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 27 Jul '11

27 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

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[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.

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[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

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