, /* 0 */ + 0x66, 0x67, 0x5d, 0x6b, /* 4 */ + 0xe3, 0x96, 0x66, 0x67, /* 8 */ + 0x5d, 0x7d, 0x83, 0x84, /* 12 */ + 0xee, 0x7d, 0x83, 0x84, /* 16 */ + 0xee, 0x39, 0x55, 0xf3, /* 20 */ + 0x2d, 0x53, 0x7e, 0x6b, /* 24 */ + 0xe3, 0x96, 0x66, 0x67, /* 28 */ + 0x5d, 0x6b, 0xe3, 0x96, /* 32 */ + 0x66, 0x67, 0x5d, 0x7d, /* 36 */ + 0x83, 0x84, 0xee, 0x7d, /* 40 */ + 0x83, 0x84, 0xee, 0x39, /* 44 */ + 0x55, 0xf3, 0x2d, 0x53, /* 48 */ + 0x7e, 0x7f, 0xff, 0x00, /* 52 */ + 0x00, 0x00, 0x00, 0x00, /* 56 */ + 0x00, 0x00, 0x00, 0x00, /* 60 */ + 0x00, 0x39, 0x55, 0xf3, /* 64 */ + 0x2d, 0x53, 0x7e, 0x39, /* 68 */ + 0x55, 0xf3, 0x2d, 0x53, /* 72 */ + 0x7e, /* 76 */ }; /* @@ -99,7 +132,8 @@ static inline unsigned int aic3x_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u8 *cache = codec->reg_cache; - if (reg >= AIC3X_CACHEREGNUM) + if ((reg >= pg1_end) || + ((reg >= pg0_end) && (reg < AIC3X_GEN_PG1_BEG))) return -1; return cache[reg]; } @@ -111,7 +145,8 @@ static inline void aic3x_write_reg_cache(struct snd_soc_codec *codec, u8 reg, u8 value) { u8 *cache = codec->reg_cache; - if (reg >= AIC3X_CACHEREGNUM) + if ((reg >= pg1_end) || + ((reg >= pg0_end) && (reg < AIC3X_GEN_PG1_BEG))) return; cache[reg] = value; } @@ -119,8 +154,8 @@ static inline void aic3x_write_reg_cache(struct snd_soc_codec *codec, /* * write to the aic3x register space */ -static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) +static int _aic3x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { u8 data[2]; @@ -132,12 +167,41 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; aic3x_write_reg_cache(codec, data[0], data[1]); + + /* adjust for page 1 before updating hardware if necessary */ + if (data[0] >= AIC3X_GEN_PG1_BEG) + data[0] -= AIC3X_GEN_PG1_BEG; + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else return -EIO; } +static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 cur_pg; + u8 reg_pg; + int ret = 0; + + cur_pg = aic3x_read_reg_cache(codec, 0); + if (reg < pg0_end) + reg_pg = 0; + else if ((reg >= AIC3X_GEN_PG1_BEG) && (reg < pg1_end)) + reg_pg = 1; + else + return -EIO; + + if (cur_pg != reg_pg) + ret = _aic3x_write(codec, 0, reg_pg); + + if (ret == 0) + ret = _aic3x_write(codec, reg, value); + + return ret; +} + /* * read from the aic3x register space */ @@ -218,6 +282,65 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +static int tlv320aic3x_dual_reg_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xffff; + return 0; +} + +static int tlv320aic3x_dual_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg_msb = kcontrol->private_value & 0xff; + int reg_lsb = (kcontrol->private_value >> 8) & 0xff; + int val = aic3x_read_reg_cache(codec, reg_msb) << 8; + + val |= aic3x_read_reg_cache(codec, reg_lsb); + + /* convert 2's complement to unsigned int */ + val ^= 0x8000; + + ucontrol->value.integer.value[0] = val; + + return 0; +} + +static int tlv320aic3x_dual_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int err; + int reg_msb = kcontrol->private_value & 0xff; + int reg_lsb = (kcontrol->private_value >> 8) & 0xff; + int val_msb, val_lsb; + + val_msb = (ucontrol->value.integer.value[0] >> 8) & 0xff; + val_lsb = ucontrol->value.integer.value[0] & 0xff; + + /* convert unsigned int to 2's complement */ + val_msb ^= 0x80; + + err = snd_soc_update_bits(codec, reg_msb, 0xff, val_msb); + if (err < 0) + return err; + err = snd_soc_update_bits(codec, reg_lsb, 0xff, val_lsb); + return err; +} + +#define TLV320AIC3X_DUAL_R(xname, page, reg_msb, reg_lsb) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = tlv320aic3x_dual_reg_info, \ + .get = tlv320aic3x_dual_reg_get, .put = tlv320aic3x_dual_reg_put, \ + .private_value = ((reg_msb) + page) | (((reg_lsb) + page) << 8) } + +#define TLV320AIC3X_PG1_DUAL_R(xname, reg_msb, reg_lsb) \ + TLV320AIC3X_DUAL_R(xname, AIC3X_GEN_PG1_BEG, reg_msb, reg_lsb) + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -344,6 +467,36 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N0", 1, 2), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N1", 3, 4), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N2", 5, 6), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N3", 7, 8), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N4", 9, 10), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient N5", 11, 12), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D1", 13, 14), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D2", 15, 16), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D4", 17, 18), + TLV320AIC3X_PG1_DUAL_R("Left Effects Coefficient D5", 19, 20), + TLV320AIC3X_PG1_DUAL_R("Left De-Emphasis Coefficient N0", 21, 22), + TLV320AIC3X_PG1_DUAL_R("Left De-Emphasis Coefficient N1", 23, 24), + TLV320AIC3X_PG1_DUAL_R("Left De-Emphasis Coefficient D1", 25, 26), + + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N0", 27, 28), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N1", 29, 30), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N2", 31, 32), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N3", 33, 34), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N4", 35, 36), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient N5", 37, 38), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D1", 39, 40), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D2", 41, 42), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D4", 43, 44), + TLV320AIC3X_PG1_DUAL_R("Right Effects Coefficient D5", 45, 46), + TLV320AIC3X_PG1_DUAL_R("Right De-Emphasis Coefficient N0", 47, 48), + TLV320AIC3X_PG1_DUAL_R("Right De-Emphasis Coefficient N1", 49, 50), + TLV320AIC3X_PG1_DUAL_R("Right De-Emphasis Coefficient D1", 51, 52), + + TLV320AIC3X_PG1_DUAL_R("3-D Attenuation Coefficient", 53, 54), }; /* Left DAC Mux */ @@ -454,6 +607,18 @@ static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = { SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; +static const struct snd_kcontrol_new aic3106_snd_controls[] = { + TLV320AIC3X_PG1_DUAL_R("Left Capture High Pass Coefficient N0", 65, 66), + TLV320AIC3X_PG1_DUAL_R("Left Capture High Pass Coefficient N1", 67, 68), + TLV320AIC3X_PG1_DUAL_R("Left Capture High Pass Coefficient D1", 69, 70), + TLV320AIC3X_PG1_DUAL_R("Right Capture High Pass Coefficient N0", + 71, 72), + TLV320AIC3X_PG1_DUAL_R("Right Capture High Pass Coefficient N1", + 73, 74), + TLV320AIC3X_PG1_DUAL_R("Right Capture High Pass Coefficient D1", + 75, 76), +}; + static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Left DAC to Left Outputs */ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0), @@ -1165,6 +1330,10 @@ static int aic3x_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + /* setup register cache sizes */ + if (codec_variant == AIC3106_CODEC) + pg1_end = AIC3106_PG1_END; + aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); aic3x_write(codec, AIC3X_RESET, SOFT_RESET); @@ -1247,6 +1416,11 @@ static int aic3x_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, aic3x_snd_controls, ARRAY_SIZE(aic3x_snd_controls)); + + if (codec_variant == AIC3106_CODEC) + snd_soc_add_controls(codec, aic3106_snd_controls, + ARRAY_SIZE(aic3106_snd_controls)); + aic3x_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -1374,6 +1548,7 @@ static int aic3x_probe(struct platform_device *pdev) printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION); setup = socdev->codec_data; + codec_variant = setup->variant; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index ac827e5..ee40e9c 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -13,7 +13,10 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_GEN_PG0_END 103 +#define AIC3X_GEN_PG1_BEG 128 +#define AIC3X_GEN_PG1_END 183 +#define AIC3106_PG1_END 205 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -199,6 +202,11 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +enum aic3x_codec_variant { + AIC3X_GENERIC_CODEC, + AIC3106_CODEC, +}; + /* GPIO API */ enum { AIC3X_GPIO1_FUNC_DISABLED = 0, @@ -284,6 +292,7 @@ int aic3x_button_pressed(struct snd_soc_codec *codec); struct aic3x_setup_data { int i2c_bus; unsigned short i2c_address; + enum aic3x_codec_variant variant; unsigned int gpio_func[2]; }; -- 1.6.0.6

2 3

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 19 May '09

19 May '09

The following changes since commit f83fba8baab9e95fff0fe2be0e1e32a1650bdd7f: Mark Brown (1): ASoC: Add debug trace for bias level transitions are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Atsushi Nemoto (1): ASoC: Add TXx9 AC link controller driver (v3) Lopez Cruz, Misael (2): ASoC: TWL4030: Add control for selecting codec operation mode ASoC: SDP3430: Connect twl4030 voice DAI to McBSP3 Peter Ujfalusi (1): ASoC: TWL4030: Fix Analog capture path for AUXR sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/twl4030.c | 49 ++++- sound/soc/omap/sdp3430.c | 87 +++++++- sound/soc/txx9/Kconfig | 29 +++ sound/soc/txx9/Makefile | 11 + sound/soc/txx9/txx9aclc-ac97.c | 255 ++++++++++++++++++++++ sound/soc/txx9/txx9aclc-generic.c | 98 +++++++++ sound/soc/txx9/txx9aclc.c | 430 +++++++++++++++++++++++++++++++++++++ sound/soc/txx9/txx9aclc.h | 83 +++++++ 10 files changed, 1033 insertions(+), 11 deletions(-) create mode 100644 sound/soc/txx9/Kconfig create mode 100644 sound/soc/txx9/Makefile create mode 100644 sound/soc/txx9/txx9aclc-ac97.c create mode 100644 sound/soc/txx9/txx9aclc-generic.c create mode 100644 sound/soc/txx9/txx9aclc.c create mode 100644 sound/soc/txx9/txx9aclc.h

2 1

[alsa-devel] [PATCH] ASoC: Add TXx9 AC link controller driver (v3)
by Atsushi Nemoto 19 May '09

19 May '09

This patch adds support for the integrated ACLC of the TXx9 family. Signed-off-by: Atsushi Nemoto <anemo(a)mba.ocn.ne.jp> --- Changes since v2: * register DAI driver as a platform driver * use platform device to obtain irq/mem/dma resources. sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/txx9/Kconfig | 29 +++ sound/soc/txx9/Makefile | 11 + sound/soc/txx9/txx9aclc-ac97.c | 255 ++++++++++++++++++++++ sound/soc/txx9/txx9aclc-generic.c | 98 +++++++++ sound/soc/txx9/txx9aclc.c | 430 +++++++++++++++++++++++++++++++++++++ sound/soc/txx9/txx9aclc.h | 83 +++++++ 8 files changed, 908 insertions(+), 0 deletions(-) create mode 100644 sound/soc/txx9/Kconfig create mode 100644 sound/soc/txx9/Makefile create mode 100644 sound/soc/txx9/txx9aclc-ac97.c create mode 100644 sound/soc/txx9/txx9aclc-generic.c create mode 100644 sound/soc/txx9/txx9aclc.c create mode 100644 sound/soc/txx9/txx9aclc.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3d2bb6f..85e985a 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/txx9/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..6039b0d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,3 +11,4 @@ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig new file mode 100644 index 0000000..ebc9327 --- /dev/null +++ b/sound/soc/txx9/Kconfig @@ -0,0 +1,29 @@ +## +## TXx9 ACLC +## +config SND_SOC_TXX9ACLC + tristate "SoC Audio for TXx9" + depends on HAS_TXX9_ACLC && TXX9_DMAC + help + This option enables support for the AC Link Controllers in TXx9 SoC. + +config HAS_TXX9_ACLC + bool + +config SND_SOC_TXX9ACLC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_TXX9ACLC_GENERIC + tristate "Generic TXx9 ACLC sound machine" + depends on SND_SOC_TXX9ACLC + select SND_SOC_TXX9ACLC_AC97 + select SND_SOC_AC97_CODEC + help + This is a generic AC97 sound machine for use in TXx9 based systems. diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile new file mode 100644 index 0000000..551f16c --- /dev/null +++ b/sound/soc/txx9/Makefile @@ -0,0 +1,11 @@ +# Platform +snd-soc-txx9aclc-objs := txx9aclc.o +snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o +obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o + +# Machine +snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c new file mode 100644 index 0000000..0f83bdb --- /dev/null +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -0,0 +1,255 @@ +/* + * TXx9 ACLC AC97 driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include "txx9aclc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#ifdef __BIG_ENDIAN +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_BE +#else +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_LE +#endif + +static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq); + +/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */ +static struct txx9aclc_soc_device *txx9aclc_soc_dev; + +static int txx9aclc_regready(struct txx9aclc_soc_device *dev) +{ + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + + return __raw_readl(drvdata->base + ACINTSTS) & ACINT_REGACCRDY; +} + +/* AC97 controller reads codec register */ +static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + u32 dat; + + if (!(__raw_readl(base + ACINTSTS) & ACINT_CODECRDY(ac97->num))) + return 0xffff; + reg |= ac97->num << 7; + dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ; + __raw_writel(dat, base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); + dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg); + dat = 0xffff; + goto done; + } + dat = __raw_readl(base + ACREGACC); + if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) { + dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n", + dat, reg); + dat = 0xffff; + goto done; + } + dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff; +done: + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); + return dat; +} + +/* AC97 controller writes to codec register */ +static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + + __raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) | + (val << ACREGACC_DAT_SHIFT), + base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + dev_err(dev->soc_dev.dev, + "ac97 write timeout (reg %#x)\n", reg); + } + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); +} + +static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY; + + __raw_writel(ACCTL_ENLINK, base + ACCTLDIS); + mmiowb(); + udelay(1); + __raw_writel(ACCTL_ENLINK, base + ACCTLEN); + /* wait for primary codec ready status */ + __raw_writel(ready, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, + (__raw_readl(base + ACINTSTS) & ready) == ready, + HZ)) { + dev_err(&ac97->dev, "primary codec is not ready " + "(status %#x)\n", + __raw_readl(base + ACINTSTS)); + } + __raw_writel(ACINT_REGACCRDY, base + ACINTSTS); + __raw_writel(ready, base + ACINTDIS); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = txx9aclc_ac97_read, + .write = txx9aclc_ac97_write, + .reset = txx9aclc_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) +{ + struct txx9aclc_plat_drvdata *drvdata = dev_id; + void __iomem *base = drvdata->base; + + __raw_writel(__raw_readl(base + ACINTMSTS), base + ACINTDIS); + wake_up(&ac97_waitq); + return IRQ_HANDLED; +} + +static int txx9aclc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + + dev->aclc_pdev = to_platform_device(dai->dev); + txx9aclc_soc_dev = dev; + return 0; +} + +static void txx9aclc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct platform_device *aclc_pdev = to_platform_device(dai->dev); + struct txx9aclc_plat_drvdata *drvdata = platform_get_drvdata(aclc_pdev); + + /* disable AC-link */ + __raw_writel(ACCTL_ENLINK, drvdata->base + ACCTLDIS); + txx9aclc_soc_dev = NULL; +} + +struct snd_soc_dai txx9aclc_ac97_dai = { + .name = "txx9aclc_ac97", + .ac97_control = 1, + .probe = txx9aclc_ac97_probe, + .remove = txx9aclc_ac97_remove, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, +}; +EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai); + +static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev) +{ + struct txx9aclc_plat_drvdata *drvdata; + struct resource *r; + int err; + int irq; + + irq = platform_get_irq(pdev, 0); + if (irq < 0) + return irq; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) + return -EBUSY; + + if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), + dev_name(&pdev->dev))) + return -EBUSY; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + platform_set_drvdata(pdev, drvdata); + drvdata->physbase = r->start; + if (sizeof(drvdata->physbase) > sizeof(r->start) && + r->start >= TXX9_DIRECTMAP_BASE && + r->start < TXX9_DIRECTMAP_BASE + 0x400000) + drvdata->physbase |= 0xf00000000ull; + drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); + if (!drvdata->base) + return -EBUSY; + err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, + IRQF_DISABLED, dev_name(&pdev->dev), drvdata); + if (err < 0) + return err; + + txx9aclc_ac97_dai.dev = &pdev->dev; + return snd_soc_register_dai(&txx9aclc_ac97_dai); +} + +static int __devexit txx9aclc_ac97_dev_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&txx9aclc_ac97_dai); + return 0; +} + +static struct platform_driver txx9aclc_ac97_driver = { + .probe = txx9aclc_ac97_dev_probe, + .remove = __devexit_p(txx9aclc_ac97_dev_remove), + .driver = { + .name = "txx9aclc-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_ac97_init(void) +{ + return platform_driver_register(&txx9aclc_ac97_driver); +} + +static void __exit txx9aclc_ac97_exit(void) +{ + platform_driver_unregister(&txx9aclc_ac97_driver); +} + +module_init(txx9aclc_ac97_init); +module_exit(txx9aclc_ac97_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c new file mode 100644 index 0000000..3175de9 --- /dev/null +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -0,0 +1,98 @@ +/* + * Generic TXx9 ACLC machine driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at ACLC (e.g. RBTX49XX boards). + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include "../codecs/ac97.h" +#include "txx9aclc.h" + +static struct snd_soc_dai_link txx9aclc_generic_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &txx9aclc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card txx9aclc_generic_card = { + .name = "Generic TXx9 ACLC Audio", + .platform = &txx9aclc_soc_platform, + .dai_link = &txx9aclc_generic_dai, + .num_links = 1, +}; + +static struct txx9aclc_soc_device txx9aclc_generic_soc_device = { + .soc_dev = { + .card = &txx9aclc_generic_card, + .codec_dev = &soc_codec_dev_ac97, + }, +}; + +static int __init txx9aclc_generic_probe(struct platform_device *pdev) +{ + struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device; + struct platform_device *soc_pdev; + int ret; + + soc_pdev = platform_device_alloc("soc-audio", -1); + if (!soc_pdev) + return -ENOMEM; + platform_set_drvdata(soc_pdev, &dev->soc_dev); + dev->soc_dev.dev = &soc_pdev->dev; + ret = platform_device_add(soc_pdev); + if (ret) { + platform_device_put(soc_pdev); + return ret; + } + platform_set_drvdata(pdev, soc_pdev); + return 0; +} + +static int __exit txx9aclc_generic_remove(struct platform_device *pdev) +{ + struct platform_device *soc_pdev = platform_get_drvdata(pdev); + + platform_device_unregister(soc_pdev); + return 0; +} + +static struct platform_driver txx9aclc_generic_driver = { + .remove = txx9aclc_generic_remove, + .driver = { + .name = "txx9aclc-generic", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_generic_init(void) +{ + return platform_driver_probe(&txx9aclc_generic_driver, + txx9aclc_generic_probe); +} + +static void __exit txx9aclc_generic_exit(void) +{ + platform_driver_unregister(&txx9aclc_generic_driver); +} + +module_init(txx9aclc_generic_init); +module_exit(txx9aclc_generic_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c new file mode 100644 index 0000000..fa33661 --- /dev/null +++ b/sound/soc/txx9/txx9aclc.c @@ -0,0 +1,430 @@ +/* + * Generic TXx9 ACLC platform driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/scatterlist.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "txx9aclc.h" + +static const struct snd_pcm_hardware txx9aclc_pcm_hardware = { + /* + * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID + * needs more works for noncoherent MIPS. + */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE, +#ifdef __BIG_ENDIAN + .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif + .period_bytes_min = 1024, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 4096, + .buffer_bytes_max = 32 * 1024, +}; + +static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_device *socdev = rtd->socdev; + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + dev_dbg(socdev->dev, + "runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd " + "runtime->min_align %ld\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + dev_dbg(socdev->dev, + "periods %d period_bytes %d stream %d\n", + params_periods(params), params_period_bytes(params), + substream->stream); + + dmadata->substream = substream; + dmadata->pos = 0; + return 0; +} + +static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + + dmadata->dma_addr = runtime->dma_addr; + dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + dmadata->period_bytes = snd_pcm_lib_period_bytes(substream); + + if (dmadata->buffer_bytes == dmadata->period_bytes) { + dmadata->frag_bytes = dmadata->period_bytes >> 1; + dmadata->frags = 2; + } else { + dmadata->frag_bytes = dmadata->period_bytes; + dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes; + } + dmadata->frag_count = 0; + dmadata->pos = 0; + return 0; +} + +static void txx9aclc_dma_complete(void *arg) +{ + struct txx9aclc_dmadata *dmadata = arg; + unsigned long flags; + + /* dma completion handler cannot submit new operations */ + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count >= 0) { + dmadata->dmacount--; + BUG_ON(dmadata->dmacount < 0); + tasklet_schedule(&dmadata->tasklet); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static struct dma_async_tx_descriptor * +txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) +{ + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct scatterlist sg; + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)), + dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1)); + sg_dma_address(&sg) = buf_dma_addr; + desc = chan->device->device_prep_slave_sg(chan, &sg, 1, + dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return NULL; + } + desc->callback = txx9aclc_dma_complete; + desc->callback_param = dmadata; + desc->tx_submit(desc); + return desc; +} + +#define NR_DMA_CHAIN 2 + +static void txx9aclc_dma_tasklet(unsigned long data) +{ + struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data; + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct snd_pcm_substream *substream = dmadata->substream; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + int i; + unsigned long flags; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count < 0) { + struct txx9aclc_soc_device *dev = + container_of(dmadata, struct txx9aclc_soc_device, + dmadata[substream->stream]); + struct txx9aclc_plat_drvdata *drvdata = + txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + chan->device->device_terminate_all(chan); + /* first time */ + for (i = 0; i < NR_DMA_CHAIN; i++) { + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + i * dmadata->frag_bytes); + if (!desc) + return; + } + dmadata->dmacount = NR_DMA_CHAIN; + chan->device->device_issue_pending(chan); + spin_lock_irqsave(&dmadata->dma_lock, flags); + __raw_writel(ctlbit, base + ACCTLEN); + dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return; + } + BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN); + while (dmadata->dmacount < NR_DMA_CHAIN) { + dmadata->dmacount++; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + + dmadata->frag_count * dmadata->frag_bytes); + if (!desc) + return; + chan->device->device_issue_pending(chan); + + spin_lock_irqsave(&dmadata->dma_lock, flags); + dmadata->frag_count++; + dmadata->frag_count %= dmadata->frags; + dmadata->pos += dmadata->frag_bytes; + dmadata->pos %= dmadata->buffer_bytes; + if ((dmadata->frag_count * dmadata->frag_bytes) % + dmadata->period_bytes == 0) + snd_pcm_period_elapsed(substream); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + unsigned long flags; + int ret = 0; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dmadata->frag_count = -1; + tasklet_schedule(&dmadata->tasklet); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + __raw_writel(ctlbit, base + ACCTLDIS); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + __raw_writel(ctlbit, base + ACCTLEN); + break; + default: + ret = -EINVAL; + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return ret; +} + +static snd_pcm_uframes_t +txx9aclc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + + return bytes_to_frames(substream->runtime, dmadata->pos); +} + +static int txx9aclc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream]; + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware); + if (ret) + return ret; + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + substream->runtime->private_data = dmadata; + return 0; +} + +static int txx9aclc_pcm_close(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct dma_chan *chan = dmadata->dma_chan; + + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + return 0; +} + +static struct snd_pcm_ops txx9aclc_pcm_ops = { + .open = txx9aclc_pcm_open, + .close = txx9aclc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = txx9aclc_pcm_hw_params, + .hw_free = txx9aclc_pcm_hw_free, + .prepare = txx9aclc_pcm_prepare, + .trigger = txx9aclc_pcm_trigger, + .pointer = txx9aclc_pcm_pointer, +}; + +static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, 64 * 1024, 4 * 1024 * 1024); +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct txx9aclc_dmadata *dmadata = param; + char devname[BUS_ID_SIZE + 2]; + + sprintf(devname, "%s.%d", dmadata->dma_res->name, + (int)dmadata->dma_res->start); + if (strcmp(dev_name(chan->device->dev), devname) == 0) { + chan->private = &dmadata->dma_slave; + return true; + } + return false; +} + +static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, + struct txx9aclc_dmadata *dmadata) +{ + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + struct txx9dmac_slave *ds = &dmadata->dma_slave; + dma_cap_mask_t mask; + + spin_lock_init(&dmadata->dma_lock); + + ds->reg_width = sizeof(u32); + if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ds->tx_reg = drvdata->physbase + ACAUDODAT; + ds->rx_reg = 0; + } else { + ds->tx_reg = 0; + ds->rx_reg = drvdata->physbase + ACAUDIDAT; + } + + /* Try to grab a DMA channel */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dmadata->dma_chan = dma_request_channel(mask, filter, dmadata); + if (!dmadata->dma_chan) { + dev_err(dev->soc_dev.dev, + "DMA channel for %s is not available\n", + dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return -EBUSY; + } + tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet, + (unsigned long)dmadata); + return 0; +} + +static int txx9aclc_pcm_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + struct resource *r; + int i; + int ret; + + dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK; + dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE; + for (i = 0; i < 2; i++) { + r = platform_get_resource(dev->aclc_pdev, IORESOURCE_DMA, i); + if (!r) { + ret = -EBUSY; + goto exit; + } + dev->dmadata[i].dma_res = r; + ret = txx9aclc_dma_init(dev, &dev->dmadata[i]); + if (ret) + goto exit; + } + return 0; + +exit: + for (i = 0; i < 2; i++) { + if (dev->dmadata[i].dma_chan) + dma_release_channel(dev->dmadata[i].dma_chan); + dev->dmadata[i].dma_chan = NULL; + } + return ret; +} + +static int txx9aclc_pcm_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + int i; + + /* disable all FIFO DMAs */ + __raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, base + ACCTLDIS); + /* dummy R/W to clear pending DMAREQ if any */ + __raw_writel(__raw_readl(base + ACAUDIDAT), base + ACAUDODAT); + + for (i = 0; i < 2; i++) { + struct txx9aclc_dmadata *dmadata = &dev->dmadata[i]; + struct dma_chan *chan = dmadata->dma_chan; + if (chan) { + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + dma_release_channel(chan); + } + dev->dmadata[i].dma_chan = NULL; + } + return 0; +} + +struct snd_soc_platform txx9aclc_soc_platform = { + .name = "txx9aclc-audio", + .probe = txx9aclc_pcm_probe, + .remove = txx9aclc_pcm_remove, + .pcm_ops = &txx9aclc_pcm_ops, + .pcm_new = txx9aclc_pcm_new, + .pcm_free = txx9aclc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(txx9aclc_soc_platform); + +static int __init txx9aclc_soc_platform_init(void) +{ + return snd_soc_register_platform(&txx9aclc_soc_platform); +} + +static void __exit txx9aclc_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&txx9aclc_soc_platform); +} + +module_init(txx9aclc_soc_platform_init); +module_exit(txx9aclc_soc_platform_exit); + +MODULE_AUTHOR("Atsushi Nemoto <anemo(a)mba.ocn.ne.jp>"); +MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h new file mode 100644 index 0000000..6769aab --- /dev/null +++ b/sound/soc/txx9/txx9aclc.h @@ -0,0 +1,83 @@ +/* + * TXx9 SoC AC Link Controller + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __TXX9ACLC_H +#define __TXX9ACLC_H + +#include <linux/interrupt.h> +#include <asm/txx9/dmac.h> + +#define ACCTLEN 0x00 /* control enable */ +#define ACCTLDIS 0x04 /* control disable */ +#define ACCTL_ENLINK 0x00000001 /* enable/disable AC-link */ +#define ACCTL_AUDODMA 0x00000100 /* AUDODMA enable/disable */ +#define ACCTL_AUDIDMA 0x00001000 /* AUDIDMA enable/disable */ +#define ACCTL_AUDOEHLT 0x00010000 /* AUDO error halt + enable/disable */ +#define ACCTL_AUDIEHLT 0x00100000 /* AUDI error halt + enable/disable */ +#define ACREGACC 0x08 /* codec register access */ +#define ACREGACC_DAT_SHIFT 0 /* data field */ +#define ACREGACC_REG_SHIFT 16 /* address field */ +#define ACREGACC_CODECID_SHIFT 24 /* CODEC ID field */ +#define ACREGACC_READ 0x80000000 /* CODEC read */ +#define ACREGACC_WRITE 0x00000000 /* CODEC write */ +#define ACINTSTS 0x10 /* interrupt status */ +#define ACINTMSTS 0x14 /* interrupt masked status */ +#define ACINTEN 0x18 /* interrupt enable */ +#define ACINTDIS 0x1c /* interrupt disable */ +#define ACINT_CODECRDY(n) (0x00000001 << (n)) /* CODECn ready */ +#define ACINT_REGACCRDY 0x00000010 /* ACREGACC ready */ +#define ACINT_AUDOERR 0x00000100 /* AUDO underrun error */ +#define ACINT_AUDIERR 0x00001000 /* AUDI overrun error */ +#define ACDMASTS 0x80 /* DMA request status */ +#define ACDMA_AUDO 0x00000001 /* AUDODMA pending */ +#define ACDMA_AUDI 0x00000010 /* AUDIDMA pending */ +#define ACAUDODAT 0xa0 /* audio out data */ +#define ACAUDIDAT 0xb0 /* audio in data */ +#define ACREVID 0xfc /* revision ID */ + +struct txx9aclc_dmadata { + struct resource *dma_res; + struct txx9dmac_slave dma_slave; + struct dma_chan *dma_chan; + struct tasklet_struct tasklet; + spinlock_t dma_lock; + int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */ + struct snd_pcm_substream *substream; + unsigned long pos; + dma_addr_t dma_addr; + unsigned long buffer_bytes; + unsigned long period_bytes; + unsigned long frag_bytes; + int frags; + int frag_count; + int dmacount; +}; + +struct txx9aclc_plat_drvdata { + void __iomem *base; + u64 physbase; +}; + +struct txx9aclc_soc_device { + struct snd_soc_device soc_dev; + struct platform_device *aclc_pdev; /* for ioresources, drvdata */ + struct txx9aclc_dmadata dmadata[2]; +}; + +static inline struct txx9aclc_plat_drvdata *txx9aclc_get_plat_drvdata( + struct txx9aclc_soc_device *sdev) +{ + return platform_get_drvdata(sdev->aclc_pdev); +} + +extern struct snd_soc_platform txx9aclc_soc_platform; +extern struct snd_soc_dai txx9aclc_ac97_dai; + +#endif /* __TXX9ACLC_H */ -- 1.5.6.5

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Re: [alsa-devel] mainline tlv320aic26 driver
by Grant Likely 19 May '09

19 May '09

BTW, Please cc: the the alsa-devel(a)alsa-project.org mailing list when asking questions. I don't know everything and others can chime in that way. On Mon, May 18, 2009 at 12:44 PM, Joey Oravec <joravec(a)drewtech.com> wrote: > Grant - > > I'm trying to create an ASoC driver for my at91 platform with a tlv320aic26, > and I have a question about your driver. I'm looking at mainline 2.6.29. > > Inside aic26_probe() it's looking for struct aic26 to from > socdev->codec_data. I see how aic26_spi_probe() kzalloc's the structure, > sets some values in aic26->codec, but I have no idea when you expected > codec_data would get set. Could you briefly explain which function is > responsible to set that pointer and when it should happen? Since the aic26 is an SPI device, it is first probed (as you found) in the aic26_spi_probe() function. On my platform, the connection between the codec and the bus driver is described in an OF Flattened Device Tree and it uses the snd-of-simple driver to match the codec and platform drivers. So, the codec_data value is pass via the of_snd_soc_register_codec() function. I don't know the best way for your fabric driver to find the aic26 spi device. It probably should use bus_find_device() and dev_get_drvdata() to resolve to the value that codec_data should be set to. g. -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 19 May '09

19 May '09

Hi, after some communications with Creative, I finally got a mergeable version of their X-Fi driver. It's not my crappy hack but by genuine Creative :) The patch is now merged into sound-unstable GIT tree topic/ctxfi branch: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git The corresponding alsa-driver snapshot tarball is: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… The driver module is named as snd-ctxfi. I tested only the driver builds with recent kernels, but unfortunately I have no test hardware now, so it'd be helpful if anyone can test it with a real hardware. Note that the ctxfi driver might conflict with the HD-audio driver. So it'd be nice if you save the output of lspci before loading the drivers and let me know if any problem happens. Also, at testing, please check the kernel message. If this conflict really happens, try to build without hd-audio, that is, run configure with --with-cards=ctxfi option in addition to others, and test whether it works with this workaround (better after reboot to reset the board info by BIOS). Once after I get positive feedbacks (or fix after negative ones), I'm going to merge it to the main tree so that it'll be in the next kernel. So, a quick test report would be greatly appreciated. thanks, Takashi

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Daniel Bankmann 19 May '09

19 May '09

Hi! I just tested the newest driver version with Open Suse 11.1 and pae-kernel. I seems not to work probably, although the latest "official" driver by Creative worked more or less good. The problem is, that always, when hearing some music or any other sound, the signal gets interrupted every few seconds. At the other side, the startup jingle for example is accompanied by a knock every few seconds. Any idea, or what information do you need to fix this?

1 0

[alsa-devel] [PATCH] snd-hda-intel: add controls to toggle DC bias on mic ports
by Nickolas Lloyd 19 May '09

19 May '09

This patch adds a mixer control for the STAC92XX boards to control the DC bias of mic ports, allowing recording from both powered and non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack Mode" to switch between Mic, Line In, and Line Out. Signed-off by: Nickolas Lloyd <ultrageek.lloyd(a)gmail.com> Note: diffed against alsa-driver-20090513 ---- diff -urpN alsa-driver/sound/pci/hda/patch_sigmatel.c alsa-driver.a/sound/pci/hda/patch_sigmatel.c --- alsa-driver/sound/pci/hda/patch_sigmatel.c 2009-05-12 22:05:31.000000000 +0000 +++ alsa-driver.a/sound/pci/hda/patch_sigmatel.c 2009-05-13 12:24:17.000000000 +0000 @@ -634,6 +634,96 @@ static int stac92xx_smux_enum_put(struct return 0; } +static unsigned int stac92xx_vref_set(struct hda_codec *codec, + hda_nid_t nid, unsigned int new_vref) +{ + unsigned int error; + unsigned int pincfg; + pincfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + pincfg &= 0xff; + pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + pincfg |= new_vref; + + if (new_vref == AC_PINCTL_VREF_HIZ) + pincfg |= AC_PINCTL_OUT_EN; + else + pincfg |= AC_PINCTL_IN_EN; + + error = snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg); + if (error < 0) + return error; + else + return 1; +} + +static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int vref; + vref = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + vref &= AC_PINCTL_VREFEN; + return vref; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref; + unsigned int error; + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = AC_PINCTL_VREF_80; + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else + new_vref = AC_PINCTL_VREF_HIZ; + + if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { + error = stac92xx_vref_set(codec, + kcontrol->private_value, new_vref); + return error; + } + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); + if (vref == AC_PINCTL_VREF_80) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 3; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -995,6 +1085,17 @@ static struct hda_verb stac9205_core_ini .private_value = verb_read | (verb_write << 16), \ } +#define DC_BIAS(xname, idx, nid) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = idx, \ + .info = stac92xx_dc_bias_info, \ + .get = stac92xx_dc_bias_get, \ + .put = stac92xx_dc_bias_put, \ + .private_value = nid, \ + } + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -2702,7 +2803,8 @@ enum { STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, - STAC_CTL_WIDGET_CLFE_SWITCH + STAC_CTL_WIDGET_CLFE_SWITCH, + STAC_CTL_WIDGET_DC_BIAS }; static struct snd_kcontrol_new stac92xx_control_templates[] = { @@ -2714,6 +2816,7 @@ static struct snd_kcontrol_new stac92xx_ STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), + DC_BIAS(NULL, 0, 0), }; /* add dynamic controls */ @@ -3165,9 +3268,9 @@ static int stac92xx_auto_create_multi_ou } if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", - (spec->mic_switch << 8) | 1); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, + "Mic Jack Mode", + spec->mic_switch); if (err < 0) return err; }

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[alsa-devel] POLLOUT warnings dropped by Pulseaudio
by Ozan Çağlayan 19 May '09

19 May '09

Hi, Pulseaudio keeps telling May 19 12:22:59 mecbuk pulseaudio[1630]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! May 19 12:22:59 mecbuk pulseaudio[1630]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. May 19 12:22:59 mecbuk pulseaudio[1630]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. on quite every system (say 8 out of 10). I remember the long discussion thread about a few problems found and fixed in alsa-lib. Is there anything left to do, any way of helping by sending some debugging output or else, to fix those issues totally? Or should I silently ignore those messages? Thanks, Ozan Caglayan

1 0

Re: [alsa-devel] [RFC] ALSA vs. dedicated char device for a USB Audio Class gadget driver
by Laurent Pinchart 19 May '09

19 May '09

Hi Hal, On Friday 15 May 2009 22:15:18 Hal Murray wrote: > > If I use a synchronous endpoint, isn't the number of samples per frame > > determined by the nominal sampling rate and the nominal SOF frequency > > ? With the SOF clock running at 1kHz, I expect a synchronous endpoint > > for a 16 bits mono 48kHz stream to deliver exactly 48 frames (96 > > bytes) per USB frame. > > Beware of the "exact" in there. In real life, crystals have a tolerance. > The USB clock will not match the audio clock perfectly. What should be 1 > kHz might be 1.00003 kHz or it might be 0.99997 kHz. > > If the USB clock in this example is slightly fast, you will get occasional > times when there are only 47 samples ready. If it's slightly slow, you > will occasionally have 49 samples. > > That's slow/fast relative to the audio clock. If the USB clock is 0.001% > slow but the audio clock is 0.002% slow, then the USB clock will be fast > for this discussion. Except that the SOF clock *is* the reference clock. It will be slower or faster than the audio clock, and the userspace application will have to perform sample rate matching so that packets will be exactly 48 bytes long. > This is a common problem in communications. There are several techniques > to cope with it. > > The simplest is to make the transport mechanism have a bit of extra > bandwidth so you can always keep up. It's just a matter of how often you > use that extra bandwidth. In this case, you would allocate enough > bandwidth for 49 samples and only use 48 most of the time. An occasional > USB frame would have 47 or 49. That's what would happen with an asynchronous endpoint. The hard bit is to find out when to send that occasional packet, as the driver doesn't have access to the audio clock. > Another approach is to use big enough FIFO so you never get in trouble. > For this to work, you need a limit on the length of the data stream. For > example, suppose the longest song you want to send is 1000 seconds and both > clocks are accurate to 50 ppm (parts per million). The max difference in > clocks is 100 ppm. 48 k samples/second * 1000 seconds is 48 million > samples total. 100 ppm means that the worst difference would be 4800 > samples. So if the USB side waits until it gets 4800 samples before it > starts forwarding data, it won't run out if the audio clock is worst-case > slow and the USB clock is worst-case fast. You also need buffering for > another 4800 samples in case the clock differences are reversed. This adds > a delay to get started while you collect the 4800 samples. That won't work here, the stream can last forever (well, not quite, but still a long time). > You can just drop or duplicate a sample whenever the clocks slip. > You can derive both clocks from a common source. > > You can lock one clock to the other with a PLL (Phase Locked Loop). The SOF clock is driven by the USB host, and the audio clock is driven by the audio codec. I can't lock one to another or derive both of them from a common source. There could even be no audio clock at all if I stream audio data from a file. > There are probably other approaches. The applicable techniques require knowledge of both the audio clock and the SOF clock in a common place. My driver has no access to the audio clock. All it knows about is the SOF clock. There are only two options I can think of. The first one is to use an asynchronous endpoint and sent the occasional smaller or bigger packet (or duplicate/drop one sample). As the driver can't access the audio clock it needs to derive the information from the amount of data present in the ALSA ring buffer. To be honest I'm not sure if that will be possible at all, as the application will write data at a non-constant rate. The second one, which sounds easier, at least on the driver side, is to use a synchronous endpoint with a fixed packet size. The application will perform rate matching (duplicating/dropping a sample or resampling the audio stream) using the audio clock and the SOF clock. What I'm still unsure about is how the application can access the audio clock and the SOF clock through ALSA, but I suppose that's possible. Best regards, Laurent Pinchart

3 5

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: fix analog right microphone path
by Peter Ujfalusi 19 May '09

19 May '09

Hello, Here is the fix for AUXR recording that Anuj reported. Bit 2 is selecting the AUXR input in the ANAMICR register and not bit 1. --- Peter Ujfalusi (1): ASoC: TWL4030: Fix Analog capture path for AUXR sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH 2/3 - RESEND] ASoC: TWL4030: Add control for selecting codec operation mode
by Lopez Cruz, Misael 19 May '09

19 May '09

Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 47 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d683d..8eca0f4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -814,6 +814,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +/* Codec operation modes */ +static const char *twl4030_op_modes_texts[] = { + "Option 2 (voice/audio)", "Option 1 (audio)" +}; + +static const struct soc_enum twl4030_op_modes_enum = + SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, + ARRAY_SIZE(twl4030_op_modes_texts), + twl4030_op_modes_texts); + +int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl4030_priv *twl4030 = codec->private_data; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val; + unsigned short mask, bitmask; + + if (twl4030->configured) { + printk(KERN_ERR "twl4030 operation mode cannot be " + "changed on-the-fly\n"); + return -EBUSY; + } + + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) + ; + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + + val = ucontrol->value.enumerated.item[0] << e->shift_l; + mask = (bitmask - 1) << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= ucontrol->value.enumerated.item[1] << e->shift_r; + mask |= (bitmask - 1) << e->shift_r; + } + + return snd_soc_update_bits(codec, e->reg, mask, val); +} + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -895,6 +937,11 @@ static const struct soc_enum twl4030_vibradir_enum = twl4030_vibradir_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + /* Codec operation mode control */ + SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, + snd_soc_get_enum_double, + snd_soc_put_twl4030_opmode_enum_double), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, -- 1.5.4.3

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[alsa-devel] [PATCH] soc: Correct print specifiers for unsigneds
by Roel Kluin 19 May '09

19 May '09

Unsigned variables should use `%u' rather than `%d'. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- This also comtains some changes that I already sent here http://lkml.org/lkml/2009/5/11/497 Sorry for that, Is there some way to split these changes? sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/tlv320aic23.c | 4 ++-- sound/soc/codecs/uda134x.c | 4 ++-- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 8 ++++---- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8753.c | 6 +++--- sound/soc/codecs/wm8900.c | 6 +++--- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 ++-- sound/soc/sh/ssi.c | 2 +- 14 files changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 7065753..9eb610c 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( * Find actual rate, compare to requested rate */ actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", rate, actual_rate); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3f4afb..a06b6bb 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, */ if ((reg < 0 || reg > 9) && (reg != 15)) { - printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } @@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; - printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, value, reg); return -EIO; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index ddefb8f..269b108 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %d", + printk(KERN_ERR "%s unkown register: reg: %u", __func__, reg); return -EINVAL; } @@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct uda134x_priv *uda134x = codec->private_data; - pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__, clk_id, freq, dir); /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0275321..e7348d3 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, if (ret < 0) return ret; dev_dbg(wm8350->dev, - "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d", + "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d", freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div, fll_div.ratio); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa6..502eefa 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->outdiv *= 2; if (factors->outdiv > 32) { dev_err(wm8400->wm8400->dev, - "Unsupported FLL output frequency %dHz\n", + "Unsupported FLL output frequency %uHz\n", Fout); return -EINVAL; } @@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->k = K / 10; dev_dbg(wm8400->wm8400->dev, - "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n", + "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n", Fref, Fout, factors->n, factors->k, factors->fratio, factors->outdiv); @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6a4cea0..c8b8dba 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source) if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8510 N value %d outwith recommended range!d\n", + "WM8510 N value %u outwith recommended range!d\n", Ndiv); pll_div.n = Ndiv; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 9f6be3d..86c4b24 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -415,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, unsigned int K, Ndiv, Nmod; int i; - pr_debug("wm8580: PLL %dHz->%dHz\n", source, target); + pr_debug("wm8580: PLL %uHz->%uHz\n", source, target); /* Scale the output frequency up; the PLL should run in the * region of 90-100MHz. @@ -447,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 5) || (Ndiv > 13)) { printk(KERN_ERR - "WM8580 N=%d outside supported range\n", Ndiv); + "WM8580 N=%u outside supported range\n", Ndiv); return -EINVAL; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f..d28eeac 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "wm8753: unsupported N = %d\n", Ndiv); + "wm8753: unsupported N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 46c5ea1..3c78945 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, } if (target > 100000000) - printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d" - " Fout=%d\n", target, Fref, Fout); + printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u" + " Fout=%u\n", target, Fref, Fout); if (div > 32) { printk(KERN_ERR "wm8900: Invalid FLL division rate %u, " - "Fref=%d, Fout=%d, target=%d\n", + "Fref=%u, Fout=%u, target=%u\n", div, Fref, Fout, target); return -EINVAL; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 40cd274..d029818 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + "WM8990 N value outwith recommended range! N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 523bad0..c79e666 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -689,7 +689,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source) Ndiv = target / source; if ((Ndiv < 5) || (Ndiv > 12)) printk(KERN_WARNING - "WM9713 PLL N value %d out of recommended range!\n", + "WM9713 PLL N value %u out of recommended range!\n", Ndiv); pll_div->n = Ndiv; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 286be31..d656a90 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -323,7 +323,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); switch (clk_id) { @@ -472,7 +472,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, ssacd |= (0x6 << 4); dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %dHz\n", + "Using SSACDD %x to supply %uHz\n", val, freq_out); break; } diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index ab680aa..3cd94c5 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -507,7 +507,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, actual = clkrate / (fsdiv * div); deviation = actual - rate; - printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", fsdiv, div, actual, deviation); deviation = abs(deviation); @@ -523,7 +523,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, break; } - printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", best_fs, best_div, best_rate); info->fs_div = best_fs; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 56fa087..b378096 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -145,7 +145,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); - pr_debug("bits: %d channels: %d\n", bits, channels); + pr_debug("bits: %u channels: %u\n", bits, channels); ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | CR_SWL_MASK);

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