July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] hda-intel and runtime power-management
by Tatulea, Dragos 28 Jul '11

28 Jul '11

Hi, I'm in the process of enabling pm runtime pm for the hda_intel driver, not being sure if this is a sane task :). Noticed that CONFIG_SND_HDA_POWER_SAVE disables codecs and chip when they are not in use, but the power gain is not noticeable on my hw. On the other hand, a full suspend (azx_suspend) does save a few hundred mW's. I am aware that this is hw specific, but I would like to take advantage of the full suspend. Suspend/resume latency is below jiffy resolution. How about a more aggressive scheme?: allow runtime power management (with azx_suspend/resume as runtime pm ops) to kick in when all streams are stopped/suspended/paused. And resume upon stream is start/resume. I don't have enough context into ALSA & sound drivers so I need a reality check: is this crazy/stupid? Thanks, Dragos

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[alsa-devel] [PATCH 5/6] sound/core/pcm_compat.c: adjust array index
by Julia Lawall 28 Jul '11

28 Jul '11

From: Julia Lawall <julia(a)diku.dk> Convert array index from the loop bound to the loop index. A simplified version of the semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e1,e2,ar; @@ for(e1 = 0; e1 < e2; e1++) { <... ar[ - e2 + e1 ] ...> } // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- Not tested. sound/core/pcm_compat.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -u -p a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -342,7 +342,7 @@ static int snd_pcm_ioctl_xfern_compat(st kfree(bufs); return -EFAULT; } - bufs[ch] = compat_ptr(ptr); + bufs[i] = compat_ptr(ptr); bufptr++; } if (dir == SNDRV_PCM_STREAM_PLAYBACK)

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Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.0 gives choppy sound
by Takashi Iwai 28 Jul '11

28 Jul '11

At Tue, 26 Jul 2011 23:14:47 +0200, Colomban Wendling wrote: > > Hi, > > Following a bug report started in Debian BTS, in bug 635362 [1], I was > asked to post here after testing upstream kernels and reproducing the bug. > > So, here's the problem: my HDA audio chip [2] started to output very > choppy sound with Linux kernel 3.0, while it worked perfectly well in > previous versions -- latest two I used being 2.6.38 and 2.6.39 from > Debian. Please note that simply booting on 2.6.39 works as usual, so > it's IMHO definitely a driver regression. Do you have 2.6.39 kernel messages? I see the warnings in your dmesg below in 3.0* kernels. Did you get it with the working kernel as well? > [ 0.000000] ------------[ cut here ]------------ > [ 0.000000] WARNING: at drivers/pci/dmar.c:634 warn_invalid_dmar+0x77/0x85() > [ 0.000000] Hardware name: MS-7636 > [ 0.000000] Your BIOS is broken; DMAR reported at address fed93000 returns all ones! > [ 0.000000] BIOS vendor: American Megatrends Inc.; Ver: V1.1; Product Version: 1.0 > [ 0.000000] Modules linked in: > [ 0.000000] Pid: 0, comm: swapper Not tainted 3.0.0 #4 > [ 0.000000] Call Trace: > [ 0.000000] [<ffffffff81043359>] ? warn_slowpath_common+0x78/0x8c > [ 0.000000] [<ffffffff810433bb>] ? warn_slowpath_fmt_taint+0x3d/0x42 > [ 0.000000] [<ffffffff81318a7e>] ? __pte+0x7/0x8 > [ 0.000000] [<ffffffff816b2a2d>] ? __early_set_fixmap+0x89/0x8d > [ 0.000000] [<ffffffff81318a7e>] ? __pte+0x7/0x8 > [ 0.000000] [<ffffffff816b2a2d>] ? __early_set_fixmap+0x89/0x8d > [ 0.000000] [<ffffffff81318a7e>] ? __pte+0x7/0x8 > [ 0.000000] [<ffffffff811b72c9>] ? warn_invalid_dmar+0x77/0x85 > [ 0.000000] [<ffffffff816b2e3a>] ? early_iounmap+0xc5/0x114 > [ 0.000000] [<ffffffff816c6f13>] ? check_zero_address+0xb0/0xeb > [ 0.000000] [<ffffffff81327736>] ? bad_to_user+0x620/0x620 > [ 0.000000] [<ffffffff816c6f5f>] ? detect_intel_iommu+0x11/0xa6 > [ 0.000000] [<ffffffff816a5b29>] ? pci_iommu_alloc+0x3f/0x67 > [ 0.000000] [<ffffffff816b280c>] ? mem_init+0x14/0xe5 > [ 0.000000] [<ffffffff8169f94e>] ? start_kernel+0x1d0/0x3c8 > [ 0.000000] [<ffffffff8169f140>] ? early_idt_handlers+0x140/0x140 > [ 0.000000] [<ffffffff8169f3c4>] ? x86_64_start_kernel+0x104/0x111 > [ 0.000000] ---[ end trace a7919e7f17c0a725 ]--- Also, you can replace the files in sound/pci/hda/* from 2.6.39 kernel tree. If this fixes the problem, it's a sound driver bug. If not, it's anything else, e.g. the problem in more core side, e.g. IOMMU like above or the timer change, etc. Takashi

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[alsa-devel] [PATCH v5 0/4] ASoC: mxs: add audio support
by Dong Aisheng 28 Jul '11

28 Jul '11

According to Liam's comment, i redo the patch set based on the for-next branch on Mark's sound-2.6 tree. The based one is [PATCH v4 00/10] ARM: mxs: add audio support. For asoc part (1-4), only one change is updating mxs_pcm_new since this API has been updated to pass in the runtime data only For mxs part, no changes made. This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Tested on MX28EVK. For the ASoC-patches (1-4): Tested-by: Wolfram Sang <w.sang(a)pengutronix.de> For the mach-mxs-Patches (5-10): Reviewed-by: Wolfram Sang <w.sang(a)pengutronix.de> Changes since v4: * ASoC: mxs: add mxs-pcm driver update mxs_pcm_new due to this API's parameter number is changed. Changes since v3: This version only has one patch changed with another two for rebase the changed one. * [PATCH v4 06/10] ARM: mxs: add saif device Drop DIGCTRL-handling based on Wolfram's comment * [PATCH v4 07/10] ARM: mxs: add sgtl5000 i2c device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device * [PATCH v4 08/10] ARM: mxs: add mxs-sgtl5000 device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (4): ASoC: mxs: add mxs-pcm driver ASoC: mxs: add mxs-saif driver ASoC: mxs: add mxs-sgtl5000 machine driver ASoC: mxs: add asoc configuration files sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 ++ sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 +++ sound/soc/mxs/mxs-saif.c | 677 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 ++++++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++++++ 9 files changed, 1406 insertions(+), 0 deletions(-) create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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[alsa-devel] [RME - hdsp] alsa-lib's hdsp.h out of sync
by Adrian Knoth 28 Jul '11

28 Jul '11

Hi! While hacking on RPM support for hdspmixer, I've noticed that alsa-lib's hdsp.h doesn't reflect the kernel's hdsp.h: RPM is missing in HDSP_IO_Type. AFAIK, somebody has to regenerate alsa-lib's hdsp.h, right? Please do so. ;) Cheers

2 3

Re: [alsa-devel] [patch] ALSA: asihpi - off by one in asihpi_hpi_ioctl()
by Eliot Blennerhassett 28 Jul '11

28 Jul '11

On 28/07/11 00:55, Dan Rosenberg wrote: > On Wed, 2011-07-27 at 15:02 +0300, Dan Carpenter wrote: >> "adapter" is used as an array index in the adapters[] array so >> the off by one would make us read past the end. >> > > Agreed. I also don't like the fact that the "pa" pointer can be set to > an arbitrary address because the index isn't checked until after its > assignment. Even though the fix to the check prevents this pointer from > being dereferenced if it's out-of-bounds, it's still wrong. :( I have to agree. I'll submit another patch that avoids doing this. > I agree. This code seems to make assumptions in more than one place > that the adapters array is fully populated with non-NULL elements. At a > glance, I can't see where such initialization occurs though. It does, in void __init asihpi_init(void) ... memset(adapters, 0, sizeof(adapters)); -- Eliot Blennerhassett AudioScience Inc.

1 0

[alsa-devel] [PATCH V2 0/3] ASoC: au1x: update PSC AC97/I2S code
by Manuel Lauss 27 Jul '11

27 Jul '11

The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Patch #3 removes the use of custom PCM_TX/RX constants in favour of the established SNDRV_PCM_SUBTREAM_PLAYBACK/CAPTURE. Changes since V1: - added patch #3 - tidied patch #1 a bit more. Run-tested on DB1200, DB1300 and DB1550 boards. Manuel Lauss (3): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things ASoC: au1x: use substream stream info directly arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 ++++++++++++------ sound/soc/au1x/dbdma2.c | 91 +++++-------------------- sound/soc/au1x/psc-ac97.c | 48 +++++++++----- sound/soc/au1x/psc-i2s.c | 42 +++++++---- sound/soc/au1x/psc.h | 11 --- 6 files changed, 133 insertions(+), 139 deletions(-) -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH V5 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 27 Jul '11

27 Jul '11

Hello, Here's V5 of the AC97/I2S ASoC patchset for early Alchemy chips and their respective evaluation boards. The patches are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V4: - incorporate feedback from Liam Girdwood. - rediff against latest ASoC changes Changes since V3: - dropped the hunk which removed the I2S constants from the au1000.h header to avoid merge conflicts with other patches. - use the context structure declared in psc.h. Follow-up patches for psc* code depend on this. Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 ++++ sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 365 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 377 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 347 +++++++++++++++++++++++ sound/soc/au1x/psc.h | 19 +- 9 files changed, 1264 insertions(+), 10 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] [PATCH 1/1] oss: rename local change_bits to avoid powerpc bitsops.h definition
by Takashi Iwai 27 Jul '11

27 Jul '11

At Wed, 27 Jul 2011 17:48:41 +0100, Andy Whitcroft wrote: > > This collides with powerpc exported functions from bitops.h. Rename the > local copy in the oss soundblaster mixer and ad1848 driver. > > Signed-off-by: Andy Whitcroft <apw(a)canonical.com> Applied now. Thanks. Takashi > --- > sound/oss/ad1848.c | 6 +++--- > sound/oss/sb_mixer.c | 6 +++--- > 2 files changed, 6 insertions(+), 6 deletions(-) > > diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c > index 4d2a6ae..8415158 100644 > --- a/sound/oss/ad1848.c > +++ b/sound/oss/ad1848.c > @@ -458,7 +458,7 @@ static int ad1848_set_recmask(ad1848_info * devc, int mask) > return mask; > } > > -static void change_bits(ad1848_info * devc, unsigned char *regval, > +static void oss_change_bits(ad1848_info * devc, unsigned char *regval, > unsigned char *muteval, int dev, int chn, int newval) > { > unsigned char mask; > @@ -516,10 +516,10 @@ static void ad1848_mixer_set_channel(ad1848_info *devc, int dev, int value, int > > if (muteregoffs != regoffs) { > muteval = ad_read(devc, muteregoffs); > - change_bits(devc, &val, &muteval, dev, channel, value); > + oss_change_bits(devc, &val, &muteval, dev, channel, value); > } > else > - change_bits(devc, &val, &val, dev, channel, value); > + oss_change_bits(devc, &val, &val, dev, channel, value); > > spin_lock_irqsave(&devc->lock,flags); > ad_write(devc, regoffs, val); > diff --git a/sound/oss/sb_mixer.c b/sound/oss/sb_mixer.c > index 2039d31..0266088 100644 > --- a/sound/oss/sb_mixer.c > +++ b/sound/oss/sb_mixer.c > @@ -232,7 +232,7 @@ static int detect_mixer(sb_devc * devc) > return 1; > } > > -static void change_bits(sb_devc * devc, unsigned char *regval, int dev, int chn, int newval) > +static void oss_change_bits(sb_devc * devc, unsigned char *regval, int dev, int chn, int newval) > { > unsigned char mask; > int shift; > @@ -284,7 +284,7 @@ int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right) > return -EINVAL; > > val = sb_getmixer(devc, regoffs); > - change_bits(devc, &val, dev, LEFT_CHN, left); > + oss_change_bits(devc, &val, dev, LEFT_CHN, left); > > if ((*devc->iomap)[dev][RIGHT_CHN].regno != regoffs) /* > * Change register > @@ -304,7 +304,7 @@ int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right) > * Read the new one > */ > } > - change_bits(devc, &val, dev, RIGHT_CHN, right); > + oss_change_bits(devc, &val, dev, RIGHT_CHN, right); > > sb_setmixer(devc, regoffs, val); > > -- > 1.7.4.1 >

1 0

[alsa-devel] Searching an argument in mailing post
by Ru Vuott 27 Jul '11

27 Jul '11

Hello, how can I do searching an particular argument in Alsa-mailing list post ?

1 0

[alsa-devel] Problem trying to add ASoC ADAV801 support to my kernel
by Wayne Tams 27 Jul '11

27 Jul '11

Hi, I am trying to add support for the ADAV801 codec to my 2.6.35.3 kernel. I came across this patch http://article.gmane.org/gmane.linux.alsa.devel/86474 obviously I had to make some modifications so it would correctly patch my source. However when I come to build the kernel I'm getting a whole host of errors and I am little unsure on how to proceed, could anyone maybe provide some guidance on the possible causes of my errors or maybe there is a more update driver available. Many thanks W wa CC sound/soc/codecs/adav80x.o ^[sound/soc/codecs/adav80x.c: In function 'adav80x_set_sysclk': sound/soc/codecs/adav80x.c:556:27: error: 'struct snd_soc_codec' has no member named 'dapm' sound/soc/codecs/adav80x.c: In function 'adav80x_set_pll': sound/soc/codecs/adav80x.c:630:27: error: 'struct snd_soc_codec' has no member named 'dapm' sound/soc/codecs/adav80x.c: In function 'adav80x_set_bias_level': sound/soc/codecs/adav80x.c:654:7: error: 'struct snd_soc_codec' has no member named 'dapm' sound/soc/codecs/adav80x.c: At top level: sound/soc/codecs/adav80x.c:698:34: error: array type has incomplete element type sound/soc/codecs/adav80x.c:700:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:700:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:701:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:701:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:702:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:702:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:703:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:703:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:704:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:704:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:705:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:705:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:706:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:706:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:707:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:707:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:709:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:709:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:710:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:710:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:711:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:711:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:712:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:712:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:713:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:713:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:714:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:714:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:716:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:716:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:719:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:719:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:720:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:720:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:721:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:721:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:722:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:722:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:723:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:723:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:724:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:724:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:725:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:725:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:726:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:726:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:728:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:728:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:729:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:729:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:730:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:730:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:731:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:731:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:732:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:732:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:733:4: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:733:4: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c:735:3: error: field name not in record or union initializer sound/soc/codecs/adav80x.c:735:3: error: (near initialization for 'adav80x_dais') sound/soc/codecs/adav80x.c: In function 'adav80x_resume': sound/soc/codecs/adav80x.c:767:2: error: implicit declaration of function 'snd_soc_cache_sync' sound/soc/codecs/adav80x.c: At top level: sound/soc/codecs/adav80x.c:777:15: error: variable 'adav80x_codec_driver' has initializer but incomplete type sound/soc/codecs/adav80x.c:778:2: error: unknown field 'probe' specified in initializer sound/soc/codecs/adav80x.c:778:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:778:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:779:2: error: unknown field 'remove' specified in initializer sound/soc/codecs/adav80x.c:779:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:779:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:780:2: error: unknown field 'suspend' specified in initializer sound/soc/codecs/adav80x.c:780:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:780:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:781:2: error: unknown field 'resume' specified in initializer sound/soc/codecs/adav80x.c:781:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:781:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:782:2: error: unknown field 'set_bias_level' specified in initializer sound/soc/codecs/adav80x.c:782:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:782:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:784:2: error: unknown field 'set_pll' specified in initializer sound/soc/codecs/adav80x.c:784:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:784:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:785:2: error: unknown field 'set_sysclk' specified in initializer sound/soc/codecs/adav80x.c:785:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:785:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:787:2: error: unknown field 'reg_word_size' specified in initializer sound/soc/codecs/adav80x.c:787:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:787:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:788:2: error: unknown field 'reg_cache_size' specified in initializer sound/soc/codecs/adav80x.c:788:103: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:788:103: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:789:2: error: unknown field 'reg_cache_default' specified in initializer sound/soc/codecs/adav80x.c:789:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:789:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:791:2: error: unknown field 'controls' specified in initializer sound/soc/codecs/adav80x.c:791:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:791:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:792:2: error: unknown field 'num_controls' specified in initializer sound/soc/codecs/adav80x.c:792:93: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:792:93: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:793:2: error: unknown field 'dapm_widgets' specified in initializer sound/soc/codecs/adav80x.c:793:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:793:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:794:2: error: unknown field 'num_dapm_widgets' specified in initializer sound/soc/codecs/adav80x.c:794:105: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:794:105: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:795:2: error: unknown field 'dapm_routes' specified in initializer sound/soc/codecs/adav80x.c:795:2: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:795:2: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c:796:2: error: unknown field 'num_dapm_routes' specified in initializer sound/soc/codecs/adav80x.c:796:102: warning: excess elements in struct initializer sound/soc/codecs/adav80x.c:796:102: warning: (near initialization for 'adav80x_codec_driver') sound/soc/codecs/adav80x.c: In function 'adav80x_bus_probe': sound/soc/codecs/adav80x.c:813:84: warning: type defaults to 'int' in type name sound/soc/codecs/adav80x.c:813:84: warning: type defaults to 'int' in type name sound/soc/codecs/adav80x.c:813:84: error: negative width in bit-field '<anonymous>' sound/soc/codecs/adav80x.c:813:84: warning: passing argument 1 of 'snd_soc_register_codec' from incompatible pointer type include/sound/soc.h:239:5: note: expected 'struct snd_soc_codec *' but argument is of type 'struct device *' sound/soc/codecs/adav80x.c:813:84: error: too many arguments to function 'snd_soc_register_codec' include/sound/soc.h:239:5: note: declared here sound/soc/codecs/adav80x.c: In function 'adav80x_bus_remove': sound/soc/codecs/adav80x.c:822:2: warning: passing argument 1 of 'snd_soc_unregister_codec' from incompatible pointer type include/sound/soc.h:240:6: note: expected 'struct snd_soc_codec *' but argument is of type 'struct device *' sound/soc/codecs/adav80x.c: At top level: sound/soc/codecs/adav80x.c:698:34: warning: 'adav80x_dais' defined but not used make[4]: *** [sound/soc/codecs/adav80x.o] Error 1 make[3]: *** [sound/soc/codecs] Error 2 make[2]: *** [sound/soc] Error 2 make[1]: *** [sound] Error 2 make[1]: Leaving directory `/home/wayne/sources/k/kernel/linux-2.6.35.3-imx' make: *** [/home/wayne/sources/k/kernel/linux-2.6.35.3-imx/.stamp_image_built] Error 2

2 2

[alsa-devel] [patch] ALSA: asihpi - off by one in asihpi_hpi_ioctl()
by Dan Carpenter 27 Jul '11

27 Jul '11

"adapter" is used as an array index in the adapters[] array so the off by one would make us read past the end. Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- 1c073b67979 "ALSA: asihpi - Remove spurious adapter index check" reverted Dan Rosenburg's check that would have prevented the overflow here. Also it moved the initialization of "pa" down a couple lines so I'm concerned there may be a bogus derereference here when we check pa->type. I don't have the hardware, so I can't test this. diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 65fcf47..7ba7073 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -183,7 +183,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) int wrflag = -1; u32 adapter = hm->h.adapter_index; - if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) { + if ((adapter >= HPI_MAX_ADAPTERS) || (!pa->type)) { hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, HPI_ADAPTER_OPEN, HPI_ERROR_BAD_ADAPTER_NUMBER);

2 1

Re: [alsa-devel] [patch] ALSA: asihpi - off by one in asihpi_hpi_ioctl()
by Takashi Iwai 27 Jul '11

27 Jul '11

At Wed, 27 Jul 2011 09:02:13 -0400, Dan Rosenberg wrote: > > > > > Also it moved the initialization of "pa" down a couple lines so I'm > > > concerned there may be a bogus derereference here when we check > > > pa->type. I don't have the hardware, so I can't test this. > > > > > > > I agree. This code seems to make assumptions in more than one place > > that the adapters array is fully populated with non-NULL elements. At a > > glance, I can't see where such initialization occurs though. > > > > I hadn't read the updated code fully, so I missed what you meant. Yes, > this is definitely a NULL dereference, since "pa" is initialized to NULL > and not changed until after this dereference. NULL dereference was already fixed today in sound git tree by commit 767cd365b22820df07b962b49ce04b220b98e537. It'll be included in the pull request in the next days, maybe tomorrow. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH 7/9] ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning
by Mark Brown 27 Jul '11

27 Jul '11

On Thu, May 12, 2011 at 04:50:10PM -0700, Stephen Boyd wrote: > Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following > warning: Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
by Mark Brown 27 Jul '11

27 Jul '11

This delay is very conservative. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 38abdff..2eb4594 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1725,6 +1725,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); + + msleep(5); } wm8994->fll[id].in = freq_in; -- 1.7.5.4

3 5

[alsa-devel] [PATCH] ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
by Liam Girdwood 27 Jul '11

27 Jul '11

Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP devices have mixer components that can alter power usage and hence require full DAPM support. This provides a generic widget IO operation wrapper in preparation for future patches that implement platform driver DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 55 ++++++++++++++++++++++++++++++++++++++++++++----- 1 files changed, 49 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 34106bc..7b7af9f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,49 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) +{ + if (w->codec) + return snd_soc_read(w->codec, reg); + return 0; +} + +static int soc_widget_write(struct snd_soc_dapm_widget *w,int reg, int val) +{ + if (w->codec) + return snd_soc_write(w->codec, reg, val); + return 0; +} + +int soc_widget_update_bits(struct snd_soc_dapm_widget *w, unsigned short reg, + unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + + old = soc_widget_read(w, reg); + new = (old & ~mask) | value; + change = old != new; + if (change) + soc_widget_write(w, reg, new); + + return change; +} + +int soc_widget_test_bits(struct snd_soc_dapm_widget *w, unsigned short reg, + unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + + old = soc_widget_read(w, reg); + new = (old & ~mask) | value; + change = old != new; + + return change; +} + + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -181,7 +224,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = snd_soc_read(w->codec, reg); + val = soc_widget_read(w, reg); val = (val >> shift) & mask; if ((invert && !val) || (!invert && val)) @@ -197,7 +240,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; @@ -227,7 +270,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); val = (val >> e->shift_l) & e->mask; for (item = 0; item < e->max; item++) { if (val == e->values[item]) @@ -669,7 +712,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - snd_soc_update_bits(w->codec, -(w->reg + 1), + soc_widget_update_bits(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -877,7 +920,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - snd_soc_update_bits(dapm->codec, reg, mask, value); + soc_widget_update_bits(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1878,7 +1921,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = snd_soc_read(w->codec, w->reg); + val = soc_widget_read(w, w->reg); val &= 1 << w->shift; if (w->invert) val = !val; -- 1.7.4.1

3 4

[alsa-devel] Event or signal for microphone connect/disconnect
by Putinei Ionut 27 Jul '11

27 Jul '11

Hi, I have 2 microphones on my laptop: 1 pluggable on the mic port and 1 on the surface . Is there any way to receive events/signal that the microphone on the mic port is connected/disconnected. I want to implement this feature in a recording program. Thank you Ionut Putinei

2 1

[alsa-devel] Rename Tegra support to Tegra20?
by Stephen Warren 27 Jul '11

27 Jul '11

Mark, Liam, The existing ASoC Tegra support is specifically for Tegra20. It turns out that Tegra30 has pretty different hardware and requires different drivers. A couple questions: * The new drivers I've written use "tegra30" in Kconfig variable names, filenames, symbol names, etc. Do you think it's a good idea to rename the existing Tegra20 driver to basically s/tegra/tegra20/ everywhere so everything is versioned, or do you think that's just unnecessary churn? (My motivation is kinda like the device tree compatible properties always being versioned) * I assume there's little point my posting Tegra30 ASoC drivers for inclusion in mainline, since there's no Tegra30 CPU support at all there yet? Thanks. -- nvpublic

2 1

[alsa-devel] [PATCH] ASoC: Add device tree binding for WM8731
by Mark Brown 27 Jul '11

27 Jul '11

Tested with the famous "hey, look! this compiles" test plan. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/devicetree/bindings/sound/wm8731.txt | 18 ++++++++++++++++++ sound/soc/codecs/wm8731.c | 12 ++++++++++++ 2 files changed, 30 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt new file mode 100644 index 0000000..15f7004 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8731.txt @@ -0,0 +1,18 @@ +WM8731 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8731" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Example: + +codec: wm8731@1a { + compatible = "wlf,wm8731"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 76b4361..5ecc05a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -22,6 +22,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -607,6 +608,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), }; +static const struct of_device_id wm8731_of_match[] = { + { .compatible = "wlf,wm8731", }, + { } +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8731_spi_probe(struct spi_device *spi) { @@ -634,10 +640,13 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } +MODULE_DEVICE_TABLE(spi, wm8731_of_match); + static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_spi_probe, .remove = __devexit_p(wm8731_spi_remove), @@ -678,10 +687,13 @@ static const struct i2c_device_id wm8731_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); +MODULE_DEVICE_TABLE(i2c, wm8731_of_match); + static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "wm8731", .owner = THIS_MODULE, + .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), -- 1.7.5.4

2 1

[alsa-devel] Supporting an unusual device
by Andrew Murray 27 Jul '11

27 Jul '11

Hello, I'm writing support for a slightly unusual device (which I intend to share later) and would like feedback on the best approach to take. The device is for a SOC which has two identical I2C connected codecs. Most codec drivers contain a snd_soc_dai structure - but as this contains instance specific data it would be necessary instead to produce (kmalloc) such structures for each instance (to ensure that both codecs can be treated and addressed independently). I.e. it seems that some codec drivers have been written with the assumption that there will only be one present on a system at any time. My view would be to implement such functionality (kmalloc'ing snd_soc_dai's) in one of the various platform specific directories as appropriate - e.g. atmel, blackfin, davinci, etc. Is this the best way forward? Are there any examples I've missed which do support multiple instances of the same codec? Also - is it recommended to expose these two codecs to the outside world as two snd_soc_card devices - or a single card with multiple dai_links? Thanks, Andrew Murray

2 2

[alsa-devel] [PATCH] ALSA: asihpi - bug fix pa use before init.
by linux＠audioscience.com 27 Jul '11

27 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Fixes bug introduced by 1c073b67. Also declare pa local to block in which it is used. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpioctl.c | 11 ++++------- 1 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 65fcf47..e0cff0c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -107,7 +107,6 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) union hpi_response_buffer_v1 *hr; u16 res_max_size; u32 uncopied_bytes; - struct hpi_adapter *pa = NULL; int err = 0; if (cmd != HPI_IOCTL_LINUX) @@ -182,6 +181,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; u32 adapter = hm->h.adapter_index; + struct hpi_adapter *pa = &adapters[adapter]; if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) { hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, @@ -197,9 +197,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } - pa = &adapters[adapter]; - - if (mutex_lock_interruptible(&adapters[adapter].mutex)) { + if (mutex_lock_interruptible(&pa->mutex)) { err = -EINTR; goto out; } @@ -235,8 +233,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) "stream buffer size %d\n", size); - mutex_unlock(&adapters - [adapter].mutex); + mutex_unlock(&pa->mutex); err = -EINVAL; goto out; } @@ -277,7 +274,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) uncopied_bytes, size); } - mutex_unlock(&adapters[adapter].mutex); + mutex_unlock(&pa->mutex); } /* on return response size must be set */ -- 1.7.0.4

2 1

Re: [alsa-devel] Bug#635362: Realtek ALC889: HDA Intel and kernel 3.0 gives choppy sound
by Julian Sikorski 27 Jul '11

27 Jul '11

W dniu 26.07.2011 23:14, Colomban Wendling pisze: > Hi, > > Following a bug report started in Debian BTS, in bug 635362 [1], I was > asked to post here after testing upstream kernels and reproducing the bug. > > So, here's the problem: my HDA audio chip [2] started to output very > choppy sound with Linux kernel 3.0, while it worked perfectly well in > previous versions -- latest two I used being 2.6.38 and 2.6.39 from > Debian. Please note that simply booting on 2.6.39 works as usual, so > it's IMHO definitely a driver regression. > > I tried Debian's 3.0 as well as mainline 3.0 and git5 snapshots, and all > 3 were affected by the problem -- though the kernel log isn't exactly > the same. > > I attach three dmesg outputs, always took after trying to play a sound > (and hearing it choppy, it happens every time): > > dmesg-3.0.0-mainline: dmesg when running 3.0 mainline > dmesg-3.0.0-git5: when running 3.0-git5 snapshot > dmesg-3.0.0-git5-2: same as dmesg-3.0.0-git5 but configured without > CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS (doesn't change anything) > > If you want to see dmesg from the 2.6.39 kernel (where sound works well) > or Debian's 3.0, see respectively [3] and [4]. > > > Thanks by advance, and if you need any further details or testing, > please let me know. > > Regards, > Colomban > > > [1] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=635362 > [2] Realtek ALC889 on an MSI H55M-E33, see lspci in Debian bug [1] for > details > [3] > http://bugs.debian.org/cgi-bin/bugreport.cgi?msg=15;filename=dmesg-2.6.39-2… > [4] > http://bugs.debian.org/cgi-bin/bugreport.cgi?msg=15;filename=dmesg-3.0.0-1-… I am probably wrong, but did you try setting CONFIG_SND_HDA_BUFFER_SIZE to a larger value? Julian

2 2

[alsa-devel] [PATCH v3] ASoC: dapm - Add API call to query the widgets of valid DAPM paths.
by Liam Girdwood 26 Jul '11

26 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets and return the path widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This modifies the existing is_connected_input_ep() and is_connected_output_ep() by adding a new parameter to store any valid widgets in a list. We also add some extra debug to provide further insight into the DAPM path and widget discovery. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++ sound/soc/soc-dapm.c | 184 ++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 176 insertions(+), 14 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5833f01..a56ead2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,11 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 80f97b2..120df5b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -605,11 +605,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -638,15 +678,35 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { - if (path->weak) - continue; if (path->walked) continue; + dev_dbg(widget->dapm->dev," %c : %s -> %s -> %s : %c\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name, + path->weak ? 'w': ' '); + + if (path->weak) + continue; + if (path->sink && path->connect) { + path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -657,7 +717,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -691,21 +752,116 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { - if (path->weak) - continue; if (path->walked) continue; + if (path->source) + dev_vdbg(widget->dapm->dev," %c : %s <- %s <- %s : %c\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name, + path->weak ? 'w': ' '); + + if (path->weak) + continue; + if (path->source && path->connect) { + path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } return con; } +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + paths = is_connected_output_ep(root, list); + dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name); + + return paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name); + paths = is_connected_input_ep(root, list); + dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name); + + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream) +{ + struct snd_soc_dapm_widget *w; + enum snd_soc_dapm_type type; + int paths; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + /* get stream root widget AIF from stream string and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (w->id != type) + continue; + + if (!strcmp(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget for %s not found\n", stream_name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_walk(dapm); + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -732,9 +888,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -745,7 +901,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) int in; if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -759,7 +915,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) int out; if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1315,9 +1471,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", -- 1.7.4.1

2 5

[alsa-devel] CX20590 - problems
by wouerner 26 Jul '11

26 Jul '11

hi, i'm from brazil, sorry bad english I have sony vaio vpceg15fb, internal mic - not work headphone jack - not work front speaker - work fine mic jack - work fine shell _> head -n 1 /proc/asound/card0/codec* ==> /proc/asound/card0/codec#0 <== Codec: Conexant CX20590 ==> /proc/asound/card0/codec#3 <== Codec: Intel CougarPoint HDMI not supported yet? i tried all tutorials in internet. thanks.

1 0

[alsa-devel] [PATCH] ASoC: core - make IO calls return a valid errno on failure.
by Liam Girdwood 26 Jul '11

26 Jul '11

IO call failures should return an errno instead of -1 Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-io.c | 6 +++--- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 159f144..fcf69f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1661,7 +1661,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, if (!platform->driver->read) { dev_err(platform->dev, "platform has no read back\n"); - return -1; + return -EINVAL; } ret = platform->driver->read(platform, reg); @@ -1677,7 +1677,7 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, { if (!platform->driver->write) { dev_err(platform->dev, "platform has no write back\n"); - return -1; + return -EINVAL; } dev_dbg(platform->dev, "write %x = %x\n", reg, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3956670..45141fa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -132,7 +132,7 @@ static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) return snd_soc_platform_read(w->platform, reg); dev_err(w->dapm->dev, "no valid widget read method\n"); - return -1; + return -EINVAL; } static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) @@ -143,7 +143,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return snd_soc_platform_write(w->platform, reg, val); dev_err(w->dapm->dev, "no valid widget write method\n"); - return -1; + return -EINVAL; } static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c..166186c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -41,7 +41,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) - return -1; + return ret; } if (codec->cache_only) { @@ -67,7 +67,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) snd_soc_codec_volatile_register(codec, reg) || codec->cache_bypass) { if (codec->cache_only) - return -1; + return -EINVAL; BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); @@ -75,7 +75,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) ret = snd_soc_cache_read(codec, reg, &val); if (ret < 0) - return -1; + return ret; return val; } -- 1.7.4.1

2 3

[alsa-devel] [PATCH] Cirrus Logic CS421x support
by Tim Howe 26 Jul '11

26 Jul '11

This update includes the changes necessary for supporting the CS421x family of codecs. Previously this file only supported the CS420x family of codecs. This file also contains init verbs to correct several issues in the CS421x hardware. Behavior between the CS421x and CS420x codec families is similar, so several functions have been reused with "if" statements to determine which codec family (CS421x or CS420x) is present. Also, this file will be updated sometime in the near future in order to add support for a system using CS421x that requires mono mix on the speaker output only. Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 786 +++++++++++++++++++++++++++++++++++++++--- 1 files changed, 745 insertions(+), 41 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7f93739..27dbd5c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include <sound/tlv.h> /* */ @@ -61,9 +62,15 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; -/* available models */ +/* available models with CS420x */ enum { CS420X_MBP53, CS420X_MBP55, @@ -72,6 +79,12 @@ enum { CS420X_MODELS }; +/* CS421x boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + /* Vendor-specific processing widget */ #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 @@ -111,21 +124,42 @@ enum { /* 0x0009 - 0x0014 -> 12 test regs */ /* 0x0015 - visibility reg */ +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -343,18 +377,44 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, { int i; hda_nid_t nid; + struct cs_spec *spec = codec->spec; nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int type; - int idx; - type = get_wcaps_type(get_wcaps(codec, nid)); - if (type != AC_WID_AUD_IN) - continue; - idx = snd_hda_get_conn_index(codec, nid, pin, 0); - if (idx >= 0) { - *idxp = idx; - return nid; + int j, nums; + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } + } else { + hda_nid_t pins[2]; + type = get_wcaps_type(get_wcaps(codec, nid)); + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } } } return 0; @@ -835,6 +895,8 @@ static int build_digital_input(struct hda_codec *codec) /* * auto-mute and auto-mic switching + * CS421x auto-output redirecting + * HP/SPK/SPDIF */ static void cs_automute(struct hda_codec *codec) @@ -842,9 +904,25 @@ static void cs_automute(struct hda_codec *codec) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int hp_present; + unsigned int spdif_present; hda_nid_t nid; int i; + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + if (is_jack_detectable(codec, nid)) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; @@ -854,11 +932,19 @@ static void cs_automute(struct hda_codec *codec) if (hp_present) break; } + + /* mute speakers if spdif or hp jack is plugged in */ for (i = 0; i < cfg->speaker_outs; i++) { nid = cfg->speaker_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); + /* detect on spdif is specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? 0 : PIN_OUT); + } } if (spec->board_config == CS420X_MBP53 || spec->board_config == CS420X_MBP55 || @@ -867,21 +953,62 @@ static void cs_automute(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); } + + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present && spec->sense_b) ? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT : 0); + + } + /* Update board GPIOs if neccessary ... */ + } } +/* + * Auto-input redirect for CS421x + * Switch max 3 inputs of a single ADC (nid 3) +*/ + static void cs_automic(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; unsigned int present; - + nid = cfg->inputs[spec->automic_idx].pin; present = snd_hda_jack_detect(codec, nid); - if (present) - change_cur_input(codec, spec->automic_idx, 0); - else - change_cur_input(codec, !spec->automic_idx, 0); + + /* specific to CS421x, single ADC */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } else { + if (present) + change_cur_input(codec, spec->automic_idx, 0); + else + change_cur_input(codec, !spec->automic_idx, 0); + } } /* @@ -911,23 +1038,28 @@ static void init_output(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); spec->hp_detect = 1; } } + + /* Speaker */ for (i = 0; i < cfg->speaker_outs; i++) snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (spec->hp_detect) + + /* SPDIF is enabled on presence detect for CS421x */ + if (spec->hp_detect || spec->spdif_detect) cs_automute(codec); } @@ -961,19 +1093,31 @@ static void init_input(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | MIC_EVENT); } - change_cur_input(codec, spec->cur_input, 1); - if (spec->mic_detect) - cs_automic(codec); - - coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ - if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ - if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is selected in - * IDX_SPDIF_CTL. - */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (spec->mic_detect) + cs_automic(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + } else { + change_cur_input(codec, spec->cur_input, 1); + if (spec->mic_detect) + cs_automic(codec); + + coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + if (is_active_pin(codec, CS_DMIC2_PIN_NID)) + coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + if (is_active_pin(codec, CS_DMIC1_PIN_NID)) + coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + * No effect if SPDIF_OUT2 is + * selected in IDX_SPDIF_CTL. + */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + } } static const struct hda_verb cs_coef_init_verbs[] = { @@ -1173,7 +1317,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1187,7 +1331,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1201,7 +1345,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1215,22 +1359,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1241,11 +1385,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1272,6 +1418,562 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: + * 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: + * The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_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 = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + unsigned int original_coef = coef; + + coef &= ~0x0003; + coef |= (vol & 0x0003); + if (original_coef == coef) + return 0; + else { + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + return 1; + } +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) { + + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_output(codec); + init_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3)) { + + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char *name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs_automute(codec); + break; + + case MIC_EVENT: + cs_automic(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", + cs421x_models[spec->board_config]); +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries @@ -1279,11 +1981,13 @@ static int patch_cs420x(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

2 2

[alsa-devel] snd_mask_min
by Tim Blechmann 26 Jul '11

26 Jul '11

hi all, i've got a problem using params_format(): it uses an inline function snd_mask_min, which is declared in sound/pcm_params.h, but this file is not included by sound/pcm.h, where it is defined. including it manually, causes a huge number of errors. it would be very helpful, if pcm.h would be self-contained. cheers, tim

2 3

[alsa-devel] [PATCH] alsa: lx6464es - initialization fixes
by Tim Blechmann 26 Jul '11

26 Jul '11

this using this fix, the driver is usable with non-rt kernels. the card initialization is moved from the prepare to the hw_params callback. also the error handling is improved. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 72 +++++++++++++++++++++-------------------- sound/pci/lx6464es/lx6464es.h | 1 + sound/pci/lx6464es/lx_core.c | 43 +++++++++++++++---------- sound/pci/lx6464es/lx_core.h | 3 +- 4 files changed, 66 insertions(+), 53 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 1bd7a54..722212e 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -103,16 +103,14 @@ static int lx_set_granularity(struct lx6464es *chip, u32 gran); static int lx_hardware_open(struct lx6464es *chip, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + int channels, snd_pcm_uframes_t period_size) { int err = 0; - struct snd_pcm_runtime *runtime = substream->runtime; - int channels = runtime->channels; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_pcm_uframes_t period_size = runtime->period_size; - - snd_printd(LXP "allocating pipe for %d channels\n", channels); + snd_printk(LXP "allocating pipe for %d channels, %lu period size\n", + channels, period_size); err = lx_pipe_allocate(chip, 0, is_capture, channels); if (err < 0) { snd_printk(KERN_ERR LXP "allocating pipe failed\n"); @@ -130,14 +128,14 @@ static int lx_hardware_open(struct lx6464es *chip, } static int lx_hardware_start(struct lx6464es *chip, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { int err = 0; - struct snd_pcm_runtime *runtime = substream->runtime; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); snd_printd(LXP "setting stream format\n"); - err = lx_stream_set_format(chip, runtime, 0, is_capture); + err = lx_stream_set_format(chip, hw_params, 0, is_capture); if (err < 0) { snd_printk(KERN_ERR LXP "setting stream format failed\n"); return err; @@ -295,13 +293,33 @@ static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream return pos; } + static int lx_pcm_prepare(struct snd_pcm_substream *substream) { struct lx6464es *chip = snd_pcm_substream_chip(substream); int err = 0; - const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_printdd("->lx_pcm_prepare\n"); + snd_printd("->lx_pcm_prepare %d\n", is_capture); + + mutex_lock(&chip->setup_mutex); + + if (chip->board_sample_rate != substream->runtime->rate) { + if (!err) + chip->board_sample_rate = substream->runtime->rate; + } + + mutex_unlock(&chip->setup_mutex); + snd_printd("<-lx_pcm_prepare\n"); + return err; +} + +static int lx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, int is_capture) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + int err = 0; + + snd_printd("->lx_pcm_hw_params\n"); mutex_lock(&chip->setup_mutex); @@ -322,14 +340,16 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) } snd_printd(LXP "opening hardware\n"); - err = lx_hardware_open(chip, substream); + err = lx_hardware_open(chip, substream, + params_channels(hw_params), + params_period_size(hw_params)); if (err < 0) { snd_printk(KERN_ERR LXP "failed to open hardware. " "Error code %d\n", err); goto exit; } - err = lx_hardware_start(chip, substream); + err = lx_hardware_start(chip, substream, hw_params); if (err < 0) { snd_printk(KERN_ERR LXP "failed to start hardware. " "Error code %d\n", err); @@ -338,26 +358,6 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) chip->hardware_running[is_capture] = 1; - if (chip->board_sample_rate != substream->runtime->rate) { - if (!err) - chip->board_sample_rate = substream->runtime->rate; - } - -exit: - mutex_unlock(&chip->setup_mutex); - return err; -} - -static int lx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params, int is_capture) -{ - struct lx6464es *chip = snd_pcm_substream_chip(substream); - int err = 0; - - snd_printdd("->lx_pcm_hw_params\n"); - - mutex_lock(&chip->setup_mutex); - /* set dma buffer */ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -367,6 +367,7 @@ static int lx_pcm_hw_params(struct snd_pcm_substream *substream, else chip->playback_stream.stream = substream; +exit: mutex_unlock(&chip->setup_mutex); return err; } @@ -389,7 +390,7 @@ static int lx_pcm_hw_free(struct snd_pcm_substream *substream) int err = 0; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_printdd("->lx_pcm_hw_free\n"); + snd_printd("->lx_pcm_hw_free\n"); mutex_lock(&chip->setup_mutex); if (chip->hardware_running[is_capture]) { @@ -477,7 +478,7 @@ static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream) int err; snd_printd(LXP "stopping: stopping stream\n"); - err = lx_stream_stop(chip, 0, is_capture); + err = lx_stream_pause(chip, 0, is_capture); if (err < 0) snd_printk(KERN_ERR LXP "couldn't stop stream\n"); else @@ -1005,6 +1006,7 @@ static int __devinit snd_lx6464es_create(struct snd_card *card, chip->card = card; chip->pci = pci; chip->irq = -1; + chip->hardware_running[0] = chip->hardware_running[1] = 0; /* initialize synchronization structs */ spin_lock_init(&chip->lock); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index aea621e..2a97a21 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -30,6 +30,7 @@ #include <sound/core.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include "lx_core.h" diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 617f98b..3a9e1dd 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -353,8 +353,11 @@ polling_successful: lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, rmh->stat_len); } - } else + lx_message_dump(rmh); + } else { snd_printk(LXP "rmh error: %08x\n", reg); + dump_stack(); + } /* clear Reg_CSM_MR */ lx_dsp_reg_write(chip, eReg_CSM, 0); @@ -362,15 +365,11 @@ polling_successful: switch (reg) { case ED_DSP_TIMED_OUT: snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); - return -ETIMEDOUT; case ED_DSP_CRASHED: snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); - return -EAGAIN; } - lx_message_dump(rmh); - return reg; } @@ -532,10 +531,22 @@ int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, err = lx_message_send_atomic(chip, &chip->rmh); spin_unlock_irqrestore(&chip->msg_lock, flags); - if (err != 0) - snd_printk(KERN_ERR "lx6464es: could not allocate pipe\n"); + if (err == 0) + return 0; - return err; + switch (err) { + case EB_ALLOCATE_AUDIO_IMPOSSIBLE: + snd_printk(KERN_ERR "lx6464es: could not allocate pipe " + "(EB_ALLOCATE_AUDIO_IMPOSSIBLE)\n"); + break; + + case EB_INVALID_AUDIO: + snd_printk(KERN_ERR "lx6464es: could not allocate pipe " + "(EB_INVALID_AUDIO)\n"); + break; + } + + return -1; } int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture) @@ -781,7 +792,8 @@ int lx_stream_set_state(struct lx6464es *chip, u32 pipe, return err; } -int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, +int lx_stream_set_format(struct lx6464es *chip, + struct snd_pcm_hw_params *params, u32 pipe, int is_capture) { int err; @@ -789,22 +801,19 @@ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - u32 channels = runtime->channels; - - if (runtime->channels != channels) - snd_printk(KERN_ERR LXP "channel count mismatch: %d vs %d", - runtime->channels, channels); + const u32 channels = params_channels(params); + const snd_pcm_format_t format = params_format(params); spin_lock_irqsave(&chip->msg_lock, flags); lx_message_init(&chip->rmh, CMD_0C_DEF_STREAM); chip->rmh.cmd[0] |= pipe_cmd; - if (runtime->sample_bits == 16) + if (snd_pcm_format_width(format) == 16) /* 16 bit format */ chip->rmh.cmd[0] |= (STREAM_FMT_16b << STREAM_FMT_OFFSET); - if (snd_pcm_format_little_endian(runtime->format)) + if (snd_pcm_format_little_endian(format)) /* little endian/intel format */ chip->rmh.cmd[0] |= (STREAM_FMT_intel << STREAM_FMT_OFFSET); @@ -974,7 +983,7 @@ int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute) chip->rmh.cmd[1] = (u32)(mute_mask >> (u64)32); /* hi part */ chip->rmh.cmd[2] = (u32)(mute_mask & (u64)0xFFFFFFFF); /* lo part */ - snd_printk("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], + snd_printd("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], chip->rmh.cmd[2]); err = lx_message_send_atomic(chip, &chip->rmh); diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 6bd9cbb..378721a 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -134,7 +134,8 @@ int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture); int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture); /* low-level stream handling */ -int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, +int lx_stream_set_format(struct lx6464es *chip, + struct snd_pcm_hw_params *runtime, u32 pipe, int is_capture); int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, int *rstate); -- 1.7.4.1

2 2

[alsa-devel] HDAudio period/buffer sizes
by pl bossart 26 Jul '11

26 Jul '11

When I look at the HDAudio spec, I see that the BDL entries need to be aligned on a 128-byte boundary, but the buffer length can be any value. When I look at the HDAudio driver (azx_pcm_open in hda_codec.c), I see that we enforce a 128-byte granularity on the period/buffer sizes, but this was added in a patch that fixed alignment issues (5f1545bc) Any technical reason why the driver is more demanding than the spec? Or is this a confusion between alignment and buffer size? I can understand that rounding to multiple of 128 bytes would be more efficient in terms of data access, but it also prevents applications from getting an interrupt when they want (eg it's impossible to use 20ms periods with a 44.1kHz sampling rate, you'll get 19.59 or 20.31ms). Thanks, -Pierre

2 1

[alsa-devel] [PATCH] ASoC: dapm - change stream event dbg to vdgb
by Liam Girdwood 26 Jul '11

26 Jul '11

Stream event debug can be noisy on larger audio devices so improve the debug SNR by changing it to the verbose level. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 120df5b..3956670 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2710,7 +2710,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, { if (!w->sname || w->dapm != dapm) continue; - dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", + dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2] ASoC: dapm - Add DAPM stream completion event.
by Liam Girdwood 26 Jul '11

26 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..5833f01 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -518,6 +518,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..64a9dd5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..159f144 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,6 +3140,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3252,6 +3253,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda0..80f97b2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2574,6 +2574,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm, event); } /** -- 1.7.4.1

2 1

[alsa-devel] M-Audio FTU issues
by Daniel Mack 25 Jul '11

25 Jul '11

So, I've looked into this today and as expected, the device normally operates in implicit feedback mode. After installing a proprietary driver on Mac OS X (that regularily crashed my kernel), the analyzer shows that for every inbound packet, an out packet is sent back with the same amount of data in it. And there are no feedback endpoints. On Tue, Jun 14, 2011 at 11:21 AM, Aurélien Leblond <blablack(a)gmail.com> wrote: > There are still the issue of samples dropping. > > Tto reproduce it: > - Open Audacity and set it up to be using Jack. > - Click Generate -> Tone -> Ok. > > "Normally" you should hear very regular pops... > What surprises me though, is that I can't hear the clicks and pops you guys describe. Instead, I hear a quite annoying distortion for both 44100 and 48000 Hz which I can't explain either. My spontanous guess is that there is a resampler in the chain which compensates the sample drift and notably reduces the audio quality. I patched the driver so that it supports implicit feedback, but that didn't change anything in that regard. Hence my question is: How do you guys test? I renderened a sine tone in different samplerates and bit depths to .wav files and play them with aplay (no pulseaudio or anything in the chain). Could you try the same and report what happens? Daniel

6 23

[alsa-devel] [PATCH V4 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 25 Jul '11

25 Jul '11

Hello, Here's V4 of the AC97/I2S ASoC patchset for early Alchemy chips and their respective evaluation boards. The patches are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V3: - dropped the hunk which removed the I2S constants from the au1000.h header to avoid merge conflicts with other patches. - use the context structure declared in psc.h. Follow-up patches for psc* code depend on this. Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 ++++ sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 365 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 374 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 342 ++++++++++++++++++++++ sound/soc/au1x/psc.h | 19 +- 9 files changed, 1256 insertions(+), 10 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 7

Re: [alsa-devel] ac97: quiet make W=1 noise by initializing all fields of snd_ac97_codec_id_vendors and snd_ac97_codec_ids.
by Takashi Iwai 25 Jul '11

25 Jul '11

At Sun, 24 Jul 2011 11:40:56 +0200, Lionel Debroux wrote: > > make W=1 bitterly complains about missing initializations in > snd_ac97_codec_id_vendors and snd_ac97_codec_ids. > Initialize the last field, to make the output of make W=1 more readable. > > Patch against current Linus tree. If we change the whole table entries, it'd be better to use C99 style init, e.g. #define _CODEC(_id, _mask, _name) \ .id = _id, .mask = _mask, .name = _name static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { _CODEC(0x41445300, 0xffffff00, "Analog Devices"),}, ... static const struct ac97_codec_id snd_ac97_codec_ids[] = { { _CODEC(0x41445303, 0xffffffff, "AD1819"), .patch = patch_ad1819 }, ... { _CODEC(0x53494c20, 0xffffffe0, "Si3036,8"), .patch = mpatch_si3036, \ .mpatch = mpatch_si3036, .flags = AC97_MODEM_PATCH }, ... thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC: Add regmap as a control type
by Mark Brown 25 Jul '11

25 Jul '11

Allow drivers to set up their own regmap API structures. This is mainly useful with MFDs where the core driver will have set up regmap at the minute, though it may make sense to push the existing regmap setup out of the core into the drivers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-io.c | 4 ++++ 2 files changed, 5 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4d04b4b..d022694 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -261,6 +261,7 @@ extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, SND_SOC_SPI, + SND_SOC_REGMAP, }; enum snd_soc_compress_type { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index b56e1c4..e471ed6 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -132,6 +132,10 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, &config); break; + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ + break; + default: return -EINVAL; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: au1x: update PSC AC97/I2S code
by Manuel Lauss 25 Jul '11

25 Jul '11

The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Run-tested on DB1200, DB1300 and DB1550 boards. Manuel Lauss (2): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 +++++++++++++------ sound/soc/au1x/dbdma2.c | 83 ++++--------------------- sound/soc/au1x/psc-ac97.c | 34 +++++++--- sound/soc/au1x/psc-i2s.c | 32 +++++++--- sound/soc/au1x/psc.h | 5 -- 6 files changed, 117 insertions(+), 117 deletions(-) -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

[alsa-devel] ASoC updates for 3.1
by Mark Brown 25 Jul '11

25 Jul '11

The following changes since commit 440085598672c0e3fde8a48495f61fea418b06d1: Merge branch 'for-3.0' into for-3.1 (2011-07-17 18:25:58 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Sorry, I thought I'd sent this out and then the release caught me on the hop a bit. Apart from Liam's changes and the new driver these are all bug fixes of one form or another - the sgtl5000 driver apparently just didn't work most of the time. Liam Girdwood (2): ASoC: dapm - add DAPM macro for external enum widgets ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context. Mark Brown (3): ASoC: Mark cache as dirty when suspending ASoC: Acknowledge WM8962 interrupts before acting on them ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver" Rajashekhara, Sudhakar (2): ASoC: davinci: fix codec start and stop functions ASoC: davinci: add missing break statement Sangbeom Kim (2): ASoC: SAMSUNG: Modify I2S driver to support idma ASoC: SAMSUNG: Add I2S0 internal dma driver Wolfram Sang (3): ASoC: core: make comments fit the code ASoC: sgtl5000: refactor registering internal ldo ASoC: sgtl5000: guide user when regulator support is needed include/sound/soc-dapm.h | 6 +++ sound/soc/codecs/sgtl5000.c | 70 +++++++++++++++++-------------------- sound/soc/codecs/wm8962.c | 6 ++-- sound/soc/davinci/davinci-vcif.c | 9 +++-- sound/soc/samsung/i2s.c | 7 ++++ sound/soc/soc-core.c | 5 ++- sound/soc/soc-dapm.c | 30 ++++++++++++++++ 7 files changed, 86 insertions(+), 47 deletions(-)

2 1

[alsa-devel] lib pcm DMA problems
by Horner, Richard 24 Jul '11

24 Jul '11

To Whom It May Concern: I see some weird behavior from the PCI audio driver for the Hauppauge analog tv card (model 44842, rev D1B2) where the card is tuning properly but audio is no longer available over PCI bus. This condition can happen within an hour or as late as after multiple months of 24x7 operation. Re-loading the TV card and ALSA driver stacks do not clear this condition. Only a warm boot clears it up. Debug prints in the PCI device sound driver, bt87x.c. "snd_bt87x_interrupt()" handler indicate it is returning only IRQ_NONE when in this state. We also get a DMA error message from pcm_lib.c which states "capture write error (DMA or IRQ trouble?)", which I presume means DMA is turned off for this device. How can I recover from this without a system reboot? Thank you so much! Here is my system info: # cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 7 model name : VIA Samuel 2 stepping : 3 cpu MHz : 533.309 cache size : 64 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr cx8 mtrr pge mmx 3dnow up bogomips : 1066.61 clflush size : 32 power management: ----------------------------------------------------------- Interrupts are still active on IRQ 5 during this state: # cat /proc/interrupts CPU0 0: 177526733 XT-PIC-XT timer 1: 2 XT-PIC-XT i8042 2: 0 XT-PIC-XT cascade 4: 1 XT-PIC-XT 5: 18834539 XT-PIC-XT bttv0, Bt87x audio 7: 0 XT-PIC-XT parport0 8: 0 XT-PIC-XT rtc0 9: 1587422 XT-PIC-XT eth0 10: 12183885 XT-PIC-XT VIA686A 11: 77875422 XT-PIC-XT uhci_hcd:usb1, uhci_hcd:usb2 12: 7 XT-PIC-XT i8042 14: 17133058 XT-PIC-XT ide0 15: 0 XT-PIC-XT ide1 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts SPU: 0 Spurious interrupts RES: 0 Rescheduling interrupts CAL: 0 Function call interrupts TLB: 0 TLB shootdowns TRM: 0 Thermal event interrupts ERR: 0 MIS: 0 Rich Horner Engineering and Technology The Nielsen Company 813-366-5060 www.nielsen.com

1 0

[alsa-devel] How to loopback two PCM interface?
by onlyfever 24 Jul '11

24 Jul '11

Hi, I configure McBSP3 and McBSP4 as PCM interface. I want to connect two PCM interface internally. The data captured from PCM1 can be directly sent to PCM2 playback interface. How can I achieve that? Please give me some advise. thank you!

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Re: [alsa-devel] sound: constify snd_ac97_bus_ops
by Takashi Iwai 24 Jul '11

24 Jul '11

At Sun, 24 Jul 2011 11:40:48 +0200, Lionel Debroux wrote: > > The fields of this struct type don't seem to be used as rvalues, besides > the initialization of struct instances. Let's make const the struct > instances. > > Maybe the Documentation/DocBook/writing-an-alsa-driver.tmpl, and > checkpatch.pl should be updated to reflect this change ? Yes, please either update or give an additional patch for document fixes. I'll apply these series not for 3.1 since this changes API. Will be applied the branches for 3.2. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: Make snd_pcm_debug_name usable outside pcm_lib
by linux＠audioscience.com 24 Jul '11

24 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Formatting a PCM name is useful for module debug too. Add snd_prefix when making function public. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- include/sound/pcm.h | 12 ++++++++++++ sound/core/pcm_lib.c | 10 +++++----- sound/pci/asihpi/asihpi.c | 21 --------------------- 3 files changed, 17 insertions(+), 26 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e1bad11..c83c754 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -507,6 +507,18 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream); void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); + +#if defined CONFIG_SND_DEBUG +void snd_pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len); +#else +static inline void +snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) +{ + *buf = 0; +} +#endif + /* * PCM library */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f134130..f82329d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -128,7 +128,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } -static void pcm_debug_name(struct snd_pcm_substream *substream, +void snd_pcm_debug_name(struct snd_pcm_substream *substream, char *name, size_t len) { snprintf(name, len, "pcmC%dD%d%c:%d", @@ -168,7 +168,7 @@ static void xrun(struct snd_pcm_substream *substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); dump_stack_on_xrun(substream); } @@ -243,7 +243,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) return; if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) return; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { entry = &log->entries[idx]; if (entry->period_size == 0) @@ -319,7 +319,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (pos >= runtime->buffer_size) { if (printk_ratelimit()) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = %ld, " "buffer size = %ld, period size = %ld\n", @@ -364,7 +364,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (xrun_debug(substream, in_interrupt ? XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd("%s_update: %s: pos=%u/%u/%u, " "hwptr=%ld/%ld/%ld/%ld\n", in_interrupt ? "period" : "hwptr", diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index b941d25..eae62eb 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -41,31 +41,10 @@ #include <sound/tlv.h> #include <sound/hwdep.h> - MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. <support(a)audioscience.com>"); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); -#if defined CONFIG_SND_DEBUG -/* copied from pcm_lib.c, hope later patch will make that version public -and this copy can be removed */ -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - snprintf(buf, size, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} -#else -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - *buf = 0; -} -#endif - #if defined CONFIG_SND_DEBUG_VERBOSE /** * snd_printddd - very verbose debug printk -- 1.7.0.4

3 4

Re: [alsa-devel] [arch-general] Panic - no sound devices after upgrade
by Adrian Knoth 23 Jul '11

23 Jul '11

On 07/23/11 16:21, Fons Adriaensen wrote: Note: I suggest to narrow the CC list a bit and better move the discussion to alsa-devel. At least, let's drop LKML and arch. >>> Seems to have solved the problem :-) :-) >> Good to know. I'll wait for your final confirmation and then commit it. > > One problem: the gain settings have changed, now having a range of 0..64. > How do these values map to the internal ones (0..0xffff) ? I *hope* there > is value corresponding to EXACTLY 0x8000 (gain = 1) !!! For the simple ALSA mixer, the driver does gain = ucontrol->value.integer.value[0]*UNITY_GAIN/64; UNITY_GAIN is 32768 (0x8000), so if you select 64 on the ALSA side, it gives you exactly what you need. Or in other words: faders to 100% in alsamixer, that's it. With hdspmixer or any other software that's using the full mixer, you can still use the fine-grained 65k steps. > BTW, why was this change made? No idea. I reckon faberman was tired of stepping through 65k mixer levels in alsamixer. ;) (disclaimer: I never used the old driver, so I don't know how it was before) Cheers

1 0

[alsa-devel] Sending headset insertion events to user space on bootup
by Brad Rubin 23 Jul '11

23 Jul '11

Hi all, I’m having an issue passing events when a headset is inserted upon bootup. I’m sending a headset insertion event to user space through snd_soc_jack_report, but user space is not yet ready to receive events and therefore misses the event. Since the kernel driver doesn’t resend this event, then there’s no way for user space to see it. User space is still booting up when the event is sent. Does anyone know of a solution so user space can receive headset insertion events on bootup? Is is possible to either allow the kernel to postpone sending events until user space is ready or to store events in some sort of cache so that user space can access them? Would it be possible to use ALSA to store the headset status in a device node? Thanks, Brad -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

[alsa-devel] ASoC Peek individual register debugfs
by Patrick Lai 23 Jul '11

23 Jul '11

Hi, Does latest soc-core.c provide debugfs interface for peeking individual register? If not, I plan to add this interface. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 3

[alsa-devel] ABE routing table without TWL6040
by hong zhang 23 Jul '11

23 Jul '11

List, My custom OMAP4 board does not have TWL6040 device. What I do is 1. comment out TWL6040 in arch/arm/mach-omap2/Kconfig 2. comment out twl6040 inside sdp4430_dai[] at sound/soc/omap/sdp4430.c I find ABE routing table is no longer useful because of changes in steps 1 and 2. run "arecord -f S16_LE -r 48000 -c 2 -D plughw:0,0 try.wav" returns error: Recording WAVE 'try.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo try.wav: No such file or directory dmesg shows dmic_trigger is called. 1. Could anyone please give me hint where I should look into? 2. Try Legacy DMIC to turn off ABE by comment out ABE in sound/soc/omap/kconfig but "No Sound card Found" when boot. Thanks. ---henry

1 0

[alsa-devel] [PATCH] sound, hda_codec: Fix compilation warnings
by Borislav Petkov 22 Jul '11

22 Jul '11

From: Borislav Petkov <borislav.petkov(a)amd.com> Convert macro to an inline function to shut up gcc: sound/pci/hda/patch_sigmatel.c: In function `stac92xx_init´: sound/pci/hda/patch_sigmatel.c:4385:3: warning: statement with no effect sound/pci/hda/patch_sigmatel.c: In function `stac92xx_resume´: sound/pci/hda/patch_sigmatel.c:4925:3: warning: statement with no effect sound/pci/hda/hda_codec.c: In function `snd_hda_mixer_amp_switch_put´: sound/pci/hda/patch_realtek.c:4176:2: warning: statement with no effect sound/pci/hda/hda_codec.c:2320:2: warning: statement with no effect sound/pci/hda/patch_realtek.c: In function `alc_resume´: sound/pci/hda/patch_realtek.c:4558:2: warning: statement with no effect sound/pci/hda/patch_realtek.c: In function `alc269_resume´: sound/pci/hda/patch_realtek.c:14878:2: warning: statement with no effect Signed-off-by: Borislav Petkov <borislav.petkov(a)amd.com> --- sound/pci/hda/hda_codec.h | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 59c9730..79ede92 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1005,8 +1005,12 @@ int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) return codec->patch_ops.check_power_status(codec, nid); return 0; } -#else -#define hda_call_check_power_status(codec, nid) 0 +#else +static inline +int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + return 0; +} #endif /* -- 1.7.4.rc2

2 1

[alsa-devel] additional problem
by David Henderson 22 Jul '11

22 Jul '11

Hi gang! I've successfully been able to compile the alsa-utils package with the "--with-alsa-inc-prefix=/opt/staging/alsa/var/share/include --with-alsa-prefix=/opt/staging/alsa/lib", but the problem I'm having now is that the compiled binaries are looking for those directories during run-time and not just compile-time. Does anyone have any thoughts on using those directories for package creation, but that the software doesn't use the '/opt/staging/alsa' prefix during run-time? Thanks, Dave

4 25

[alsa-devel] [PATCH V3 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 22 Jul '11

22 Jul '11

Hello, These 3 patches implement ASoC drivers for the AC97 and I2S controllers found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 +++ arch/mips/include/asm/mach-au1x00/au1000.h | 31 -- sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 375 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 381 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 348 ++++++++++++++++++++++ sound/soc/au1x/psc.h | 22 +- 10 files changed, 1282 insertions(+), 41 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 3

[alsa-devel] [PATCH] regulator: Make core more chatty about some errors
by Mark Brown 22 Jul '11

22 Jul '11

Prevent some head scratching by making the core log about some rare but possible errors with invalid voltage ranges and modes being set. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/core.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/drivers/regulator/core.c b/drivers/regulator/core.c index f59821f..3700d09 100644 --- a/drivers/regulator/core.c +++ b/drivers/regulator/core.c @@ -151,8 +151,11 @@ static int regulator_check_voltage(struct regulator_dev *rdev, if (*min_uV < rdev->constraints->min_uV) *min_uV = rdev->constraints->min_uV; - if (*min_uV > *max_uV) + if (*min_uV > *max_uV) { + rdev_err(rdev, "unsupportable voltage range: %d-%duV\n", + min_uV, max_uV); return -EINVAL; + } return 0; } @@ -205,8 +208,11 @@ static int regulator_check_current_limit(struct regulator_dev *rdev, if (*min_uA < rdev->constraints->min_uA) *min_uA = rdev->constraints->min_uA; - if (*min_uA > *max_uA) + if (*min_uA > *max_uA) { + rdev_err(rdev, "unsupportable current range: %d-%duA\n", + min_uA, max_uA); return -EINVAL; + } return 0; } @@ -221,6 +227,7 @@ static int regulator_mode_constrain(struct regulator_dev *rdev, int *mode) case REGULATOR_MODE_STANDBY: break; default: + rdev_err(rdev, "invalid mode %x specified\n", *mode); return -EINVAL; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: dapm - Add API call to query the widgets of valid DAPM paths.
by Liam Girdwood 22 Jul '11

22 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets and return the path widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This modifies the existing is_connected_input_ep() and is_connected_output_ep() by adding a new parameter to store any valid widgets in a list. We also add some extra debug to provide further insight into the DAPM path and widget discovery. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++ sound/soc/soc-dapm.c | 184 +++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 178 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..198997d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,11 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda0..4205a97 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -605,11 +605,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -638,15 +678,38 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + if (path->weak) continue; if (path->walked) continue; - if (path->sink && path->connect) { + + if (path->sink) { + + dev_dbg(widget->dapm->dev," %c : %s -> %s -> %s\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name); + + if (!path->connect) + continue; + path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -657,7 +720,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -691,21 +755,117 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + if (path->weak) continue; if (path->walked) continue; - if (path->source && path->connect) { + if (path->source) { + + dev_dbg(widget->dapm->dev," %c : %s <- %s <- %s\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name); + + if (!path->connect) + continue; + path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } return con; } +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + paths = is_connected_output_ep(root, list); + dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name); + + return paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name); + paths = is_connected_input_ep(root, list); + dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name); + + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial widget specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream) +{ + struct snd_soc_dapm_widget *w; + enum snd_soc_dapm_type type; + int paths; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + /* get stream root widget AIF from stream string and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (w->id != type) + continue; + + if (strstr(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget for %s not found\n", stream_name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_walk(dapm); + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -732,9 +892,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -745,7 +905,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) int in; if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -759,7 +919,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) int out; if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1315,9 +1475,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", -- 1.7.4.1

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