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Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Takashi Iwai 22 Jan '13

22 Jan '13

At Sat, 19 Jan 2013 00:33:52 -0500, Miro Hodak wrote: > > On Fri, Jan 18, 2013 at 1:45 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > OK, thanks for quick tests! > > > > > > Takashi > > > > Testing latest updates in sound-unstable: > > commit f741c8a1f649379f8f3eb81119df2676c46075cf > Merge: d821c1e 164a7ad > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Fri Jan 18 18:30:00 2013 +0100 > > Playback and capture appear to work correctly with one possible caveats: > > Item labeled "Front" controls the level of sound through the speakers. I > would expect that it would only control sound from front HP. This happens > whether Independent HP is on and off. When you look at other controls, there are "Surround", "Center", and "LFE", and these are mixer controls for other two line-out jacks. In that sense, "Front" is corresponding to the same group, the line out jack. You have an individual "Headphone" output. Isn't it obvious? Takashi

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[alsa-devel] [asoc:topic/ioremap 1/1] sound/soc/codecs/jz4740.c:364:2: error: implicit declaration of function 'devm_ioremap_resource'
by kbuild test robot 22 Jan '13

22 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/ioremap head: bc96ea9b07adce5ec678899ea2604bd38677938e commit: bc96ea9b07adce5ec678899ea2604bd38677938e [1/1] ASoC: Convert to devm_ioremap_resource() config: make ARCH=i386 allyesconfig All error/warnings: sound/soc/codecs/jz4740.c: In function 'jz4740_codec_probe': >> sound/soc/codecs/jz4740.c:364:2: error: implicit declaration of function 'devm_ioremap_resource' [-Werror=implicit-function-declaration] sound/soc/codecs/jz4740.c:364:7: warning: assignment makes pointer from integer without a cast [enabled by default] cc1: some warnings being treated as errors vim +/devm_ioremap_resource +364 sound/soc/codecs/jz4740.c 358 jz4740_codec = devm_kzalloc(&pdev->dev, sizeof(*jz4740_codec), 359 GFP_KERNEL); 360 if (!jz4740_codec) 361 return -ENOMEM; 362 363 mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); > 364 base = devm_ioremap_resource(&pdev->dev, mem); 365 if (IS_ERR(base)) 366 return PTR_ERR(base); 367 --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

1 0

[alsa-devel] HDA independent headphone
by Raymond Yau 22 Jan '13

22 Jan '13

> > > Actually, it works fine in Skype when using "AD1989B Analog (hw: 0,2)". I > > > was confused because I changed only "Speakers" to that device and tried > > > making a test sound. That did not work, apparently test sound goes to > > > whatever "Ringing" is set to. But when making actual calls, it works as > > > expected and I can take advantage of the Independent HP feature. > > > > Hm, there should be nothing to touch the indep-HP mixer control from > > PCM, so I have no idea why it went wrong. > > > > BTW, I guess we have still an issue regarding indep HP. When it's ON, > > should the auto-mute be disabled, no? > > Although one can disable the automute manually, it'd be maybe more > > user-friendly if the driver does it by itself. > > > > The patch is below, already merged in test/hda-gen-parser branch (and > test/hda-migrate and master branch, too), as usual. > Does it mean that the value of the "front headphone jack" kcontrol is no longer updated when the independent headphone is switch on ? > > --- > From: Takashi Iwai <tiwai(a)suse.de> > Subject: [PATCH] ALSA: hda - Disable HP auto-mute during independent HP mode > > Both the HP auto-mute and the independent HP mode conflict with each > other. Make HP auto-mute disabled (only for the affected HP jack) > during the driver is in HP independent mode. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > sound/pci/hda/hda_generic.c | 19 ++++++++++++++++--- > 1 file changed, 16 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c > index 63d12ef..258fb5e 100644 > --- a/sound/pci/hda/hda_generic.c > +++ b/sound/pci/hda/hda_generic.c > @@ -1889,6 +1889,13 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol, > *dacp = 0; > else > *dacp = spec->alt_dac_nid; > + > + /* update HP auto-mute state too */ > + if (spec->hp_automute_hook) > + spec->hp_automute_hook(codec, NULL); > + else > + snd_hda_gen_hp_automute(codec, NULL); > + > ret = 1; > } > unlock: > @@ -3467,10 +3474,16 @@ static void call_update_outputs(struct hda_codec *codec) > void snd_hda_gen_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack) > { > struct hda_gen_spec *spec = codec->spec; > + hda_nid_t *pins = spec->autocfg.hp_pins; > + int num_pins = ARRAY_SIZE(spec->autocfg.hp_pins); > + > + /* No detection for the first HP jack during indep-HP mode */ > + if (spec->indep_hp_enabled) { > + pins++; > + num_pins--; > + } > > - spec->hp_jack_present = > - detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), > - spec->autocfg.hp_pins); > + spec->hp_jack_present = detect_jacks(codec, num_pins, pins); > if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo)) > return; > call_update_outputs(codec); > -- > 1.8.1.1 >

2 1

[alsa-devel] [PATCH 33/33] ASoC: Convert to devm_ioremap_resource()
by Thierry Reding 22 Jan '13

22 Jan '13

Convert all uses of devm_request_and_ioremap() to the newly introduced devm_ioremap_resource() which provides more consistent error handling. devm_ioremap_resource() provides its own error messages so all explicit error messages can be removed from the failure code paths. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/cirrus/ep93xx-ac97.c | 7 ++++--- sound/soc/cirrus/ep93xx-i2s.c | 6 +++--- sound/soc/codecs/jz4740.c | 6 +++--- sound/soc/fsl/imx-audmux.c | 6 +++--- sound/soc/fsl/imx-ssi.c | 7 +++---- sound/soc/kirkwood/kirkwood-i2s.c | 8 +++----- sound/soc/mxs/mxs-saif.c | 8 +++----- sound/soc/pxa/mmp-sspa.c | 6 +++--- 8 files changed, 25 insertions(+), 29 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index f3f50e6..1738d28 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -11,6 +11,7 @@ */ #include <linux/delay.h> +#include <linux/err.h> #include <linux/io.h> #include <linux/init.h> #include <linux/module.h> @@ -367,9 +368,9 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (!res) return -ENODEV; - info->regs = devm_request_and_ioremap(&pdev->dev, res); - if (!info->regs) - return -ENXIO; + info->regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(info->regs)) + return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); if (!irq) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 3365d4e..323ed69 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -380,9 +380,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) if (!res) return -ENODEV; - info->regs = devm_request_and_ioremap(&pdev->dev, res); - if (!info->regs) - return -ENXIO; + info->regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(info->regs)) + return PTR_ERR(info->regs); info->mclk = clk_get(&pdev->dev, "mclk"); if (IS_ERR(info->mclk)) { diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index d991529..5f607b3 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -361,9 +361,9 @@ static int jz4740_codec_probe(struct platform_device *pdev) return -ENOMEM; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, mem); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(base)) + return PTR_ERR(base); jz4740_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, &jz4740_codec_regmap_config); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..c21ac9c 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -252,9 +252,9 @@ static int imx_audmux_probe(struct platform_device *pdev) of_match_device(imx_audmux_dt_ids, &pdev->dev); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - audmux_base = devm_request_and_ioremap(&pdev->dev, res); - if (!audmux_base) - return -EADDRNOTAVAIL; + audmux_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audmux_base)) + return PTR_ERR(audmux_base); pinctrl = devm_pinctrl_get_select_default(&pdev->dev); if (IS_ERR(pinctrl)) { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 3b48042..55464a5 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -550,10 +550,9 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_get_resource; } - ssi->base = devm_request_and_ioremap(&pdev->dev, res); - if (!ssi->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; + ssi->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(ssi->base)) { + ret = PTR_ERR(ssi->base); goto failed_register; } diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 282d8b1..c74c890 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -472,11 +472,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - priv->io = devm_request_and_ioremap(&pdev->dev, mem); - if (!priv->io) { - dev_err(&pdev->dev, "devm_request_and_ioremap failed\n"); - return -ENOMEM; - } + priv->io = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(priv->io)) + return PTR_ERR(priv->io); priv->irq = platform_get_irq(pdev, 0); if (priv->irq <= 0) { diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 365d9d2..b327709 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -724,11 +724,9 @@ static int mxs_saif_probe(struct platform_device *pdev) iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - saif->base = devm_request_and_ioremap(&pdev->dev, iores); - if (!saif->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - return -ENODEV; - } + saif->base = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(saif->base)) + return PTR_ERR(saif->base); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) { diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 41c3a09..9140c4a 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -429,9 +429,9 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res); - if (priv->sspa->mmio_base == NULL) - return -ENODEV; + priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(priv->sspa->mmio_base)) + return PTR_ERR(priv->sspa->mmio_base); priv->sspa->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(priv->sspa->clk)) -- 1.8.1.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: Release firmware on error
by Charles Keepax 22 Jan '13

22 Jan '13

This patch correctly releases the firmware if the magic string in the firmware header does not match. Change-Id: I2d08f7fff92b2f0e0f2ab1fb937007e3783f815e Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4196f2d..b6b6548 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -396,7 +396,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) hdr = (void*)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); - return -EINVAL; + goto out_fw; } adsp_dbg(dsp, "%s: v%d.%d.%d\n", file, -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/5] ASoC: tegra: Some fixes in codec files
by Sachin Kamat 22 Jan '13

22 Jan '13

Compile tested using tegra_defconfig on the for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Sachin Kamat (5): ASoC: tegra_wm9712: Remove __devinitconst attribute ASoC: tegra20_ac97: Remove __devinitconst attribute ASoC: tegra: Make 'ahub_auxdata' static ASoC: tegra: Use NULL instead of 0 for pointers ASoC: tegra: Staticize some functions in tegra30_i2s.c sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 6 +++--- sound/soc/tegra/tegra30_i2s.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 2 +- 4 files changed, 7 insertions(+), 7 deletions(-) -- 1.7.4.1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixup for Acer AO725 laptop
by Takashi Iwai 21 Jan '13

21 Jan '13

Acer AO725 needs the same fixup as AO756. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- it's targeted for 3.8-rc5 inclusion sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6369b23..cf38861 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6266,6 +6266,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), -- 1.8.1.1

1 0

[alsa-devel] [PATCH 2/2] ALSA: hda - Add Conexant CX20755/20756/20757 codec IDs
by Takashi Iwai 21 Jan '13

21 Jan '13

These are just compatible with other CX2075x codecs. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- it's targeted for 3.8-rc5 inclusion sound/pci/hda/patch_conexant.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dd798c3..009b77a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4636,6 +4636,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f15111, .name = "CX20753/4", .patch = patch_conexant_auto }, + { .id = 0x14f15113, .name = "CX20755", + .patch = patch_conexant_auto }, + { .id = 0x14f15114, .name = "CX20756", + .patch = patch_conexant_auto }, + { .id = 0x14f15115, .name = "CX20757", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -4659,6 +4665,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150b9"); MODULE_ALIAS("snd-hda-codec-id:14f1510f"); MODULE_ALIAS("snd-hda-codec-id:14f15110"); MODULE_ALIAS("snd-hda-codec-id:14f15111"); +MODULE_ALIAS("snd-hda-codec-id:14f15113"); +MODULE_ALIAS("snd-hda-codec-id:14f15114"); +MODULE_ALIAS("snd-hda-codec-id:14f15115"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.8.1.1

1 0

[alsa-devel] Invitation to connect on LinkedIn
by Putinei Ionut via LinkedIn 21 Jan '13

21 Jan '13

LinkedIn ------------ Putinei Ionut requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, Find out why I use LinkedIn. Stay in touch and build your professional network. - Putinei Accept invitation from Putinei Ionut http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Putinei Ionut http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/rso/227244299/bCFj/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] [PATCH] ASoC: arizona: Support clearing clocks
by Mark Brown 21 Jan '13

21 Jan '13

Some systems may wish to support switching between telephony and CD audio clock rates but this is restricted by enforcement of constraints on the current DAI clock. Support setting clocks to zero and don't enforce any constraints in that case in order to facilitate this use case. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 4c2b626..8c56488 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -585,6 +585,10 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, case 147456000: val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT; break; + case 0: + dev_dbg(arizona->dev, "%s cleared\n", name); + *clk = freq; + return 0; default: return -EINVAL; } @@ -803,6 +807,9 @@ static int arizona_startup(struct snd_pcm_substream *substream, return 0; } + if (base_rate == 0) + return 0; + if (base_rate % 8000) constraint = &arizona_44k1_constraint; else -- 1.7.10.4

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[alsa-devel] question on snd_soc_dai_link_event
by Sebastien LEDUC 21 Jan '13

21 Jan '13

Hi, I have one question about current implementation of function snd_soc_dai_link_event Currently, on power up, it calls hw_params on the source and sink dai drivers, and digital_mute(0) on the sink dai driver I wonder why it does not call the full hw_params / prepare / trigger sequence on both dai drivers to start the 2 DAIs. Thanks in advance Regards, Sebastien

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[alsa-devel] Focusrite Scarlett 18i6 mixer
by Robin Gareus 20 Jan '13

20 Jan '13

Hi, I've started to reverse engineer the 18i6 mixer protocol and came up with a proof-of-concept test script which includes documentation: https://github.com/x42/scarlettmixer Except for a few fixed controls and switches, the 18i6's mixer and router is [likely] too complex to be exposed via ALSA's mixer interface. AFAICT it'll require a dedicated control application - like hdspmixer - in order to make full use of it. The current proof-of-concept uses pyUSB to send URB control messages and therefore requires root access. It will also not work while the device is in use (by snd-usb-audio). A dedicated ioctl interface - similar to SNDRV_HDSP_ - should be added to ALSA. However, it is yet unclear to me how much of the protocol should be moved to kernel-space. Anyway, the script can be used to configure the device, save settings to it and subsequently use those settings with snd-usb-audio and Martin's patch. Thoughts, comments? robin PS. the same protocol may apply to other Focusrite Scalett USB2 devices such as the 8i6 - some of the factory-reset traces hint at that, but I would not know.

1 1

[alsa-devel] Must MIDI SysEx data contain the start byte 0xf0 and the end byte 0xf7?
by Joerg-Cyril.Hoehle＠t-systems.com 20 Jan '13

20 Jan '13

Hi, I've read about SysEx continuations aka. divided SysEx http://www.allegro.cc/forums/thread/607445 http://www.sonicspot.com/guide/midifiles.html What it means is that within a MIDI file, a long sysex could be split across several events with different delta times, giving the receiver some processing time in between (e.g. flash EEPROM etc.). The delta times tell when to send the parts. Now, is that truth? How to do that with ALSA? The documentation on snd_seq_ev_set_sysex insists: "the sysex data must contain the start byte 0xf0 and the end byte 0xf7." http://www.alsa-project.org/alsa-doc/alsa-lib/group___seq_middle.html#ga104… Would it be the sole business of the low-level ALSA driver to slowly send large (and complete) sysex data, out of application control (e.g. the delta times would get lost)? Should apps always coalesce partial sysex data and hand out the complete blob to ALSA's snd_seq_ev_set_sysex? Is divided SysEx too rare to worry about? I'm asking because we've seen a user of Wine & KORG stumble upon a non-F7-terminated SysEx. Regards, Jörg Höhle

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[alsa-devel] [PATCH] ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory
by Mark Brown 20 Jan '13

20 Jan '13

With some buses the transfers may DMAed, especially for larger blocks. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 625e973..3e4e40e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3131,9 +3131,12 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (!codec->using_regmap) return -EINVAL; - data = ucontrol->value.bytes.data; len = params->num_regs * codec->val_bytes; + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a @@ -3146,10 +3149,6 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, val &= params->mask; - data = kmemdup(data, len, GFP_KERNEL); - if (!data) - return -ENOMEM; - switch (codec->val_bytes) { case 1: ((u8 *)data)[0] &= ~params->mask; @@ -3171,8 +3170,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(codec->control_data, params->base, data, len); - if (params->mask) - kfree(data); + kfree(data); return ret; } -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: wm2200: Implement EQ and LHPF coefficient configuration
by Mark Brown 20 Jan '13

20 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 732b3dd..1be61e6 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1141,6 +1141,12 @@ SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SND_SOC_BYTES_MASK("EQL Coefficients", WM2200_EQL_1, 20, WM2200_EQL_ENA), +SND_SOC_BYTES_MASK("EQR Coefficients", WM2200_EQR_1, 20, WM2200_EQR_ENA), + +SND_SOC_BYTES("LHPF1 Coefficeints", WM2200_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficeints", WM2200_HPLPF2_2, 1), + SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L, WM2200_OUT1_OSR_SHIFT, 1, 0), SOC_SINGLE("OUT2 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_2L, -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Make region identification errors more informative
by Mark Brown 20 Jan '13

20 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f956c35..267a6f1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -773,7 +773,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) break; default: - adsp_err(dsp, "Unknown region type %x\n", type); + adsp_err(dsp, "%s.%d: Unknown region type %x at %d\n", + file, blocks, type, pos); break; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Use GFP_DMA for things that may be DMAed
by Mark Brown 20 Jan '13

20 Jan '13

Normally kmalloc() returns things that are DMA safe so not visible on all platforms but we do need to explicitly request DMA safe memory. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f263bfd..f956c35 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -410,7 +410,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) if (reg) { buf = kmemdup(region->data, le32_to_cpu(region->len), - GFP_KERNEL); + GFP_KERNEL | GFP_DMA); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; @@ -557,7 +557,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", term, be32_to_cpu(val)); - alg = kzalloc((term - pos) * 2, GFP_KERNEL); + alg = kzalloc((term - pos) * 2, GFP_KERNEL | GFP_DMA); if (!alg) return -ENOMEM; @@ -779,7 +779,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) if (reg) { buf = kmemdup(blk->data, le32_to_cpu(blk->len), - GFP_KERNEL); + GFP_KERNEL | GFP_DMA); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; -- 1.7.10.4

1 0

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Takashi Iwai 19 Jan '13

19 Jan '13

At Thu, 17 Jan 2013 21:40:43 +0100, Manolo Díaz wrote: > > El jue, 17 ene 2013 a las 21:16 horas > Takashi Iwai escribió: > > >At Thu, 17 Jan 2013 19:12:24 +0100, > >Manolo Díaz wrote: > >> > >> El jue, 17 ene 2013 a las 16:32 horas > >> Takashi Iwai escribió: > >> > >> >At Thu, 17 Jan 2013 09:59:20 -0500, > >> >Miro Hodak wrote: > >> >> > >> >> On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau <superquad.vortex2(a)gmail.com>wrote: > >> >> > >> >> > > >> >> > > > >> >> > > OK, now as far as Auto-Mute goes, should it do what expected when > >> >> > enabled - i.e. muting the speakers when headphones are connected to the > >> >> > front panel? I have tried that, does not seem to work. > >> >> > > > >> >> > > >> >> > Do your computer chassis have a hda front audio panel with Jack detection ? > >> >> > > >> >> > The front headphone Jack should return true when headphone Jack is plugged > >> >> > > >> >> > control.35 { > >> >> > iface CARD > >> >> > name 'Front Headphone Jack' > >> >> > > >> >> > value false > >> >> > comment { > >> >> > access read > >> >> > type BOOLEAN > >> >> > count 1 > >> >> > } > >> >> > } > >> >> > > >> >> > All ADI codecs support impedance measurement > >> >> > > >> >> > Try the modified hda-jack-sense-test which need hda-analyzer > >> >> > > >> >> > http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115 > >> >> > > >> >> > >> >> Hmmm, ran into some python issues. Even hda-analyzer itself is not working: > >> >> ... > >> >> File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile > >> >> raise ValueError, "wrong proc file format (%s)" % msg > >> >> ValueError: wrong proc file format (integer decode 'State of AFG node > >> >> 0x01:' ('Node ')) > >> > > >> >Ah, it's a new proc file entry that has been added recently. > >> >hda-analyzer needs to be fixed. > >> > > >> >In anyway, only for checking the presence unsolicited events, you can > >> >do without extra program. Suppose you built your kernel with the > >> >tracing support. See "Tracepoints" option in > >> >Documentation/sound/alsa/HD-Audio.txt. > >> > > >> > # echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable > >> > > >> > (plug and unplug your headphone jacks) > >> > > >> > # cat /sys/kernel/debug/tracing/trace > >> > > >> >If you see some hda_unsol_event lines there, it implies that the jack > >> >detection basically works in hardware level. > >> > > >> > > >> >Takashi > >> > >> I Suppose that when auto-mute is enabled an HP independent is disabled, > >> inserting the HP jack should disable sound from front speakers. If so > >> then auto-mute doesn't work. Jack insertion/extraction events are well > >> detected for front speakers and HP. > >> > >> #~ cat /sys/kernel/debug/tracing/trace > >> # tracer: nop > >> # > >> # entries-in-buffer/entries-written: 4/4 #P:2 > >> # > >> # _-----=> irqs-off > >> # / _----=> need-resched > >> # | / _---=> hardirq/softirq > >> # || / _--=> preempt-depth > >> # ||| / delay > >> # TASK-PID CPU# |||| TIMESTAMP FUNCTION > >> # | | | |||| | | > >> <idle>-0 [001] d.h. 137.881919: hda_unsol_event: [0] res=8000000, res_ex=10 > >> <idle>-0 [001] d.h. 162.511102: hda_unsol_event: [0] res=8000000, res_ex=10 > >> <idle>-0 [001] d.h. 246.529813: hda_unsol_event: [0] res=4000000, res_ex=10 > >> <idle>-0 [001] d.h. 272.381719: hda_unsol_event: [0] res=4000000, res_ex=10 > >> > >> branch tested: test/hda-gen-parser > > > >Please test with either master or test/hda-migrate branch. > >test/hda-gen-parser branch doesn't contain the patch for AD codecs. > > > >In anyway, please attach alsa-info.sh outputs before and after > >plugging the headphone with the latest sound-unstable tree (and a > >proper branch). > > > > > >thanks, > > > >Takashi > > branch tested: master > > commit d89012bef4ad6d33c1050dd2924f010f5072ff52 > Merge: ad92fbe 36c9db7 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Jan 17 17:47:19 2013 +0100 > > Just the same: auto-mute doesn't work. > > alsa-info output show no differences (attached). Thanks. It turned out that simply missing unsol_event handlers in the codec driver code. The patch below should fix the issue. It's already merged in test/hda-migrate and master branches. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix missing unsol event handler in some codec drivers This resulted in non-working auto-mute behavior, of course... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_ca0110.c | 1 + sound/pci/hda/patch_cmedia.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 02fe0d1..6feaec4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -595,6 +595,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = ad198x_free, + .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, .suspend = ad198x_suspend, diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 8d09325..db7635c 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -36,6 +36,7 @@ static const struct hda_codec_ops ca0110_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = snd_hda_gen_free, + .unsol_event = snd_hda_jack_unsol_event, }; static int ca0110_parse_auto_config(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 04dd3b6..087cabb 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -576,6 +576,7 @@ static const struct hda_codec_ops cmi_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = snd_hda_gen_free, + .unsol_event = snd_hda_jack_unsol_event, }; static int cmi_parse_auto_config(struct hda_codec *codec) -- 1.8.1

4 17

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Raymond Yau 19 Jan '13

19 Jan '13

> > > Testing latest updates in sound-unstable: > > commit f741c8a1f649379f8f3eb81119df2676c46075cf > Merge: d821c1e 164a7ad > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Fri Jan 18 18:30:00 2013 +0100 > > Playback and capture appear to work correctly with one possible caveats: > > Item labeled "Front" controls the level of sound through the speakers. I would expect that it would only control sound from front HP. This happens whether Independent HP is on and off. > Is it feasible to add a hint which disable the creation of virtual master playback volume and switch ?

2 1

[alsa-devel] [PATCH 0/3] You guessed it, it's generic parser patches
by David Henningsson 18 Jan '13

18 Jan '13

The third one seems to apply to the regular sound tree too. David Henningsson (3): ALSA: hda - make sure there are enough input labels and paths ALSA: hda - don't compare with yourself in fill_input_pin_labels ALSA: hda - Don't add unnecessary indices on HDMI and SPDIF sound/pci/hda/hda_auto_parser.c | 22 ++++++++++++---------- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_generic.h | 6 +++--- 3 files changed, 17 insertions(+), 15 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ALSA: hda - Fix mute led for another HP machine
by David Henningsson 18 Jan '13

18 Jan '13

This machine also has the "HP_Mute_LED_0_A" string in DMI information. Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1096789 Tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f519627..6369b23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6251,6 +6251,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1977, "HP Pavilion 14", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/4] Fixes for missing boost controls in generic parser
by Takashi Iwai 18 Jan '13

18 Jan '13

Hi, this is a series of patches to fix the missing boost controls in the generic parser. I found the problem not only on AD1988 but some others, and while debugging this issue, more issues have been revealed, thus it became more volumes than expected. All patches are found in sound-unstable tree test/hda-gen-parser branch. And, now both test/hda-gen-parser branch and test/hda-migrate branch become identical. It doesn't matter which branch you take. Also, it's already merged in master branch of sound-unstable tree. So.. at this point, it'd be helpful if anyone can test this tree, and let me know any issues. thanks, Takashi

2 6

Re: [alsa-devel] [PATCH] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Hebbar, Gururaja 18 Jan '13

18 Jan '13

On Fri, Jan 18, 2013 at 14:13:40, Michal Bachraty wrote: > On Friday, January 18, 2013 08:22:53 Hebbar, Gururaja wrote: > > On Tue, Jan 15, 2013 at 20:11:00, Michal Bachraty wrote: > > > Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> > > > > Even simple patches need a simple changelog. > > Add a brief description about what & why it is done? > > > > Hi Gurujaja, > > > > - u32 rotate = (32 - word_length) / 4; > This implementation is wrong, but it works only for 16, or 32 bit audio data. > (rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in > 4 bits (max value 0x7)and then masks them . That data are sended to i2s bus. > For 24 bit or 20 bit or other data formats, this code rotates data badly and > you hear somethink like noise. You need to use > > > + u32 rotate = (word_length / 4) & 0x7; > to proper data rotation. Correct. Now can you add the above info into the patch and re-submit. That way when users sees this patch, they will get all the info needed. > > Regards, > Michal > > Regards, Gururaja

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock
by Chris Rattray 18 Jan '13

18 Jan '13

Sets the ADSP1 clock rate to match the system clock rate. To support this the codec driver provides details of register containing the system clock control bits. Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9a3f220..56bd6ba 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -103,6 +103,13 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ +/* + * ADSP1 Control 31 + */ +#define ADSP1_CLK_SEL_MASK 0x0007 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_SHIFT 0 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_WIDTH 3 /* CLK_SEL_ENA */ + #define ADSP2_CONTROL 0 #define ADSP2_CLOCKING 1 #define ADSP2_STATUS1 4 @@ -817,12 +824,38 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; int ret; + int val; switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, ADSP1_SYS_ENA); + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + if(dsp->sysclk_reg) { + ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", + ret); + return ret; + } + + val = (val & dsp->sysclk_mask) + >> dsp->sysclk_shift; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP1_CONTROL_31, + ADSP1_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", + ret); + return ret; + } + } + ret = wm_adsp_load(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41206d7..cb8871a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -40,6 +40,9 @@ struct wm_adsp { struct regmap *regmap; int base; + int sysclk_reg; + int sysclk_mask; + int sysclk_shift; struct list_head alg_regions; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Michal Bachraty 18 Jan '13

18 Jan '13

Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 13f8d73..a03bece 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -626,7 +626,7 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (32 - word_length) / 4; + u32 rotate = (word_length / 4) & 0x7; u32 mask = (1ULL << word_length) - 1; /* -- 1.7.9.5

2 1

[alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Miro Hodak 18 Jan '13

18 Jan '13

Commit identified in the subject (ALSA: HDA - Add Independent Headphone for all models of ad1988/ad1989) causes problem with audio on at least two Asus boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop). The symptoms: 1. XFCE and Gnome2 audio mixers crash whenever sound is started or stopped. The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed. 2. On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are misconfigured since muting the Master channel does not turn the sound off. Either Master is mislabeled or it does not work correctly. Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit fixes these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box. We do not understand enough of alsa in kernel to propose a patch, but hope that we have provided enough information that it will be easy to write. We are happy to provide more information and test any proposed patches. Alsa-info for both boards can be found below. Best regards, Miro Hodak and Manolo Diaz Results of alsa-info Asus Rampage II Gene: Before the commit: http://pastebin.com/gCZurk6D After the commit (autodetect): http://pastebin.com/3hNX2g1c After the commit (6stack_dig model): http://pastebin.com/dL0UGeTW More recent kernel: 3.6.11-gentoo (autodetect): http://pastebin.com/0gjnhYA2 3.6.11-gentoo (6stack_dig model): http://pastebin.com/K9bsWBWW Asus M3A79-T DELUXE Before the commit: http://pastebin.com/hyUBmpDL After the commit (autodetect) : http://pastebin.com/gzGg60bM After the commit (6stack_dig model): http://pastebin.com/y88jCMhM

7 53

[alsa-devel] Problems capturing and playback via the loopback device
by Erik Kazandjian 18 Jan '13

18 Jan '13

Hi, I'm new to alsa and I'm trying to do the following. I have configured my rythm box to use the loopback device. I've written a little program to capture everything that comes on the other side of the loopback and save that into a file. If I play a song on rythm box I see actual packets on the couterside of the loopback however I save these packets in a file and try to play them back using my sound card I hear no sound. I use the following .asoundrc pcm.loopplayback { type hw card 1 device 1 subdevice 0 } pcm.loopcapture { type hw card 1 device 0 subdevice 0 } pcm.play48 { type plug slave { pcm loopplayback rate 44100 channels 2 } } attached is my test program to do the capturing. Anyone has any idea what I'm doing wrong ? Kind regards Erik

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock
by Chris Rattray 18 Jan '13

18 Jan '13

Sets the ADSP1 clock rate to match the system clock rate. To support this the codec driver provides details of register containing the system clock control bits. Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 38 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9a3f220..739dab6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -75,6 +75,8 @@ #define ADSP1_CONTROL_30 0x24 #define ADSP1_CONTROL_31 0x26 +#define ADSP1_CLOCKING 38 + /* * ADSP1 Control 19 */ @@ -103,6 +105,13 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ +/* + * ADSP1 Control 31 + */ +#define ADSP1_CLK_SEL_MASK 0x0007 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_SHIFT 0 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_WIDTH 3 /* CLK_SEL_ENA */ + #define ADSP2_CONTROL 0 #define ADSP2_CLOCKING 1 #define ADSP2_STATUS1 4 @@ -817,12 +826,38 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; int ret; + int val; switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, ADSP1_SYS_ENA); + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + if(dsp->sysclk_reg) { + ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", + ret); + return ret; + } + + val = (val & dsp->sysclk_mask) + >> dsp->sysclk_shift; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP1_CLOCKING, + ADSP1_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", + ret); + return ret; + } + } + ret = wm_adsp_load(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41206d7..cb8871a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -40,6 +40,9 @@ struct wm_adsp { struct regmap *regmap; int base; + int sysclk_reg; + int sysclk_mask; + int sysclk_shift; struct list_head alg_regions; -- 1.7.2.5

2 2

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Miro Hodak 18 Jan '13

18 Jan '13

On Thu, Jan 17, 2013 at 3:40 PM, Manolo Díaz <diaz.manolo(a)gmail.com> wrote: > > branch tested: master > > commit d89012bef4ad6d33c1050dd2924f010f5072ff52 > Merge: ad92fbe 36c9db7 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Jan 17 17:47:19 2013 +0100 > > Just the same: auto-mute doesn't work. > > alsa-info output show no differences (attached). > > Jack inserting event correctly detected again. > > #~ cat /sys/kernel/debug/tracing/trace > # tracer: nop > # > # entries-in-buffer/entries-written: 1/1 #P:2 > # > # _-----=> irqs-off > # / _----=> need-resched > # | / _---=> hardirq/softirq > # || / _--=> preempt-depth > # ||| / delay > # TASK-PID CPU# |||| TIMESTAMP FUNCTION > # | | | |||| | | > <idle>-0 [001] d.h. 198.843063: hda_unsol_event: [0] > res=4000000, res_ex=10 > > > Regards, > -- > Manolo Díaz > Testing master branch, I got the similar results as Manolo commit 920544e4bc95d5fb4465eb58e419a2c61074cd7e Merge: 910fb91 ae12700 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Jan 16 16:26:03 2013 +0100 # cat /sys/kernel/debug/tracing/trace # tracer: nop # # entries-in-buffer/entries-written: 1/1 #P:8 # # _-----=> irqs-off # / _----=> need-resched # | / _---=> hardirq/softirq # || / _--=> preempt-depth # ||| / delay # TASK-PID CPU# |||| TIMESTAMP FUNCTION # | | | |||| | | <idle>-0 [000] d.h. 341.307150: hda_unsol_event: [0] res=4000000, res_ex=10 No difference in alsa-info before and after headphones are plugged in (other than timestamp): http://pastebin.com/J1PiRia8 Should I test with the test/hda-gen-parser branch?

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[alsa-devel] [PATCH RFC] ALSA: hda - Use generic parser for STAC/IDT codec driver
by Takashi Iwai 17 Jan '13

17 Jan '13

Finally we reached here. All codecs driver (except for CA0132, which has really device-specific requirements) have been converted to use the generic parser. This patch appears bigger than others since it also involves with the code shuffling, but mostly the cut-off of parser codes and adapt to the generic parser flags. Most of fixup codecs haven't been changed but just removed a few unnecessary codes. The only missing stuff is the SPDIF mux control. It'll be added again later. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 + sound/pci/hda/patch_sigmatel.c | 5219 ++++++++-------------------------------- 2 files changed, 1019 insertions(+), 4201 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 4004d40..4466bf6 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -112,6 +112,7 @@ config SND_HDA_CODEC_ANALOG config SND_HDA_CODEC_SIGMATEL bool "Build IDT/Sigmatel HD-audio codec support" default y + select SND_HDA_GENERIC help Say Y here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f975735..9d2dfad 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -39,14 +39,11 @@ #include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" +#include "hda_generic.h" enum { - STAC_VREF_EVENT = 1, - STAC_INSERT_EVENT, + STAC_VREF_EVENT = 8, STAC_PWR_EVENT, - STAC_HP_EVENT, - STAC_LO_EVENT, - STAC_MIC_EVENT, }; enum { @@ -115,7 +112,6 @@ enum { STAC_HP_DV4, STAC_HP_DV5, STAC_HP_HDX, - STAC_HP_DV4_1222NR, STAC_92HD71BXX_HP, STAC_92HD71BXX_NO_DMIC, STAC_92HD71BXX_NO_SMUX, @@ -173,30 +169,14 @@ enum { STAC_9872_MODELS }; -struct sigmatel_mic_route { - hda_nid_t pin; - signed char mux_idx; - signed char dmux_idx; -}; - -#define MAX_PINS_NUM 16 -#define MAX_ADCS_NUM 4 -#define MAX_DMICS_NUM 4 - struct sigmatel_spec { - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; + struct hda_gen_spec gen; unsigned int eapd_switch: 1; - unsigned int surr_switch: 1; - unsigned int alt_switch: 1; - unsigned int hp_detect: 1; - unsigned int spdif_mute: 1; - unsigned int check_volume_offset:1; - unsigned int auto_mic:1; unsigned int linear_tone_beep:1; unsigned int headset_jack:1; /* 4-pin headset jack (hp + mono mic) */ unsigned int volknob_init:1; /* special volume-knob initialization */ + unsigned int powerdown_adcs:1; /* gpio lines */ unsigned int eapd_mask; @@ -218,6 +198,7 @@ struct sigmatel_spec { /* analog loopback */ const struct snd_kcontrol_new *aloopback_ctl; + unsigned int aloopback; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -225,458 +206,130 @@ struct sigmatel_spec { unsigned int power_map_bits; unsigned int num_pwrs; const hda_nid_t *pwr_nids; - const hda_nid_t *dac_list; - - /* playback */ - struct hda_input_mux *mono_mux; - unsigned int cur_mmux; - struct hda_multi_out multiout; - hda_nid_t dac_nids[5]; - hda_nid_t hp_dacs[5]; - hda_nid_t speaker_dacs[5]; - - int volume_offset; - - /* capture */ - const hda_nid_t *adc_nids; - unsigned int num_adcs; - const hda_nid_t *mux_nids; - unsigned int num_muxes; - const hda_nid_t *dmic_nids; - unsigned int num_dmics; - const hda_nid_t *dmux_nids; - unsigned int num_dmuxes; - const hda_nid_t *smux_nids; - unsigned int num_smuxes; - unsigned int num_analog_muxes; - - const unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */ - const unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */ - unsigned int num_caps; /* number of capture volume/switch elements */ - - struct sigmatel_mic_route ext_mic; - struct sigmatel_mic_route int_mic; - struct sigmatel_mic_route dock_mic; - - const char * const *spdif_labels; - - hda_nid_t dig_in_nid; - hda_nid_t mono_nid; + unsigned int active_adcs; + + /* beep widgets */ hda_nid_t anabeep_nid; hda_nid_t digbeep_nid; - - /* pin widgets */ - const hda_nid_t *pin_nids; - unsigned int num_pins; - - /* codec specific stuff */ - const struct hda_verb *init; - const struct snd_kcontrol_new *mixer; - - /* capture source */ - struct hda_input_mux *dinput_mux; - unsigned int cur_dmux[2]; - struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - struct hda_input_mux *sinput_mux; - unsigned int cur_smux[2]; - unsigned int cur_amux; - hda_nid_t *amp_nids; - unsigned int powerdown_adcs; - - /* i/o switches */ - unsigned int io_switch[2]; - unsigned int clfe_swap; - hda_nid_t line_switch; /* shared line-in for input and output */ - hda_nid_t mic_switch; /* shared mic-in for input and output */ - hda_nid_t hp_switch; /* NID of HP as line-out */ - unsigned int aloopback; - - struct hda_pcm pcm_rec[2]; /* PCM information */ - - /* dynamic controls and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_dimux; - struct hda_input_mux private_imux; - struct hda_input_mux private_smux; - struct hda_input_mux private_mono_mux; - - /* auto spec */ - unsigned auto_pin_cnt; - hda_nid_t auto_pin_nids[MAX_PINS_NUM]; - unsigned auto_adc_cnt; - hda_nid_t auto_adc_nids[MAX_ADCS_NUM]; - hda_nid_t auto_mux_nids[MAX_ADCS_NUM]; - hda_nid_t auto_dmux_nids[MAX_ADCS_NUM]; - unsigned long auto_capvols[MAX_ADCS_NUM]; - unsigned auto_dmic_cnt; - hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; - - struct hda_vmaster_mute_hook vmaster_mute; }; #define AC_VERB_IDT_SET_POWER_MAP 0x7ec #define AC_VERB_IDT_GET_POWER_MAP 0xfec -static const hda_nid_t stac9200_adc_nids[1] = { - 0x03, -}; - -static const hda_nid_t stac9200_mux_nids[1] = { - 0x0c, -}; - -static const hda_nid_t stac9200_dac_nids[1] = { - 0x02, -}; - static const hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0a, 0x0b, 0x0c, 0xd, 0x0e, 0x0f, 0x10, 0x11 }; -static const hda_nid_t stac92hd73xx_slave_dig_outs[2] = { - 0x26, 0, -}; - -static const hda_nid_t stac92hd73xx_adc_nids[2] = { - 0x1a, 0x1b -}; - -#define STAC92HD73XX_NUM_DMICS 2 -static const hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { - 0x13, 0x14, 0 -}; - -#define STAC92HD73_DAC_COUNT 5 - -static const hda_nid_t stac92hd73xx_mux_nids[2] = { - 0x20, 0x21, -}; - -static const hda_nid_t stac92hd73xx_dmux_nids[2] = { - 0x20, 0x21, -}; - -static const hda_nid_t stac92hd73xx_smux_nids[2] = { - 0x22, 0x23, -}; - -#define STAC92HD73XX_NUM_CAPS 2 -static const unsigned long stac92hd73xx_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), -}; -#define stac92hd73xx_capsws stac92hd73xx_capvols - -#define STAC92HD83_DAC_COUNT 3 - static const hda_nid_t stac92hd83xxx_pwr_nids[7] = { 0x0a, 0x0b, 0x0c, 0xd, 0x0e, 0x0f, 0x10 }; -static const hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { - 0x1e, 0, -}; - -static const hda_nid_t stac92hd83xxx_dmic_nids[] = { - 0x11, 0x20, -}; - static const hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f }; -static const hda_nid_t stac92hd71bxx_adc_nids[2] = { - 0x12, 0x13, -}; - -static const hda_nid_t stac92hd71bxx_mux_nids[2] = { - 0x1a, 0x1b -}; - -static const hda_nid_t stac92hd71bxx_dmux_nids[2] = { - 0x1c, 0x1d, -}; - -static const hda_nid_t stac92hd71bxx_smux_nids[2] = { - 0x24, 0x25, -}; - -#define STAC92HD71BXX_NUM_DMICS 2 -static const hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { - 0x18, 0x19, 0 -}; - -static const hda_nid_t stac92hd71bxx_dmic_5port_nids[STAC92HD71BXX_NUM_DMICS] = { - 0x18, 0 -}; - -static const hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { - 0x22, 0 -}; - -#define STAC92HD71BXX_NUM_CAPS 2 -static const unsigned long stac92hd71bxx_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), -}; -#define stac92hd71bxx_capsws stac92hd71bxx_capvols - -static const hda_nid_t stac925x_adc_nids[1] = { - 0x03, -}; - -static const hda_nid_t stac925x_mux_nids[1] = { - 0x0f, -}; - -static const hda_nid_t stac925x_dac_nids[1] = { - 0x02, -}; - -#define STAC925X_NUM_DMICS 1 -static const hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = { - 0x15, 0 -}; - -static const hda_nid_t stac925x_dmux_nids[1] = { - 0x14, -}; - -static const unsigned long stac925x_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT), -}; -static const unsigned long stac925x_capsws[] = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), -}; - -static const hda_nid_t stac922x_adc_nids[2] = { - 0x06, 0x07, -}; - -static const hda_nid_t stac922x_mux_nids[2] = { - 0x12, 0x13, -}; - -#define STAC922X_NUM_CAPS 2 -static const unsigned long stac922x_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), -}; -#define stac922x_capsws stac922x_capvols - -static const hda_nid_t stac927x_slave_dig_outs[2] = { - 0x1f, 0, -}; - -static const hda_nid_t stac927x_adc_nids[3] = { - 0x07, 0x08, 0x09 -}; - -static const hda_nid_t stac927x_mux_nids[3] = { - 0x15, 0x16, 0x17 -}; - -static const hda_nid_t stac927x_smux_nids[1] = { - 0x21, -}; - -static const hda_nid_t stac927x_dac_nids[6] = { - 0x02, 0x03, 0x04, 0x05, 0x06, 0 -}; - -static const hda_nid_t stac927x_dmux_nids[1] = { - 0x1b, -}; - -#define STAC927X_NUM_DMICS 2 -static const hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { - 0x13, 0x14, 0 -}; - -#define STAC927X_NUM_CAPS 3 -static const unsigned long stac927x_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT), -}; -static const unsigned long stac927x_capsws[] = { - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), -}; - -static const char * const stac927x_spdif_labels[5] = { - "Digital Playback", "ADAT", "Analog Mux 1", - "Analog Mux 2", "Analog Mux 3" -}; - -static const hda_nid_t stac9205_adc_nids[2] = { - 0x12, 0x13 -}; - -static const hda_nid_t stac9205_mux_nids[2] = { - 0x19, 0x1a -}; - -static const hda_nid_t stac9205_dmux_nids[1] = { - 0x1d, -}; - -static const hda_nid_t stac9205_smux_nids[1] = { - 0x21, -}; - -#define STAC9205_NUM_DMICS 2 -static const hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { - 0x17, 0x18, 0 -}; - -#define STAC9205_NUM_CAPS 2 -static const unsigned long stac9205_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT), -}; -static const unsigned long stac9205_capsws[] = { - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT), -}; - -static const hda_nid_t stac9200_pin_nids[8] = { - 0x08, 0x09, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, -}; - -static const hda_nid_t stac925x_pin_nids[8] = { - 0x07, 0x08, 0x0a, 0x0b, - 0x0c, 0x0d, 0x10, 0x11, -}; - -static const hda_nid_t stac922x_pin_nids[10] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x15, 0x1b, -}; - -static const hda_nid_t stac92hd73xx_pin_nids[13] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x22, 0x23 -}; - -#define STAC92HD71BXX_NUM_PINS 13 -static const hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x00, - 0x00, 0x14, 0x18, 0x19, 0x1e, - 0x1f, 0x20, 0x27 -}; -static const hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x18, 0x19, 0x1e, - 0x1f, 0x20, 0x27 -}; - -static const hda_nid_t stac927x_pin_nids[14] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x21, 0x22, 0x23, -}; - -static const hda_nid_t stac9205_pin_nids[12] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x16, 0x17, 0x18, - 0x21, 0x22, -}; -static int stac_add_event(struct hda_codec *codec, hda_nid_t nid, - unsigned char type, int data); -static int stac_add_hp_bass_switch(struct hda_codec *codec); -static void stac92xx_auto_set_pinctl(struct hda_codec *codec, - hda_nid_t nid, int pin_type); -static int hp_bnb2011_with_dock(struct hda_codec *codec); -static int hp_blike_system(u32 subsystem_id); -static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity); - -static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +/* + * PCM hooks + */ +static void stac_playback_pcm_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->dinput_mux, uinfo); + if (action == HDA_GEN_PCM_ACT_OPEN && spec->stream_delay) + msleep(spec->stream_delay); } -static int stac92xx_dmux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_capture_pcm_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int i, idx = 0; - ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx]; - return 0; -} + if (!spec->powerdown_adcs) + return; -static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + for (i = 0; i < spec->gen.num_all_adcs; i++) { + if (spec->gen.all_adcs[i] == hinfo->nid) { + idx = i; + break; + } + } - return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol, - spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + msleep(40); + snd_hda_codec_write(codec, hinfo->nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + spec->active_adcs |= (1 << idx); + break; + case HDA_GEN_PCM_ACT_CLOSE: + snd_hda_codec_write(codec, hinfo->nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + spec->active_adcs &= ~(1 << idx); + break; + } } -static int stac92xx_smux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->sinput_mux, uinfo); -} +/* + * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a + * funky external mute control using GPIO pins. + */ -static int stac92xx_smux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, + unsigned int dir_mask, unsigned int data) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + unsigned int gpiostate, gpiomask, gpiodir; - ucontrol->value.enumerated.item[0] = spec->cur_smux[smux_idx]; - return 0; + snd_printdd("%s msk %x dir %x gpio %x\n", __func__, mask, dir_mask, data); + + gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); + + gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_MASK, 0); + gpiomask |= mask; + + gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + gpiodir |= dir_mask; + + /* Configure GPIOx as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, gpiomask); + snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ + + msleep(1); + + snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } -static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* hook for controlling mic-mute LED GPIO */ +static void stac_capture_led_hook(struct hda_codec *codec, bool enable) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *smux = &spec->private_smux; - unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - int err, val; - hda_nid_t nid; - - err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, - spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]); - if (err < 0) - return err; + bool mute = !enable; - if (spec->spdif_mute) { - if (smux_idx == 0) - nid = spec->multiout.dig_out_nid; - else - nid = codec->slave_dig_outs[smux_idx - 1]; - if (spec->cur_smux[smux_idx] == smux->num_items - 1) - val = HDA_AMP_MUTE; + if (spec->mic_mute_led_on != mute) { + spec->mic_mute_led_on = mute; + if (mute) + spec->gpio_data |= spec->mic_mute_led_gpio; else - val = 0; - /* un/mute SPDIF out */ - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, val); + spec->gpio_data &= ~spec->mic_mute_led_gpio; + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); } - return 0; } static int stac_vrefout_set(struct hda_codec *codec, @@ -702,141 +355,233 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } -static unsigned int stac92xx_vref_set(struct hda_codec *codec, - hda_nid_t nid, unsigned int new_vref) -{ - int error; - unsigned int pincfg; - pincfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - - pincfg &= 0xff; - pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - pincfg |= new_vref; - - if (new_vref == AC_PINCTL_VREF_HIZ) - pincfg |= AC_PINCTL_OUT_EN; - else - pincfg |= AC_PINCTL_IN_EN; - - error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg); - if (error < 0) - return error; - else - return 1; -} - -static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int vref; - vref = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - vref &= AC_PINCTL_VREFEN; - return vref; -} - -static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* update mute-LED accoring to the master switch */ +static void stac_update_led_status(struct hda_codec *codec, int enabled) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} + int muted = !enabled; -static int stac92xx_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + if (!spec->gpio_led) + return; - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} + /* LED state is inverted on these systems */ + if (spec->gpio_led_polarity) + muted = !muted; -static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - const struct hda_input_mux *imux = spec->input_mux; - unsigned int idx, prev_idx, didx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - prev_idx = spec->cur_mux[adc_idx]; - if (prev_idx == idx) - return 0; - if (idx < spec->num_analog_muxes) { - snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[idx].index); - if (prev_idx >= spec->num_analog_muxes && - spec->mux_nids[adc_idx] != spec->dmux_nids[adc_idx]) { - imux = spec->dinput_mux; - /* 0 = analog */ - snd_hda_codec_write_cache(codec, - spec->dmux_nids[adc_idx], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[0].index); - } + if (!spec->vref_mute_led_nid) { + if (muted) + spec->gpio_data |= spec->gpio_led; + else + spec->gpio_data &= ~spec->gpio_led; + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); } else { - imux = spec->dinput_mux; - /* first dimux item is hardcoded to select analog imux, - * so lets skip it - */ - didx = idx - spec->num_analog_muxes + 1; - snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[didx].index); + spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; + stac_vrefout_set(codec, spec->vref_mute_led_nid, + spec->vref_led); } - spec->cur_mux[adc_idx] = idx; - return 1; } -static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +/* vmaster hook to update mute LED */ +static void stac_vmaster_hook(void *private_data, int val) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->mono_mux, uinfo); + stac_update_led_status(private_data, val); } -static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* automute hook to handle GPIO mute and EAPD updates */ +static void stac_update_outputs(struct hda_codec *codec) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = spec->cur_mmux; - return 0; -} + if (spec->gpio_mute) + spec->gen.master_mute = + !(snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); -static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; + snd_hda_gen_update_outputs(codec); - return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol, - spec->mono_nid, &spec->cur_mmux); + if (spec->eapd_mask && spec->eapd_switch) { + unsigned int val = spec->gpio_data; + if (spec->gen.speaker_muted) + val &= ~spec->eapd_mask; + else + val |= spec->eapd_mask; + if (spec->gpio_data != val) + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, + val); + } } -#define stac92xx_aloopback_info snd_ctl_boolean_mono_info - -static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, + bool enable, bool do_write) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct sigmatel_spec *spec = codec->spec; + unsigned int idx, val; - ucontrol->value.integer.value[0] = !!(spec->aloopback & - (spec->aloopback_mask << idx)); - return 0; + for (idx = 0; idx < spec->num_pwrs; idx++) { + if (spec->pwr_nids[idx] == nid) + break; + } + if (idx >= spec->num_pwrs) + return; + + idx = 1 << idx; + + val = spec->power_map_bits; + if (enable) + val &= ~idx; + else + val |= idx; + + /* power down unused output ports */ + if (val != spec->power_map_bits) { + spec->power_map_bits = val; + if (do_write) + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, val); + } +} + +/* update power bit per jack plug/unplug */ +static void jack_update_power(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + if (!spec->num_pwrs) + return; + + if (jack && jack->nid) { + stac_toggle_power_map(codec, jack->nid, + snd_hda_jack_detect(codec, jack->nid), + true); + return; + } + + /* update all jacks */ + for (i = 0; i < spec->num_pwrs; i++) { + hda_nid_t nid = spec->pwr_nids[i]; + jack = snd_hda_jack_tbl_get(codec, nid); + if (!jack || !jack->action) + continue; + if (jack->action == STAC_PWR_EVENT || + jack->action <= HDA_GEN_LAST_EVENT) + stac_toggle_power_map(codec, nid, + snd_hda_jack_detect(codec, nid), + false); + } + + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_IDT_SET_POWER_MAP, + spec->power_map_bits); +} + +static void stac_hp_automute(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + snd_hda_gen_hp_automute(codec, jack); + jack_update_power(codec, jack); +} + +static void stac_line_automute(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + snd_hda_gen_line_automute(codec, jack); + jack_update_power(codec, jack); +} + +static void stac_vref_event(struct hda_codec *codec, struct hda_jack_tbl *event) +{ + unsigned int data; + + data = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + /* toggle VREF state based on GPIOx status */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, + !!(data & (1 << event->private_data))); +} + +/* initialize the power map and enable the power event to jacks that + * haven't been assigned to automute + */ +static void stac_init_power_map(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pwrs; i++) { + hda_nid_t nid = spec->pwr_nids[i]; + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + def_conf = get_defcfg_connect(def_conf); + if (snd_hda_jack_tbl_get(codec, nid)) + continue; + if (def_conf == AC_JACK_PORT_COMPLEX && + !(spec->vref_mute_led_nid == nid || + is_jack_detectable(codec, nid))) { + snd_hda_jack_detect_enable_callback(codec, nid, + STAC_PWR_EVENT, + jack_update_power); + } else { + if (def_conf == AC_JACK_PORT_NONE) + stac_toggle_power_map(codec, nid, false, false); + else + stac_toggle_power_map(codec, nid, true, false); + } + } +} + +/* + */ + +static inline bool get_int_hint(struct hda_codec *codec, const char *key, + int *valp) +{ + return !snd_hda_get_int_hint(codec, key, valp); +} + +/* override some hints from the hwdep entry */ +static void stac_store_hints(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int val; + + if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { + spec->eapd_mask = spec->gpio_dir = spec->gpio_data = + spec->gpio_mask; + } + if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) + spec->gpio_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) + spec->eapd_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) + spec->gpio_mute &= spec->gpio_mask; + val = snd_hda_get_bool_hint(codec, "eapd_switch"); + if (val >= 0) + spec->eapd_switch = val; +} + +/* + * loopback controls + */ + +#define stac_aloopback_info snd_ctl_boolean_mono_info + +static int stac_aloopback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = !!(spec->aloopback & + (spec->aloopback_mask << idx)); + return 0; } -static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int stac_aloopback_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -875,399 +620,466 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct hda_verb stac9200_core_init[] = { - /* set dac0mux for dac converter */ - { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - -static const struct hda_verb stac9200_eapd_init[] = { - /* set dac0mux for dac converter */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {} -}; - -static const struct hda_verb dell_eq_core_init[] = { - /* set master volume to max value without distortion - * and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, - {} -}; - -static const struct hda_verb stac92hd73xx_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - -static const struct hda_verb stac92hd83xxx_core_init[] = { - /* power state controls amps */ - { 0x01, AC_VERB_SET_EAPD, 1 << 2}, - {} -}; - -static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = { - { 0x22, 0x785, 0x43 }, - { 0x22, 0x782, 0xe0 }, - { 0x22, 0x795, 0x00 }, - {} -}; - -static const struct hda_verb stac92hd71bxx_core_init[] = { - /* set master volume and direct control */ - { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - -static const struct hda_verb stac92hd71bxx_unmute_core_init[] = { - /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {} -}; - -static const struct hda_verb stac925x_core_init[] = { - /* set dac0mux for dac converter */ - { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* mute the master volume */ - { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - {} -}; - -static const struct hda_verb stac922x_core_init[] = { - /* set master volume and direct control */ - { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - -static const struct hda_verb d965_core_init[] = { - /* unmute node 0x1b */ - { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* select node 0x03 as DAC */ - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, - {} -}; - -static const struct hda_verb dell_3st_core_init[] = { - /* don't set delta bit */ - {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, - /* unmute node 0x1b */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* select node 0x03 as DAC */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, - {} -}; - -static const struct hda_verb stac927x_core_init[] = { - /* set master volume and direct control */ - { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* enable analog pc beep path */ - { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, - {} -}; - -static const struct hda_verb stac927x_volknob_core_init[] = { - /* don't set delta bit */ - {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, - /* enable analog pc beep path */ - {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, - {} -}; - -static const struct hda_verb stac9205_core_init[] = { - /* set master volume and direct control */ - { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* enable analog pc beep path */ - { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, - {} -}; - -#define STAC_MONO_MUX \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Mono Mux", \ - .count = 1, \ - .info = stac92xx_mono_mux_enum_info, \ - .get = stac92xx_mono_mux_enum_get, \ - .put = stac92xx_mono_mux_enum_put, \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Analog Loopback", \ .count = cnt, \ - .info = stac92xx_aloopback_info, \ - .get = stac92xx_aloopback_get, \ - .put = stac92xx_aloopback_put, \ + .info = stac_aloopback_info, \ + .get = stac_aloopback_get, \ + .put = stac_aloopback_put, \ .private_value = verb_read | (verb_write << 16), \ } -#define DC_BIAS(xname, idx, nid) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = idx, \ - .info = stac92xx_dc_bias_info, \ - .get = stac92xx_dc_bias_get, \ - .put = stac92xx_dc_bias_put, \ - .private_value = nid, \ - } +/* + * Mute LED handling on HP laptops + */ -static const struct snd_kcontrol_new stac9200_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - { } /* end */ -}; +/* check whether it's a HP laptop with a docking port */ +static bool hp_bnb2011_with_dock(struct hda_codec *codec) +{ + if (codec->vendor_id != 0x111d7605 && + codec->vendor_id != 0x111d76d1) + return false; -static const struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - {} -}; + switch (codec->subsystem_id) { + case 0x103c1618: + case 0x103c1619: + case 0x103c161a: + case 0x103c161b: + case 0x103c161c: + case 0x103c161d: + case 0x103c161e: + case 0x103c161f: -static const struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), - {} -}; + case 0x103c162a: + case 0x103c162b: -static const struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - {} -}; + case 0x103c1630: + case 0x103c1631: + case 0x103c1633: + case 0x103c1634: + case 0x103c1635: -static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) -}; + case 0x103c3587: + case 0x103c3588: + case 0x103c3589: + case 0x103c358a: -static const struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT), - { } /* end */ -}; + case 0x103c3667: + case 0x103c3668: + case 0x103c3669: -static const struct snd_kcontrol_new stac9205_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), - {} -}; - -static const struct snd_kcontrol_new stac927x_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), - {} -}; - -static struct snd_kcontrol_new stac_dmux_mixer = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - /* count set later */ - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, -}; + return true; + } + return false; +} -static struct snd_kcontrol_new stac_smux_mixer = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Source", - /* count set later */ - .info = stac92xx_smux_enum_info, - .get = stac92xx_smux_enum_get, - .put = stac92xx_smux_enum_put, -}; +static bool hp_blike_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c1520: + case 0x103c1521: + case 0x103c1523: + case 0x103c1524: + case 0x103c1525: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + case 0x103c172a: + case 0x103c172b: + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c7007: + case 0x103c7008: + return true; + } + return false; +} -static const char * const slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "Bass Speaker", "IEC958", "PCM", - NULL -}; +static void set_hp_led_gpio(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio; -static void stac92xx_update_led_status(struct hda_codec *codec, int enabled); + if (spec->gpio_led) + return; -static void stac92xx_vmaster_hook(void *private_data, int val) -{ - stac92xx_update_led_status(private_data, val); + gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + gpio &= AC_GPIO_IO_COUNT; + if (gpio > 3) + spec->gpio_led = 0x08; /* GPIO 3 */ + else + spec->gpio_led = 0x01; /* GPIO 0 */ } -static void stac92xx_free_kctls(struct hda_codec *codec); - -static int stac92xx_build_controls(struct hda_codec *codec) +/* + * This method searches for the mute LED GPIO configuration + * provided as OEM string in SMBIOS. The format of that string + * is HP_Mute_LED_P_G or HP_Mute_LED_P + * where P can be 0 or 1 and defines mute LED GPIO control state (low/high) + * that corresponds to the NOT muted state of the master volume + * and G is the index of the GPIO to use as the mute LED control (0..9) + * If _G portion is missing it is assigned based on the codec ID + * + * So, HP B-series like systems may have HP_Mute_LED_0 (current models) + * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * -- kunal + */ +static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) { struct sigmatel_spec *spec = codec->spec; - unsigned int vmaster_tlv[4]; - int err; - int i; + const struct dmi_device *dev = NULL; - if (spec->mixer) { - err = snd_hda_add_new_ctls(codec, spec->mixer); - if (err < 0) - return err; + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { + get_int_hint(codec, "gpio_led_polarity", + &spec->gpio_led_polarity); + return 1; } - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (!spec->auto_mic && spec->num_dmuxes > 0 && - snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { - stac_dmux_mixer.count = spec->num_dmuxes; - err = snd_hda_ctl_add(codec, 0, - snd_ctl_new1(&stac_dmux_mixer, codec)); - if (err < 0) - return err; - } - if (spec->num_smuxes > 0) { - int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid); - struct hda_input_mux *smux = &spec->private_smux; - /* check for mute support on SPDIF out */ - if (wcaps & AC_WCAP_OUT_AMP) { - snd_hda_add_imux_item(smux, "Off", 0, NULL); - spec->spdif_mute = 1; + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { + if (sscanf(dev->name, "HP_Mute_LED_%d_%x", + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { + unsigned int max_gpio; + max_gpio = snd_hda_param_read(codec, codec->afg, + AC_PAR_GPIO_CAP); + max_gpio &= AC_GPIO_IO_COUNT; + if (spec->gpio_led < max_gpio) + spec->gpio_led = 1 << spec->gpio_led; + else + spec->vref_mute_led_nid = spec->gpio_led; + return 1; + } + if (sscanf(dev->name, "HP_Mute_LED_%d", + &spec->gpio_led_polarity) == 1) { + set_hp_led_gpio(codec); + return 1; + } + /* BIOS bug: unfilled OEM string */ + if (strstr(dev->name, "HP_Mute_LED_P_G")) { + set_hp_led_gpio(codec); + if (default_polarity >= 0) + spec->gpio_led_polarity = default_polarity; + else + spec->gpio_led_polarity = 1; + return 1; } - stac_smux_mixer.count = spec->num_smuxes; - err = snd_hda_ctl_add(codec, 0, - snd_ctl_new1(&stac_smux_mixer, codec)); - if (err < 0) - return err; } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_dig_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid, - spec->autocfg.dig_out_type[0]); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) { - err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; + /* + * Fallback case - if we don't find the DMI strings, + * we statically set the GPIO - if not a B-series system + * and default polarity is provided + */ + if (!hp_blike_system(codec->subsystem_id) && + (default_polarity == 0 || default_polarity == 1)) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = default_polarity; + return 1; } + return 0; +} - /* if we have no master control, let's create it */ - snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], - HDA_OUTPUT, vmaster_tlv); - /* correct volume offset */ - vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; - /* minimum value is actually mute */ - vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; +/* + * PC beep controls + */ - err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch", true, - &spec->vmaster_mute.sw_kctl); - if (err < 0) - return err; +/* create PC beep volume controls */ +static int stac_auto_create_beep_ctls(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); + struct snd_kcontrol_new *knew; + static struct snd_kcontrol_new abeep_mute_ctl = + HDA_CODEC_MUTE(NULL, 0, 0, 0); + static struct snd_kcontrol_new dbeep_mute_ctl = + HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0); + static struct snd_kcontrol_new beep_vol_ctl = + HDA_CODEC_VOLUME(NULL, 0, 0, 0); - if (spec->gpio_led) { - spec->vmaster_mute.hook = stac92xx_vmaster_hook; - err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); - if (err < 0) - return err; + /* check for mute support for the the amp */ + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { + const struct snd_kcontrol_new *temp; + if (spec->anabeep_nid == nid) + temp = &abeep_mute_ctl; + else + temp = &dbeep_mute_ctl; + knew = snd_hda_gen_add_kctl(&spec->gen, + "Beep Playback Switch", temp); + if (!knew) + return -ENOMEM; + knew->private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT); } - if (spec->aloopback_ctl && - snd_hda_get_bool_hint(codec, "loopback") == 1) { - err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl); - if (err < 0) - return err; + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + knew = snd_hda_gen_add_kctl(&spec->gen, + "Beep Playback Volume", + &beep_vol_ctl); + if (!knew) + return -ENOMEM; + knew->private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT); } + return 0; +} - stac92xx_free_kctls(codec); /* no longer needed */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP +#define stac_dig_beep_switch_info snd_ctl_boolean_mono_info - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); - if (err < 0) - return err; +static int stac_dig_beep_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = codec->beep->enabled; + return 0; +} - return 0; +static int stac_dig_beep_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); } -static const struct hda_pintbl ref9200_pin_configs[] = { - { 0x08, 0x01c47010 }, - { 0x09, 0x01447010 }, - { 0x0d, 0x0221401f }, - { 0x0e, 0x01114010 }, - { 0x0f, 0x02a19020 }, - { 0x10, 0x01a19021 }, - { 0x11, 0x90100140 }, - { 0x12, 0x01813122 }, - {} +static const struct snd_kcontrol_new stac_dig_beep_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Beep Playback Switch", + .info = stac_dig_beep_switch_info, + .get = stac_dig_beep_switch_get, + .put = stac_dig_beep_switch_put, }; -static const struct hda_pintbl gateway9200_m4_pin_configs[] = { - { 0x08, 0x400000fe }, - { 0x09, 0x404500f4 }, - { 0x0d, 0x400100f0 }, - { 0x0e, 0x90110010 }, - { 0x0f, 0x400100f1 }, - { 0x10, 0x02a1902e }, - { 0x11, 0x500000f2 }, - { 0x12, 0x500000f3 }, +static int stac_beep_switch_ctl(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &stac_dig_beep_ctrl)) + return -ENOMEM; + return 0; +} +#endif + +/* + */ + +static const struct hda_verb stac9200_core_init[] = { + /* set dac0mux for dac converter */ + { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {} }; -static const struct hda_pintbl gateway9200_m4_2_pin_configs[] = { - { 0x08, 0x400000fe }, - { 0x09, 0x404500f4 }, - { 0x0d, 0x400100f0 }, - { 0x0e, 0x90110010 }, - { 0x0f, 0x400100f1 }, - { 0x10, 0x02a1902e }, - { 0x11, 0x500000f2 }, - { 0x12, 0x500000f3 }, +static const struct hda_verb stac9200_eapd_init[] = { + /* set dac0mux for dac converter */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {} }; -/* - STAC 9200 pin configs for - 102801A8 - 102801DE - 102801E8 -*/ -static const struct hda_pintbl dell9200_d21_pin_configs[] = { - { 0x08, 0x400001f0 }, - { 0x09, 0x400001f1 }, - { 0x0d, 0x02214030 }, - { 0x0e, 0x01014010 }, - { 0x0f, 0x02a19020 }, - { 0x10, 0x01a19021 }, - { 0x11, 0x90100140 }, - { 0x12, 0x01813122 }, +static const struct hda_verb dell_eq_core_init[] = { + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, {} }; -/* - STAC 9200 pin configs for - 102801C0 - 102801C1 -*/ -static const struct hda_pintbl dell9200_d22_pin_configs[] = { - { 0x08, 0x400001f0 }, - { 0x09, 0x400001f1 }, - { 0x0d, 0x0221401f }, - { 0x0e, 0x01014010 }, - { 0x0f, 0x01813020 }, - { 0x10, 0x02a19021 }, - { 0x11, 0x90100140 }, - { 0x12, 0x400001f2 }, +static const struct hda_verb stac92hd73xx_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -/* +static const struct hda_verb stac92hd83xxx_core_init[] = { + /* power state controls amps */ + { 0x01, AC_VERB_SET_EAPD, 1 << 2}, + {} +}; + +static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = { + { 0x22, 0x785, 0x43 }, + { 0x22, 0x782, 0xe0 }, + { 0x22, 0x795, 0x00 }, + {} +}; + +static const struct hda_verb stac92hd71bxx_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + {} +}; + +static const struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + +static const struct hda_verb stac925x_core_init[] = { + /* set dac0mux for dac converter */ + { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* mute the master volume */ + { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + {} +}; + +static const struct hda_verb stac922x_core_init[] = { + /* set master volume and direct control */ + { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + {} +}; + +static const struct hda_verb d965_core_init[] = { + /* unmute node 0x1b */ + { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + +static const struct hda_verb dell_3st_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* unmute node 0x1b */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + +static const struct hda_verb stac927x_core_init[] = { + /* set master volume and direct control */ + { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + +static const struct hda_verb stac927x_volknob_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* enable analog pc beep path */ + {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + +static const struct hda_verb stac9205_core_init[] = { + /* set master volume and direct control */ + { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + +static const struct snd_kcontrol_new stac92hd73xx_6ch_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3); + +static const struct snd_kcontrol_new stac92hd73xx_8ch_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4); + +static const struct snd_kcontrol_new stac92hd73xx_10ch_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5); + +static const struct snd_kcontrol_new stac92hd71bxx_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2); + +static const struct snd_kcontrol_new stac9205_loopback = + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1); + +static const struct snd_kcontrol_new stac927x_loopback = + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1); + +static const struct hda_pintbl ref9200_pin_configs[] = { + { 0x08, 0x01c47010 }, + { 0x09, 0x01447010 }, + { 0x0d, 0x0221401f }, + { 0x0e, 0x01114010 }, + { 0x0f, 0x02a19020 }, + { 0x10, 0x01a19021 }, + { 0x11, 0x90100140 }, + { 0x12, 0x01813122 }, + {} +}; + +static const struct hda_pintbl gateway9200_m4_pin_configs[] = { + { 0x08, 0x400000fe }, + { 0x09, 0x404500f4 }, + { 0x0d, 0x400100f0 }, + { 0x0e, 0x90110010 }, + { 0x0f, 0x400100f1 }, + { 0x10, 0x02a1902e }, + { 0x11, 0x500000f2 }, + { 0x12, 0x500000f3 }, + {} +}; + +static const struct hda_pintbl gateway9200_m4_2_pin_configs[] = { + { 0x08, 0x400000fe }, + { 0x09, 0x404500f4 }, + { 0x0d, 0x400100f0 }, + { 0x0e, 0x90110010 }, + { 0x0f, 0x400100f1 }, + { 0x10, 0x02a1902e }, + { 0x11, 0x500000f2 }, + { 0x12, 0x500000f3 }, + {} +}; + +/* + STAC 9200 pin configs for + 102801A8 + 102801DE + 102801E8 +*/ +static const struct hda_pintbl dell9200_d21_pin_configs[] = { + { 0x08, 0x400001f0 }, + { 0x09, 0x400001f1 }, + { 0x0d, 0x02214030 }, + { 0x0e, 0x01014010 }, + { 0x0f, 0x02a19020 }, + { 0x10, 0x01a19021 }, + { 0x11, 0x90100140 }, + { 0x12, 0x01813122 }, + {} +}; + +/* + STAC 9200 pin configs for + 102801C0 + 102801C1 +*/ +static const struct hda_pintbl dell9200_d22_pin_configs[] = { + { 0x08, 0x400001f0 }, + { 0x09, 0x400001f1 }, + { 0x0d, 0x0221401f }, + { 0x0e, 0x01014010 }, + { 0x0f, 0x01813020 }, + { 0x10, 0x02a19021 }, + { 0x11, 0x90100140 }, + { 0x12, 0x400001f2 }, + {} +}; + +/* STAC 9200 pin configs for 102801C4 (Dell Dimension E310) 102801C5 @@ -1431,17 +1243,13 @@ static void stac9200_fixup_panasonic(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gpio_mask = spec->gpio_dir = 0x09; spec->gpio_data = 0x00; - break; - case HDA_FIXUP_ACT_PROBE: /* CF-74 has no headphone detection, and the driver should *NOT* * do detection and HP/speaker toggle because the hardware does it. */ - spec->hp_detect = 0; - break; + spec->gen.suppress_auto_mute = 1; } } @@ -1862,7 +1670,6 @@ static void stac92hd73xx_fixup_dell(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; snd_hda_apply_pincfgs(codec, dell_m6_pin_configs); - spec->num_smuxes = 0; spec->eapd_switch = 0; } @@ -1883,43 +1690,34 @@ static void stac92hd73xx_fixup_dell_eq(struct hda_codec *codec, static void stac92hd73xx_fixup_dell_m6_amic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - if (action != HDA_FIXUP_ACT_PRE_PROBE) return; stac92hd73xx_fixup_dell(codec); snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); - spec->num_dmics = 0; } /* Digital Mics */ static void stac92hd73xx_fixup_dell_m6_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - if (action != HDA_FIXUP_ACT_PRE_PROBE) return; stac92hd73xx_fixup_dell(codec); snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); - spec->num_dmics = 1; } /* Both */ static void stac92hd73xx_fixup_dell_m6_both(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - if (action != HDA_FIXUP_ACT_PRE_PROBE) return; stac92hd73xx_fixup_dell(codec); snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); - spec->num_dmics = 1; } static void stac92hd73xx_fixup_alienware_m17x(struct hda_codec *codec, @@ -1931,19 +1729,14 @@ static void stac92hd73xx_fixup_alienware_m17x(struct hda_codec *codec, return; snd_hda_apply_pincfgs(codec, alienware_m17x_pin_configs); - spec->num_dmics = STAC92HD73XX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); spec->eapd_switch = 0; } static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PROBE) - return; - spec->hp_detect = 0; + if (action == HDA_FIXUP_ACT_PRE_PROBE) + codec->no_jack_detect = 1; } static const struct hda_fixup stac92hd73xx_fixups[] = { @@ -2348,6 +2141,54 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { {} /* terminator */ }; +/* HP dv7 bass switch - GPIO5 */ +#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info +static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); + return 0; +} + +static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_data; + + gpio_data = (spec->gpio_data & ~0x20) | + (ucontrol->value.integer.value[0] ? 0x20 : 0); + if (gpio_data == spec->gpio_data) + return 0; + spec->gpio_data = gpio_data; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); + return 1; +} + +static const struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = stac_hp_bass_gpio_info, + .get = stac_hp_bass_gpio_get, + .put = stac_hp_bass_gpio_put, +}; + +static int stac_add_hp_bass_switch(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!snd_hda_gen_add_kctl(&spec->gen, "Bass Speaker Playback Switch", + &stac_hp_bass_sw_ctrl)) + return -ENOMEM; + + spec->gpio_mask |= 0x20; + spec->gpio_dir |= 0x20; + spec->gpio_data |= 0x20; + return 0; +} + static const struct hda_pintbl ref92hd71bxx_pin_configs[] = { { 0x0a, 0x02214030 }, { 0x0b, 0x02a19040 }, @@ -2420,66 +2261,29 @@ static void stac92hd71bxx_fixup_ref(struct hda_codec *codec, spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0; } -static void stac92hd71bxx_fixup_no_dmic(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - spec->num_dmics = 0; - spec->num_smuxes = 0; - spec->num_dmuxes = 0; -} - -static void stac92hd71bxx_fixup_no_smux(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - spec->num_dmics = 1; - spec->num_smuxes = 0; - spec->num_dmuxes = 1; -} - static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct sigmatel_spec *spec = codec->spec; + struct hda_jack_tbl *jack; if (action != HDA_FIXUP_ACT_PRE_PROBE) return; /* Enable VREF power saving on GPIO1 detect */ - stac_add_event(codec, codec->afg, STAC_VREF_EVENT, 0x02); snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); - snd_hda_jack_detect_enable(codec, codec->afg, 0); + snd_hda_jack_detect_enable_callback(codec, codec->afg, + STAC_VREF_EVENT, + stac_vref_event); + jack = snd_hda_jack_tbl_get(codec, codec->afg); + if (jack) + jack->private_data = 0x02; + spec->gpio_mask |= 0x02; /* enable internal microphone */ snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); - stac92xx_auto_set_pinctl(codec, 0x0e, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); - - stac92hd71bxx_fixup_no_dmic(codec, fix, action); -} - -static void stac92hd71bxx_fixup_hp_dv4_1222nr(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - spec->num_dmics = 1; - /* I don't know if it needs 1 or 2 smuxes - will wait for - * bug reports to fix if needed - */ - spec->num_smuxes = 1; - spec->num_dmuxes = 1; } static void stac92hd71bxx_fixup_hp_dv4(struct hda_codec *codec, @@ -2495,21 +2299,14 @@ static void stac92hd71bxx_fixup_hp_dv4(struct hda_codec *codec, static void stac92hd71bxx_fixup_hp_dv5(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; unsigned int cap; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); - stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); break; case HDA_FIXUP_ACT_PROBE: - /* HP dv6 gives the headphone pin as a line-out. Thus we - * need to set hp_detect flag here to force to enable HP - * detection. - */ - spec->hp_detect = 1; /* enable bass on HP dv7 */ cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP); cap &= AC_GPIO_IO_COUNT; @@ -2527,9 +2324,6 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; spec->gpio_led = 0x08; - spec->num_dmics = 1; - spec->num_smuxes = 1; - spec->num_dmuxes = 1; } @@ -2559,7 +2353,7 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, } } - if (find_mute_led_cfg(codec, spec->default_polarity)) + if (find_mute_led_cfg(codec, 1)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); @@ -2574,20 +2368,14 @@ static const struct hda_fixup stac92hd71bxx_fixups[] = { [STAC_DELL_M4_1] = { .type = HDA_FIXUP_PINS, .v.pins = dell_m4_1_pin_configs, - .chained = true, - .chain_id = STAC_92HD71BXX_NO_SMUX, }, [STAC_DELL_M4_2] = { .type = HDA_FIXUP_PINS, .v.pins = dell_m4_2_pin_configs, - .chained = true, - .chain_id = STAC_92HD71BXX_NO_DMIC, }, [STAC_DELL_M4_3] = { .type = HDA_FIXUP_PINS, .v.pins = dell_m4_3_pin_configs, - .chained = true, - .chain_id = STAC_92HD71BXX_NO_SMUX, }, [STAC_HP_M4] = { .type = HDA_FIXUP_FUNC, @@ -2613,23 +2401,9 @@ static const struct hda_fixup stac92hd71bxx_fixups[] = { .chained = true, .chain_id = STAC_92HD71BXX_HP, }, - [STAC_HP_DV4_1222NR] = { + [STAC_92HD71BXX_HP] = { .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_hp_dv4_1222nr, - .chained = true, - .chain_id = STAC_HP_DV4, - }, - [STAC_92HD71BXX_NO_DMIC] = { - .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_no_dmic, - }, - [STAC_92HD71BXX_NO_SMUX] = { - .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_no_smux, - }, - [STAC_92HD71BXX_HP] = { - .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_hp, + .v.func = stac92hd71bxx_fixup_hp, }, }; @@ -2642,7 +2416,7 @@ static const struct hda_model_fixup stac92hd71bxx_models[] = { { .id = STAC_HP_DV4, .name = "hp-dv4" }, { .id = STAC_HP_DV5, .name = "hp-dv5" }, { .id = STAC_HP_HDX, .name = "hp-hdx" }, - { .id = STAC_HP_DV4_1222NR, .name = "hp-dv4-1222nr" }, + { .id = STAC_HP_DV4, .name = "hp-dv4-1222nr" }, {} }; @@ -2652,8 +2426,6 @@ static const struct snd_pci_quirk stac92hd71bxx_fixup_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, - "HP dv4-1222nr", STAC_HP_DV4_1222NR), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, @@ -3234,11 +3006,9 @@ static const struct hda_pintbl dell_3st_pin_configs[] = { static void stac927x_fixup_ref_no_jd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - /* no jack detecion for ref-no-jd model */ - if (action == HDA_FIXUP_ACT_PROBE) - spec->hp_detect = 0; + if (action == HDA_FIXUP_ACT_PRE_PROBE) + codec->no_jack_detect = 1; } static void stac927x_fixup_ref(struct hda_codec *codec, @@ -3266,13 +3036,9 @@ static void stac927x_fixup_dell_dmic(struct hda_codec *codec, spec->eapd_mask = spec->gpio_mask = 0x04; spec->gpio_dir = spec->gpio_data = 0x04; } - spec->dmic_nids = stac927x_dmic_nids; - spec->num_dmics = STAC927X_NUM_DMICS; snd_hda_add_verbs(codec, dell_3st_core_init); spec->volknob_init = 1; - spec->dmux_nids = stac927x_dmux_nids; - spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); } static void stac927x_fixup_volknob(struct hda_codec *codec, @@ -3511,18 +3277,20 @@ static void stac9205_fixup_dell_m43(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct sigmatel_spec *spec = codec->spec; - int err; + struct hda_jack_tbl *jack; if (action == HDA_FIXUP_ACT_PRE_PROBE) { snd_hda_apply_pincfgs(codec, dell_9205_m43_pin_configs); /* Enable unsol response for GPIO4/Dock HP connection */ - err = stac_add_event(codec, codec->afg, STAC_VREF_EVENT, 0x01); - if (err < 0) - return; snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); - snd_hda_jack_detect_enable(codec, codec->afg, 0); + snd_hda_jack_detect_enable_callback(codec, codec->afg, + STAC_VREF_EVENT, + stac_vref_event); + jack = snd_hda_jack_tbl_get(codec, codec->afg); + if (jack) + jack->private_data = 0x01; spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01; @@ -3610,2687 +3378,163 @@ static const struct snd_pci_quirk stac9205_fixup_tbl[] = { "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, - "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, - "Dell Vostro 1500", STAC_9205_DELL_M42), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229, - "Dell Vostro 1700", STAC_9205_DELL_M42), - /* Gateway */ - SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), - SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), - {} /* terminator */ -}; - -/* - * Analog playback callbacks - */ -static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - if (spec->stream_delay) - msleep(spec->stream_delay); - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); -} - -static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital playback callbacks - */ -static int stac92xx_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int stac92xx_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); -} - - -/* - * Analog capture callbacks - */ -static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = spec->adc_nids[substream->number]; - - if (spec->powerdown_adcs) { - msleep(40); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); - return 0; -} - -static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = spec->adc_nids[substream->number]; - - snd_hda_codec_cleanup_stream(codec, nid); - if (spec->powerdown_adcs) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - return 0; -} - -static const struct hda_pcm_stream stac92xx_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in stac92xx_build_pcms */ - .ops = { - .open = stac92xx_dig_playback_pcm_open, - .close = stac92xx_dig_playback_pcm_close, - .prepare = stac92xx_dig_playback_pcm_prepare, - .cleanup = stac92xx_dig_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream stac92xx_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in stac92xx_build_pcms */ -}; - -static const struct hda_pcm_stream stac92xx_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, - .nid = 0x02, /* NID to query formats and rates */ - .ops = { - .open = stac92xx_playback_pcm_open, - .prepare = stac92xx_playback_pcm_prepare, - .cleanup = stac92xx_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0x06, /* NID to query formats and rates */ - .ops = { - .open = stac92xx_playback_pcm_open, - .prepare = stac92xx_playback_pcm_prepare, - .cleanup = stac92xx_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream stac92xx_pcm_analog_capture = { - .channels_min = 2, - .channels_max = 2, - /* NID + .substreams is set in stac92xx_build_pcms */ - .ops = { - .prepare = stac92xx_capture_pcm_prepare, - .cleanup = stac92xx_capture_pcm_cleanup - }, -}; - -static int stac92xx_build_pcms(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "STAC92xx Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dac_nids[0]; - if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && - spec->autocfg.line_outs == 2) - info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap = - snd_pcm_2_1_chmaps; - - info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs; - - if (spec->alt_switch) { - codec->num_pcms++; - info++; - info->name = "STAC92xx Analog Alt"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_alt_playback; - } - - if (spec->multiout.dig_out_nid || spec->dig_in_nid) { - codec->num_pcms++; - info++; - info->name = "STAC92xx Digital"; - info->pcm_type = spec->autocfg.dig_out_type[0]; - if (spec->multiout.dig_out_nid) { - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - } - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } - } - - return 0; -} - -static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) - -{ - snd_hda_set_pin_ctl_cache(codec, nid, pin_type); -} - -#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info - -static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = !!spec->hp_switch; - return 0; -} - -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid); - -static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - int nid = kcontrol->private_value; - - spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0; - - /* check to be sure that the ports are up to date with - * switch changes - */ - stac_issue_unsol_event(codec, nid); - - return 1; -} - -static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int i; - static const char * const texts[] = { - "Mic In", "Line In", "Line Out" - }; - - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value; - - if (nid == spec->mic_switch || nid == spec->line_switch) - i = 3; - else - i = 2; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = i; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= i) - uinfo->value.enumerated.item = i-1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value; - unsigned int vref = stac92xx_vref_get(codec, nid); - - if (vref == snd_hda_get_default_vref(codec, nid)) - ucontrol->value.enumerated.item[0] = 0; - else if (vref == AC_PINCTL_VREF_GRD) - ucontrol->value.enumerated.item[0] = 1; - else if (vref == AC_PINCTL_VREF_HIZ) - ucontrol->value.enumerated.item[0] = 2; - - return 0; -} - -static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int new_vref = 0; - int error; - hda_nid_t nid = kcontrol->private_value; - - if (ucontrol->value.enumerated.item[0] == 0) - new_vref = snd_hda_get_default_vref(codec, nid); - else if (ucontrol->value.enumerated.item[0] == 1) - new_vref = AC_PINCTL_VREF_GRD; - else if (ucontrol->value.enumerated.item[0] == 2) - new_vref = AC_PINCTL_VREF_HIZ; - else - return 0; - - if (new_vref != stac92xx_vref_get(codec, nid)) { - error = stac92xx_vref_set(codec, nid, new_vref); - return error; - } - - return 0; -} - -static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - char *texts[2]; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - if (kcontrol->private_value == spec->line_switch) - texts[0] = "Line In"; - else - texts[0] = "Mic In"; - texts[1] = "Line Out"; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 2; - uinfo->count = 1; - - if (uinfo->value.enumerated.item >= 2) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value; - int io_idx = (nid == spec->mic_switch) ? 1 : 0; - - ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx]; - return 0; -} - -static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value; - int io_idx = (nid == spec->mic_switch) ? 1 : 0; - unsigned short val = !!ucontrol->value.enumerated.item[0]; - - spec->io_switch[io_idx] = val; - - if (val) - stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); - else { - unsigned int pinctl = AC_PINCTL_IN_EN; - if (io_idx) /* set VREF for mic */ - pinctl |= snd_hda_get_default_vref(codec, nid); - stac92xx_auto_set_pinctl(codec, nid, pinctl); - } - - /* check the auto-mute again: we need to mute/unmute the speaker - * appropriately according to the pin direction - */ - if (spec->hp_detect) - stac_issue_unsol_event(codec, nid); - - return 1; -} - -#define stac92xx_clfe_switch_info snd_ctl_boolean_mono_info - -static int stac92xx_clfe_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = spec->clfe_swap; - return 0; -} - -static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int val = !!ucontrol->value.integer.value[0]; - - if (spec->clfe_swap == val) - return 0; - - spec->clfe_swap = val; - - snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, - spec->clfe_swap ? 0x4 : 0x0); - - return 1; -} - -#define STAC_CODEC_HP_SWITCH(xname) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .info = stac92xx_hp_switch_info, \ - .get = stac92xx_hp_switch_get, \ - .put = stac92xx_hp_switch_put, \ - } - -#define STAC_CODEC_IO_SWITCH(xname, xpval) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .info = stac92xx_io_switch_info, \ - .get = stac92xx_io_switch_get, \ - .put = stac92xx_io_switch_put, \ - .private_value = xpval, \ - } - -#define STAC_CODEC_CLFE_SWITCH(xname, xpval) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .info = stac92xx_clfe_switch_info, \ - .get = stac92xx_clfe_switch_get, \ - .put = stac92xx_clfe_switch_put, \ - .private_value = xpval, \ - } - -enum { - STAC_CTL_WIDGET_VOL, - STAC_CTL_WIDGET_MUTE, - STAC_CTL_WIDGET_MUTE_BEEP, - STAC_CTL_WIDGET_MONO_MUX, - STAC_CTL_WIDGET_HP_SWITCH, - STAC_CTL_WIDGET_IO_SWITCH, - STAC_CTL_WIDGET_CLFE_SWITCH, - STAC_CTL_WIDGET_DC_BIAS -}; - -static const struct snd_kcontrol_new stac92xx_control_templates[] = { - HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), - HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0), - STAC_MONO_MUX, - STAC_CODEC_HP_SWITCH(NULL), - STAC_CODEC_IO_SWITCH(NULL, 0), - STAC_CODEC_CLFE_SWITCH(NULL, 0), - DC_BIAS(NULL, 0, 0), -}; - -/* add dynamic controls */ -static struct snd_kcontrol_new * -stac_control_new(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - const char *name, - unsigned int subdev) -{ - struct snd_kcontrol_new *knew; - - knew = snd_array_new(&spec->kctls); - if (!knew) - return NULL; - *knew = *ktemp; - knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) { - /* roolback */ - memset(knew, 0, sizeof(*knew)); - spec->kctls.alloced--; - return NULL; - } - knew->subdevice = subdev; - return knew; -} - -static struct snd_kcontrol_new * -add_control_temp(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) -{ - struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, - HDA_SUBDEV_AMP_FLAG); - if (!knew) - return NULL; - knew->index = idx; - knew->private_value = val; - return knew; -} - -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) -{ - return add_control_temp(spec, ktemp, idx, name, val) ? 0 : -ENOMEM; -} - -static inline int stac92xx_add_control_idx(struct sigmatel_spec *spec, - int type, int idx, const char *name, - unsigned long val) -{ - return stac92xx_add_control_temp(spec, - &stac92xx_control_templates[type], - idx, name, val); -} - - -/* add dynamic controls */ -static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, - const char *name, unsigned long val) -{ - return stac92xx_add_control_idx(spec, type, 0, name, val); -} - -static const struct snd_kcontrol_new stac_input_src_temp = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, -}; - -static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, - hda_nid_t nid, int idx) -{ - int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int control = 0; - struct sigmatel_spec *spec = codec->spec; - char name[22]; - - if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { - if (spec->headset_jack && snd_hda_get_input_pin_attr(def_conf) - != INPUT_PIN_ATTR_DOCK) - return 0; - if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD - && nid == spec->line_switch) - control = STAC_CTL_WIDGET_IO_SWITCH; - else if (snd_hda_query_pin_caps(codec, nid) - & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT)) - control = STAC_CTL_WIDGET_DC_BIAS; - else if (nid == spec->mic_switch) - control = STAC_CTL_WIDGET_IO_SWITCH; - } - - if (control) { - snd_hda_get_pin_label(codec, nid, &spec->autocfg, - name, sizeof(name), NULL); - return stac92xx_add_control(codec->spec, control, - strcat(name, " Jack Mode"), nid); - } - - return 0; -} - -static int stac92xx_add_input_source(struct sigmatel_spec *spec) -{ - struct snd_kcontrol_new *knew; - struct hda_input_mux *imux = &spec->private_imux; - - if (spec->auto_mic) - return 0; /* no need for input source */ - if (!spec->num_adcs || imux->num_items <= 1) - return 0; /* no need for input source control */ - knew = stac_control_new(spec, &stac_input_src_temp, - stac_input_src_temp.name, 0); - if (!knew) - return -ENOMEM; - knew->count = spec->num_adcs; - return 0; -} - -/* check whether the line-input can be used as line-out */ -static hda_nid_t check_line_out_switch(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t nid; - unsigned int pincap; - int i; - - if (cfg->line_out_type != AUTO_PIN_LINE_OUT) - return 0; - for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].type == AUTO_PIN_LINE_IN) { - nid = cfg->inputs[i].pin; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_OUT) - return nid; - } - } - return 0; -} - -static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid); - -/* check whether the mic-input can be used as line-out */ -static hda_nid_t check_mic_out_switch(struct hda_codec *codec, hda_nid_t *dac) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int def_conf, pincap; - int i; - - *dac = 0; - if (cfg->line_out_type != AUTO_PIN_LINE_OUT) - return 0; - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - if (cfg->inputs[i].type != AUTO_PIN_MIC) - continue; - def_conf = snd_hda_codec_get_pincfg(codec, nid); - /* some laptops have an internal analog microphone - * which can't be used as a output */ - if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_OUT) { - *dac = get_unassigned_dac(codec, nid); - if (*dac) - return nid; - } - } - } - return 0; -} - -static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - for (i = 0; i < spec->multiout.num_dacs; i++) { - if (spec->multiout.dac_nids[i] == nid) - return 1; - } - - return 0; -} - -static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - if (is_in_dac_nids(spec, nid)) - return 1; - for (i = 0; i < spec->autocfg.hp_outs; i++) - if (spec->hp_dacs[i] == nid) - return 1; - for (i = 0; i < spec->autocfg.speaker_outs; i++) - if (spec->speaker_dacs[i] == nid) - return 1; - return 0; -} - -static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int j, conn_len; - hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac; - unsigned int wcaps, wtype; - - conn_len = snd_hda_get_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); - /* 92HD88: trace back up the link of nids to find the DAC */ - while (conn_len == 1 && (get_wcaps_type(get_wcaps(codec, conn[0])) - != AC_WID_AUD_OUT)) { - nid = conn[0]; - conn_len = snd_hda_get_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); - } - for (j = 0; j < conn_len; j++) { - wcaps = get_wcaps(codec, conn[j]); - wtype = get_wcaps_type(wcaps); - /* we check only analog outputs */ - if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) - continue; - /* if this route has a free DAC, assign it */ - if (!check_all_dac_nids(spec, conn[j])) { - if (conn_len > 1) { - /* select this DAC in the pin's input mux */ - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, j); - } - return conn[j]; - } - } - - /* if all DACs are already assigned, connect to the primary DAC, - unless we're assigning a secondary headphone */ - fallback_dac = spec->multiout.dac_nids[0]; - if (spec->multiout.hp_nid) { - for (j = 0; j < cfg->hp_outs; j++) - if (cfg->hp_pins[j] == nid) { - fallback_dac = spec->multiout.hp_nid; - break; - } - } - - if (conn_len > 1) { - for (j = 0; j < conn_len; j++) { - if (conn[j] == fallback_dac) { - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, j); - break; - } - } - } - return 0; -} - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid); -static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid); - -/* - * Fill in the dac_nids table from the parsed pin configuration - * This function only works when every pin in line_out_pins[] - * contains atleast one DAC in its connection list. Some 92xx - * codecs are not connected directly to a DAC, such as the 9200 - * and 9202/925x. For those, dac_nids[] must be hard-coded. - */ -static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - hda_nid_t nid, dac; - - for (i = 0; i < cfg->line_outs; i++) { - nid = cfg->line_out_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (!dac) { - if (spec->multiout.num_dacs > 0) { - /* we have already working output pins, - * so let's drop the broken ones again - */ - cfg->line_outs = spec->multiout.num_dacs; - break; - } - /* error out, no available DAC found */ - snd_printk(KERN_ERR - "%s: No available DAC for pin 0x%x\n", - __func__, nid); - return -ENODEV; - } - add_spec_dacs(spec, dac); - } - - for (i = 0; i < cfg->hp_outs; i++) { - nid = cfg->hp_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = dac; - else - add_spec_extra_dacs(spec, dac); - } - spec->hp_dacs[i] = dac; - } - - for (i = 0; i < cfg->speaker_outs; i++) { - nid = cfg->speaker_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) - add_spec_extra_dacs(spec, dac); - spec->speaker_dacs[i] = dac; - } - - /* add line-in as output */ - nid = check_line_out_switch(codec); - if (nid) { - dac = get_unassigned_dac(codec, nid); - if (dac) { - snd_printdd("STAC: Add line-in 0x%x as output %d\n", - nid, cfg->line_outs); - cfg->line_out_pins[cfg->line_outs] = nid; - cfg->line_outs++; - spec->line_switch = nid; - add_spec_dacs(spec, dac); - } - } - /* add mic as output */ - nid = check_mic_out_switch(codec, &dac); - if (nid && dac) { - snd_printdd("STAC: Add mic-in 0x%x as output %d\n", - nid, cfg->line_outs); - cfg->line_out_pins[cfg->line_outs] = nid; - cfg->line_outs++; - spec->mic_switch = nid; - add_spec_dacs(spec, dac); - } - - snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", - spec->multiout.num_dacs, - spec->multiout.dac_nids[0], - spec->multiout.dac_nids[1], - spec->multiout.dac_nids[2], - spec->multiout.dac_nids[3], - spec->multiout.dac_nids[4]); - - return 0; -} - -/* create volume control/switch for the given prefx type */ -static int create_controls_idx(struct hda_codec *codec, const char *pfx, - int idx, hda_nid_t nid, int chs) -{ - struct sigmatel_spec *spec = codec->spec; - char name[32]; - int err; - - if (!spec->check_volume_offset) { - unsigned int caps, step, nums, db_scale; - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - step = (caps & AC_AMPCAP_STEP_SIZE) >> - AC_AMPCAP_STEP_SIZE_SHIFT; - step = (step + 1) * 25; /* in .01dB unit */ - nums = (caps & AC_AMPCAP_NUM_STEPS) >> - AC_AMPCAP_NUM_STEPS_SHIFT; - db_scale = nums * step; - /* if dB scale is over -64dB, and finer enough, - * let's reduce it to half - */ - if (db_scale > 6400 && nums >= 0x1f) - spec->volume_offset = nums / 2; - spec->check_volume_offset = 1; - } - - sprintf(name, "%s Playback Volume", pfx); - err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name, - HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, - spec->volume_offset)); - if (err < 0) - return err; - sprintf(name, "%s Playback Switch", pfx); - err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); - if (err < 0) - return err; - return 0; -} - -#define create_controls(codec, pfx, nid, chs) \ - create_controls_idx(codec, pfx, 0, nid, chs) - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (spec->multiout.num_dacs > 4) { - printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); - return 1; - } else { - snd_BUG_ON(spec->multiout.dac_nids != spec->dac_nids); - spec->dac_nids[spec->multiout.num_dacs] = nid; - spec->multiout.num_dacs++; - } - return 0; -} - -static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { - if (!spec->multiout.extra_out_nid[i]) { - spec->multiout.extra_out_nid[i] = nid; - return 0; - } - } - printk(KERN_WARNING "stac92xx: No space for extra DAC 0x%x\n", nid); - return 1; -} - -/* Create output controls - * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT) - */ -static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, - const hda_nid_t *pins, - const hda_nid_t *dac_nids, - int type) -{ - struct sigmatel_spec *spec = codec->spec; - static const char * const chname[4] = { - "Front", "Surround", NULL /*CLFE*/, "Side" - }; - hda_nid_t nid; - int i, err; - unsigned int wid_caps; - - for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { - if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { - if (is_jack_detectable(codec, pins[i])) - spec->hp_detect = 1; - } - nid = dac_nids[i]; - if (!nid) - continue; - if (type != AUTO_PIN_HP_OUT && i == 2) { - /* Center/LFE */ - err = create_controls(codec, "Center", nid, 1); - if (err < 0) - return err; - err = create_controls(codec, "LFE", nid, 2); - if (err < 0) - return err; - - wid_caps = get_wcaps(codec, nid); - - if (wid_caps & AC_WCAP_LR_SWAP) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_CLFE_SWITCH, - "Swap Center/LFE Playback Switch", nid); - - if (err < 0) - return err; - } - - } else { - const char *name; - int idx; - switch (type) { - case AUTO_PIN_HP_OUT: - name = "Headphone"; - idx = i; - break; - case AUTO_PIN_SPEAKER_OUT: - if (num_outs <= 2) { - name = i ? "Bass Speaker" : "Speaker"; - idx = 0; - break; - } - /* Fall through in case of multi speaker outs */ - default: - name = chname[i]; - idx = 0; - break; - } - err = create_controls_idx(codec, name, idx, nid, 3); - if (err < 0) - return err; - } - } - return 0; -} - -static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, - unsigned int dir_mask, unsigned int data); - -/* hook for controlling mic-mute LED GPIO */ -static int stac92xx_capture_sw_put_led(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - int err; - bool mute; - - err = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - if (err <= 0) - return err; - mute = !(ucontrol->value.integer.value[0] && - ucontrol->value.integer.value[1]); - if (spec->mic_mute_led_on != mute) { - spec->mic_mute_led_on = mute; - if (mute) - spec->gpio_data |= spec->mic_mute_led_gpio; - else - spec->gpio_data &= ~spec->mic_mute_led_gpio; - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } - return err; -} - -static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol, - unsigned long sw, int idx) -{ - struct sigmatel_spec *spec = codec->spec; - struct snd_kcontrol_new *knew; - int err; - - err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, - "Capture Volume", vol); - if (err < 0) - return err; - - knew = add_control_temp(spec, - &stac92xx_control_templates[STAC_CTL_WIDGET_MUTE], - idx, "Capture Switch", sw); - if (!knew) - return -ENOMEM; - /* add a LED hook for some HP laptops */ - if (spec->mic_mute_led_gpio) - knew->put = stac92xx_capture_sw_put_led; - - return 0; -} - -/* add playback controls from the parsed DAC table */ -static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid; - int err; - int idx; - - err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, - spec->multiout.dac_nids, - cfg->line_out_type); - if (err < 0) - return err; - - if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_HP_SWITCH, - "Headphone as Line Out Switch", - cfg->hp_pins[cfg->hp_outs - 1]); - if (err < 0) - return err; - } - - for (idx = 0; idx < cfg->num_inputs; idx++) { - if (cfg->inputs[idx].type > AUTO_PIN_LINE_IN) - break; - nid = cfg->inputs[idx].pin; - err = stac92xx_add_jack_mode_control(codec, nid, idx); - if (err < 0) - return err; - } - - return 0; -} - -/* add playback controls for Speaker and HP outputs */ -static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, - struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int err; - - err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins, - spec->hp_dacs, AUTO_PIN_HP_OUT); - if (err < 0) - return err; - - err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, - spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT); - if (err < 0) - return err; - - return 0; -} - -/* labels for mono mux outputs */ -static const char * const stac92xx_mono_labels[4] = { - "DAC0", "DAC1", "Mixer", "DAC2" -}; - -/* create mono mux for mono out on capable codecs */ -static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *mono_mux = &spec->private_mono_mux; - int i, num_cons; - hda_nid_t con_lst[ARRAY_SIZE(stac92xx_mono_labels)]; - - num_cons = snd_hda_get_connections(codec, - spec->mono_nid, - con_lst, - HDA_MAX_NUM_INPUTS); - if (num_cons <= 0 || num_cons > ARRAY_SIZE(stac92xx_mono_labels)) - return -EINVAL; - - for (i = 0; i < num_cons; i++) - snd_hda_add_imux_item(mono_mux, stac92xx_mono_labels[i], i, - NULL); - - return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX, - "Mono Mux", spec->mono_nid); -} - -/* create PC beep volume controls */ -static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, - hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); - int err, type = STAC_CTL_WIDGET_MUTE_BEEP; - - if (spec->anabeep_nid == nid) - type = STAC_CTL_WIDGET_MUTE; - - /* check for mute support for the the amp */ - if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { - err = stac92xx_add_control(spec, type, - "Beep Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - - /* check to see if there is volume support for the amp */ - if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "Beep Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - return 0; -} - -#ifdef CONFIG_SND_HDA_INPUT_BEEP -#define stac92xx_dig_beep_switch_info snd_ctl_boolean_mono_info - -static int stac92xx_dig_beep_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = codec->beep->enabled; - return 0; -} - -static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); -} - -static const struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = stac92xx_dig_beep_switch_info, - .get = stac92xx_dig_beep_switch_get, - .put = stac92xx_dig_beep_switch_put, -}; - -static int stac92xx_beep_switch_ctl(struct hda_codec *codec) -{ - return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl, - 0, "Beep Playback Switch", 0); -} -#endif - -static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i, j, err = 0; - - for (i = 0; i < spec->num_muxes; i++) { - hda_nid_t nid; - unsigned int wcaps; - unsigned long val; - - nid = spec->mux_nids[i]; - wcaps = get_wcaps(codec, nid); - if (!(wcaps & AC_WCAP_OUT_AMP)) - continue; - - /* check whether already the same control was created as - * normal Capture Volume. - */ - val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - for (j = 0; j < spec->num_caps; j++) { - if (spec->capvols[j] == val) - break; - } - if (j < spec->num_caps) - continue; - - err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, i, - "Mux Capture Volume", val); - if (err < 0) - return err; - } - return 0; -}; - -static const char * const stac92xx_spdif_labels[3] = { - "Digital Playback", "Analog Mux 1", "Analog Mux 2", -}; - -static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *spdif_mux = &spec->private_smux; - const char * const *labels = spec->spdif_labels; - int i, num_cons; - hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - - num_cons = snd_hda_get_connections(codec, - spec->smux_nids[0], - con_lst, - HDA_MAX_NUM_INPUTS); - if (num_cons <= 0) - return -EINVAL; - - if (!labels) - labels = stac92xx_spdif_labels; - - for (i = 0; i < num_cons; i++) - snd_hda_add_imux_item(spdif_mux, labels[i], i, NULL); - - return 0; -} - -/* labels for dmic mux inputs */ -static const char * const stac92xx_dmic_labels[5] = { - "Analog Inputs", "Digital Mic 1", "Digital Mic 2", - "Digital Mic 3", "Digital Mic 4" -}; - -static hda_nid_t get_connected_node(struct hda_codec *codec, hda_nid_t mux, - int idx) -{ - hda_nid_t conn[HDA_MAX_NUM_INPUTS]; - int nums; - nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); - if (idx >= 0 && idx < nums) - return conn[idx]; - return 0; -} - -/* look for NID recursively */ -#define get_connection_index(codec, mux, nid) \ - snd_hda_get_conn_index(codec, mux, nid, 1) - -/* create a volume assigned to the given pin (only if supported) */ -/* return 1 if the volume control is created */ -static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, - const char *label, int idx, int direction) -{ - unsigned int caps, nums; - char name[32]; - int err; - - if (direction == HDA_OUTPUT) - caps = AC_WCAP_OUT_AMP; - else - caps = AC_WCAP_IN_AMP; - if (!(get_wcaps(codec, nid) & caps)) - return 0; - caps = query_amp_caps(codec, nid, direction); - nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; - if (!nums) - return 0; - snprintf(name, sizeof(name), "%s Capture Volume", label); - err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction)); - if (err < 0) - return err; - return 1; -} - -/* create playback/capture controls for input pins on dmic capable codecs */ -static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->private_imux; - struct hda_input_mux *dimux = &spec->private_dimux; - int err, i; - unsigned int def_conf; - - snd_hda_add_imux_item(dimux, stac92xx_dmic_labels[0], 0, NULL); - - for (i = 0; i < spec->num_dmics; i++) { - hda_nid_t nid; - int index, type_idx; - char label[32]; - - nid = spec->dmic_nids[i]; - if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) - continue; - def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) - continue; - - index = get_connection_index(codec, spec->dmux_nids[0], nid); - if (index < 0) - continue; - - snd_hda_get_pin_label(codec, nid, &spec->autocfg, - label, sizeof(label), NULL); - snd_hda_add_imux_item(dimux, label, index, &type_idx); - if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) - snd_hda_add_imux_item(imux, label, index, &type_idx); - - err = create_elem_capture_vol(codec, nid, label, type_idx, - HDA_INPUT); - if (err < 0) - return err; - if (!err) { - err = create_elem_capture_vol(codec, nid, label, - type_idx, HDA_OUTPUT); - if (err < 0) - return err; - if (!err) { - nid = get_connected_node(codec, - spec->dmux_nids[0], index); - if (nid) - err = create_elem_capture_vol(codec, - nid, label, - type_idx, HDA_INPUT); - if (err < 0) - return err; - } - } - } - - return 0; -} - -static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *fixed, hda_nid_t *ext, hda_nid_t *dock) -{ - unsigned int cfg; - unsigned int type; - - if (!nid) - return 0; - cfg = snd_hda_codec_get_pincfg(codec, nid); - type = get_defcfg_device(cfg); - switch (snd_hda_get_input_pin_attr(cfg)) { - case INPUT_PIN_ATTR_INT: - if (*fixed) - return 1; /* already occupied */ - if (type != AC_JACK_MIC_IN) - return 1; /* invalid type */ - *fixed = nid; - break; - case INPUT_PIN_ATTR_UNUSED: - break; - case INPUT_PIN_ATTR_DOCK: - if (*dock) - return 1; /* already occupied */ - if (type != AC_JACK_MIC_IN && type != AC_JACK_LINE_IN) - return 1; /* invalid type */ - *dock = nid; - break; - default: - if (*ext) - return 1; /* already occupied */ - if (type != AC_JACK_MIC_IN) - return 1; /* invalid type */ - *ext = nid; - break; - } - return 0; -} - -static int set_mic_route(struct hda_codec *codec, - struct sigmatel_mic_route *mic, - hda_nid_t pin) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - - mic->pin = pin; - if (pin == 0) - return 0; - for (i = 0; i < cfg->num_inputs; i++) { - if (pin == cfg->inputs[i].pin) - break; - } - if (i < cfg->num_inputs && cfg->inputs[i].type == AUTO_PIN_MIC) { - /* analog pin */ - i = get_connection_index(codec, spec->mux_nids[0], pin); - if (i < 0) - return -1; - mic->mux_idx = i; - mic->dmux_idx = -1; - if (spec->dmux_nids) - mic->dmux_idx = get_connection_index(codec, - spec->dmux_nids[0], - spec->mux_nids[0]); - } else if (spec->dmux_nids) { - /* digital pin */ - i = get_connection_index(codec, spec->dmux_nids[0], pin); - if (i < 0) - return -1; - mic->dmux_idx = i; - mic->mux_idx = -1; - if (spec->mux_nids) - mic->mux_idx = get_connection_index(codec, - spec->mux_nids[0], - spec->dmux_nids[0]); - } - return 0; -} - -/* return non-zero if the device is for automatic mic switch */ -static int stac_check_auto_mic(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t fixed, ext, dock; - int i; - - fixed = ext = dock = 0; - for (i = 0; i < cfg->num_inputs; i++) - if (check_mic_pin(codec, cfg->inputs[i].pin, - &fixed, &ext, &dock)) - return 0; - for (i = 0; i < spec->num_dmics; i++) - if (check_mic_pin(codec, spec->dmic_nids[i], - &fixed, &ext, &dock)) - return 0; - if (!fixed || (!ext && !dock)) - return 0; /* no input to switch */ - if (!is_jack_detectable(codec, ext)) - return 0; /* no unsol support */ - if (set_mic_route(codec, &spec->ext_mic, ext) || - set_mic_route(codec, &spec->int_mic, fixed) || - set_mic_route(codec, &spec->dock_mic, dock)) - return 0; /* something is wrong */ - return 1; -} - -/* create playback/capture controls for input pins */ -static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->private_imux; - int i, j; - const char *label; - - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - int index, err, type_idx; - - index = -1; - for (j = 0; j < spec->num_muxes; j++) { - index = get_connection_index(codec, spec->mux_nids[j], - nid); - if (index >= 0) - break; - } - if (index < 0) - continue; - - label = hda_get_autocfg_input_label(codec, cfg, i); - snd_hda_add_imux_item(imux, label, index, &type_idx); - - err = create_elem_capture_vol(codec, nid, - label, type_idx, - HDA_INPUT); - if (err < 0) - return err; - } - spec->num_analog_muxes = imux->num_items; - - if (imux->num_items) { - /* - * Set the current input for the muxes. - * The STAC9221 has two input muxes with identical source - * NID lists. Hopefully this won't get confused. - */ - for (i = 0; i < spec->num_muxes; i++) { - snd_hda_codec_write_cache(codec, spec->mux_nids[i], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[0].index); - } - } - - return 0; -} - -static void stac92xx_auto_init_multi_out(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->autocfg.line_outs; i++) { - hda_nid_t nid = spec->autocfg.line_out_pins[i]; - stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); - } -} - -static void stac92xx_auto_init_hp_out(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->autocfg.hp_outs; i++) { - hda_nid_t pin; - pin = spec->autocfg.hp_pins[i]; - if (pin) /* connect to front */ - stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); - } - for (i = 0; i < spec->autocfg.speaker_outs; i++) { - hda_nid_t pin; - pin = spec->autocfg.speaker_pins[i]; - if (pin) /* connect to front */ - stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN); - } -} - -static int is_dual_headphones(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i, valid_hps; - - if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT || - spec->autocfg.hp_outs <= 1) - return 0; - valid_hps = 0; - for (i = 0; i < spec->autocfg.hp_outs; i++) { - hda_nid_t nid = spec->autocfg.hp_pins[i]; - unsigned int cfg = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_location(cfg) & AC_JACK_LOC_SEPARATE) - continue; - valid_hps++; - } - return (valid_hps > 1); -} - - -static int stac92xx_parse_auto_config(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t dig_out = 0, dig_in = 0; - int hp_swap = 0; - int i, err; - - if ((err = snd_hda_parse_pin_def_config(codec, - &spec->autocfg, - spec->dmic_nids)) < 0) - return err; - if (! spec->autocfg.line_outs) - return -EINVAL; /* can't find valid pin config */ - - /* If we have no real line-out pin and multiple hp-outs, HPs should - * be set up as multi-channel outputs. - */ - if (is_dual_headphones(codec)) { - /* Copy hp_outs to line_outs, backup line_outs in - * speaker_outs so that the following routines can handle - * HP pins as primary outputs. - */ - snd_printdd("stac92xx: Enabling multi-HPs workaround\n"); - memcpy(spec->autocfg.speaker_pins, spec->autocfg.line_out_pins, - sizeof(spec->autocfg.line_out_pins)); - spec->autocfg.speaker_outs = spec->autocfg.line_outs; - memcpy(spec->autocfg.line_out_pins, spec->autocfg.hp_pins, - sizeof(spec->autocfg.hp_pins)); - spec->autocfg.line_outs = spec->autocfg.hp_outs; - spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; - spec->autocfg.hp_outs = 0; - hp_swap = 1; - } - if (spec->autocfg.mono_out_pin) { - int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & - (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); - u32 caps = query_amp_caps(codec, - spec->autocfg.mono_out_pin, dir); - hda_nid_t conn_list[1]; - - /* get the mixer node and then the mono mux if it exists */ - if (snd_hda_get_connections(codec, - spec->autocfg.mono_out_pin, conn_list, 1) && - snd_hda_get_connections(codec, conn_list[0], - conn_list, 1) > 0) { - - int wcaps = get_wcaps(codec, conn_list[0]); - int wid_type = get_wcaps_type(wcaps); - /* LR swap check, some stac925x have a mux that - * changes the DACs output path instead of the - * mono-mux path. - */ - if (wid_type == AC_WID_AUD_SEL && - !(wcaps & AC_WCAP_LR_SWAP)) - spec->mono_nid = conn_list[0]; - } - if (dir) { - hda_nid_t nid = spec->autocfg.mono_out_pin; - - /* most mono outs have a least a mute/unmute switch */ - dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "Mono Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); - if (err < 0) - return err; - /* check for volume support for the amp */ - if ((caps & AC_AMPCAP_NUM_STEPS) - >> AC_AMPCAP_NUM_STEPS_SHIFT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_VOL, - "Mono Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); - if (err < 0) - return err; - } - } - - stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, - AC_PINCTL_OUT_EN); - } - - if (!spec->multiout.num_dacs) { - err = stac92xx_auto_fill_dac_nids(codec); - if (err < 0) - return err; - err = stac92xx_auto_create_multi_out_ctls(codec, - &spec->autocfg); - if (err < 0) - return err; - } - - /* setup analog beep controls */ - if (spec->anabeep_nid > 0) { - err = stac92xx_auto_create_beep_ctls(codec, - spec->anabeep_nid); - if (err < 0) - return err; - } - - /* setup digital beep controls and input device */ -#ifdef CONFIG_SND_HDA_INPUT_BEEP - if (spec->digbeep_nid > 0) { - hda_nid_t nid = spec->digbeep_nid; - unsigned int caps; - - err = stac92xx_auto_create_beep_ctls(codec, nid); - if (err < 0) - return err; - err = snd_hda_attach_beep_device(codec, nid); - if (err < 0) - return err; - if (codec->beep) { - /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = spec->linear_tone_beep; - /* if no beep switch is available, make its own one */ - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - if (!(caps & AC_AMPCAP_MUTE)) { - err = stac92xx_beep_switch_ctl(codec); - if (err < 0) - return err; - } - } - } -#endif - - err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); - if (err < 0) - return err; - - /* All output parsing done, now restore the swapped hp pins */ - if (hp_swap) { - memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, - sizeof(spec->autocfg.hp_pins)); - spec->autocfg.hp_outs = spec->autocfg.line_outs; - spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; - spec->autocfg.line_outs = 0; - } - - if (stac_check_auto_mic(codec)) { - spec->auto_mic = 1; - /* only one capture for auto-mic */ - spec->num_adcs = 1; - spec->num_caps = 1; - spec->num_muxes = 1; - } - - for (i = 0; i < spec->num_caps; i++) { - err = stac92xx_add_capvol_ctls(codec, spec->capvols[i], - spec->capsws[i], i); - if (err < 0) - return err; - } - - err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); - if (err < 0) - return err; - - if (spec->mono_nid > 0) { - err = stac92xx_auto_create_mono_output_ctls(codec); - if (err < 0) - return err; - } - if (spec->num_dmics > 0 && !spec->dinput_mux) - if ((err = stac92xx_auto_create_dmic_input_ctls(codec, - &spec->autocfg)) < 0) - return err; - if (spec->num_muxes > 0) { - err = stac92xx_auto_create_mux_input_ctls(codec); - if (err < 0) - return err; - } - if (spec->num_smuxes > 0) { - err = stac92xx_auto_create_spdif_mux_ctls(codec); - if (err < 0) - return err; - } - - err = stac92xx_add_input_source(spec); - if (err < 0) - return err; - - spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->multiout.max_channels > 2) - spec->surr_switch = 1; - - /* find digital out and in converters */ - for (i = codec->start_nid; i < codec->start_nid + codec->num_nodes; i++) { - unsigned int wid_caps = get_wcaps(codec, i); - if (wid_caps & AC_WCAP_DIGITAL) { - switch (get_wcaps_type(wid_caps)) { - case AC_WID_AUD_OUT: - if (!dig_out) - dig_out = i; - break; - case AC_WID_AUD_IN: - if (!dig_in) - dig_in = i; - break; - } - } - } - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = dig_out; - if (dig_in && spec->autocfg.dig_in_pin) - spec->dig_in_nid = dig_in; - - if (spec->kctls.list) - spec->mixers[spec->num_mixers++] = spec->kctls.list; - - spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) - spec->dinput_mux = &spec->private_dimux; - spec->sinput_mux = &spec->private_smux; - spec->mono_mux = &spec->private_mono_mux; - return 0; -} - -/* add playback controls for HP output */ -static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, - struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin = cfg->hp_pins[0]; - - if (! pin) - return 0; - - if (is_jack_detectable(codec, pin)) - spec->hp_detect = 1; - - return 0; -} - -/* add playback controls for LFE output */ -static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, - struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int err; - hda_nid_t lfe_pin = 0x0; - int i; - - /* - * search speaker outs and line outs for a mono speaker pin - * with an amp. If one is found, add LFE controls - * for it. - */ - for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) { - hda_nid_t pin = spec->autocfg.speaker_pins[i]; - unsigned int wcaps = get_wcaps(codec, pin); - wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); - if (wcaps == AC_WCAP_OUT_AMP) - /* found a mono speaker with an amp, must be lfe */ - lfe_pin = pin; - } - - /* if speaker_outs is 0, then speakers may be in line_outs */ - if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) { - for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { - hda_nid_t pin = spec->autocfg.line_out_pins[i]; - unsigned int defcfg; - defcfg = snd_hda_codec_get_pincfg(codec, pin); - if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { - unsigned int wcaps = get_wcaps(codec, pin); - wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); - if (wcaps == AC_WCAP_OUT_AMP) - /* found a mono speaker with an amp, - must be lfe */ - lfe_pin = pin; - } - } - } - - if (lfe_pin) { - err = create_controls(codec, "LFE", lfe_pin, 1); - if (err < 0) - return err; - } - - return 0; -} - -static int stac9200_parse_auto_config(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int err; - - if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) - return err; - - if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) - return err; - - if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0) - return err; - - if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0) - return err; - - if (spec->num_muxes > 0) { - err = stac92xx_auto_create_mux_input_ctls(codec); - if (err < 0) - return err; - } - - err = stac92xx_add_input_source(spec); - if (err < 0) - return err; - - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = 0x05; - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = 0x04; - - if (spec->kctls.list) - spec->mixers[spec->num_mixers++] = spec->kctls.list; - - spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; - - return 0; -} - -/* - * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a - * funky external mute control using GPIO pins. - */ - -static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, - unsigned int dir_mask, unsigned int data) -{ - unsigned int gpiostate, gpiomask, gpiodir; - - snd_printdd("%s msk %x dir %x gpio %x\n", __func__, mask, dir_mask, data); - - gpiostate = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); - - gpiomask = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= mask; - - gpiodir = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= dir_mask; - - /* Configure GPIOx as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); - - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ - - msleep(1); - - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ -} - -static int stac_add_event(struct hda_codec *codec, hda_nid_t nid, - unsigned char type, int data) -{ - struct hda_jack_tbl *event; - - event = snd_hda_jack_tbl_new(codec, nid); - if (!event) - return -ENOMEM; - event->action = type; - event->private_data = data; - - return 0; -} - -static void handle_unsol_event(struct hda_codec *codec, - struct hda_jack_tbl *event); - -/* check if given nid is a valid pin and no other events are assigned - * to it. If OK, assign the event, set the unsol flag, and returns 1. - * Otherwise, returns zero. - */ -static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int type) -{ - struct hda_jack_tbl *event; - - if (!is_jack_detectable(codec, nid)) - return 0; - event = snd_hda_jack_tbl_new(codec, nid); - if (!event) - return -ENOMEM; - if (event->action && event->action != type) - return 0; - event->action = type; - event->callback = handle_unsol_event; - snd_hda_jack_detect_enable(codec, nid, 0); - return 1; -} - -static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) -{ - int i; - for (i = 0; i < cfg->hp_outs; i++) - if (cfg->hp_pins[i] == nid) - return 1; /* nid is a HP-Out */ - for (i = 0; i < cfg->line_outs; i++) - if (cfg->line_out_pins[i] == nid) - return 1; /* nid is a line-Out */ - return 0; /* nid is not a HP-Out */ -}; - -static void stac92xx_power_down(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - /* power down inactive DACs */ - const hda_nid_t *dac; - for (dac = spec->dac_list; *dac; dac++) - if (!check_all_dac_nids(spec, *dac)) - snd_hda_codec_write(codec, *dac, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); -} - -static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, - int enable); - -static inline bool get_int_hint(struct hda_codec *codec, const char *key, - int *valp) -{ - return !snd_hda_get_int_hint(codec, key, valp); -} - -/* override some hints from the hwdep entry */ -static void stac_store_hints(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int val; - - val = snd_hda_get_bool_hint(codec, "hp_detect"); - if (val >= 0) - spec->hp_detect = val; - if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { - spec->eapd_mask = spec->gpio_dir = spec->gpio_data = - spec->gpio_mask; - } - if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) - spec->gpio_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) - spec->gpio_dir &= spec->gpio_mask; - if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) - spec->eapd_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) - spec->gpio_mute &= spec->gpio_mask; - val = snd_hda_get_bool_hint(codec, "eapd_switch"); - if (val >= 0) - spec->eapd_switch = val; -} - -static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins, - const hda_nid_t *pins) -{ - while (num_pins--) - stac_issue_unsol_event(codec, *pins++); -} - -/* fake event to set up pins */ -static void stac_fake_hp_events(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (spec->autocfg.hp_outs) - stac_issue_unsol_events(codec, spec->autocfg.hp_outs, - spec->autocfg.hp_pins); - if (spec->autocfg.line_outs && - spec->autocfg.line_out_pins[0] != spec->autocfg.hp_pins[0]) - stac_issue_unsol_events(codec, spec->autocfg.line_outs, - spec->autocfg.line_out_pins); -} - -static int stac92xx_init(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int gpio; - int i; - - if (spec->init) - snd_hda_sequence_write(codec, spec->init); - - snd_hda_apply_verbs(codec); - - /* power down adcs initially */ - if (spec->powerdown_adcs) - for (i = 0; i < spec->num_adcs; i++) - snd_hda_codec_write(codec, - spec->adc_nids[i], 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* override some hints */ - stac_store_hints(codec); - - /* set up GPIO */ - gpio = spec->gpio_data; - /* turn on EAPD statically when spec->eapd_switch isn't set. - * otherwise, unsol event will turn it on/off dynamically - */ - if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); - - /* set up pins */ - if (spec->hp_detect) { - /* Enable unsolicited responses on the HP widget */ - for (i = 0; i < cfg->hp_outs; i++) { - hda_nid_t nid = cfg->hp_pins[i]; - enable_pin_detect(codec, nid, STAC_HP_EVENT); - } - if (cfg->line_out_type == AUTO_PIN_LINE_OUT && - cfg->speaker_outs > 0) { - /* enable pin-detect for line-outs as well */ - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - enable_pin_detect(codec, nid, STAC_LO_EVENT); - } - } - - /* force to enable the first line-out; the others are set up - * in unsol_event - */ - stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], - AC_PINCTL_OUT_EN); - /* fake event to set up pins */ - stac_fake_hp_events(codec); - } else { - stac92xx_auto_init_multi_out(codec); - stac92xx_auto_init_hp_out(codec); - for (i = 0; i < cfg->hp_outs; i++) - stac_toggle_power_map(codec, cfg->hp_pins[i], 1); - } - if (spec->auto_mic) { - /* initialize connection to analog input */ - if (spec->dmux_nids) - snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, 0); - if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT)) - stac_issue_unsol_event(codec, spec->ext_mic.pin); - if (enable_pin_detect(codec, spec->dock_mic.pin, - STAC_MIC_EVENT)) - stac_issue_unsol_event(codec, spec->dock_mic.pin); - } - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - int type = cfg->inputs[i].type; - unsigned int pinctl, conf; - if (type == AUTO_PIN_MIC) { - /* for mic pins, force to initialize */ - pinctl = snd_hda_get_default_vref(codec, nid); - pinctl |= AC_PINCTL_IN_EN; - stac92xx_auto_set_pinctl(codec, nid, pinctl); - } else { - pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - /* if PINCTL already set then skip */ - /* Also, if both INPUT and OUTPUT are set, - * it must be a BIOS bug; need to override, too - */ - if (!(pinctl & AC_PINCTL_IN_EN) || - (pinctl & AC_PINCTL_OUT_EN)) { - pinctl &= ~AC_PINCTL_OUT_EN; - pinctl |= AC_PINCTL_IN_EN; - stac92xx_auto_set_pinctl(codec, nid, pinctl); - } - } - conf = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { - if (enable_pin_detect(codec, nid, STAC_INSERT_EVENT)) - stac_issue_unsol_event(codec, nid); - } - } - for (i = 0; i < spec->num_dmics; i++) - stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], - AC_PINCTL_IN_EN); - if (cfg->dig_out_pins[0]) - stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0], - AC_PINCTL_OUT_EN); - if (cfg->dig_in_pin) - stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, - AC_PINCTL_IN_EN); - for (i = 0; i < spec->num_pwrs; i++) { - hda_nid_t nid = spec->pwr_nids[i]; - unsigned int pinctl, def_conf; - - def_conf = snd_hda_codec_get_pincfg(codec, nid); - def_conf = get_defcfg_connect(def_conf); - if (def_conf == AC_JACK_PORT_NONE) { - /* power off unused ports */ - stac_toggle_power_map(codec, nid, 0); - continue; - } - if (def_conf == AC_JACK_PORT_FIXED) { - /* no need for jack detection for fixed pins */ - stac_toggle_power_map(codec, nid, 1); - continue; - } - /* power on when no jack detection is available */ - /* or when the VREF is used for controlling LED */ - if (!spec->hp_detect || - spec->vref_mute_led_nid == nid || - !is_jack_detectable(codec, nid)) { - stac_toggle_power_map(codec, nid, 1); - continue; - } - - if (is_nid_out_jack_pin(cfg, nid)) - continue; /* already has an unsol event */ - - pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - /* outputs are only ports capable of power management - * any attempts on powering down a input port cause the - * referenced VREF to act quirky. - */ - if (pinctl & AC_PINCTL_IN_EN) { - stac_toggle_power_map(codec, nid, 1); - continue; - } - if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) { - stac_issue_unsol_event(codec, nid); - continue; - } - /* none of the above, turn the port OFF */ - stac_toggle_power_map(codec, nid, 0); - } - - /* sync mute LED */ - if (spec->gpio_led) { - if (spec->vmaster_mute.hook) - snd_hda_sync_vmaster_hook(&spec->vmaster_mute); - else /* the very first init call doesn't have vmaster yet */ - stac92xx_update_led_status(codec, false); - } - - /* sync the power-map */ - if (spec->num_pwrs) - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_IDT_SET_POWER_MAP, - spec->power_map_bits); - if (spec->dac_list) - stac92xx_power_down(codec); - return 0; -} - -static void stac92xx_free_kctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (spec->kctls.list) { - struct snd_kcontrol_new *kctl = spec->kctls.list; - int i; - for (i = 0; i < spec->kctls.used; i++) - kfree(kctl[i].name); - } - snd_array_free(&spec->kctls); -} - -static void stac92xx_shutup_pins(struct hda_codec *codec) -{ - unsigned int i, def_conf; - - if (codec->bus->shutdown) - return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); - def_conf = snd_hda_codec_get_pincfg(codec, pin->nid); - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) - snd_hda_set_pin_ctl(codec, pin->nid, 0); - } -} - -static void stac92xx_shutup(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - stac92xx_shutup_pins(codec); - - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); -} - -static void stac92xx_free(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (! spec) - return; - - kfree(spec); - snd_hda_detach_beep_device(codec); -} - -static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, - unsigned int flag) -{ - unsigned int old_ctl, pin_ctl; - - pin_ctl = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - - if (pin_ctl & AC_PINCTL_IN_EN) { - /* - * we need to check the current set-up direction of - * shared input pins since they can be switched via - * "xxx as Output" mixer switch - */ - struct sigmatel_spec *spec = codec->spec; - if (nid == spec->line_switch || nid == spec->mic_switch) - return; - } - - old_ctl = pin_ctl; - /* if setting pin direction bits, clear the current - direction bits first */ - if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) - pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - - pin_ctl |= flag; - if (old_ctl != pin_ctl) - snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl); -} - -static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, - unsigned int flag) -{ - unsigned int pin_ctl = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - if (pin_ctl & flag) - snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag); -} - -static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) -{ - if (!nid) - return 0; - return snd_hda_jack_detect(codec, nid); -} - -static void stac92xx_line_out_detect(struct hda_codec *codec, - int presence) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - - if (cfg->speaker_outs == 0) - return; - - for (i = 0; i < cfg->line_outs; i++) { - if (presence) - break; - presence = get_pin_presence(codec, cfg->line_out_pins[i]); - if (presence) { - unsigned int pinctl; - pinctl = snd_hda_codec_read(codec, - cfg->line_out_pins[i], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pinctl & AC_PINCTL_IN_EN) - presence = 0; /* mic- or line-input */ - } - } - - if (presence) { - /* disable speakers */ - for (i = 0; i < cfg->speaker_outs; i++) - stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], - AC_PINCTL_OUT_EN); - if (spec->eapd_mask && spec->eapd_switch) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); - } else { - /* enable speakers */ - for (i = 0; i < cfg->speaker_outs; i++) - stac92xx_set_pinctl(codec, cfg->speaker_pins[i], - AC_PINCTL_OUT_EN); - if (spec->eapd_mask && spec->eapd_switch) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data | - spec->eapd_mask); - } -} - -/* return non-zero if the hp-pin of the given array index isn't - * a jack-detection target - */ -static int no_hp_sensing(struct sigmatel_spec *spec, int i) -{ - struct auto_pin_cfg *cfg = &spec->autocfg; - - /* ignore sensing of shared line and mic jacks */ - if (cfg->hp_pins[i] == spec->line_switch) - return 1; - if (cfg->hp_pins[i] == spec->mic_switch) - return 1; - /* ignore if the pin is set as line-out */ - if (cfg->hp_pins[i] == spec->hp_switch) - return 1; - return 0; -} - -static void stac92xx_hp_detect(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i, presence; - - presence = 0; - if (spec->gpio_mute) - presence = !(snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); - - for (i = 0; i < cfg->hp_outs; i++) { - if (presence) - break; - if (no_hp_sensing(spec, i)) - continue; - presence = get_pin_presence(codec, cfg->hp_pins[i]); - if (presence) { - unsigned int pinctl; - pinctl = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pinctl & AC_PINCTL_IN_EN) - presence = 0; /* mic- or line-input */ - } - } - - if (presence) { - /* disable lineouts */ - if (spec->hp_switch) - stac92xx_reset_pinctl(codec, spec->hp_switch, - AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->line_outs; i++) - stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], - AC_PINCTL_OUT_EN); - } else { - /* enable lineouts */ - if (spec->hp_switch) - stac92xx_set_pinctl(codec, spec->hp_switch, - AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->line_outs; i++) - stac92xx_set_pinctl(codec, cfg->line_out_pins[i], - AC_PINCTL_OUT_EN); - } - stac92xx_line_out_detect(codec, presence); - /* toggle hp outs */ - for (i = 0; i < cfg->hp_outs; i++) { - unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; - if (no_hp_sensing(spec, i)) - continue; - if (1 /*presence*/) - stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); -#if 0 /* FIXME */ -/* Resetting the pinctl like below may lead to (a sort of) regressions - * on some devices since they use the HP pin actually for line/speaker - * outs although the default pin config shows a different pin (that is - * wrong and useless). - * - * So, it's basically a problem of default pin configs, likely a BIOS issue. - * But, disabling the code below just works around it, and I'm too tired of - * bug reports with such devices... - */ - else - stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val); -#endif /* FIXME */ - } -} - -static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, - int enable) -{ - struct sigmatel_spec *spec = codec->spec; - unsigned int idx, val; - - for (idx = 0; idx < spec->num_pwrs; idx++) { - if (spec->pwr_nids[idx] == nid) - break; - } - if (idx >= spec->num_pwrs) - return; - - idx = 1 << idx; - - val = spec->power_map_bits; - if (enable) - val &= ~idx; - else - val |= idx; - - /* power down unused output ports */ - if (val != spec->power_map_bits) { - spec->power_map_bits = val; - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_IDT_SET_POWER_MAP, val); - } -} - -static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid) -{ - stac_toggle_power_map(codec, nid, get_pin_presence(codec, nid)); -} + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, + "Dell Inspiron", STAC_9205_DELL_M44), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, + "Dell Vostro 1500", STAC_9205_DELL_M42), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229, + "Dell Vostro 1700", STAC_9205_DELL_M42), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), + SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), + {} /* terminator */ +}; -/* get the pin connection (fixed, none, etc) */ -static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int cfg; + int err; - cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); - return get_defcfg_connect(cfg); -} + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + if (err < 0) + return err; -static int stac92xx_connected_ports(struct hda_codec *codec, - const hda_nid_t *nids, int num_nids) -{ - struct sigmatel_spec *spec = codec->spec; - int idx, num; - unsigned int def_conf; - - for (num = 0; num < num_nids; num++) { - for (idx = 0; idx < spec->num_pins; idx++) - if (spec->pin_nids[idx] == nids[num]) - break; - if (idx >= spec->num_pins) - break; - def_conf = stac_get_defcfg_connect(codec, idx); - if (def_conf == AC_JACK_PORT_NONE) - break; - } - return num; -} + /* add hooks */ + spec->gen.pcm_playback_hook = stac_playback_pcm_hook; + spec->gen.pcm_capture_hook = stac_capture_pcm_hook; -static void stac92xx_mic_detect(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_mic_route *mic; + spec->gen.automute_hook = stac_update_outputs; + spec->gen.hp_automute_hook = stac_hp_automute; + spec->gen.line_automute_hook = stac_line_automute; - if (get_pin_presence(codec, spec->ext_mic.pin)) - mic = &spec->ext_mic; - else if (get_pin_presence(codec, spec->dock_mic.pin)) - mic = &spec->dock_mic; - else - mic = &spec->int_mic; - if (mic->dmux_idx >= 0) - snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, - mic->dmux_idx); - if (mic->mux_idx >= 0) - snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, - mic->mux_idx); -} + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + if (err < 0) + return err; -static void handle_unsol_event(struct hda_codec *codec, - struct hda_jack_tbl *event) -{ - struct sigmatel_spec *spec = codec->spec; - int data; + /* minimum value is actually mute */ + spec->gen.vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; - switch (event->action) { - case STAC_HP_EVENT: - case STAC_LO_EVENT: - stac92xx_hp_detect(codec); - break; - case STAC_MIC_EVENT: - stac92xx_mic_detect(codec); - break; + /* setup analog beep controls */ + if (spec->anabeep_nid > 0) { + err = stac_auto_create_beep_ctls(codec, + spec->anabeep_nid); + if (err < 0) + return err; } - switch (event->action) { - case STAC_HP_EVENT: - case STAC_LO_EVENT: - case STAC_MIC_EVENT: - case STAC_INSERT_EVENT: - case STAC_PWR_EVENT: - if (spec->num_pwrs > 0) - stac92xx_pin_sense(codec, event->nid); - - switch (codec->subsystem_id) { - case 0x103c308f: - if (event->nid == 0xb) { - int pin = AC_PINCTL_IN_EN; - - if (get_pin_presence(codec, 0xa) - && get_pin_presence(codec, 0xb)) - pin |= AC_PINCTL_VREF_80; - if (!get_pin_presence(codec, 0xb)) - pin |= AC_PINCTL_VREF_80; - - /* toggle VREF state based on mic + hp pin - * status - */ - stac92xx_auto_set_pinctl(codec, 0x0a, pin); + /* setup digital beep controls and input device */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (spec->digbeep_nid > 0) { + hda_nid_t nid = spec->digbeep_nid; + unsigned int caps; + + err = stac_auto_create_beep_ctls(codec, nid); + if (err < 0) + return err; + err = snd_hda_attach_beep_device(codec, nid); + if (err < 0) + return err; + if (codec->beep) { + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = spec->linear_tone_beep; + /* if no beep switch is available, make its own one */ + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + if (!(caps & AC_AMPCAP_MUTE)) { + err = stac_beep_switch_ctl(codec); + if (err < 0) + return err; } } - break; - case STAC_VREF_EVENT: - data = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - /* toggle VREF state based on GPIOx status */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, - !!(data & (1 << event->private_data))); - break; } -} +#endif -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) -{ - struct hda_jack_tbl *event = snd_hda_jack_tbl_get(codec, nid); - if (!event) - return; - handle_unsol_event(codec, event); + if (spec->gpio_led) + spec->gen.vmaster_mute.hook = stac_vmaster_hook; + + if (spec->aloopback_ctl && + snd_hda_get_bool_hint(codec, "loopback") == 1) { + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, spec->aloopback_ctl)) + return -ENOMEM; + } + + stac_init_power_map(codec); + + return 0; } -static void set_hp_led_gpio(struct hda_codec *codec) + +static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; unsigned int gpio; + int i; - if (spec->gpio_led) - return; + /* override some hints */ + stac_store_hints(codec); - gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); - gpio &= AC_GPIO_IO_COUNT; - if (gpio > 3) - spec->gpio_led = 0x08; /* GPIO 3 */ - else - spec->gpio_led = 0x01; /* GPIO 0 */ -} + /* set up GPIO */ + gpio = spec->gpio_data; + /* turn on EAPD statically when spec->eapd_switch isn't set. + * otherwise, unsol event will turn it on/off dynamically + */ + if (!spec->eapd_switch) + gpio |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); -/* - * This method searches for the mute LED GPIO configuration - * provided as OEM string in SMBIOS. The format of that string - * is HP_Mute_LED_P_G or HP_Mute_LED_P - * where P can be 0 or 1 and defines mute LED GPIO control state (low/high) - * that corresponds to the NOT muted state of the master volume - * and G is the index of the GPIO to use as the mute LED control (0..9) - * If _G portion is missing it is assigned based on the codec ID - * - * So, HP B-series like systems may have HP_Mute_LED_0 (current models) - * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings - * - * - * The dv-series laptops don't seem to have the HP_Mute_LED* strings in - * SMBIOS - at least the ones I have seen do not have them - which include - * my own system (HP Pavilion dv6-1110ax) and my cousin's - * HP Pavilion dv9500t CTO. - * Need more information on whether it is true across the entire series. - * -- kunal - */ -static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) -{ - struct sigmatel_spec *spec = codec->spec; - const struct dmi_device *dev = NULL; + snd_hda_gen_init(codec); - if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { - get_int_hint(codec, "gpio_led_polarity", - &spec->gpio_led_polarity); - return 1; - } + /* sync the power-map */ + if (spec->num_pwrs) + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, + spec->power_map_bits); - while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { - if (sscanf(dev->name, "HP_Mute_LED_%d_%x", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { - unsigned int max_gpio; - max_gpio = snd_hda_param_read(codec, codec->afg, - AC_PAR_GPIO_CAP); - max_gpio &= AC_GPIO_IO_COUNT; - if (spec->gpio_led < max_gpio) - spec->gpio_led = 1 << spec->gpio_led; - else - spec->vref_mute_led_nid = spec->gpio_led; - return 1; - } - if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - set_hp_led_gpio(codec); - return 1; - } - /* BIOS bug: unfilled OEM string */ - if (strstr(dev->name, "HP_Mute_LED_P_G")) { - set_hp_led_gpio(codec); - if (default_polarity >= 0) - spec->gpio_led_polarity = default_polarity; - else - spec->gpio_led_polarity = 1; - return 1; + /* power down inactive ADCs */ + if (spec->powerdown_adcs) { + for (i = 0; i < spec->gen.num_all_adcs; i++) { + if (spec->active_adcs & (1 << i)) + continue; + snd_hda_codec_write(codec, spec->gen.all_adcs[i], 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); } } - /* - * Fallback case - if we don't find the DMI strings, - * we statically set the GPIO - if not a B-series system - * and default polarity is provided - */ - if (!hp_blike_system(codec->subsystem_id) && - (default_polarity == 0 || default_polarity == 1)) { - set_hp_led_gpio(codec); - spec->gpio_led_polarity = default_polarity; - return 1; + /* power down unused DACs */ + for (i = 0; i < spec->gen.num_all_dacs; i++) { + if (!snd_hda_get_nid_path(codec, spec->gen.all_dacs[i], 0)) + snd_hda_codec_write(codec, spec->gen.all_dacs[i], 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); } + return 0; } -static int hp_blike_system(u32 subsystem_id) +static void stac_shutup(struct hda_codec *codec) { - switch (subsystem_id) { - case 0x103c1520: - case 0x103c1521: - case 0x103c1523: - case 0x103c1524: - case 0x103c1525: - case 0x103c1722: - case 0x103c1723: - case 0x103c1724: - case 0x103c1725: - case 0x103c1726: - case 0x103c1727: - case 0x103c1728: - case 0x103c1729: - case 0x103c172a: - case 0x103c172b: - case 0x103c307e: - case 0x103c307f: - case 0x103c3080: - case 0x103c3081: - case 0x103c7007: - case 0x103c7008: - return 1; - } - return 0; + struct sigmatel_spec *spec = codec->spec; + + snd_hda_shutup_pins(codec); + + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); +} + +static void stac_free(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!spec) + return; + + snd_hda_gen_spec_free(&spec->gen); + kfree(spec); + snd_hda_detach_beep_device(codec); } #ifdef CONFIG_PROC_FS @@ -6341,101 +3585,69 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #endif #ifdef CONFIG_PM -static int stac92xx_resume(struct hda_codec *codec) +static int stac_resume(struct hda_codec *codec) { - stac92xx_init(codec); + codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); - /* fake event to set up pins again to override cached values */ - stac_fake_hp_events(codec); return 0; } -static int stac92xx_suspend(struct hda_codec *codec) -{ - stac92xx_shutup(codec); - return 0; -} - -static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - unsigned int afg_power_state = power_state; - struct sigmatel_spec *spec = codec->spec; - - if (power_state == AC_PWRST_D3) { - if (spec->vref_mute_led_nid) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state - */ - afg_power_state = AC_PWRST_D1; - } - /* this delay seems necessary to avoid click noise at power-down */ - msleep(100); - } - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - afg_power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state, true); -} -#else -#define stac92xx_suspend NULL -#define stac92xx_resume NULL -#define stac92xx_set_power_state NULL -#endif /* CONFIG_PM */ - -/* update mute-LED accoring to the master switch */ -static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) -{ - struct sigmatel_spec *spec = codec->spec; - int muted = !enabled; - - if (!spec->gpio_led) - return; - - /* LED state is inverted on these systems */ - if (spec->gpio_led_polarity) - muted = !muted; - - if (!spec->vref_mute_led_nid) { - if (muted) - spec->gpio_data |= spec->gpio_led; - else - spec->gpio_data &= ~spec->gpio_led; - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } else { - spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; - stac_vrefout_set(codec, spec->vref_mute_led_nid, - spec->vref_led); +static int stac_suspend(struct hda_codec *codec) +{ + stac_shutup(codec); + return 0; +} + +static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + unsigned int afg_power_state = power_state; + struct sigmatel_spec *spec = codec->spec; + + if (power_state == AC_PWRST_D3) { + if (spec->vref_mute_led_nid) { + /* with vref-out pin used for mute led control + * codec AFG is prevented from D3 state + */ + afg_power_state = AC_PWRST_D1; + } + /* this delay seems necessary to avoid click noise at power-down */ + msleep(100); } + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + afg_power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } +#else +#define stac_suspend NULL +#define stac_resume NULL +#define stac_set_power_state NULL +#endif /* CONFIG_PM */ -static const struct hda_codec_ops stac92xx_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, +static const struct hda_codec_ops stac_patch_ops = { + .build_controls = snd_hda_gen_build_controls, + .build_pcms = snd_hda_gen_build_pcms, + .init = stac_init, + .free = stac_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM - .suspend = stac92xx_suspend, - .resume = stac92xx_resume, + .suspend = stac_suspend, + .resume = stac_resume, #endif - .reboot_notify = stac92xx_shutup, + .reboot_notify = stac_shutup, }; -static int alloc_stac_spec(struct hda_codec *codec, int num_pins, - const hda_nid_t *pin_nids) +static int alloc_stac_spec(struct hda_codec *codec) { struct sigmatel_spec *spec; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; codec->no_trigger_sense = 1; /* seems common with STAC/IDT codecs */ - spec->num_pins = num_pins; - spec->pin_nids = pin_nids; - snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32); return 0; } @@ -6444,40 +3656,28 @@ static int patch_stac9200(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac9200_pin_nids), - stac9200_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac9200_models, stac9200_fixup_tbl, - stac9200_fixups); + codec->patch_ops = stac_patch_ops; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = stac9200_dac_nids; - spec->adc_nids = stac9200_adc_nids; - spec->mux_nids = stac9200_mux_nids; - spec->num_muxes = 1; - spec->num_dmics = 0; - spec->num_adcs = 1; - spec->num_pwrs = 0; snd_hda_add_verbs(codec, stac9200_eapd_init); - spec->mixer = stac9200_mixer; - + snd_hda_pick_fixup(codec, stac9200_models, stac9200_fixup_tbl, + stac9200_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac9200_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6488,56 +3688,28 @@ static int patch_stac925x(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac925x_pin_nids), - stac925x_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac925x_models, stac925x_fixup_tbl, - stac925x_fixups); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = stac925x_dac_nids; - spec->adc_nids = stac925x_adc_nids; - spec->mux_nids = stac925x_mux_nids; - spec->num_muxes = 1; - spec->num_adcs = 1; - spec->num_pwrs = 0; - switch (codec->vendor_id) { - case 0x83847632: /* STAC9202 */ - case 0x83847633: /* STAC9202D */ - case 0x83847636: /* STAC9251 */ - case 0x83847637: /* STAC9251D */ - spec->num_dmics = STAC925X_NUM_DMICS; - spec->dmic_nids = stac925x_dmic_nids; - spec->num_dmuxes = ARRAY_SIZE(stac925x_dmux_nids); - spec->dmux_nids = stac925x_dmux_nids; - break; - default: - spec->num_dmics = 0; - break; - } + codec->patch_ops = stac_patch_ops; snd_hda_add_verbs(codec, stac925x_core_init); - spec->mixer = stac925x_mixer; - spec->num_caps = 1; - spec->capvols = stac925x_capvols; - spec->capsws = stac925x_capsws; + snd_hda_pick_fixup(codec, stac925x_models, stac925x_fixup_tbl, + stac925x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6546,86 +3718,66 @@ static int patch_stac925x(struct hda_codec *codec) static int patch_stac92hd73xx(struct hda_codec *codec) { struct sigmatel_spec *spec; - hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; int err; int num_dacs; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac92hd73xx_pin_nids), - stac92hd73xx_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 0; - codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; - - snd_hda_pick_fixup(codec, stac92hd73xx_models, stac92hd73xx_fixup_tbl, - stac92hd73xx_fixups); - - num_dacs = snd_hda_get_connections(codec, 0x0a, - conn, STAC92HD73_DAC_COUNT + 2) - 1; + num_dacs = snd_hda_get_num_conns(codec, 0x0a) - 1; if (num_dacs < 3 || num_dacs > 5) { printk(KERN_WARNING "hda_codec: Could not determine " "number of channels defaulting to DAC count\n"); - num_dacs = STAC92HD73_DAC_COUNT; + num_dacs = 5; } switch (num_dacs) { case 0x3: /* 6 Channel */ - spec->aloopback_ctl = stac92hd73xx_6ch_loopback; + spec->aloopback_ctl = &stac92hd73xx_6ch_loopback; break; case 0x4: /* 8 Channel */ - spec->aloopback_ctl = stac92hd73xx_8ch_loopback; + spec->aloopback_ctl = &stac92hd73xx_8ch_loopback; break; case 0x5: /* 10 Channel */ - spec->aloopback_ctl = stac92hd73xx_10ch_loopback; + spec->aloopback_ctl = &stac92hd73xx_10ch_loopback; break; } - spec->multiout.dac_nids = spec->dac_nids; spec->aloopback_mask = 0x01; spec->aloopback_shift = 8; spec->digbeep_nid = 0x1c; - spec->mux_nids = stac92hd73xx_mux_nids; - spec->adc_nids = stac92hd73xx_adc_nids; - spec->dmic_nids = stac92hd73xx_dmic_nids; - spec->dmux_nids = stac92hd73xx_dmux_nids; - spec->smux_nids = stac92hd73xx_smux_nids; - - spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - - spec->num_caps = STAC92HD73XX_NUM_CAPS; - spec->capvols = stac92hd73xx_capvols; - spec->capsws = stac92hd73xx_capsws; /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; - spec->num_dmics = STAC92HD73XX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); spec->eapd_switch = 1; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; + spec->gen.own_eapd_ctl = 1; + + codec->patch_ops = stac_patch_ops; + + snd_hda_pick_fixup(codec, stac92hd73xx_models, stac92hd73xx_fixup_tbl, + stac92hd73xx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); if (!spec->volknob_init) snd_hda_add_verbs(codec, stac92hd73xx_core_init); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac92hd7x_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -6633,145 +3785,6 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return 0; } -static int hp_bnb2011_with_dock(struct hda_codec *codec) -{ - if (codec->vendor_id != 0x111d7605 && - codec->vendor_id != 0x111d76d1) - return 0; - - switch (codec->subsystem_id) { - case 0x103c1618: - case 0x103c1619: - case 0x103c161a: - case 0x103c161b: - case 0x103c161c: - case 0x103c161d: - case 0x103c161e: - case 0x103c161f: - - case 0x103c162a: - case 0x103c162b: - - case 0x103c1630: - case 0x103c1631: - - case 0x103c1633: - case 0x103c1634: - case 0x103c1635: - - case 0x103c3587: - case 0x103c3588: - case 0x103c3589: - case 0x103c358a: - - case 0x103c3667: - case 0x103c3668: - case 0x103c3669: - - return 1; - } - return 0; -} - -static void stac92hd8x_add_pin(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int i; - - spec->auto_pin_nids[spec->auto_pin_cnt] = nid; - spec->auto_pin_cnt++; - - if (get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) { - for (i = 0; i < ARRAY_SIZE(stac92hd83xxx_dmic_nids); i++) { - if (nid == stac92hd83xxx_dmic_nids[i]) { - spec->auto_dmic_nids[spec->auto_dmic_cnt] = nid; - spec->auto_dmic_cnt++; - } - } - } -} - -static void stac92hd8x_add_adc(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - - spec->auto_adc_nids[spec->auto_adc_cnt] = nid; - spec->auto_adc_cnt++; -} - -static void stac92hd8x_add_mux(struct hda_codec *codec, hda_nid_t nid) -{ - int i, j; - struct sigmatel_spec *spec = codec->spec; - - for (i = 0; i < spec->auto_adc_cnt; i++) { - if (get_connection_index(codec, - spec->auto_adc_nids[i], nid) >= 0) { - /* mux and volume for adc_nids[i] */ - if (!spec->auto_mux_nids[i]) { - spec->auto_mux_nids[i] = nid; - /* 92hd codecs capture volume is in mux */ - spec->auto_capvols[i] = HDA_COMPOSE_AMP_VAL(nid, - 3, 0, HDA_OUTPUT); - } - for (j = 0; j < spec->auto_dmic_cnt; j++) { - if (get_connection_index(codec, nid, - spec->auto_dmic_nids[j]) >= 0) { - /* dmux for adc_nids[i] */ - if (!spec->auto_dmux_nids[i]) - spec->auto_dmux_nids[i] = nid; - break; - } - } - break; - } - } -} - -static void stac92hd8x_fill_auto_spec(struct hda_codec *codec) -{ - hda_nid_t nid, end_nid; - unsigned int wid_caps, wid_type; - struct sigmatel_spec *spec = codec->spec; - - end_nid = codec->start_nid + codec->num_nodes; - - for (nid = codec->start_nid; nid < end_nid; nid++) { - wid_caps = get_wcaps(codec, nid); - wid_type = get_wcaps_type(wid_caps); - - if (wid_type == AC_WID_PIN) - stac92hd8x_add_pin(codec, nid); - - if (wid_type == AC_WID_AUD_IN && !(wid_caps & AC_WCAP_DIGITAL)) - stac92hd8x_add_adc(codec, nid); - } - - for (nid = codec->start_nid; nid < end_nid; nid++) { - wid_caps = get_wcaps(codec, nid); - wid_type = get_wcaps_type(wid_caps); - - if (wid_type == AC_WID_AUD_SEL) - stac92hd8x_add_mux(codec, nid); - } - - spec->pin_nids = spec->auto_pin_nids; - spec->num_pins = spec->auto_pin_cnt; - spec->adc_nids = spec->auto_adc_nids; - spec->num_adcs = spec->auto_adc_cnt; - spec->capvols = spec->auto_capvols; - spec->capsws = spec->auto_capvols; - spec->num_caps = spec->auto_adc_cnt; - spec->mux_nids = spec->auto_mux_nids; - spec->num_muxes = spec->auto_adc_cnt; - spec->dmux_nids = spec->auto_dmux_nids; - spec->num_dmuxes = spec->auto_adc_cnt; - spec->dmic_nids = spec->auto_dmic_nids; - spec->num_dmics = spec->auto_dmic_cnt; -} - static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -6783,7 +3796,7 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_data |= spec->gpio_led; } else { codec->patch_ops.set_power_state = - stac92xx_set_power_state; + stac_set_power_state; } } @@ -6792,6 +3805,8 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_dir |= spec->mic_mute_led_gpio; spec->mic_mute_led_on = true; spec->gpio_data |= spec->mic_mute_led_gpio; + + spec->gen.capture_switch_hook = stac_capture_led_hook; } } @@ -6800,36 +3815,34 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, 0, NULL); /* pins filled later */ + err = alloc_stac_spec(codec); if (err < 0) return err; codec->epss = 0; /* longer delay needed for D3 */ - stac92hd8x_fill_auto_spec(codec); spec = codec->spec; spec->linear_tone_beep = 0; - codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; + spec->gen.own_eapd_ctl = 1; + spec->digbeep_nid = 0x21; spec->pwr_nids = stac92hd83xxx_pwr_nids; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); - spec->multiout.dac_nids = spec->dac_nids; + spec->default_polarity = -1; /* no default cfg */ - snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, - stac92hd83xxx_fixups); + codec->patch_ops = stac_patch_ops; snd_hda_add_verbs(codec, stac92hd83xxx_core_init); - spec->default_polarity = -1; /* no default cfg */ - - codec->patch_ops = stac92xx_patch_ops; + snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, + stac92hd83xxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); stac_setup_gpio(codec); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } @@ -6840,133 +3853,31 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return 0; } -static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, - hda_nid_t dig0pin) -{ - struct sigmatel_spec *spec = codec->spec; - int idx; - - for (idx = 0; idx < spec->num_pins; idx++) - if (spec->pin_nids[idx] == dig0pin) - break; - if ((idx + 2) >= spec->num_pins) - return 0; - - /* dig1pin case */ - if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE) - return 2; - - /* dig0pin + dig2pin case */ - if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE) - return 2; - if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE) - return 1; - else - return 0; -} - -/* HP dv7 bass switch - GPIO5 */ -#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info -static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); - return 0; -} - -static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int gpio_data; - - gpio_data = (spec->gpio_data & ~0x20) | - (ucontrol->value.integer.value[0] ? 0x20 : 0); - if (gpio_data == spec->gpio_data) - return 0; - spec->gpio_data = gpio_data; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); - return 1; -} - -static const struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = stac_hp_bass_gpio_info, - .get = stac_hp_bass_gpio_get, - .put = stac_hp_bass_gpio_put, -}; - -static int stac_add_hp_bass_switch(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!stac_control_new(spec, &stac_hp_bass_sw_ctrl, - "Bass Speaker Playback Switch", 0)) - return -ENOMEM; - - spec->gpio_mask |= 0x20; - spec->gpio_dir |= 0x20; - spec->gpio_data |= 0x20; - return 0; -} - static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; const struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err; - err = alloc_stac_spec(codec, STAC92HD71BXX_NUM_PINS, - stac92hd71bxx_pin_nids_4port); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 0; - codec->patch_ops = stac92xx_patch_ops; - switch (codec->vendor_id) { - case 0x111d76b6: - case 0x111d76b7: - break; - case 0x111d7603: - case 0x111d7608: - /* On 92HD75Bx 0x27 isn't a pin nid */ - spec->num_pins--; - /* fallthrough */ - default: - spec->pin_nids = stac92hd71bxx_pin_nids_6port; - } - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac92hd71bxx_models, stac92hd71bxx_fixup_tbl, - stac92hd71bxx_fixups); + codec->patch_ops = stac_patch_ops; /* GPIO0 = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; spec->gpio_data = 0x01; - spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nids = stac92hd71bxx_dmux_nids; - - spec->num_caps = STAC92HD71BXX_NUM_CAPS; - spec->capvols = stac92hd71bxx_capvols; - spec->capsws = stac92hd71bxx_capsws; - switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: unmute_init++; - /* fallthru */ - case 0x111d76b4: /* 6 Port without Analog Mixer */ - case 0x111d76b5: - codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_nids, - STAC92HD71BXX_NUM_DMICS); break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -6977,21 +3888,11 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); - spec->dmic_nids = stac92hd71bxx_dmic_5port_nids; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_5port_nids, - STAC92HD71BXX_NUM_DMICS - 1); break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ - /* fallthru */ - default: - codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_nids, - STAC92HD71BXX_NUM_DMICS); break; } @@ -7001,32 +3902,24 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); - spec->aloopback_ctl = stac92hd71bxx_loopback; + spec->aloopback_ctl = &stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; spec->powerdown_adcs = 1; spec->digbeep_nid = 0x26; - spec->mux_nids = stac92hd71bxx_mux_nids; - spec->adc_nids = stac92hd71bxx_adc_nids; - spec->smux_nids = stac92hd71bxx_smux_nids; + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); - - spec->multiout.dac_nids = spec->dac_nids; - spec->default_polarity = 1; - + snd_hda_pick_fixup(codec, stac92hd71bxx_models, stac92hd71bxx_fixup_tbl, + stac92hd71bxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); stac_setup_gpio(codec); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } @@ -7042,42 +3935,18 @@ static int patch_stac922x(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac922x_pin_nids), - stac922x_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac922x_models, stac922x_fixup_tbl, - stac922x_fixups); - - spec->adc_nids = stac922x_adc_nids; - spec->mux_nids = stac922x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac922x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac922x_adc_nids); - spec->num_dmics = 0; - spec->num_pwrs = 0; + codec->patch_ops = stac_patch_ops; - spec->num_caps = STAC922X_NUM_CAPS; - spec->capvols = stac922x_capvols; - spec->capsws = stac922x_capsws; - - spec->multiout.dac_nids = spec->dac_nids; - snd_hda_add_verbs(codec, stac922x_core_init); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - err = stac92xx_parse_auto_config(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - - codec->patch_ops = stac92xx_patch_ops; - /* Fix Mux capture level; max to 2 */ snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, (0 << AC_AMPCAP_OFFSET_SHIFT) | @@ -7085,6 +3954,16 @@ static int patch_stac922x(struct hda_codec *codec) (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | (0 << AC_AMPCAP_MUTE_SHIFT)); + snd_hda_pick_fixup(codec, stac922x_models, stac922x_fixup_tbl, + stac922x_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + err = stac_parse_auto_config(codec); + if (err < 0) { + stac_free(codec); + return err; + } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -7095,58 +3974,40 @@ static int patch_stac927x(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac927x_pin_nids), - stac927x_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; - codec->slave_dig_outs = stac927x_slave_dig_outs; - - snd_hda_pick_fixup(codec, stac927x_models, stac927x_fixup_tbl, - stac927x_fixups); + spec->gen.own_eapd_ctl = 1; spec->digbeep_nid = 0x23; - spec->adc_nids = stac927x_adc_nids; - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->smux_nids = stac927x_smux_nids; - spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids); - spec->spdif_labels = stac927x_spdif_labels; - spec->dac_list = stac927x_dac_nids; - spec->multiout.dac_nids = spec->dac_nids; /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = 0x01; spec->gpio_dir = spec->gpio_data = 0x01; - spec->num_dmics = 0; - - spec->num_caps = STAC927X_NUM_CAPS; - spec->capvols = stac927x_capvols; - spec->capsws = stac927x_capsws; - - spec->num_pwrs = 0; - spec->aloopback_ctl = stac927x_loopback; + spec->aloopback_ctl = &stac927x_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->eapd_switch = 1; + codec->patch_ops = stac_patch_ops; + + snd_hda_pick_fixup(codec, stac927x_models, stac927x_fixup_tbl, + stac927x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); if (!spec->volknob_init) snd_hda_add_verbs(codec, stac927x_core_init); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac927x_proc_hook; /* @@ -7171,40 +4032,21 @@ static int patch_stac9205(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac9205_pin_nids), - stac9205_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; - - snd_hda_pick_fixup(codec, stac9205_models, stac9205_fixup_tbl, - stac9205_fixups); + spec->gen.own_eapd_ctl = 1; spec->digbeep_nid = 0x23; - spec->adc_nids = stac9205_adc_nids; - spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids); - spec->mux_nids = stac9205_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); - spec->smux_nids = stac9205_smux_nids; - spec->num_smuxes = ARRAY_SIZE(stac9205_smux_nids); - spec->dmic_nids = stac9205_dmic_nids; - spec->num_dmics = STAC9205_NUM_DMICS; - spec->dmux_nids = stac9205_dmux_nids; - spec->num_dmuxes = ARRAY_SIZE(stac9205_dmux_nids); - spec->num_pwrs = 0; snd_hda_add_verbs(codec, stac9205_core_init); - spec->aloopback_ctl = stac9205_loopback; - - spec->num_caps = STAC9205_NUM_CAPS; - spec->capvols = stac9205_capvols; - spec->capsws = stac9205_capsws; + spec->aloopback_ctl = &stac9205_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - spec->multiout.dac_nids = spec->dac_nids; /* GPIO0 High = EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; @@ -7213,16 +4055,18 @@ static int patch_stac9205(struct hda_codec *codec) /* Turn on/off EAPD per HP plugging */ spec->eapd_switch = 1; + codec->patch_ops = stac_patch_ops; + + snd_hda_pick_fixup(codec, stac9205_models, stac9205_fixup_tbl, + stac9205_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac9205_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7240,24 +4084,6 @@ static const struct hda_verb stac9872_core_init[] = { {} }; -static const hda_nid_t stac9872_pin_nids[] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, - 0x11, 0x13, 0x14, -}; - -static const hda_nid_t stac9872_adc_nids[] = { - 0x8 /*,0x6*/ -}; - -static const hda_nid_t stac9872_mux_nids[] = { - 0x15 -}; - -static const unsigned long stac9872_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), -}; -#define stac9872_capsws stac9872_capvols - static const struct hda_pintbl stac9872_vaio_pin_configs[] = { { 0x0a, 0x03211020 }, { 0x0b, 0x411111f0 }, @@ -7294,36 +4120,27 @@ static int patch_stac9872(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac9872_pin_nids), - stac9872_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac9872_models, stac9872_fixup_tbl, - stac9872_fixups); + codec->patch_ops = stac_patch_ops; - spec->multiout.dac_nids = spec->dac_nids; - spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); - spec->adc_nids = stac9872_adc_nids; - spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); - spec->mux_nids = stac9872_mux_nids; - spec->num_caps = 1; - spec->capvols = stac9872_capvols; - spec->capsws = stac9872_capsws; snd_hda_add_verbs(codec, stac9872_core_init); + snd_hda_pick_fixup(codec, stac9872_models, stac9872_fixup_tbl, + stac9872_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return -EINVAL; } - spec->input_mux = &spec->private_imux; - codec->patch_ops = stac92xx_patch_ops; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); -- 1.8.1

1 0

[alsa-devel] [PATCH RFC 0/9] Final bits for HD-audio generic parser
by Takashi Iwai 17 Jan '13

17 Jan '13

Hi, this is the final bits for improving / fixing the HD-audio generic parser for migrating patch_sigmatel.c. After these, patch_sigmatel.c is converted to use the generic parser, and all done! As usual, the patches are found in sound-unstable git tree test/hda-gen-parser branch (it's already merged into master and test/hda-migrate branch, too). Takashi

1 9

[alsa-devel] [PATCH 01/19] ALSA: add Yamaha YMU831 codec support
by Yoichi Yuasa 17 Jan '13

17 Jan '13

Signed-off-by: Yoichi Yuasa <yuasa(a)linux-mips.org> --- sound/soc/codecs/Kconfig | 6 ++++++ sound/soc/codecs/Makefile | 1 + sound/soc/codecs/ymu831/Makefile | 3 +++ 3 files changed, 10 insertions(+) create mode 100644 sound/soc/codecs/ymu831/Makefile diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..bb405b87 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_YMU831 if SPI_MASTER help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -494,6 +495,11 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate +config SND_SOC_YMU831 + tristate + select SND_HWDEP + + # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..1d31783 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -236,6 +236,7 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_YMU831) += ymu831/ # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/ymu831/Makefile b/sound/soc/codecs/ymu831/Makefile new file mode 100644 index 0000000..5b3f66d --- /dev/null +++ b/sound/soc/codecs/ymu831/Makefile @@ -0,0 +1,3 @@ +snd-soc-ymu831-objs := \ + +obj-$(CONFIG_SND_SOC_YMU831) += snd-soc-ymu831.o -- 1.7.9.5

3 41

[alsa-devel] [PATCH v2 0/3] Tegra AC97 v2
by Lucas Stach 17 Jan '13

17 Jan '13

v2 of the Tegra AC97 series. Unmerged ASoC related patches only. The machine driver now instantiates the codec driver by itself, without usage of DT. Other fixups for review feedback also included. I'll post a patch for the AC97 clock once Prashant patches land in a branch. Lucas Stach (3): ASoC: tegra: add ac97 host driver ASoC: tegra: add ac97 host controller to device tree ASoC: tegra: add tegra machine driver using wm9712 codec .../bindings/sound/nvidia,tegra-audio-wm9712.txt | 51 +++ .../bindings/sound/nvidia,tegra20-ac97.txt | 22 + arch/arm/boot/dts/tegra20.dtsi | 9 + sound/soc/tegra/Kconfig | 19 + sound/soc/tegra/Makefile | 4 + sound/soc/tegra/tegra20_ac97.c | 480 +++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.h | 95 ++++ sound/soc/tegra/tegra_wm9712.c | 179 ++++++++ 8 files changed, 859 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt create mode 100644 sound/soc/tegra/tegra20_ac97.c create mode 100644 sound/soc/tegra/tegra20_ac97.h create mode 100644 sound/soc/tegra/tegra_wm9712.c -- 1.8.0.1

3 13

[alsa-devel] [PATCH alsa-utils] speaker-test: increase the maximum supported rate to 384000
by Antonio Ospite 17 Jan '13

17 Jan '13

There are some devices around supporting this sample rate so let's make speaker-test capable to deal with them. Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- speaker-test/speaker-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 3c89c38..de42a0c 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -999,7 +999,7 @@ int main(int argc, char *argv[]) { case 'r': rate = atoi(optarg); rate = rate < 4000 ? 4000 : rate; - rate = rate > 196000 ? 196000 : rate; + rate = rate > 384000 ? 384000 : rate; break; case 'c': channels = atoi(optarg); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: Use actual rather than desired BCLK when calculating LRCLK
by Mark Brown 17 Jan '13

17 Jan '13

Otherwise we'll get the wrong LRCLK if we need to pick a higher BCLK than is required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c3592db..ef62c43 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -685,7 +685,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, } sr_val = i; - lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + lrclk = rates[bclk] / params_rate(params); arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v3 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
by mengdong.lin＠intel.com 17 Jan '13

17 Jan '13

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI or Display Port. So this patch adds fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the pins' connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..59abe73 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1687,6 +1687,30 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + /* override 3 pins connection list */ + snd_hda_override_conn_list(codec, 0x05, 3, list); + snd_hda_override_conn_list(codec, 0x06, 3, list); + snd_hda_override_conn_list(codec, 0x07, 3, list); +} + + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1720,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + + if (codec->vendor_id == 0x80862807) + intel_haswell_fixup_connect_list(codec); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ...and three more fixes for the new generic parser
by David Henningsson 16 Jan '13

16 Jan '13

These regressions were found when running the test tool over even more alsa-infos. Btw, would you mind updating the hda-migrate branch too? It's useful to have your latest fixes from the hda-gen-parser branch when I'm running pre-release QA testing. David Henningsson (3): ALSA: hda - do not add non-existing Mic boost controls ALSA: hda - force different capture controls if amp caps differ ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs sound/pci/hda/hda_generic.c | 31 ++++++++++++++++++++++++++----- 1 file changed, 26 insertions(+), 5 deletions(-) -- 1.7.9.5

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[alsa-devel] alsa-util fix (Re: [PATCH] alsa-lib: Fix build with automake-1.13)
by Marko Lindqvist 16 Jan '13

16 Jan '13

On 4 January 2013 01:40, Marko Lindqvist <cazfi74(a)gmail.com> wrote: > Automake-1.13 removed long obsolete AM_CONFIG_HEADER completely ( > http://lists.gnu.org/archive/html/automake/2012-12/msg00038.html ) and > errors out upon seeing it. > > Attached patch replaces it with proper AC_CONFIG_HEADERS. > > > I'm not subscribed to the list. I'm mass-submitting automake-1.13 fixes. Please see what you can do so I don't need to do the same when automake-1.14 comes out: http://cazfi.livejournal.com/195108.html - ML

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[alsa-devel] [PATCH] ASoC: fsi: Remove __devinitconst
by Thierry Reding 16 Jan '13

16 Jan '13

__devinitconst and friends have recently been removed and must not be used anymore. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9157612..c724026 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2160,7 +2160,7 @@ static struct fsi_core fsi2_core = { .b_mclk = B_MST_CTLR, }; -static struct of_device_id fsi_of_match[] __devinitconst = { +static struct of_device_id fsi_of_match[] = { { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, {}, -- 1.8.1

2 1

[alsa-devel] [PATCH] ASoC: wm8978: SND_SOC_DAIFMT_NB_NF become default clock inversion
by Kuninori Morimoto 16 Jan '13

16 Jan '13

Current wm8978 driver return -EINVAL if it doesn't have SND_SOC_DAIFMT_INV_MASK flags on snd_soc_dai_ops :: set_fmt. But default settings should be treated as SND_SOC_DAIFMT_NB_NF (= no clock inversion). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8978.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index f347af3..ba01981 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -692,8 +692,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; case SND_SOC_DAIFMT_IB_IF: iface |= 0x180; break; @@ -703,8 +701,9 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_IF: iface |= 0x80; break; + case SND_SOC_DAIFMT_NB_NF: default: - return -EINVAL; + break; } snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/3] Fixes for the new generic parser
by David Henningsson 16 Jan '13

16 Jan '13

Hi Takashi, I've run my hda-emu summary test tool to check for errors in the 110 codecs we have of certified systems (those in codecs/canonical) against the test/hda-migrate branch. After these fixes and the one I sent yesterday, there are no remaining regressions (as seen by this test tool) compared to the current sound tree. David Henningsson (3): ALSA: hda - Check array bounds in get_input_path ALSA: hda - fix wrong adc_idx in generic parser ALSA: hda - initialize channel counts correctly sound/pci/hda/hda_generic.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: arizona: Don't request FLL lock IRQ
by Mark Brown 16 Jan '13

16 Jan '13

We only log the result and since the interrupt triggers on loss of lock during shutdown this may lead to spurious interrupts during shutdown delaying the process. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 19 ------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 20 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9da5eee..b3621d6 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1008,17 +1008,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_lock(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "Lock status changed\n"); - - complete(&fll->lock); - - return IRQ_HANDLED; -} - static irqreturn_t arizona_fll_clock_ok(int irq, void *data) { struct arizona_fll *fll = data; @@ -1258,7 +1247,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, { int ret; - init_completion(&fll->lock); init_completion(&fll->ok); fll->id = id; @@ -1269,13 +1257,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, - arizona_fll_lock, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", - id, ret); - } - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, arizona_fll_clock_ok, fll); if (ret != 0) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b3b97f56..67fd49b 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -200,7 +200,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion lock; struct completion ok; unsigned int fref; unsigned int fout; -- 1.7.10.4

1 0

[alsa-devel] Fujitsu Lifebook s7110 (ALC262) no sound
by Raymond Yau 16 Jan '13

16 Jan '13

> after patching the clean 12.04 install with teh last alsa version (see comment #13) and with external speakers plugged (3.5 mm jack), I have got the follwoing Alsa log: http://www.alsa-project.org/db/?f=778415ad65d437d9ff5e5dd341e0e556ff848bc1 http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=3d… -/* - * fujitsu model - * 0x14 = headphone/spdif-out, 0x15 = internal speaker, - * 0x1b = port replicator headphone out - */ Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Surround Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x032110f0: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 2 0x0c 0x0d* Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Front Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0xb7231110: [Fixed] HP Out at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Dock Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x2321101f: [Jack] HP Out at Sep Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 2 0x0c* 0x0d If seem that BIOS did not setup the internal speaker node and need pin fixup Pulseaudio expect headphone Jack and speaker phantom port https://launchpadlibrarian.net/128561206/pulseverbose2.log

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[alsa-devel] [PATCH v3 0/4] RME HDSP fixes
by Adrian Knoth 16 Jan '13

16 Jan '13

Hi! Changes compared to v2: - use snd_printd to print iobox detection time - use KERN_ERR in case no iobox was found - define HDSP_S300 as a dummy for (0x100 | HDSP_S200) Adrian Knoth (4): ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes ALSA: hdsp - Implement generic function to toggle settings ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings ALSA: hdsp - Remove obsolete settings functions sound/pci/rme9652/hdsp.c | 462 ++++++++++++---------------------------------- 1 file changed, 117 insertions(+), 345 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH] ASoC: maintainers: Update email address.
by Liam Girdwood 16 Jan '13

16 Jan '13

From: Liam Girdwood <liam.r.girdwood(a)intel.com> I'm no longer at TI. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)intel.com> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 915564e..3c2e275 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7082,7 +7082,7 @@ F: include/uapi/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) -M: Liam Girdwood <lrg(a)ti.com> +M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2 0/4] RME HDSP fixes
by Adrian Knoth 15 Jan '13

15 Jan '13

Hi, here's v2 of my previous series of patches for the RME HDSP. Changes to the previous version: - correct time value in verbose output ("iobox found after %d ms") - define and use constants for iobox detection (no more 0x10a0) - hdsp_toggle_setting: only change register if newval != oldval Adrian Knoth (4): ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes ALSA: hdsp - Implement generic function to toggle settings ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings ALSA: hdsp - Remove obsolete settings functions sound/pci/rme9652/hdsp.c | 461 ++++++++++++---------------------------------- 1 file changed, 116 insertions(+), 345 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH (sound-unstable)] ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
by David Henningsson 15 Jan '13

15 Jan '13

Obvious copy-paste error. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b28e403..e6cdad7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3794,7 +3794,7 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) } for (i = 0; i < codec->amp_cache.buf.used; i++) { struct hda_amp_info *amp; - amp = snd_array_elem(&codec->cmd_cache.buf, i); + amp = snd_array_elem(&codec->amp_cache.buf, i); amp->head.dirty = 1; } } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Make it work with sound-unstable.git
by David Henningsson 15 Jan '13

15 Jan '13

To be able to build the new code (so we can run aggregate tests on it) the following patch needs to be applied. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 8 +++++++ hda-ctlsh.c | 5 ++++ include/linux/ctype.h | 64 +++++++++++++++++++++++++++++++++++++++++++++++++ snd-wrapper.c | 32 +++++++++++++++++++++++++ 4 files changed, 109 insertions(+) create mode 100644 include/linux/ctype.h diff --git a/configure.ac b/configure.ac index 3103eb5..80b68a6 100644 --- a/configure.ac +++ b/configure.ac @@ -182,6 +182,14 @@ if test "$OLD_HDA_CMD" = "1"; then AC_DEFINE(OLD_HDA_CMD) fi +AC_MSG_CHECKING(for user_mutex) +if grep -q 'struct mutex user_mutex' hda/hda_codec.h; then + AC_DEFINE(HAVE_USER_MUTEX) + AC_MSG_RESULT(yes) +else + AC_MSG_RESULT(no) +fi + AC_MSG_CHECKING(for new pm_notify callback) ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS -Iinclude" diff --git a/hda-ctlsh.c b/hda-ctlsh.c index 5ad95fc..04fc318 100644 --- a/hda-ctlsh.c +++ b/hda-ctlsh.c @@ -537,6 +537,11 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) return -ENOENT; } +int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) +{ + return -ENOENT; +} + #ifdef HAVE_USER_PINCFGS static void get_drv_pincfgs(void) { diff --git a/include/linux/ctype.h b/include/linux/ctype.h new file mode 100644 index 0000000..8acfe31 --- /dev/null +++ b/include/linux/ctype.h @@ -0,0 +1,64 @@ +#ifndef _LINUX_CTYPE_H +#define _LINUX_CTYPE_H + +/* + * NOTE! This ctype does not handle EOF like the standard C + * library is required to. + */ + +#define _U 0x01 /* upper */ +#define _L 0x02 /* lower */ +#define _D 0x04 /* digit */ +#define _C 0x08 /* cntrl */ +#define _P 0x10 /* punct */ +#define _S 0x20 /* white space (space/lf/tab) */ +#define _X 0x40 /* hex digit */ +#define _SP 0x80 /* hard space (0x20) */ + +extern const unsigned char _ctype[]; + +#define __ismask(x) (_ctype[(int)(unsigned char)(x)]) + +#define isalnum(c) ((__ismask(c)&(_U|_L|_D)) != 0) +#define isalpha(c) ((__ismask(c)&(_U|_L)) != 0) +#define iscntrl(c) ((__ismask(c)&(_C)) != 0) +#define isdigit(c) ((__ismask(c)&(_D)) != 0) +#define isgraph(c) ((__ismask(c)&(_P|_U|_L|_D)) != 0) +#define islower(c) ((__ismask(c)&(_L)) != 0) +#define isprint(c) ((__ismask(c)&(_P|_U|_L|_D|_SP)) != 0) +#define ispunct(c) ((__ismask(c)&(_P)) != 0) +/* Note: isspace() must return false for %NUL-terminator */ +#define isspace(c) ((__ismask(c)&(_S)) != 0) +#define isupper(c) ((__ismask(c)&(_U)) != 0) +#define isxdigit(c) ((__ismask(c)&(_D|_X)) != 0) + +#define isascii(c) (((unsigned char)(c))<=0x7f) +#define toascii(c) (((unsigned char)(c))&0x7f) + +static inline unsigned char __tolower(unsigned char c) +{ + if (isupper(c)) + c -= 'A'-'a'; + return c; +} + +static inline unsigned char __toupper(unsigned char c) +{ + if (islower(c)) + c -= 'a'-'A'; + return c; +} + +#define tolower(c) __tolower(c) +#define toupper(c) __toupper(c) + +/* + * Fast implementation of tolower() for internal usage. Do not use in your + * code. + */ +static inline char _tolower(const char c) +{ + return c | 0x20; +} + +#endif diff --git a/snd-wrapper.c b/snd-wrapper.c index 69cd19c..c2649d9 100644 --- a/snd-wrapper.c +++ b/snd-wrapper.c @@ -27,6 +27,35 @@ #include "hda/hda_codec.h" #include "hda-types.h" #include "hda-log.h" +#include <linux/ctype.h> + +/* Copied from lib/ctype.c */ +const unsigned char _ctype[] = { +_C,_C,_C,_C,_C,_C,_C,_C, /* 0-7 */ +_C,_C|_S,_C|_S,_C|_S,_C|_S,_C|_S,_C,_C, /* 8-15 */ +_C,_C,_C,_C,_C,_C,_C,_C, /* 16-23 */ +_C,_C,_C,_C,_C,_C,_C,_C, /* 24-31 */ +_S|_SP,_P,_P,_P,_P,_P,_P,_P, /* 32-39 */ +_P,_P,_P,_P,_P,_P,_P,_P, /* 40-47 */ +_D,_D,_D,_D,_D,_D,_D,_D, /* 48-55 */ +_D,_D,_P,_P,_P,_P,_P,_P, /* 56-63 */ +_P,_U|_X,_U|_X,_U|_X,_U|_X,_U|_X,_U|_X,_U, /* 64-71 */ +_U,_U,_U,_U,_U,_U,_U,_U, /* 72-79 */ +_U,_U,_U,_U,_U,_U,_U,_U, /* 80-87 */ +_U,_U,_U,_P,_P,_P,_P,_P, /* 88-95 */ +_P,_L|_X,_L|_X,_L|_X,_L|_X,_L|_X,_L|_X,_L, /* 96-103 */ +_L,_L,_L,_L,_L,_L,_L,_L, /* 104-111 */ +_L,_L,_L,_L,_L,_L,_L,_L, /* 112-119 */ +_L,_L,_L,_P,_P,_P,_P,_C, /* 120-127 */ +0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, /* 128-143 */ +0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, /* 144-159 */ +_S|_SP,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P, /* 160-175 */ +_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P, /* 176-191 */ +_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U, /* 192-207 */ +_U,_U,_U,_U,_U,_U,_U,_P,_U,_U,_U,_U,_U,_U,_U,_L, /* 208-223 */ +_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L, /* 224-239 */ +_L,_L,_L,_L,_L,_L,_L,_P,_L,_L,_L,_L,_L,_L,_L,_L}; /* 240-255 */ + int snd_pcm_format_width(int format) { @@ -46,6 +75,9 @@ int snd_pcm_format_width(int format) int snd_hda_create_hwdep(struct hda_codec *codec) { +#ifdef HAVE_USER_MUTEX + mutex_init(&codec->user_mutex); +#endif return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] A couple of pending HD-audio patches
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, these are yet another couple of HD-audio patches, one for merging to 3.8, and another an enhancement for 3.9. Takashi

1 2

[alsa-devel] [PATCH 00/12] Use standard fixup table for STAC/IDT codecs
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, this is a series of patches to convert the STAC/IDT codec parser to use the standard fixup table. Obviously it's a step forward for the migration of patch_sigmatel.c to the generic parser. Takashi

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[alsa-devel] [PATCH RFC 0/2] Yet more patches for HD-audio generic parser
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, these two patches have slipped from the previous patch series. Takashi

1 2

[alsa-devel] sound/pci/hda/patch_sigmatel.c:2219:24: warning: unused variable 'spec'
by Fengguang Wu 15 Jan '13

15 Jan '13

Hi Takashi, FYI, there are new compile warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 629965377895ebe6a546184911d588fd29596dc8 commit: 8018ee4e722f6c408d475262b54dc8a73fdd349f ALSA: hda - Use standard fixup table for IDT92HD71Bxx date: 2 hours ago config: make ARCH=i386 defconfig All warnings: sound/pci/hda/patch_sigmatel.c: In function 'stac92hd71bxx_fixup_hp': >> sound/pci/hda/patch_sigmatel.c:2219:24: warning: unused variable 'spec' [-Wunused-variable] vim +/spec +2219 sound/pci/hda/patch_sigmatel.c 8018ee4e Takashi Iwai 2013-01-14 2216 static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, 8018ee4e Takashi Iwai 2013-01-14 2217 const struct hda_fixup *fix, int action) 8018ee4e Takashi Iwai 2013-01-14 2218 { 8018ee4e Takashi Iwai 2013-01-14 @2219 struct sigmatel_spec *spec = codec->spec; 8018ee4e Takashi Iwai 2013-01-14 2220 8018ee4e Takashi Iwai 2013-01-14 2221 if (action != HDA_FIXUP_ACT_PRE_PROBE) 8018ee4e Takashi Iwai 2013-01-14 2222 return; 8018ee4e Takashi Iwai 2013-01-14 2223 8018ee4e Takashi Iwai 2013-01-14 2224 if (hp_blike_system(codec->subsystem_id)) { 8018ee4e Takashi Iwai 2013-01-14 2225 unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); 8018ee4e Takashi Iwai 2013-01-14 2226 if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || 8018ee4e Takashi Iwai 2013-01-14 2227 get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] sound/pci/hda/patch_sigmatel.c:2260:26: sparse: obsolete array initializer, use C99 syntax
by Fengguang Wu 15 Jan '13

15 Jan '13

Hi Takashi, FYI, there are new sparse warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 629965377895ebe6a546184911d588fd29596dc8 commit: 8018ee4e722f6c408d475262b54dc8a73fdd349f ALSA: hda - Use standard fixup table for IDT92HD71Bxx date: 2 hours ago >> sound/pci/hda/patch_sigmatel.c:2260:26: sparse: obsolete array initializer, use C99 syntax sound/pci/hda/patch_sigmatel.c:2266:26: sparse: obsolete array initializer, use C99 syntax vim +2260 sound/pci/hda/patch_sigmatel.c 8018ee4e Takashi Iwai 2013-01-14 2249 static const struct hda_fixup stac92hd71bxx_fixups[] = { 8018ee4e Takashi Iwai 2013-01-14 2250 [STAC_92HD71BXX_REF] = { 8018ee4e Takashi Iwai 2013-01-14 2251 .type = HDA_FIXUP_FUNC, 8018ee4e Takashi Iwai 2013-01-14 2252 .v.func = stac92hd71bxx_fixup_ref, 8018ee4e Takashi Iwai 2013-01-14 2253 }, 8018ee4e Takashi Iwai 2013-01-14 2254 [STAC_DELL_M4_1] = { 8018ee4e Takashi Iwai 2013-01-14 2255 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2256 .v.pins = dell_m4_1_pin_configs, 8018ee4e Takashi Iwai 2013-01-14 2257 .chained = true, 8018ee4e Takashi Iwai 2013-01-14 2258 .chain_id = STAC_92HD71BXX_NO_SMUX, 8018ee4e Takashi Iwai 2013-01-14 2259 }, 8018ee4e Takashi Iwai 2013-01-14 @2260 [STAC_DELL_M4_2] { 8018ee4e Takashi Iwai 2013-01-14 2261 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2262 .v.pins = dell_m4_2_pin_configs, 8018ee4e Takashi Iwai 2013-01-14 2263 .chained = true, 8018ee4e Takashi Iwai 2013-01-14 2264 .chain_id = STAC_92HD71BXX_NO_DMIC, 8018ee4e Takashi Iwai 2013-01-14 2265 }, 8018ee4e Takashi Iwai 2013-01-14 2266 [STAC_DELL_M4_3] { 8018ee4e Takashi Iwai 2013-01-14 2267 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2268 .v.pins = dell_m4_3_pin_configs, --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] [PATCH 1/4] ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes
by Adrian Knoth 15 Jan '13

15 Jan '13

The current iobox detection code reportedly fails for various users, so simply do what the Win32 driver does instead. Patch originally by Karl Grill <kgrill(a)chello.at> and then modified to comply with kernel coding guidelines + current HEAD. Signed-off-by: Karl Grill <kgrill(a)chello.at> Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4fae81f..bcad9ba 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -154,10 +154,11 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_BIGENDIAN_MODE 0x200 #define HDSP_RD_MULTIPLE 0x400 #define HDSP_9652_ENABLE_MIXER 0x800 +#define HDSP_DUNNO 0x1000 #define HDSP_TDO 0x10000000 -#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0) -#define HDSP_S_LOAD (HDSP_PROGRAM|HDSP_CONFIG_MODE_1) +#define HDSP_S_PROGRAM (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_0) +#define HDSP_S_LOAD (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_1) /* Control Register bits */ @@ -671,13 +672,23 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { + int i; + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; - if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { - snd_printk("Hammerfall-DSP: no IO box connected!\n"); - hdsp->state &= ~HDSP_FirmwareLoaded; - return -EIO; + for (i = 0; i < 500; i++) { + if (0 == (hdsp_read(hdsp, HDSP_statusRegister) & + HDSP_ConfigError)) { + if (i) { + snd_printk("Hammerfall-DSP: IO box found after %d ms\n", + i); + } + return 0; + } + msleep(20); } - return 0; + snd_printk("Hammerfall-DSP: no IO box connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, @@ -728,6 +739,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); return -EIO; } @@ -737,17 +749,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { hdsp_write(hdsp, HDSP_fifoData, cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout during firmware loading\n"); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); return -EIO; } } - ssleep(3); - - if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { - snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); - return -EIO; - } + hdsp_fifo_wait(hdsp, 3, HDSP_LONG_WAIT); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); + ssleep(3); #ifdef SNDRV_BIG_ENDIAN hdsp->control2_register = HDSP_BIGENDIAN_MODE; #else @@ -773,24 +783,51 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); - hdsp_write (hdsp, HDSP_fifoData, 0); - if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) - return -EIO; + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); - hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + } + + hdsp_write(hdsp, HDSP_control2Reg, 0x820); hdsp_write (hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { - hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); - hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) - hdsp->io_type = RPM; - else - hdsp->io_type = Multiface; - } else { + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) == 0) { hdsp->io_type = Digiface; + snd_printk("Hammerfall-DSP: Digiface found\n"); + return 0; } + + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) == 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + + hdsp->io_type = RPM; + snd_printk("Hammerfall-DSP: RPM found\n"); + return 0; } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) -- 1.7.10.4

2 6

[alsa-devel] ASoC fixes for atmel
by Mark Brown 15 Jan '13

15 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-atmel-pinctrl for you to fetch changes up to ea03c81521bde526570e1dec96eaa21fe5ac84a2: ASoC: atmel-ssc: add pinctrl selection to driver (2013-01-12 00:00:33 +0000) ---------------------------------------------------------------- ASoC: atmel: Fixes for pinctrl Due to a series of problems with the handling of Atmel, a combination of making changes that make other branches instantly buggy and a general failure to deal with the resulting issues effectively, v3.8 Atmel audio currently won't work at all for DT boards without adding pinctrl definitions and a request for those. ---------------------------------------------------------------- Bo Shen (2): ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - mute LEDs on HP: also check PCI SSID
by David Henningsson 14 Jan '13

14 Jan '13

If the codec subvendor is not set to HP, maybe the PCI SSID is set to HP. If so, there might still be a valid mute LED string in BIOS/UEFI. Cc: stable(a)vger.kernel.org Buglink: https://bugs.launchpad.net/bugs/1099372 Reported-and-tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a86547c..40c74a2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4948,7 +4948,8 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) &spec->gpio_led_polarity); return 1; } - if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP || + codec->bus->pci->subsystem_vendor == PCI_VENDOR_ID_HP) { while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (sscanf(dev->name, "HP_Mute_LED_%d_%x", -- 1.7.9.5

2 3

[alsa-devel] [RFC PATCH] ALSA: usb - fix race in creation of M-Audio Fast track pro driver
by David Henningsson 14 Jan '13

14 Jan '13

A patch in the 3.2 kernel caused regression with hotplugging the M-Audio Fast track pro, or sound after suspend. I don't have the device so I haven't done a full analysis, but it seems userspace (both udev and pulseaudio) got confused when a card was created, immediately destroyed, and then created again. However, at least one person in the bug report (martin djfun) reports that this patch resolves the issue for him. It also leaves a message in the log: "snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is a bit misleading. It is better than non-working audio, but maybe there's a more elegant solution? BugLink: https://bugs.launchpad.net/bugs/1095315 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/quirks.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) ...oh, and if you end up taking this patch as it is, you should probably add cc to stable (3.2+). diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acc12f0..e71fc8b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -387,11 +387,13 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) * rules */ err = usb_driver_set_configuration(dev, 2); - if (err < 0) { + if (err < 0) snd_printdd("error usb_driver_set_configuration: %d\n", err); - return -ENODEV; - } + /* Always return an error, so that we stop creating a device + that will just be destroyed and recreated with a new + configuration */ + return -ENODEV; } else snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n"); -- 1.7.9.5

3 4

[alsa-devel] [PATCH 0/3] M-Audio C400 corrections
by Eldad Zack 14 Jan '13

14 Jan '13

Hi Takashi, The following are specific fixes for the M-Audio FT C400. * Use the clock selector instead of the internal clock and add names for the clocks instead of Unit 129 and Unit 130. * Fix an issue where the latency would change every time the stream is started. Uses skip_packets. I think they could go in current mainline, since it is only for this device, but I also checked that they apply cleanly against for-next. Cheers, Eldad Eldad Zack (3): ALSA: usb-audio: correct M-Audio C400 clock source quirk ALSA: usb-audio: M-Audio FT C400 skip packet quirk ALSA: usb-audio: selector map for M-Audio FT C400 sound/usb/mixer_maps.c | 13 +++++++++++++ sound/usb/quirks-table.h | 4 ++-- sound/usb/quirks.c | 11 +++++++++++ 3 files changed, 26 insertions(+), 2 deletions(-) -- 1.7.12.4

2 4

Re: [alsa-devel] [RFC PATCH 4/4] alsa/soc: add hdmi audio codec based on cdf
by Sachin Kamat 14 Jan '13

14 Jan '13

+CC: ALSA mailing list, Mark Brown On 13 January 2013 18:22, Rahul Sharma <rahul.sharma(a)samsung.com> wrote: > This patch registers hdmi-audio codec to the ALSA framework. This is the second > client to the hdmi panel. Once notified by the CDF Core it proceeds towards > audio setting and audio control. It also subscribes for hpd notification to > implement hpd related audio requirements. > > Signed-off-by: Rahul Sharma <rahul.sharma(a)samsung.com> > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/exynos_hdmi_audio.c | 307 +++++++++++++++++++++++++++++++++++ > 3 files changed, 313 insertions(+) > create mode 100644 sound/soc/codecs/exynos_hdmi_audio.c > > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index b92759a..93f3f6b 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -496,3 +496,7 @@ config SND_SOC_ML26124 > > config SND_SOC_TPA6130A2 > tristate > + > +config SND_SOC_EXYNOS_HDMI_AUDIO > + tristate > + default y Do you want to enable this by default? Shouldn't this be depending on HDMI support? > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index 9bd4d95..bfe93e6 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -112,6 +112,7 @@ snd-soc-wm9705-objs := wm9705.o > snd-soc-wm9712-objs := wm9712.o > snd-soc-wm9713-objs := wm9713.o > snd-soc-wm-hubs-objs := wm_hubs.o > +snd-soc-exynos-hdmi-audio-objs := exynos_hdmi_audio.o > > # Amp > snd-soc-max9877-objs := max9877.o > @@ -230,6 +231,7 @@ obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o > obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o > obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o > obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o > +obj-$(CONFIG_SND_SOC_EXYNOS_HDMI_AUDIO) += snd-soc-exynos-hdmi-audio.o > > # Amp > obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o > diff --git a/sound/soc/codecs/exynos_hdmi_audio.c b/sound/soc/codecs/exynos_hdmi_audio.c > new file mode 100644 > index 0000000..50e8564 > --- /dev/null > +++ b/sound/soc/codecs/exynos_hdmi_audio.c > @@ -0,0 +1,307 @@ > +/* > + * ALSA SoC codec driver for HDMI audio on Samsung Exynos processors. > + * Copyright (C) 2012 Samsung corp. Copyright (c) 2012 (-13?) Samsung Electronics Co., Ltd. > + * Author: Rahul Sharma <rahul.sharma(a)samsung.com> > + * > + * This program is free software; you can redistribute it and/or > + * modify it under the terms of the GNU General Public License > + * version 2 as published by the Free Software Foundation. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + */ > +#include <linux/module.h> > +#include <linux/delay.h> > +#include <sound/soc.h> > +#include <video/display.h> > +#include <video/exynos_hdmi.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <linux/of_platform.h> > +#include <linux/platform_device.h> > + > +#undef dev_info > + > +#define dev_info(dev, format, arg...) \ > + dev_printk(KERN_CRIT, dev, format, ##arg) You may directly use dev_crit instead of dev_printk. > + > +static struct snd_soc_codec_driver hdmi_codec; > + > +/* platform device pointer for eynos hdmi audio device. */ > +static struct platform_device *exynos_hdmi_audio_pdev; > + > +struct hdmi_audio_context { > + struct platform_device *pdev; > + atomic_t plugged; > + struct display_entity_audio_params audio_params; > + struct display_entity *entity; > + struct display_entity_notifier notf; > + struct display_event_subscriber subscriber; > +}; > + > +static int exynos_hdmi_audio_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct hdmi_audio_context *ctx = snd_soc_codec_get_drvdata(codec); > + int ret; > + > + dev_info(codec->dev, "[%d] %s\n", __LINE__, __func__); How about making this a debug message as it does not convey anything useful? > + > + ctx->audio_params.type = DISPLAY_ENTITY_AUDIO_I2S; > + > + switch (params_channels(params)) { > + case 6: > + case 4: > + case 2: > + case 1: > + ctx->audio_params.channels = params_channels(params); > + break; > + default: > + dev_err(codec->dev, "%d channels not supported\n", > + params_channels(params)); > + return -EINVAL; > + } > + > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S8: > + ctx->audio_params.bits_per_sample = 8; > + break; > + case SNDRV_PCM_FORMAT_S16_LE: > + ctx->audio_params.bits_per_sample = 12; > + break; > + case SNDRV_PCM_FORMAT_S24_LE: > + ctx->audio_params.bits_per_sample = 16; > + break; > + default: > + dev_err(codec->dev, "Format(%d) not supported\n", > + params_format(params)); > + return -EINVAL; > + } > + > + switch (params_rate(params)) { > + case 32000: > + case 44100: > + case 88200: > + case 176400: > + case 48000: > + case 96000: > + case 192000: > + ctx->audio_params.sf = params_rate(params); > + break; > + default: > + dev_err(codec->dev, "%d Rate supported\n", > + params_rate(params)); > + return -EINVAL; > + } > + > + /* checking here to cache audioparms for hpd plug handling */ > + if (!atomic_read(&ctx->plugged)) > + return -EINVAL; > + > + ret = > + display_entity_hdmi_init_audio(ctx->entity, &ctx->audio_params); > + return ret; > +} > + > +static int exynos_hdmi_audio_trigger(struct snd_pcm_substream *substream, > + int cmd, struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct hdmi_audio_context *ctx = snd_soc_codec_get_drvdata(codec); > + int ret; > + > + dev_info(codec->dev, "[%d] %s\n", __LINE__, __func__); ditto > + > + /* checking here to cache audioparms for hpd plug handling */ > + if (!atomic_read(&ctx->plugged)) > + return -EINVAL; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + ret = display_entity_hdmi_set_audiostate(ctx->entity, > + DISPLAY_ENTITY_AUDIOSTATE_ON); > + if (ret) { > + dev_err(codec->dev, "audio enable failed.\n"); > + return -EINVAL; > + } > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + ret = display_entity_hdmi_set_audiostate(ctx->entity, > + DISPLAY_ENTITY_AUDIOSTATE_OFF); > + break; > + default: > + ret = -EINVAL; > + break; > + } > + > + return ret; > +} > + > +static const struct snd_soc_dai_ops exynos_hdmi_audio_dai_ops = { > + .hw_params = exynos_hdmi_audio_hw_params, > + .trigger = exynos_hdmi_audio_trigger, > +}; > + > +static struct snd_soc_dai_driver hdmi_codec_dai = { > + .name = "exynos-hdmi-audio", > + .playback = { > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_32000 | > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | > + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | > + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | > + SNDRV_PCM_FMTBIT_S24_LE, > + }, > + .ops = &exynos_hdmi_audio_dai_ops, > +}; > + > +void hdmi_audio_event_notify(struct display_entity *entity, > + enum display_entity_event_type type, > + unsigned int value, void *context) > +{ > + struct hdmi_audio_context *ctx = (struct hdmi_audio_context *)context; > + > + if (type == DISPLAY_ENTITY_HDMI_HOTPLUG) { > + dev_info(&ctx->pdev->dev, "[%d][%s] hpd(%d)\n", __LINE__, > + __func__, value); > + atomic_set(&ctx->plugged, !!value); > + } > +} > + > +int exynos_hdmi_audio_notification(struct display_entity_notifier *notf, > + struct display_entity *entity, int status) > +{ > + struct hdmi_audio_context *ctx = container_of(notf, > + struct hdmi_audio_context, notf); > + struct exynos_hdmi_control_ops *exynos_ops = > + (struct exynos_hdmi_control_ops *)entity->private; > + int hpd; > + > + if (status != DISPLAY_ENTITY_NOTIFIER_CONNECT && entity) > + return -EINVAL; > + > + dev_info(&ctx->pdev->dev, "[%d][%s]\n", __LINE__, __func__); > + > + ctx->entity = entity; > + > + ctx->subscriber.context = ctx; > + ctx->subscriber.notify = hdmi_audio_event_notify; > + > + display_entity_subscribe_event(entity, &ctx->subscriber); > + > + exynos_ops->get_hpdstate(entity, &hpd); > + atomic_set(&ctx->plugged, !!hpd); > + > + return 0; > +} > + > +static __devinit int hdmi_codec_probe(struct platform_device *pdev) __devinit is not necessary. +{ > + int ret; > + struct hdmi_audio_context *ctx; > + struct device_node *dev_node; > + struct platform_device *disp_pdev; > + > + dev_info(&pdev->dev, "[%d][%s]\n", __LINE__, __func__); > + > + ret = snd_soc_register_codec(&pdev->dev, &hdmi_codec, > + &hdmi_codec_dai, 1); > + > + if (ret) { > + dev_err(&pdev->dev, "register_codec failed (%d)\n", ret); > + return ret; > + } > + > + ctx = devm_kzalloc(&pdev->dev, sizeof(struct hdmi_audio_context), > + GFP_KERNEL); > + if (ctx == NULL) > + return -ENOMEM; > + > + ctx->pdev = pdev; > + atomic_set(&ctx->plugged, 0); > + platform_set_drvdata(pdev, ctx); > + > + dev_node = of_find_compatible_node(NULL, NULL, > + "samsung,exynos5-hdmi"); > + if (!dev_node) { > + dev_err(&pdev->dev, "[%d][%s] dt node not found.\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + > + disp_pdev = of_find_device_by_node(dev_node); > + if (!disp_pdev) { > + dev_err(&pdev->dev, "[ERROR][%d][%s] No pdev\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + > + ctx->notf.dev = &disp_pdev->dev; > + ctx->notf.notify = exynos_hdmi_audio_notification; > + > + ret = display_entity_register_notifier(&ctx->notf); > + if (ret) { > + dev_err(&pdev->dev, "[%d][%s] entity registe failed.\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + return ret; > +} > + > +static __devexit int hdmi_codec_remove(struct platform_device *pdev) __devexit is not necessary. > +{ > + dev_info(&pdev->dev, " %s:%s:%d", __FILE__, __func__, __LINE__); > + mdelay(1000); > + > + snd_soc_unregister_codec(&pdev->dev); > + return 0; > +} > + > +static struct platform_driver hdmi_codec_driver = { > + .driver = { > + .name = "exynos-hdmi-audio-codec", > + .owner = THIS_MODULE, > + }, > + > + .probe = hdmi_codec_probe, > + .remove = __devexit_p(hdmi_codec_remove), > +}; > + > +static int __init hdmi_codec_init(void) > +{ > + int ret; > + > + ret = platform_driver_register(&hdmi_codec_driver); > + if (ret < 0) > + return -EINVAL; > + > + exynos_hdmi_audio_pdev = platform_device_register_simple > + ("exynos-hdmi-audio-codec", -1, NULL, 0); > + if (IS_ERR_OR_NULL(exynos_hdmi_audio_pdev)) { > + ret = PTR_ERR(exynos_hdmi_audio_pdev); > + platform_driver_unregister(&hdmi_codec_driver); > + return ret; > + } > + > + return 0; > +} > +static void __exit hdmi_codec_exit(void) > +{ > + platform_driver_unregister(&hdmi_codec_driver); > + platform_device_unregister(exynos_hdmi_audio_pdev); > +} > + > +module_init(hdmi_codec_init); > +module_exit(hdmi_codec_exit); > + > +MODULE_AUTHOR("Rahul Sharma <rahul.sharma(a)samsung.com>"); > +MODULE_DESCRIPTION("ASoC EXYNOS HDMI codec driver"); > +MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:" DRV_NAME); > -- > 1.8.0 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html -- With warm regards, Sachin

2 1

[alsa-devel] [PATCH 0/2] ASoC: fsi: add DT support
by Kuninori Morimoto 14 Jan '13

14 Jan '13

Hi Mark These patches adds device tree support on FSI driver These are based on mark/topic/fsi branch Kuninori Morimoto (2): ASoC: fsi: don't use platform info pointer on probe() ASoC: fsi: add device tree support .../devicetree/bindings/sound/renesas,fsi.txt | 23 +++++ sound/soc/sh/fsi.c | 98 ++++++++++++++++---- 2 files changed, 104 insertions(+), 17 deletions(-) Best regards --- Kuninori Morimoto

2 7

Re: [alsa-devel] [PATCH v2 3/3] ASoC: tegra: add tegra machine driver using wm9712 codec
by Mark Brown 14 Jan '13

14 Jan '13

On Tue, Jan 08, 2013 at 11:52:33PM +0100, Lucas Stach wrote: > Am Dienstag, den 08.01.2013, 15:17 -0700 schrieb Stephen Warren: > > Hmmm. I thought that could be auto-instantiated based on probing the > > AC'97 bus for whatever was there? > AC97 is in principle probeable by reading the first regs of the codec, > but all the infrastructure to do proper hotplugging on the AC97 bus is > missing in the kernel. As I didn't want to be the one to add all this > infrastructure to a dying audio platform, I opted to do the same thing > as all the other platform using AC97 and instantiate the codec from the > machine driver. That's about the size of it, yes. We should write the device tree as though we'd got better infrastructure here since this isn't something that ought to be required but right now this is how you do AC'97 in ASoC. It'd be really nice if someone had the time to work on the AC'97 infrastructure here but I'd not hold my breath given that it's rarely used in new designs.

1 0

[alsa-devel] [PATCH 1/8] regmap: Add support for 24 bit wide register addresses
by Lars-Peter Clausen 13 Jan '13

13 Jan '13

Since regmap already has support for formatting 24 bit wide values, so adding support for 24 bit wide registers is pretty much straight forward. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/base/regmap/regmap.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 42d5cb0..26af93a 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -500,6 +500,12 @@ struct regmap *regmap_init(struct device *dev, } break; + case 24: + if (reg_endian != REGMAP_ENDIAN_BIG) + goto err_map; + map->format.format_reg = regmap_format_24; + break; + case 32: switch (reg_endian) { case REGMAP_ENDIAN_BIG: -- 1.8.0

2 17

[alsa-devel] [PATCH] ASoC: simple-card: add asoc_simple_dai for initializing
by Kuninori Morimoto 13 Jan '13

13 Jan '13

Current simple-card driver calls asoc_simple_card_dai_init() if platform had a asoc_simple_card_dai_init pointer. And, this initialization function works only when platform has an applicable initial value for each dai settings. And basically, almost all sound card requires certain initialization. This means that almost all platform has initialization settings, and driver do nothing if it doesn't have settings. And additionally, current simple-card supports sysclk settings but it was only for codec. In order to abolish deviation between cpu and codec, and in order to simplify processing, this patch adds asoc_simple_dai, and removed pointless struct asoc_simple_dai_init_info which was trigger of calling asoc_simple_card_dai_init(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This patch is based on both mark/topic/simple-card and mark/topic/fsi. I used "git merge". Please let me know if I should rebase it. >> Simon Please give this patch your Acked-by, since it modifies mach-shmobile. arch/arm/mach-shmobile/board-ap4evb.c | 35 +++++++------- arch/arm/mach-shmobile/board-armadillo800eva.c | 34 +++++++------- arch/arm/mach-shmobile/board-kzm9g.c | 20 ++++---- arch/arm/mach-shmobile/board-mackerel.c | 35 +++++++------- arch/sh/boards/mach-ecovec24/setup.c | 19 ++++---- arch/sh/boards/mach-se/7724/setup.c | 21 ++++----- include/sound/simple_card.h | 12 ++--- sound/soc/generic/simple-card.c | 58 ++++++++++++------------ 8 files changed, 116 insertions(+), 118 deletions(-) diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 4c97903..08294fa 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -686,21 +686,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { @@ -809,19 +809,18 @@ static struct platform_device lcdc1_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 042a21e..22a1cc5 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -807,21 +807,21 @@ static struct platform_device fsi_device = { }; /* FSI-WM8978 */ -static struct asoc_simple_dai_init_info fsi_wm8978_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 12288000, -}; - static struct asoc_simple_card_info fsi_wm8978_info = { .name = "wm8978", .card = "FSI2A-WM8978", - .cpu_dai = "fsia-dai", .codec = "wm8978.0-001a", .platform = "sh_fsi2", - .codec_dai = "wm8978-hifi", - .init = &fsi_wm8978_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "wm8978-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, + .sysclk = 12288000, + }, }; static struct platform_device fsi_wm8978_device = { @@ -833,18 +833,18 @@ static struct platform_device fsi_wm8978_device = { }; /* FSI-HDMI */ -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index 9ca7ce5..f170aed 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -526,21 +526,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4648_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4648_info = { .name = "AK4648", .card = "FSI2A-AK4648", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4648_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4648_device = { diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index b5d210b..3fd716d 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -502,19 +502,18 @@ static struct platform_device hdmi_lcdc_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { @@ -893,21 +892,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 8ebe4c7..065e9600 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -897,21 +897,20 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi_da7210_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi_da7210_info = { .name = "DA7210", .card = "FSIB-DA7210", - .cpu_dai = "fsib-dai", .codec = "da7210.0-001a", .platform = "sh_fsi.0", - .codec_dai = "da7210-hifi", - .init = &fsi_da7210_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "da7210-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, }; static struct platform_device fsi_da7210_device = { diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 975608f..4010e63 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -299,22 +299,21 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi_ak4642_info = { .name = "AK4642", .card = "FSIA-AK4642", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4642_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4642_device = { diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 4b62b8d..6c74527 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -14,21 +14,21 @@ #include <sound/soc.h> -struct asoc_simple_dai_init_info { +struct asoc_simple_dai { + const char *name; unsigned int fmt; - unsigned int cpu_daifmt; - unsigned int codec_daifmt; unsigned int sysclk; }; struct asoc_simple_card_info { const char *name; const char *card; - const char *cpu_dai; const char *codec; const char *platform; - const char *codec_dai; - struct asoc_simple_dai_init_info *init; /* for snd_link.init */ + + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; /* used in simple-card.c */ struct snd_soc_dai_link snd_link; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bc050ec..6cf8355 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,33 +16,38 @@ #define asoc_simple_get_card_info(p) \ container_of(p->dai_link, struct asoc_simple_card_info, snd_link) +static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, + struct asoc_simple_dai *set, + unsigned int daifmt) +{ + int ret = 0; + + daifmt |= set->fmt; + + if (!ret && daifmt) + ret = snd_soc_dai_set_fmt(dai, daifmt); + + if (!ret && set->sysclk) + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + + return ret; +} + static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); - struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; - unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + unsigned int daifmt = info->daifmt; int ret; - if (codec_daifmt) { - ret = snd_soc_dai_set_fmt(codec, codec_daifmt); - if (ret < 0) - return ret; - } - - if (iinfo->sysclk) { - ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + if (ret < 0) + return ret; - if (cpu_daifmt) { - ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + if (ret < 0) + return ret; return 0; } @@ -59,10 +64,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!cinfo->name || !cinfo->card || - !cinfo->cpu_dai || !cinfo->codec || !cinfo->platform || - !cinfo->codec_dai) { + !cinfo->cpu_dai.name || + !cinfo->codec_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -72,14 +77,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai; - - /* enable snd_link.init if cinfo has settings */ - if (cinfo->init) - cinfo->snd_link.init = asoc_simple_card_dai_init; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; + cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Remove suspend/resume soc driver operations
by Peter Ujfalusi 13 Jan '13

13 Jan '13

With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320dac33.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 782b0cd..4f35839 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1452,20 +1452,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static int dac33_soc_suspend(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int dac33_soc_resume(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, @@ -1476,8 +1462,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .reg_cache_default = dac33_reg, .probe = dac33_soc_probe, .remove = dac33_soc_remove, - .suspend = dac33_soc_suspend, - .resume = dac33_soc_resume, .controls = dac33_snd_controls, .num_controls = ARRAY_SIZE(dac33_snd_controls), -- 1.8.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Provide explicit trace of coefficient writes
by Mark Brown 12 Jan '13

12 Jan '13

Helpful for debugging. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4a233b8..14af8e2 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -775,6 +775,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) return -ENOMEM; } + adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", + file, blocks, le32_to_cpu(blk->len), + reg); ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/2 v2] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 12 Jan '13

12 Jan '13

Hi Mark, Olof, Arnd, This series goes on top of Linus' v3.8-rc3 and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. This material was designed to enter Mark's fixes queue, but as discussed with Olof, we can imagine merging everything through arm-soc or split the series (of 2 patches) and let them progress upstream separated (option that I do not like even if I know that the consequences are not so dramatic). So please, Olof, if you feel confortable with this series, tell us what you prefer and we will make our best to make this material go forward... Thanks for your help, best regards, v2: - remove the fix for typo in SSC status property: already merged upstream Bo Shen (2): ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-) -- 1.8.0

4 7

[alsa-devel] sound/pci/hda/hda_hwdep.c:487 parse_hints() warn: inconsistent returns mutex:&codec->user_mutex: locked (478) unlocked (447, 449, 452, 457, 487)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: bea06a147981e88465f566acd6c6e98b297f7712 ALSA: hda - Protect user-defined arrays via mutex date: 3 hours ago sound/pci/hda/hda_hwdep.c:487 parse_hints() warn: inconsistent returns mutex:&codec->user_mutex: locked (478) unlocked (447,449,452,457,487) vim +487 sound/pci/hda/hda_hwdep.c 43b62713 Takashi Iwai 2009-03-02 441 char *key, *val; 43b62713 Takashi Iwai 2009-03-02 442 struct hda_hint *hint; bea06a14 Takashi Iwai 2013-01-10 443 int err = 0; 1e1be432 Takashi Iwai 2008-07-30 444 e7d2860b André Goddard Rosa 2009-12-14 445 buf = skip_spaces(buf); 43b62713 Takashi Iwai 2009-03-02 446 if (!*buf || *buf == '#' || *buf == '\n') 4ea6fbc8 Takashi Iwai 2009-06-17 @447 return 0; 43b62713 Takashi Iwai 2009-03-02 448 if (*buf == '=') 43b62713 Takashi Iwai 2009-03-02 @449 return -EINVAL; 43b62713 Takashi Iwai 2009-03-02 450 key = kstrndup_noeol(buf, 1024); 43b62713 Takashi Iwai 2009-03-02 451 if (!key) 1e1be432 Takashi Iwai 2008-07-30 @452 return -ENOMEM; 43b62713 Takashi Iwai 2009-03-02 453 /* extract key and val */ 43b62713 Takashi Iwai 2009-03-02 454 val = strchr(key, '='); 43b62713 Takashi Iwai 2009-03-02 455 if (!val) { 43b62713 Takashi Iwai 2009-03-02 456 kfree(key); 43b62713 Takashi Iwai 2009-03-02 @457 return -EINVAL; 43b62713 Takashi Iwai 2009-03-02 458 } 43b62713 Takashi Iwai 2009-03-02 459 *val++ = 0; e7d2860b André Goddard Rosa 2009-12-14 460 val = skip_spaces(val); 43b62713 Takashi Iwai 2009-03-02 461 remove_trail_spaces(key); 43b62713 Takashi Iwai 2009-03-02 462 remove_trail_spaces(val); bea06a14 Takashi Iwai 2013-01-10 463 mutex_lock(&codec->user_mutex); 43b62713 Takashi Iwai 2009-03-02 464 hint = get_hint(codec, key); 43b62713 Takashi Iwai 2009-03-02 465 if (hint) { 43b62713 Takashi Iwai 2009-03-02 466 /* replace */ 43b62713 Takashi Iwai 2009-03-02 467 kfree(hint->key); 43b62713 Takashi Iwai 2009-03-02 468 hint->key = key; 43b62713 Takashi Iwai 2009-03-02 469 hint->val = val; bea06a14 Takashi Iwai 2013-01-10 470 goto unlock; 43b62713 Takashi Iwai 2009-03-02 471 } 43b62713 Takashi Iwai 2009-03-02 472 /* allocate a new hint entry */ 43b62713 Takashi Iwai 2009-03-02 473 if (codec->hints.used >= MAX_HINTS) 43b62713 Takashi Iwai 2009-03-02 474 hint = NULL; 43b62713 Takashi Iwai 2009-03-02 475 else 43b62713 Takashi Iwai 2009-03-02 476 hint = snd_array_new(&codec->hints); 1e1be432 Takashi Iwai 2008-07-30 477 if (!hint) { 1e1be432 Takashi Iwai 2008-07-30 @478 return -ENOMEM; bea06a14 Takashi Iwai 2013-01-10 479 goto unlock; 1e1be432 Takashi Iwai 2008-07-30 480 } 43b62713 Takashi Iwai 2009-03-02 481 hint->key = key; 43b62713 Takashi Iwai 2009-03-02 482 hint->val = val; bea06a14 Takashi Iwai 2013-01-10 483 unlock: bea06a14 Takashi Iwai 2013-01-10 484 mutex_unlock(&codec->user_mutex); bea06a14 Takashi Iwai 2013-01-10 485 if (err) bea06a14 Takashi Iwai 2013-01-10 486 kfree(key); bea06a14 Takashi Iwai 2013-01-10 @487 return err; 4ea6fbc8 Takashi Iwai 2009-06-17 488 } 4ea6fbc8 Takashi Iwai 2009-06-17 489 4ea6fbc8 Takashi Iwai 2009-06-17 490 static ssize_t hints_store(struct device *dev, --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

2 1

[alsa-devel] sound/pci/hda/hda_generic.c:1529 indep_hp_put() warn: inconsistent returns mutex:&spec->pcm_mutex: locked (1526) unlocked (1517, 1529)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: a5de6f4d021608c50ff5ff51d208b5030da92957 ALSA: hda - Implement independent HP control date: 33 hours ago sound/pci/hda/hda_generic.c:1529 indep_hp_put() warn: inconsistent returns mutex:&spec->pcm_mutex: locked (1526) unlocked (1517,1529) vim +1529 sound/pci/hda/hda_generic.c a5de6f4d Takashi Iwai 2012-12-21 1511 struct hda_gen_spec *spec = codec->spec; a5de6f4d Takashi Iwai 2012-12-21 1512 unsigned int select = ucontrol->value.enumerated.item[0]; a5de6f4d Takashi Iwai 2012-12-21 1513 a5de6f4d Takashi Iwai 2012-12-21 1514 mutex_lock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 1515 if (spec->active_streams) { a5de6f4d Takashi Iwai 2012-12-21 1516 mutex_unlock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 @1517 return -EBUSY; a5de6f4d Takashi Iwai 2012-12-21 1518 } a5de6f4d Takashi Iwai 2012-12-21 1519 a5de6f4d Takashi Iwai 2012-12-21 1520 if (spec->indep_hp_enabled != select) { a5de6f4d Takashi Iwai 2012-12-21 1521 spec->indep_hp_enabled = select; a5de6f4d Takashi Iwai 2012-12-21 1522 if (spec->indep_hp_enabled) a5de6f4d Takashi Iwai 2012-12-21 1523 spec->multiout.hp_out_nid[0] = 0; a5de6f4d Takashi Iwai 2012-12-21 1524 else a5de6f4d Takashi Iwai 2012-12-21 1525 spec->multiout.hp_out_nid[0] = spec->alt_dac_nid; a5de6f4d Takashi Iwai 2012-12-21 @1526 return 1; a5de6f4d Takashi Iwai 2012-12-21 1527 } a5de6f4d Takashi Iwai 2012-12-21 1528 mutex_unlock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 @1529 return 0; a5de6f4d Takashi Iwai 2012-12-21 1530 } a5de6f4d Takashi Iwai 2012-12-21 1531 a5de6f4d Takashi Iwai 2012-12-21 1532 static const struct snd_kcontrol_new indep_hp_ctl = { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] sound/pci/hda/hda_codec.c:3433 snd_hda_codec_write_cache() error: we previously assumed 'c' could be null (see line 3430)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: 95836563e7f636fc57732f5e687ca2f170eb674f ALSA: hda - Introduce cache & flush cmd / amp writes date: 33 hours ago sound/pci/hda/hda_codec.c:3433 snd_hda_codec_write_cache() error: we previously assumed 'c' could be null (see line 3430) vim +/c +3433 sound/pci/hda/hda_codec.c aa2936f5 Takashi Iwai 2009-05-26 3424 /* parm may contain the verb stuff for get/set amp */ aa2936f5 Takashi Iwai 2009-05-26 3425 verb = verb | (parm >> 8); aa2936f5 Takashi Iwai 2009-05-26 3426 parm &= 0xff; aa2936f5 Takashi Iwai 2009-05-26 3427 key = build_cmd_cache_key(nid, verb); aa2936f5 Takashi Iwai 2009-05-26 3428 mutex_lock(&codec->bus->cmd_mutex); aa2936f5 Takashi Iwai 2009-05-26 3429 c = get_alloc_hash(&codec->cmd_cache, key); aa2936f5 Takashi Iwai 2009-05-26 @3430 if (c) aa2936f5 Takashi Iwai 2009-05-26 3431 c->val = parm; 95836563 Takashi Iwai 2012-12-13 3432 if (codec->cached_write) 95836563 Takashi Iwai 2012-12-13 @3433 c->dirty = 1; aa2936f5 Takashi Iwai 2009-05-26 3434 mutex_unlock(&codec->bus->cmd_mutex); aa2936f5 Takashi Iwai 2009-05-26 3435 return 0; b3ac5636 Takashi Iwai 2007-08-10 3436 } --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [RESEND PATCH] ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word
by Ricardo Neri 12 Jan '13

12 Jan '13

As the IEC-60958 channel status word is set by ANDing and ORing with the appropriate definitions, the word bytes need to be initialized to zero to avoid misconfiguration due to previous hw_params calls. Signed-off-by: Ricardo Neri <rneri(a)dextratech.com> --- sound/soc/omap/omap-hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index f59c69f..75ad378 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -110,6 +110,8 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, /* * fill the IEC-60958 channel status word */ + /* initialize the word bytes */ + memset(iec->status, 0, sizeof(iec->status)); /* specify IEC-60958-3 (commercial use) */ iec->status[0] &= ~IEC958_AES0_PROFESSIONAL; -- 1.7.10.4

2 2

[alsa-devel] [patch] ASoC: pcm: delete some dead code
by Dan Carpenter 12 Jan '13

12 Jan '13

I've removed several unreachable returns. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..1d6af4d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1729,20 +1729,16 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* startup must always be called for new BEs */ ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { + if (ret < 0) goto disconnect; - return ret; - } /* keep going if FE state is > open */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN) return 0; ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) { + if (ret < 0) goto close; - return ret; - } /* keep going if FE state is > hw_params */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS) @@ -1750,10 +1746,8 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) { + if (ret < 0) goto hw_free; - return ret; - } /* run the stream event for each BE */ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP);

2 1

[alsa-devel] [PATCH] ASoC: fsl: fix multiple definition of init_module
by Shawn Guo 12 Jan '13

12 Jan '13

With commit f2818d0 (ASoC: fsl: fix miscompilation of snd-soc-imx-pcm), we will see the following build error when building modules with CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig. CC [M] sound/soc/fsl/phycore-ac97.o LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o LD [M] sound/soc/fsl/snd-soc-fsl-utils.o LD [M] sound/soc/fsl/snd-soc-imx-ssi.o LD [M] sound/soc/fsl/snd-soc-imx-audmux.o LD [M] sound/soc/fsl/snd-soc-imx-pcm.o sound/soc/fsl/imx-pcm-dma.o: In function `init_module': imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module' sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module': imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module' sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1 Instead of using bool for SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to fix the original issue, we should completely remove SND_SOC_IMX_PCM and have imx-pcm.o statically linked with imx-pcm-fiq.o or imx-pcm-dma.o. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Mark, To be clear, I think it's a v3.8 material. Shawn sound/soc/fsl/Kconfig | 9 ++------- sound/soc/fsl/Makefile | 5 ++++- sound/soc/fsl/imx-pcm.c | 3 --- 3 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3b98159..a210c8d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,18 +108,13 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM - tristate - config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ - select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_DMAENGINE_PCM - select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index afd3479..ec14579 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,7 +41,10 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index d5cd9ef..0c9f188 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,7 +31,6 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -80,7 +79,6 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } -EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -102,7 +100,6 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } -EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: twl6040: Updates for 3.9
by Peter Ujfalusi 12 Jan '13

12 Jan '13

Hi Mark, Liam, Changes since v1: - patch 2 for twl6040_resume() has been replaced with a simpler one Cover letter from v1: Small fixe from Misael for twl6040 codec driver and devm_* conversion, switch to system workqueue and dead code removal. Regards, Peter --- Misael Lopez Cruz (1): ASoC: twl6040: Convert PLUGINT to no-suspend irq Peter Ujfalusi (4): ASoC: twl6040: Only set the bias_level once in twl6040_resume() ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation sound/soc/codecs/twl6040.c | 62 +++++++--------------------------------------- 1 file changed, 9 insertions(+), 53 deletions(-) -- 1.8.1

2 6

[alsa-devel] [PATCH] ASoC: twl4030: Remove suspend/resume soc driver operations
by Peter Ujfalusi 12 Jan '13

12 Jan '13

With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 63b280b..5611b7b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2267,18 +2267,6 @@ static struct snd_soc_dai_driver twl4030_dai[] = { }, }; -static int twl4030_soc_suspend(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int twl4030_soc_resume(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int twl4030_soc_probe(struct snd_soc_codec *codec) { struct twl4030_priv *twl4030; @@ -2316,8 +2304,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .probe = twl4030_soc_probe, .remove = twl4030_soc_remove, - .suspend = twl4030_soc_suspend, - .resume = twl4030_soc_resume, .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, -- 1.8.1

2 1

Re: [alsa-devel] [PATCH v2 06/10] ASoC: tegra: remove auxdata
by Stephen Warren 11 Jan '13

11 Jan '13

On 01/11/2013 01:01 AM, Prashant Gaikwad wrote: > Configlink clock information is added to device tree. Get the clocks > using device node. Remove AUXDATA. Cc'ing in the ASoC maintainers as an FYI and for Acks; this patch needs to go through the Tegra tree due to dependencies. > > Signed-off-by: Prashant Gaikwad <pgaikwad(a)nvidia.com> > --- > sound/soc/tegra/tegra30_ahub.c | 14 ++------------ > 1 files changed, 2 insertions(+), 12 deletions(-) > > diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c > index bb31c41..2355630 100644 > --- a/sound/soc/tegra/tegra30_ahub.c > +++ b/sound/soc/tegra/tegra30_ahub.c > @@ -299,15 +299,6 @@ static const char * const configlink_clocks[] = { > "spdif_in", > }; > > -struct of_dev_auxdata ahub_auxdata[] = { > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), > - {} > -}; > - > #define LAST_REG(name) \ > (TEGRA30_AHUB_##name + \ > (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) > @@ -451,7 +442,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) > * Ensure that here. > */ > for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { > - clk = clk_get_sys(NULL, configlink_clocks[i]); > + clk = clk_get(&pdev->dev, configlink_clocks[i]); > if (IS_ERR(clk)) { > dev_err(&pdev->dev, "Can't get clock %s\n", > configlink_clocks[i]); > @@ -569,8 +560,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) > goto err_pm_disable; > } > > - of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, > - &pdev->dev); > + of_platform_populate(pdev->dev.of_node, NULL, NULL, &pdev->dev); > > return 0; > >

2 1

[alsa-devel] [PATCH 0/5] ASoC: twl6040: Updates for 3.9
by Peter Ujfalusi 11 Jan '13

11 Jan '13

Hi Mark, Liam, Small fixes from Misael for twl6040 codec driver and devm_* conversion, switch to system workqueue and dead code removal. Regards, Peter --- Misael Lopez Cruz (2): ASoC: twl6040: Convert PLUGINT to no-suspend irq ASoC: twl6040: Prevent extra power transitions during resume Peter Ujfalusi (3): ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation sound/soc/codecs/twl6040.c | 67 +++++++++------------------------------------- 1 file changed, 13 insertions(+), 54 deletions(-) -- 1.8.1

2 12

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 11 Jan '13

11 Jan '13

At Fri, 11 Jan 2013 08:47:25 +0100, Jens Axboe wrote: > > On 2013-01-10 21:19, Takashi Iwai wrote: > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH v2] ALSA: usb-audio: Fix NULL dereference by access to > > non-existing substream > > > > The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for > > audioformat mismatch] introduced the correction of parameters to be > > set for sync EP. But since the new code assumes that the sync EP is > > always paired with the data EP of another direction, it triggers Oops > > when a device only with a single direction is used. > > > > This patch adds a proper check of sync EP type and the presence of the > > paired substream for avoiding the crash. > > > > Reported-by: Jens Axboe <axboe(a)kernel.dk> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > Confirmed, it works. You can add my tested-by too. Thanks Takashi! Thanks for quick testing. Applied it now. Takashi

1 0

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 13:49:22 +0100, Jens Axboe wrote: > > On 2013-01-10 11:21, Takashi Iwai wrote: > > At Thu, 10 Jan 2013 09:23:48 +0100, > > Jens Axboe wrote: > >> > >> Hi, > >> > >> I have a USB DAC that I use for music. Just upgraded my workstation from > >> 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > >> an immediate oops. I have attached the picture. It's crashing here: > >> > >> (gdb) l *snd_usb_pcm_prepare+0x153 > >> 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > >> 465 snd_pcm_format_t pcm_format) > >> 466 { > >> 467 int i; > >> 468 int score = 0; > >> 469 > >> 470 if (fp->channels < 1) { > >> 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > >> 472 return 0; > >> 473 } > >> 474 > >> > >> 'fp' is clearly NULL. > > > > Hmm... it's a function called only from a single place in > > configure_sync_endpoint() and it's in list_for_each_entry(): > > > > list_for_each_entry(fp, &sync_subs->fmt_list, list) { > > int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, > > subs->cur_rate, subs->pcm_format); > > > > so it must be hardly NULL in normal situations. > > The only scenario I can imagine right now is that the whole structure > > is still uninitialized while it's called. If so, it's a race problem, > > and shouldn't be new to 3.8. A patch like below might paper over the > > problem although it's far from perfect. > > > > > >> Let me know if you want a bisect. > > > > Yeah, that'll be really helpful. > > Here it is, it's from the one introducing the audioformat lookup. > Confirmed that 3.8-rc3 with this backed out works fine, too. So should > be fairly confident in that result. Thanks! I'm slowly understanding what's happening there. Could you try the patch below? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: usb-audio: Fix NULL dereference by access to non-existing substream The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-by: Jens Axboe <axboe(a)kernel.dk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..21c0001 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -511,6 +511,17 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) struct snd_usb_substream *sync_subs = &subs->stream->substream[subs->direction ^ 1]; + if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA || + !subs->stream) { + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + } + /* Try to find the best matching audioformat. */ list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, -- 1.8.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Fix sense of regulator bypass mode
by Mark Brown 10 Jan '13

10 Jan '13

Enable bypass when the regulator is idle, not when it is in use. This is consistent with what the few existing users actually want. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc18d78..17dba18 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, true); + ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to bypass %s: %d\n", @@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, false); + ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to unbypass %s: %d\n", @@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } + + if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + ret = regulator_allow_bypass(w->regulator, true); + if (ret != 0) + dev_warn(w->dapm->dev, + "ASoC: Failed to unbypass %s: %d\n", + w->name, ret); + } break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP -- 1.7.10.4

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.8
by Mark Brown 10 Jan '13

10 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-3.8-rc2 for you to fetch changes up to 49a170bcf28af5ef372cdbed7750ba8d4162bb0e: Merge remote-tracking branch 'asoc/fix/wm5100' into tmp (2013-01-10 12:22:30 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.8 Nothing terribly exciting here except for the DOUBLE_RANGE fix which just hadn't worked before, nobody noticed due to lack of use. ---------------------------------------------------------------- Axel Lin (5): ASoC: sta529: Fix update register bits in sta529_set_dai_fmt ASoC: wm2200: Fix setting dai format in wm2200_set_fmt ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt() ASoC: arizona: Do proper shift for setting AIF rate ASoC: cs42l52: Catch no-match case in cs42l52_get_clk Chuansheng Liu (2): ASoC: core: fix the memory leak in case of device_add() failure ASoC: core: fix the memory leak in case of remove_aux_dev() Daniel Mack (2): ASoC: cs4271: fix sparse warning ASoC: cs4271: fix property check Fabio Estevam (2): ASoC: soc-core: Remove unused 'ret' variable ASoC: sgtl5000: Fix maximum value for microphone gain MR.Swami.Reddy(a)ti.com (2): ASoC: lm49453: Fix adc, mic and sidetone volume ranges ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B Mark Brown (20): ASoC: core: Fix SOC_DOUBLE_RANGE() macros ASoC: arizona: Correct FLL source definitions ASoC: wm5102: Improve speaker enable performance ASoC: wm2200: Remove DSP B and left justified AIF modes ASoC: arizona: Remove DSP B and left justified AIF modes ASoC: wm5100: Remove DSP B and left justified formats ASoC: wm2000: Fix sense of speech clarity enable ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/cs4271' into tmp Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp Merge remote-tracking branch 'asoc/fix/lm49453' into tmp Merge remote-tracking branch 'asoc/fix/pxa' into tmp Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp Merge remote-tracking branch 'asoc/fix/sta529' into tmp Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Merge remote-tracking branch 'asoc/fix/wm5100' into tmp Mike Dunn (2): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset Patrick Lai (1): ASoC: pcm: allow backend hardware to be freed in pause state arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 + arch/arm/mach-pxa/pxa27x.c | 4 +- include/sound/cs4271.h | 2 +- include/sound/soc.h | 10 ++- sound/arm/pxa2xx-ac97-lib.c | 26 ++++++- sound/soc/codecs/arizona.c | 9 +-- sound/soc/codecs/arizona.h | 18 ++--- sound/soc/codecs/cs4271.c | 6 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/lm49453.c | 106 ++++++++++----------------- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta529.c | 9 ++- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2200.c | 8 +- sound/soc/codecs/wm5100.c | 6 -- sound/soc/codecs/wm5102.c | 48 +++++++++++- sound/soc/codecs/wm_adsp.c | 23 +++++- sound/soc/soc-core.c | 35 ++++++++- sound/soc/soc-pcm.c | 1 + 19 files changed, 201 insertions(+), 125 deletions(-)

2 1

[alsa-devel] [PATCH RFC 0/8] Use generic parser for HD-audio codec drivers
by Takashi Iwai 10 Jan '13

10 Jan '13

Hi, this is the next series of patches to convert various codec drivers to use the new generic parser code. Most of changes are obvious, but some places it gets tricky. As mentioned in another mail, the patches are found in sound-unstable git tree test/hda-migrate branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git Takashi

2 10

[alsa-devel] [PATCH] ASoC: ak4642: add Device Tree support
by Kuninori Morimoto 10 Jan '13

10 Jan '13

Support for loading the ak4642 codec module via devicetree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/ak4642.txt | 17 ++++++++++ sound/soc/codecs/ak4642.c | 33 ++++++++++++++++++-- 2 files changed, 48 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt new file mode 100644 index 0000000..623d4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.txt @@ -0,0 +1,17 @@ +AK4642 I2C transmitter + +This device supports I2C mode only. + +Required properties: + + - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" + - reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4648: ak4648@0x12 { + compatible = "asahi-kasei,ak4642"; + reg = <0x12>; + }; +}; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f0cdab..c78794d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/initval.h> @@ -513,12 +514,31 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct device_node *np = i2c->dev.of_node; + const struct snd_soc_codec_driver *driver; + + driver = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4642_of_match, &i2c->dev); + if (of_id) + driver = of_id->data; + } else { + driver = (struct snd_soc_codec_driver *)id->driver_data; + } + + if (!driver) { + dev_err(&i2c->dev, "no driver\n"); + return -EINVAL; + } + return snd_soc_register_codec(&i2c->dev, - (struct snd_soc_codec_driver *)id->driver_data, - &ak4642_dai, 1); + driver, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -527,6 +547,14 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } +static struct of_device_id ak4642_of_match[] __devinitconst = { + { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4642_of_match); + static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, @@ -539,6 +567,7 @@ static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", .owner = THIS_MODULE, + .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, .remove = ak4642_i2c_remove, -- 1.7.9.5

2 1

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 09:23:48 +0100, Jens Axboe wrote: > > Hi, > > I have a USB DAC that I use for music. Just upgraded my workstation from > 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > an immediate oops. I have attached the picture. It's crashing here: > > (gdb) l *snd_usb_pcm_prepare+0x153 > 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > 465 snd_pcm_format_t pcm_format) > 466 { > 467 int i; > 468 int score = 0; > 469 > 470 if (fp->channels < 1) { > 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > 472 return 0; > 473 } > 474 > > 'fp' is clearly NULL. Hmm... it's a function called only from a single place in configure_sync_endpoint() and it's in list_for_each_entry(): list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, subs->cur_rate, subs->pcm_format); so it must be hardly NULL in normal situations. The only scenario I can imagine right now is that the whole structure is still uninitialized while it's called. If so, it's a race problem, and shouldn't be new to 3.8. A patch like below might paper over the problem although it's far from perfect. > Let me know if you want a bisect. Yeah, that'll be really helpful. Or, at least, could you try to copy sound/usb/* from 3.7 to 3.8 and see whether it's really a regression there? thanks, Takashi --- diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ad181d5..3bfc564 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -81,8 +81,7 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) */ static void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp) + int stream) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -96,6 +95,13 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->speed = snd_usb_get_speed(subs->dev); snd_usb_set_pcm_ops(as->pcm, stream); +} + +static void snd_usb_add_substream_format(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; list_add_tail(&fp->list, &subs->fmt_list); subs->formats |= fp->formats; @@ -342,7 +348,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp); + snd_usb_add_substream_format(as, stream, fp); return add_chmap(as->pcm, stream, subs); } @@ -370,7 +376,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strcpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_PLAYBACK); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_CAPTURE); + snd_usb_add_substream_format(as, stream, fp); list_add(&as->list, &chip->pcm_list); chip->pcm_devs++;

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Make ebox44_table static
by Sachin Kamat 10 Jan '13

10 Jan '13

Fixes the following sparse warning: sound/usb/mixer_quirks.c:1209:23: warning: symbol 'ebox44_table' was not declared. Should it be static? Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/mixer_quirks.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0422b13..15520de 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -struct std_mono_table ebox44_table[] = { +static struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ALSA: hdspm: Fix wordclock status on AES32
by Andre Schramm 09 Jan '13

09 Jan '13

Use correct bitmask for AES32 cards to determine wordclock lock state, add missing bitmask for sync check and make output of the corresponding control and /proc coherent. Signed-off-by: Andre Schramm <andre.schramm(a)iosono-sound.com> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e02e06..36e247f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); */ /* status */ #define HDSPM_AES32_wcLock 0x0200000 +#define HDSPM_AES32_wcSync 0x0100000 #define HDSPM_AES32_wcFreq_bit 22 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ @@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) switch (hdspm->io_type) { case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_wcSync) - return 2; - else if (status & HDSPM_wcLock) - return 1; + if (status & HDSPM_AES32_wcLock) { + if (status & HDSPM_AES32_wcSync) + return 2; + else + return 1; + } return 0; break; @@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, unsigned int status; unsigned int status2; unsigned int timecode; + unsigned int wcLock, wcSync; int pref_syncref; char *autosync_ref; int x; @@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Status:\n"); + wcLock = status & HDSPM_AES32_wcLock; + wcSync = wcLock && (status & HDSPM_AES32_wcSync); + snd_iprintf(buffer, "Word: %s Frequency: %d\n", - (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock", + (wcLock) ? (wcSync ? "Sync " : "Lock") : "No Lock", HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF)); for (x = 0; x < 8; x++) { -- 1.7.1

3 4

[alsa-devel] [PATCH] Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
by David Henningsson 09 Jan '13

09 Jan '13

This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. The original patch was meant to remove clicking, but in fact caused even more clicking instead. Thanks to c4pp4 for doing most of the work with this bug. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- @c4pp4: Is this consistent with what you have done to reduce the clicking? Will you recommend upstream Linux to take the patch below? sound/pci/hda/patch_conexant.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60890bf..dd798c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM -static int conexant_suspend(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); - return 0; -} -#endif - static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; /* -- 1.7.9.5

2 7

Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 09 Jan '13

09 Jan '13

At Tue, 08 Jan 2013 21:25:21 +0100, Vincent Blut wrote: > > Le mardi 08 janvier 2013 à 13:56 +0100, Takashi Iwai a écrit : > > At Tue, 08 Jan 2013 13:28:55 +0100, > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Disable runtime D3 for Intel CPT & co > > > > We've got a few bug reports that the runtime D3 results in the dead > > HD-audio controller. It seems that the problem is in a deeper level > > than the sound driver itself, so as a temporal solution, disable the > > feature for these controllers again. > > > > Reported-by: Vincent Blut <vincent.debian(a)free.fr> > > Reported-by: Maurizio Avogadro <mavoga(a)gmail.com> > > Cc: <stable(a)vger.kernel.org> [v3.7] > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/hda_intel.c | 13 ++++++++----- > > 1 file changed, 8 insertions(+), 5 deletions(-) > > > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index cca8727..0b6aeba 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -573,9 +573,12 @@ enum { > > #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ > > > > /* quirks for Intel PCH */ > > -#define AZX_DCAPS_INTEL_PCH \ > > +#define AZX_DCAPS_INTEL_PCH_NOPM \ > > (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ > > - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) > > + AZX_DCAPS_COUNT_LPIB_DELAY) > > + > > +#define AZX_DCAPS_INTEL_PCH \ > > + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) > > > > /* quirks for ATI SB / AMD Hudson */ > > #define AZX_DCAPS_PRESET_ATI_SB \ > > @@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci) > > static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > > /* CPT */ > > { PCI_DEVICE(0x8086, 0x1c20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* PBG */ > > { PCI_DEVICE(0x8086, 0x1d20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Panther Point */ > > { PCI_DEVICE(0x8086, 0x1e20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Lynx Point */ > > { PCI_DEVICE(0x8086, 0x8c20), > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > Well, my sound controller behaves nicely with the above quirk. > > Tested-by: Vincent Blut <vincent.debian(a)free.fr> OK, I queued the patch now. thanks, Takashi

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Re: [alsa-devel] [PATCH v2 1/3] ASoC: tegra: add ac97 host driver
by Stephen Warren 08 Jan '13

08 Jan '13

On 01/08/2013 03:22 PM, Lucas Stach wrote: > Am Dienstag, den 08.01.2013, 15:10 -0700 schrieb Stephen Warren: >> On 01/04/2013 06:18 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c >>> +static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) >>> +{ >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN); >>> + >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN | >>> + TEGRA20_AC97_CTRL_STM_EN, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN | >>> + TEGRA20_AC97_CTRL_STM_EN); >>> +} >> >> That sets both PCM_DAC_EN and STM_EN, but ... >> >>> +static inline void tegra20_ac97_stop_playback(struct tegra20_ac97 *ac97) >>> +{ >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, 0); >>> + >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN, 0); >>> +} >> >> ... that only clears DAC_EN. Should it clear STM_EN too? >> > To be honest I have no idea what STM really is, seems to be some form of > packed format selection. If not set playback outputs only more or less > random noise. Only PCM_EN controls FIFO and DMA operations though, so > it's ok to just disable this to stop playback. I'd be tempted to just hard-code those bits on in probe() then, but it's not a big deal, so not worth spinning the patch for that. I guess I meant to say, Reviewed-by: Stephen Warren <swarren(a)nvidia.com>

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[alsa-devel] [PATCH 0/3] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 08 Jan '13

08 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Changelog: v2: - don't specify the LPM flag DRIVE_HIGH in #define for the mfp config - patches 2 and 3 are combined into one patch - mfp level and direction are set only once, during driver initialization Mike Dunn (3): ALSA: pxa2xx: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset ALSA: pxa27x: rename pxa27x_assert_ac97reset() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 24 ++++++++++++------ sound/arm/pxa2xx-ac97-lib.c | 34 +++++++++++++++++++++----- 3 files changed, 46 insertions(+), 15 deletions(-) -- 1.7.8.6

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[alsa-devel] [asoc:topic/adsp 4/4] sound/soc/codecs/wm_adsp.c:664:29: sparse: incorrect type in assignment (different modifiers)
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 25326bb263894a528288716901bd434372927844 [4/4] ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks sparse warnings: + sound/soc/codecs/wm_adsp.c:664:29: sparse: incorrect type in assignment (different modifiers) sound/soc/codecs/wm_adsp.c:664:29: expected struct wm_adsp_region *mem sound/soc/codecs/wm_adsp.c:664:29: got struct wm_adsp_region const * sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_load_coeff': sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type [enabled by default] vim +664 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 648 case WMFW_NAME_TEXT: 2159ad93 Mark Brown 2012-10-11 649 case WMFW_INFO_TEXT: 2159ad93 Mark Brown 2012-10-11 650 break; 2159ad93 Mark Brown 2012-10-11 651 case WMFW_ABSOLUTE: 2159ad93 Mark Brown 2012-10-11 652 region_name = "register"; 2159ad93 Mark Brown 2012-10-11 653 reg = offset; 2159ad93 Mark Brown 2012-10-11 654 break; 25326bb2 Mark Brown 2013-01-08 655 25326bb2 Mark Brown 2013-01-08 656 case WMFW_ADSP1_DM: 25326bb2 Mark Brown 2013-01-08 657 case WMFW_ADSP1_ZM: 25326bb2 Mark Brown 2013-01-08 658 case WMFW_ADSP2_XM: 25326bb2 Mark Brown 2013-01-08 659 case WMFW_ADSP2_YM: 25326bb2 Mark Brown 2013-01-08 660 adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", 25326bb2 Mark Brown 2013-01-08 661 file, blocks, le32_to_cpu(blk->len), 25326bb2 Mark Brown 2013-01-08 662 type, le32_to_cpu(blk->id)); 25326bb2 Mark Brown 2013-01-08 663 25326bb2 Mark Brown 2013-01-08 @664 mem = wm_adsp_find_region(dsp, type); 25326bb2 Mark Brown 2013-01-08 665 if (!mem) { 25326bb2 Mark Brown 2013-01-08 666 adsp_err(dsp, "No base for region %x\n", type); 25326bb2 Mark Brown 2013-01-08 667 break; 25326bb2 Mark Brown 2013-01-08 668 } 25326bb2 Mark Brown 2013-01-08 669 25326bb2 Mark Brown 2013-01-08 670 reg = 0; 25326bb2 Mark Brown 2013-01-08 671 list_for_each_entry(alg_region, 25326bb2 Mark Brown 2013-01-08 672 &dsp->alg_regions, list) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [asoc:topic/adsp 4/4] sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 25326bb263894a528288716901bd434372927844 [4/4] ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:465:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm_adsp.c:480:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_load_coeff': sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type [enabled by default] vim +/const +664 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 648 case WMFW_NAME_TEXT: 2159ad93 Mark Brown 2012-10-11 649 case WMFW_INFO_TEXT: 2159ad93 Mark Brown 2012-10-11 650 break; 2159ad93 Mark Brown 2012-10-11 651 case WMFW_ABSOLUTE: 2159ad93 Mark Brown 2012-10-11 652 region_name = "register"; 2159ad93 Mark Brown 2012-10-11 653 reg = offset; 2159ad93 Mark Brown 2012-10-11 654 break; 25326bb2 Mark Brown 2013-01-08 655 25326bb2 Mark Brown 2013-01-08 656 case WMFW_ADSP1_DM: 25326bb2 Mark Brown 2013-01-08 657 case WMFW_ADSP1_ZM: 25326bb2 Mark Brown 2013-01-08 658 case WMFW_ADSP2_XM: 25326bb2 Mark Brown 2013-01-08 659 case WMFW_ADSP2_YM: 25326bb2 Mark Brown 2013-01-08 660 adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", 25326bb2 Mark Brown 2013-01-08 661 file, blocks, le32_to_cpu(blk->len), 25326bb2 Mark Brown 2013-01-08 662 type, le32_to_cpu(blk->id)); 25326bb2 Mark Brown 2013-01-08 663 25326bb2 Mark Brown 2013-01-08 @664 mem = wm_adsp_find_region(dsp, type); 25326bb2 Mark Brown 2013-01-08 665 if (!mem) { 25326bb2 Mark Brown 2013-01-08 666 adsp_err(dsp, "No base for region %x\n", type); 25326bb2 Mark Brown 2013-01-08 667 break; 25326bb2 Mark Brown 2013-01-08 668 } 25326bb2 Mark Brown 2013-01-08 669 25326bb2 Mark Brown 2013-01-08 670 reg = 0; 25326bb2 Mark Brown 2013-01-08 671 list_for_each_entry(alg_region, 25326bb2 Mark Brown 2013-01-08 672 &dsp->alg_regions, list) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [asoc:topic/adsp 3/4] sound/soc/codecs/wm_adsp.c:464:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t'
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 7265971f1bda27fdf97a19db3608ae15e2cdc7d2 [3/4] ASoC: wm_asdp: Validate sanity of algorithm count config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:464:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm_adsp.c:479:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] vim +464 sound/soc/codecs/wm_adsp.c aab034fe Mark Brown 2012-10-26 448 aab034fe Mark Brown 2012-10-26 449 pos = sizeof(adsp2_id) / 2; aab034fe Mark Brown 2012-10-26 450 term = pos + ((sizeof(*adsp2_alg) * algs) / 2); aab034fe Mark Brown 2012-10-26 451 break; aab034fe Mark Brown 2012-10-26 452 aab034fe Mark Brown 2012-10-26 453 default: aab034fe Mark Brown 2012-10-26 454 BUG_ON(NULL == "Unknown DSP type"); aab034fe Mark Brown 2012-10-26 455 return -EINVAL; aab034fe Mark Brown 2012-10-26 456 } aab034fe Mark Brown 2012-10-26 457 aab034fe Mark Brown 2012-10-26 458 if (algs == 0) { aab034fe Mark Brown 2012-10-26 459 adsp_err(dsp, "No algorithms\n"); aab034fe Mark Brown 2012-10-26 460 return -EINVAL; aab034fe Mark Brown 2012-10-26 461 } aab034fe Mark Brown 2012-10-26 462 7265971f Mark Brown 2012-12-19 463 if (algs > 1024) { 7265971f Mark Brown 2012-12-19 @464 adsp_err(dsp, "Algorithm count %d excessive\n", algs); 7265971f Mark Brown 2012-12-19 465 print_hex_dump_bytes(dev_name(dsp->dev), DUMP_PREFIX_OFFSET, 7265971f Mark Brown 2012-12-19 466 buf, buf_size); 7265971f Mark Brown 2012-12-19 467 return -EINVAL; 7265971f Mark Brown 2012-12-19 468 } 7265971f Mark Brown 2012-12-19 469 aab034fe Mark Brown 2012-10-26 470 /* Read the terminator first to validate the length */ aab034fe Mark Brown 2012-10-26 471 ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); aab034fe Mark Brown 2012-10-26 472 if (ret != 0) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [asoc:topic/adsp 1/4] sound/soc/codecs/wm_adsp.c:446:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int'
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: aab034fe60460c0e352a64e94f2323edc12ff296 [1/4] ASoC: wm_adsp: Add support for parsing algorithms config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:446:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] vim +446 sound/soc/codecs/wm_adsp.c aab034fe Mark Brown 2012-10-26 430 } aab034fe Mark Brown 2012-10-26 431 aab034fe Mark Brown 2012-10-26 432 if (algs == 0) { aab034fe Mark Brown 2012-10-26 433 adsp_err(dsp, "No algorithms\n"); aab034fe Mark Brown 2012-10-26 434 return -EINVAL; aab034fe Mark Brown 2012-10-26 435 } aab034fe Mark Brown 2012-10-26 436 aab034fe Mark Brown 2012-10-26 437 /* Read the terminator first to validate the length */ aab034fe Mark Brown 2012-10-26 438 ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); aab034fe Mark Brown 2012-10-26 439 if (ret != 0) { aab034fe Mark Brown 2012-10-26 440 adsp_err(dsp, "Failed to read algorithm list end: %d\n", aab034fe Mark Brown 2012-10-26 441 ret); aab034fe Mark Brown 2012-10-26 442 return ret; aab034fe Mark Brown 2012-10-26 443 } aab034fe Mark Brown 2012-10-26 444 aab034fe Mark Brown 2012-10-26 445 if (be32_to_cpu(val) != 0xbedead) aab034fe Mark Brown 2012-10-26 @446 adsp_warn(dsp, "Algorithm list end %zx 0x%x != 0xbeadead\n", aab034fe Mark Brown 2012-10-26 447 term, be32_to_cpu(val)); aab034fe Mark Brown 2012-10-26 448 aab034fe Mark Brown 2012-10-26 449 alg = kzalloc((term - pos) * 2, GFP_KERNEL); aab034fe Mark Brown 2012-10-26 450 if (!alg) aab034fe Mark Brown 2012-10-26 451 return -ENOMEM; aab034fe Mark Brown 2012-10-26 452 aab034fe Mark Brown 2012-10-26 453 ret = regmap_raw_read(regmap, mem->base + pos, alg, (term - pos) * 2); aab034fe Mark Brown 2012-10-26 454 if (ret != 0) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep the attributions in the case of of_alias_get_id is not negative. sound/soc/mxs/mxs-saif.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..8f17094 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; + else + saif->id = ret; + /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +687,14 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; + else + saif->master_id = ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Factor out calculation of memory base addresses
by Mark Brown 08 Jan '13

08 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6bff303..ed9a6c4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -156,6 +156,26 @@ static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, return NULL; } +static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, + unsigned int offset) +{ + switch (region->type) { + case WMFW_ADSP1_PM: + return region->base + (offset * 3); + case WMFW_ADSP1_DM: + return region->base + (offset * 2); + case WMFW_ADSP2_XM: + return region->base + (offset * 2); + case WMFW_ADSP2_YM: + return region->base + (offset * 2); + case WMFW_ADSP1_ZM: + return region->base + (offset * 2); + default: + WARN_ON(NULL != "Unknown memory region type"); + return offset; + } +} + static int wm_adsp_load(struct wm_adsp *dsp) { const struct firmware *firmware; @@ -283,27 +303,27 @@ static int wm_adsp_load(struct wm_adsp *dsp) case WMFW_ADSP1_PM: BUG_ON(!mem); region_name = "PM"; - reg = mem->base + (offset * 3); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_DM: BUG_ON(!mem); region_name = "DM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_XM: BUG_ON(!mem); region_name = "XM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_YM: BUG_ON(!mem); region_name = "YM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_ZM: BUG_ON(!mem); region_name = "ZM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; default: adsp_warn(dsp, -- 1.7.10.4

1 1

Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 08 Jan '13

08 Jan '13

At Fri, 28 Dec 2012 15:25:40 +0100, Vincent Blut wrote: > > Hi, > > Since I updated to Linux 3.7.1, listening to some audio/video bits > frequently cause the following: > > [ 7896.166946] hda-intel: azx_get_response timeout, switching to polling > mode: last cmd=0x020c0000 > [ 7897.173444] hda-intel: No response from codec, disabling MSI: last > cmd=0x020c0000 > [ 7898.179932] hda_intel: azx_get_response timeout, switching to > single_cmd mode: last cmd=0x020c0000 > [ 7898.179983] hda-codec: out of range cmd 0:0:20:400:fffff7ff > [ 9445.034371] plugin-containe[5873]: segfault at 7f44bb95e639 ip > 00007f44e454bca0 sp 00007f44c91165f8 error 4 in > libc-2.13.so[7f44e442c000+180000] > > It seems to be a codec probing failure (?). This is really fatal because > the sound become very choppy and can't recover until I reboot. > I'll try to play with 'probe_mask' kernel parameter to see if I can > narrow the correct codec slots! > > By the way I can't reproduce this on 3.6.9, so is there something that > changed in this area in 3.7.1? If it's new in 3.7, this could be a regression by runtime D3. Try to pass power_save_controller=0 option to snd-hda-intel module (or change it via sysfs dynamically). thanks, Takashi

4 10

[alsa-devel] [PATCH] ALSA: hda - print power state for AFG node in proc file
by David Henningsson 08 Jan '13

08 Jan '13

It seems useful, and power states are required for AFG nodes, so I see no reason not to print it. As a bonus, also print the AFG nid. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_proc.c | 2 ++ 1 file changed, 2 insertions(+) (Feel free to rephrase "State of AFG node 0x%02x:\n" if you like.) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 045e5d3..d6c8416 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -603,6 +603,8 @@ static void print_codec_info(struct snd_info_entry *entry, print_amp_caps(buffer, codec, codec->afg, HDA_INPUT); snd_iprintf(buffer, "Default Amp-Out caps: "); print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT); + snd_iprintf(buffer, "State of AFG node 0x%02x:\n", codec->afg); + print_power_state(buffer, codec, codec->afg); nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); if (! nid || nodes < 0) { -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..a920206 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,9 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +684,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH RESEND v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

2 3

[alsa-devel] [PATCH v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

2 5

Re: [alsa-devel] About Alsa arecord
by Raymond Yau 08 Jan '13

08 Jan '13

> Good morning! > Are you saying I should modify the source code of alsa-driver-1.0.25? >> >> > Hello! >> > There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? >> >> In alc_auto_create_input_ctls , add the connection from node 0x0b of mixer ADC as input source similar to those pin complex >> >> Those Mic/line/CD playback volume and switches of node 0x0b affect the signal of the new input source ”mixer” > > Yes, also need a switch which mute the input from 0x0b for all output jacks to prevent feedback if using mic the current hda driver only support stereo capture and you have to implement multi channel capturing similar to multi channel playback in hda_codec.c snd_hda_multi_in_analog_prepare for (i = 1; i < num_adcs; i++) { snd_hda_codec_setup_stream(codec, adc_nids[i], stream_tag, i * 2, format); or try multi plugin of two stereo capture devices

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
by Mark Brown 07 Jan '13

07 Jan '13

Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5d8bb1d..a765458 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/4] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 07 Jan '13

07 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Mike Dunn (4): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset bug work-around code ALSA: pxa27x: ac97 controller driver requests gpio ALSA: pxa27x: rename pxa27x_assert_ac97() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 38 +++++++++++++++++++------- sound/arm/pxa2xx-ac97-lib.c | 26 +++++++++++++----- 3 files changed, 50 insertions(+), 17 deletions(-) -- 1.7.8.6

4 21

[alsa-devel] [PATCH v2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..71ae23d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,6 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } +/* update mute-LED according to the speaker mute state via mic1 VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + /* update mute-LED according to the speaker mute state via mic2 VREF pin */ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { @@ -6043,6 +6067,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6196,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 1

[alsa-devel] Alsa snapshot directory layout changed
by David Henningsson 07 Jan '13

07 Jan '13

I have a script that picks daily snapshots from ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-snapshot.tar.gz ...and I just noticed that the directory layout has changed, so what was "alsa-driver" is now "alsa-driver/alsa". Is this expected, i e, should I update my script to follow this new layout, or was it a mistake? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 3

[alsa-devel] [PATCH] ALSA: pxa2xx: fix ac97 cold reset for pxa27x
by Mike Dunn 07 Jan '13

07 Jan '13

Currently, ac97 reset is broken on PXA27x. Through trial-and-error (the pxa270 developer's manual is mostly incoherent on the topic of ac97 reset), I fixed it by setting the WARM_RST bit in the GCR register at the end of the cold reset, and then skipping the warm reset if the link is already up and running. It appears that setting the WARM_RST bit is a necessary final step during cold reset. I think that the PXA25x and PXA3xx may currently be working correctly because WARM_RST is set within the warm reset routine, and the codec drivers always follow a cold reset with a warm reset during their initialization. so this combination effectively completes a cold reset. This doesn't work on the PXA27x because its warm reset routine contains additional code for working around a hardware bug in the warm reset sequence, which causes the reset sequence to fail. I only have a PXA27x platform for testing, so this patch only affects the PXA27x. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- As an aside... I don't understand how always following a cold reset with a warm reset (as the codec drivers do) makes sense. Unless I'm mistaken, a warm reset can only occur on a link that's currently down. All the timing diagrams I've seen for reset (warm and cold) show the link down (BITCLK stopped) at the start of the reset sequence. The SYNC line is used to effect the warm reset, and SYNC is already pulsing every frame on a link that's up and running, so it seems physically impossible to do a warm reset on a link that's currently up. But I'm no ac97 expert, so I may just be embarassing myself. Any insight gratefully received. Thanks! sound/arm/pxa2xx-ac97-lib.c | 15 +++++++++++++++ 1 files changed, 15 insertions(+), 0 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fc0ae9..aa242ff 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -138,6 +138,10 @@ static inline void pxa_ac97_warm_pxa27x(void) { gsr_bits = 0; + /* don't attempt warm reset if link is up */ + if (GSR & (GSR_SCR | GSR_PCR)) + return; + /* warm reset broken on Bulverde, so manually keep AC97 reset high */ pxa27x_assert_ac97reset(reset_gpio, 1); udelay(10); @@ -148,6 +152,8 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { + unsigned int timeout; + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -159,6 +165,15 @@ static inline void pxa_ac97_cold_pxa27x(void) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); + + /* + * Setting the WARM_RST bit is a necessary part of a cold reset, at + * least on the buggy and poorly documented PXA27x ac97 controller. + */ + GCR |= GCR_WARM_RST; + timeout = 100; /* wait for the codec-ready bit to be set */ + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); } #endif -- 1.7.8.6

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[alsa-devel] [PATCH 1/2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) I know this is a copy-paste, but I didn't really find a better and more condensed way to do it. Perhaps if we get a third variation it would make sense to write a more generic version. Also, Alsa-info is available at https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1096789/+attachment/34… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..2e6803e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,7 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } -/* update mute-LED according to the speaker mute state via mic2 VREF pin */ +/* update mute-LED according to the speaker mute state via mic VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -6043,6 +6066,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6195,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 4

[alsa-devel] About Alsa arecord
by FF 07 Jan '13

07 Jan '13

Hello! There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? Thank you!Best regards.

2 1

[alsa-devel] Roland-Boss RC-300 support
by Jean-Samuel Reynaud 06 Jan '13

06 Jan '13

Hi All, I have a Boss RC-300 loop station with an USB connector. The windows driver add an audio device but it was not working under linux. So please find attached, the patch for supporting audio in this device. Note that midi is not working yet but I'm working on it... Regards,

2 1

Re: [alsa-devel] Missing exactly 3 of 8 audio packets?
by Daniel Griscom 06 Jan '13

06 Jan '13

Dear Daniel (M) and Clemens, I'd like to conclude this thread if I could. It seems clear that configuring our USB audio card's descriptor with a bInterval value of 1 (a packet every microframe) was too aggressive, and was part of what caused the packet stuttering issue. (Right?) We almost certainly should increase bInterval; I'll try 2, and if that seems stable will probably go to 3 (4 microframes) for some margin. However, I'm still concerned that the symptoms don't sound like a simple starvation issue, where I'd expect to have a missing frame every once in a while. Instead, we were seeing reliable operation interspersed with periods where lots of packets were being consistently omitted. To me, this suggests that there's some other issue going on, either in the USB stack or in the USB hardware itself. Given that, I'm worried that increasing bInterval will just move the symptoms around, making it harder to duplicate but not truly becoming robust. Any thoughts on this? Should I try chasing the issue into the Linux USB development mailing list, or will they just say "set bInterval to a larger value!"? Thanks, Dan -- Daniel T. Griscom griscom(a)suitable.com Suitable Systems http://www.suitable.com/ 1 Centre Street, Suite 204 (781) 665-0053 Wakefield, MA 01880-2400

2 1

Re: [alsa-devel] [PATCH v3 0/5] ARM: OMAP4+: HDMI: Update platform devices for audio
by Ricardo Neri 05 Jan '13

05 Jan '13

Hi Mark, Tomi, Just wondering if you had a chance to take a look at these patches. Thanks! Ricardo On 12/17/2012 07:30 PM, Ricardo Neri wrote: > Hi Mark, Tomi, Liam, Tony, > > This set aims to be the version 3 of my previous submission[1] and aims to > address the comments that Mark and Tomi kindly provided on such submission. > > The creation of the platform device for the HDMI audio interface from within the > OMAPDSS HDMI driver that was previously submitted[2] is resubmitted to be > complemented with code to relocate to arch/arm/mach-omap2/display.c the > creation of the platform devices for the HDMI ASoC codec and card drivers. This > series does not break the HDMI audio functionality in any patch. > > Also, the names of the platform devices are changed to give them more logical > and more descriptive names. > > As the commit > > commit 14840b9a83c6a56629db2ba0ec247503e975f143 > Author: Ricardo Neri <ricardo.n...(a)ti.com> > Date: Tue Nov 6 00:19:17 2012 -0600 > > OMAPDSS: HDMI: Create platform device for audio support > > is reverted in Tomi's git://gitorious.org/linux-omap-dss2/linux.git master > branch, this series applies cleanly. > > Changes from v1: > *Put in a single series all the patches related to platform device updates. > *Now HDMI audio works correctly in every patch. > *Remove reference to the TPD12S015 HDMI companion chip as the ASoC drivers > are not aware of this and other chips could be used in the future. > > Changes from v2: > *Split in two patches the renaming and the relocation of the platform devices. > *Create a separate patch to pass only the address offset of the DMA data port > to the audio interface platform device. > *Keep the name hdmi-audio-codec to not refer to explicitly to OMAP. The codec > can be made generic in a different patch series submitted to alsa-devel. > > BR, > > Ricardo > > > [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg80785.html > [2]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html > > Ricardo Neri (5): > ARM: OMAP4+: HDMI: Rename platform devices for ASoC drivers > ARM: OMAP4: Assign IDs to DSS HDMI devices > ARM4: OMAP4+: HDMI: Relocate devices for audio codec and card > ARM: OMAP4+: HDMI: Relocate the device for audio interface > ARM: OMAP4+: HDMI: Refine the DMA port resource for audio > > arch/arm/mach-omap2/board-4430sdp.c | 9 ++--- > arch/arm/mach-omap2/board-omap4panda.c | 9 ++--- > arch/arm/mach-omap2/devices.c | 31 ---------------- > arch/arm/mach-omap2/display.c | 31 ++++++++++++++++ > drivers/video/omap2/dss/hdmi.c | 62 ++++++++++++++++++++++++++++++++ > sound/soc/omap/omap-hdmi-card.c | 4 +-- > sound/soc/omap/omap-hdmi.c | 5 ++- > sound/soc/omap/omap-hdmi.h | 2 -- > 8 files changed, 103 insertions(+), 50 deletions(-) > > -- > 1.7.10.4 >

2 2

[alsa-devel] [PATCH] ASoC: wm2000: Fix sense of speech clarity enable
by Mark Brown 04 Jan '13

04 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f..12bcae6 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); if (wm2000->speech_clarity) - ret &= ~WM2000_SPEECH_CLARITY; - else ret |= WM2000_SPEECH_CLARITY; + else + ret &= ~WM2000_SPEECH_CLARITY; wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: arizona: Allow runtime reconfiguration of the output mode
by Mark Brown 04 Jan '13

04 Jan '13

Some systems use external analogue switches to connect more analogue devices to the CODEC than are supported by the device. In some systems this requires changing the switched output from single ended to differential mode dynamically at runtime. Add a new function arizona_set_output_mode() to support this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 3 +++ 2 files changed, 37 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7724b4b..6d98cf4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1176,6 +1176,40 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, } EXPORT_SYMBOL_GPL(arizona_init_fll); +/** + * arizona_set_output_mode - Set the mode of the specified output + * + * @codec: Device to configure + * @output: Output number + * @diff: True to set the output to differential mode + * + * Some systems use external analogue switches to connect more + * analogue devices to the CODEC than are supported by the device. In + * some systems this requires changing the switched output from single + * ended to differential mode dynamically at runtime, an operation + * supported using this function. + * + * Most systems have a single static configuration and should use + * platform data instead. + */ +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) +{ + unsigned int reg, val; + + if (output < 1 || output > 6) + return -EINVAL; + + reg = ARIZONA_OUTPUT_PATH_CONFIG_1L + (output - 1) * 8; + + if (diff) + val = ARIZONA_OUT1_MONO; + else + val = 0; + + return snd_soc_update_bits(codec, reg, ARIZONA_OUT1_MONO, val); +} +EXPORT_SYMBOL_GPL(arizona_set_output_mode); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 68dfae5..697ff6f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,4 +213,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_dai(struct arizona_priv *priv, int dai); +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, + bool diff); + #endif -- 1.7.10.4

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Re: [alsa-devel] draft -- ALSA firewire + digi003
by Robin Gareus 04 Jan '13

04 Jan '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Clemens, Hi ALSA-devs, I'm working with Damien Zammit on the Digidesign-003 driver which is based on your alsa-kprivate dice code. We were able to work out most of the issues so far but are now stuck with timing problems. More specifically: the jitter of async message timing is quite large and the AMDTP ISO stream has regular drop-outs. Comparing 1394 data dumps, OSX sends async messages at regular 125 (+-5us) intervals. On GNU/Linux we've seen a delta -40us, + 120us. For async messages this is not a big deal. But.. Just playing audio via aplay: there are short dropouts every couple of seconds but otherwise the devices plays fine indefinitely. Interestingly, when running cyclictest while playing sound: there are no dropouts but the stream drops after ~10-15 seconds. Any hints why that may be or how to debug the issue further? In order to rule out kernel related issues (3.0.0 is quite old), we've rebased the work (and alsa-kprivate's firewire-kernel-streaming branch) onto alsa-driver.git. The issue remains the same with 3.2.0-4-rt Alas, neither of us has detailed knowledge nor specs for AMDTP, not to mention in-depth knowledge about Linux' firewire stack timings. Any help or leads are appreciated. As we believe we are close to a fully functional driver for the 003, we are very keen to resolve the following two separate issues: 1) The timing problem we have observed with AMDTP over firewire. 2) A framework for the capture stream in ALSA firewire. The rebased driver code is at git://gareus.org/alsa-driver.git http://gareus.org/gitweb/?p=alsa-driver.git The broken-out digi003 driver is available from: https://github.com/zamaudio/snd-firewire-003 https://github.com/x42/003amdtp 1394 bus data dumps of both OSX as well as Linux can be found at http://www.zamaudio.com/?p=715 Cheers! robin -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJQ5t9CAAoJEKCQvOAs9X8ELbsP/Re0wxZKU6v2rbBVhV/FC6zr JYmx/jAsmoC7vtVZMKE4gGcCreszX58oU7RG323qR9iKSEgXeqtuThZSTT6l3N39 KpHW4T094FyQxmvxRYrZnRlHgWexvLfJT/YiuTUX50L/CKgS75ajRfLqADZaHSy5 Oc+CEy50Jmoi/9/dTuM0pVPVvt5APFqLyAisw1/5wzhYMTHF0mBByn5AHXBp7bIl SVg7IoJuTNd4dYgo3PtKw9OqhjMGZ1g5DNHA9OaE7qY12r3TMMKm7Sq2iWS4HQoq RMQLOVtyMOVUKyddK8ubWi0Cm2lYjHUXUNSLyZLcdWYTWbk5eWo6Ze83Tbl33h4T ngiWt+DWpqKpm+U0UQDOU6p+DSAosKHsVcPPPkYVkwXKAkHctXKlPUst5NHEhhK3 5mbVQECHE9+BBy9qY2eimDZ/p3+5C2kIMofDga6jiuFWfHMqKp30uacUHwP4xY38 jHLaTmpX7VNwNYGnFLvGve709ejNjL8ByFKrUKpV1nAMTO4QP4SipHuApBF3BWS7 hAGBD6mKQAPl6AhvuJaxtCdTQc/tkAF0Ep1LClthsHnyhqPVCX+L5sb/z7vVmCkp cFxGXJq6TujcW4kkKPfZCNsnHqea8lgY6tEonc8aRmcIZvF+L4Yp+p4/r9OxL6ma xIIgIVDp1X9HxVayBDWL =lVsY -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Remove DSP B and left justified AIF modes
by Mark Brown 04 Jan '13

04 Jan '13

These are not supported. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a4d7fb9..7724b4b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -503,15 +503,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mode = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mode = 1; - break; case SND_SOC_DAIFMT_I2S: mode = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mode = 3; - break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); -- 1.7.10.4

1 1

Re: [alsa-devel] sound: usb-audio: [PATCH]x2 full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 04 Jan '13

04 Jan '13

On 23 December 2012 14:36, Damien Zammit <damien.zammit(a)gmail.com> wrote: > I have fixed both the kernel panic and the broken pipe issue with midi. > I believe this completes the patch, I just don't know how to submit it to you. Please find attached the patch you accepted previously: (Version 5) Plus an incremental patch (Version 5-2) ***Please ignore patch (Version 6)*** I hope this is the easiest way for you to review it. It compiles, runs, and works on the hardware now without errors. Thanks, Damien

2 6

Re: [alsa-devel] [PATCH] sound: Add support for Creative BT-D1 via usb sound quirks
by Takashi Iwai 04 Jan '13

04 Jan '13

At Thu, 03 Jan 2013 12:59:07 +0100, Alexander Schremmer wrote: > > On 03.01.2013 10:58, Takashi Iwai wrote: > > > Thanks for the patch. The changes look OK, but could you fix trivial > > coding issues reported by scripts/checkpatch.pl and resend? > > Oops, my sending workflow ate the tabs, here it is again (hopefully now > working correctly): > > > From: Alexander Schremmer <alex(a)alexanderweb.de> > > Support the Creative BT-D1 Bluetooth USB audio device. Before this > patch, Linux had trouble finding the correct USB descriptors and bailed > out with these messages: > > no or invalid class specific endpoint descriptor > > Now it still prints these messages on hotplug: > > snd-usb-audio: probe of ...:1.0 failed with error -5 > snd-usb-audio: probe of ...:1.2 failed with error -5 > snd-usb-audio: probe of ...:1.3 failed with error -5 > > But the device works correctly, including the HID support. > > The patch is diff'ed against 3.8-rc1 but should apply to older kernels > as well. > > Signed-off-by: Alexander Schremmer <alex(a)alexanderweb.de> Thanks, applied. Takashi > > --- > diff -uprN linux-3.8-rc1/sound/usb/quirks-table.h > linux-3.8-rc1-mod/sound/usb/quirks-table.h > --- linux-3.8-rc1/sound/usb/quirks-table.h 2012-12-22 02:19:00.000000000 > +0100 > +++ linux-3.8-rc1-mod/sound/usb/quirks-table.h 2012-12-26 > 11:41:28.000000000 +0100 > @@ -50,6 +50,28 @@ > } > }, > > +{ > + /* Creative BT-D1 */ > + USB_DEVICE(0x041e, 0x0005), > + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > + .ifnum = 1, > + .type = QUIRK_AUDIO_FIXED_ENDPOINT, > + .data = &(const struct audioformat) { > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels = 2, > + .iface = 1, > + .altsetting = 1, > + .altset_idx = 1, > + .endpoint = 0x03, > + .ep_attr = USB_ENDPOINT_XFER_ISOC, > + .attributes = 0, > + .rates = SNDRV_PCM_RATE_CONTINUOUS, > + .rate_min = 48000, > + .rate_max = 48000, > + } > + } > +}, > + > /* Creative/Toshiba Multimedia Center SB-0500 */ > { > USB_DEVICE(0x041e, 0x3048), >

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[alsa-devel] [sound:for-linus 4/4] sound/oss/pas2_card.c:338:5: warning: too many arguments for format
by kbuild test robot 04 Jan '13

04 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: b56458b36c8922d1d57c2320151194059dfb9a31 commit: b56458b36c8922d1d57c2320151194059dfb9a31 [4/4] sound: oss/pas2: Fix possible access out of array config: make ARCH=x86_64 allyesconfig All warnings: sound/oss/pas2_card.c: In function 'attach_pas_card': sound/oss/pas2_card.c:338:5: warning: too many arguments for format [-Wformat-extra-args] vim +338 sound/oss/pas2_card.c ^1da177e Linus Torvalds 2005-04-16 322 return pas_model; ^1da177e Linus Torvalds 2005-04-16 323 } ^1da177e Linus Torvalds 2005-04-16 324 ^1da177e Linus Torvalds 2005-04-16 325 static void __init attach_pas_card(struct address_info *hw_config) ^1da177e Linus Torvalds 2005-04-16 326 { ^1da177e Linus Torvalds 2005-04-16 327 pas_irq = hw_config->irq; ^1da177e Linus Torvalds 2005-04-16 328 ^1da177e Linus Torvalds 2005-04-16 329 if (detect_pas_hw(hw_config)) ^1da177e Linus Torvalds 2005-04-16 330 { ^1da177e Linus Torvalds 2005-04-16 331 ^1da177e Linus Torvalds 2005-04-16 332 if ((pas_model = pas_read(0xFF88))) ^1da177e Linus Torvalds 2005-04-16 333 { ^1da177e Linus Torvalds 2005-04-16 334 char temp[100]; ^1da177e Linus Torvalds 2005-04-16 335 b56458b3 Asim Kadav 2013-01-03 336 if (pas_model < 0 || b56458b3 Asim Kadav 2013-01-03 337 pas_model >= ARRAY_SIZE(pas_model_names)) { b56458b3 Asim Kadav 2013-01-03 @338 printk(KERN_ERR, "pas2 unrecognized model.\n"); b56458b3 Asim Kadav 2013-01-03 339 return; b56458b3 Asim Kadav 2013-01-03 340 } ^1da177e Linus Torvalds 2005-04-16 341 sprintf(temp, ^1da177e Linus Torvalds 2005-04-16 342 "%s rev %d", pas_model_names[(int) pas_model], ^1da177e Linus Torvalds 2005-04-16 343 pas_read(0x2789)); ^1da177e Linus Torvalds 2005-04-16 344 conf_printf(temp, hw_config); ^1da177e Linus Torvalds 2005-04-16 345 } ^1da177e Linus Torvalds 2005-04-16 346 if (config_pas_hw(hw_config)) --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [Tiny6410/Mini6410] Sound support
by Alexander Nestorov 03 Jan '13

03 Jan '13

I'm working on a Tiny6410 device (S3C6410) on top of a 3.7 kernel and sound isn't working with the default settings. I also wasn't able to make it work enablig the related stuff to AC97 in menuconfig so I'm wondering if there is actually any support for this device? Looking at the code in sound/soc, there is something for the SMDK6410 but I'm not really sure if that's compatible with my device or even if that code is fully functional. Can I get some help with this, please? I'll appreciate any kind of help. Regards PS: I'm including Kukjin Kim and Sangbeom Kim as of recomendation of <broonie> from #alsa-soc at Freenode -- alexandernst

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[alsa-devel] [PATCH v2] ASoC: pxa2xx-ac97: fix oops when codec_driver probe runs
by Mike Dunn 03 Jan '13

03 Jan '13

Change log: v2: pxa2xx_ac97_hw_probe() is called *before* snd_soc_register_dais(), and likewise pxa2xx_ac97_hw_remove() is called after snd_soc_unregister_dais(). Thanks much Mark. This patch fixes a kernel oops that occurs when the snd_soc_codec_driver probe function runs. The exception occurs when the codec calls soc_ac97_ops.reset(). The pxa2xx-ac97's clocks have not been initialized at this point, but the cpu's reset function (pxa_ac97_cold_pxa27x()) expects them to be, resulting in a NULL pointer dereference in the pxa's clk_enable(). The pxa2xx-ac97's clocks are initialized in pxa2xx_ac97_hw_probe(). To ensure that the clocks are initialized when the codec_driver probe runs, the calls to pxa2xx_ac97_hw_{probe,remove}() are moved from the cpu's dai_driver probe/remove functions to those of its platform_driver. This is necessary because snd_soc_instantiate_card() always calls the codec_driver probe before any of the dai_drivers' probes. This change leaves nothing in the cpu's dai_driver probe/remove functions, so those functions are eliminated. Tested on the pxa27x; palmtreo680 machine using the wm9712 codec. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- sound/soc/pxa/pxa2xx-ac97.c | 27 ++++++++++----------------- 1 files changed, 10 insertions(+), 17 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009..ba04f49 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -104,17 +104,6 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -184,8 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, .playback = { @@ -236,15 +223,20 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } - /* Punt most of the init to the SoC probe; we may need the machine - * driver to do interesting things with the clocking to get us up - * and running. - */ + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret < 0) { + dev_err(&pdev->dev, "pxa2xx_ac97_hw_probe failed with %d\n", + ret); + return ret; + } + return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver)); } @@ -252,6 +244,7 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); + pxa2xx_ac97_hw_remove(pdev); return 0; } -- 1.7.8.6

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[alsa-devel] [PATCH] ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrol
by David Henningsson 03 Jan '13

03 Jan '13

The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted, but "Mute-LED Mode" being "On" refers to the LED being on, not the volume being on. Therefore "On" and "Off" should be switched. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8353c77..b8fb0a5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts[] = { - "Off", "On", "Follow Master" + "On", "Off", "Follow Master" }; unsigned int index; -- 1.7.9.5

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Re: [alsa-devel] [PATCH] sound: Add support for Creative BT-D1 via usb sound quirks
by Takashi Iwai 03 Jan '13

03 Jan '13

At Sat, 29 Dec 2012 14:26:25 +0100, Alexander Schremmer wrote: > > From: Alexander Schremmer <alex(a)alexanderweb.de> > > Support the Creative BT-D1 Bluetooth USB audio device. Before this > patch, Linux had trouble finding the correct USB descriptors and bailed > out with these messages: > > no or invalid class specific endpoint descriptor > > Now it still prints these messages on hotplug: > > snd-usb-audio: probe of ...:1.0 failed with error -5 > snd-usb-audio: probe of ...:1.2 failed with error -5 > snd-usb-audio: probe of ...:1.3 failed with error -5 > > But the device works correctly, including the HID support. > > The patch is diff'ed against 3.8-rc1 but should apply to older kernels > as well. > > Signed-off-by: Alexander Schremmer <alex(a)alexanderweb.de> Thanks for the patch. The changes look OK, but could you fix trivial coding issues reported by scripts/checkpatch.pl and resend? Takashi > > --- > diff -uprN linux-3.8-rc1/sound/usb/quirks-table.h > linux-3.8-rc1-mod/sound/usb/quirks-table.h > --- linux-3.8-rc1/sound/usb/quirks-table.h 2012-12-22 > 02:19:00.000000000 +0100 > +++ linux-3.8-rc1-mod/sound/usb/quirks-table.h 2012-12-26 > 11:41:28.000000000 +0100 > @@ -50,6 +50,28 @@ > } > }, > > +{ > + /* Creative BT-D1 */ > + USB_DEVICE(0x041e, 0x0005), > + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > + .ifnum = 1, > + .type = QUIRK_AUDIO_FIXED_ENDPOINT, > + .data = &(const struct audioformat) { > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels = 2, > + .iface = 1, > + .altsetting = 1, > + .altset_idx = 1, > + .endpoint = 0x03, > + .ep_attr = USB_ENDPOINT_XFER_ISOC, > + .attributes = 0, > + .rates = SNDRV_PCM_RATE_CONTINUOUS, > + .rate_min = 48000, > + .rate_max = 48000, > + } > + } > +}, > + > /* Creative/Toshiba Multimedia Center SB-0500 */ > { > USB_DEVICE(0x041e, 0x3048), >

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Re: [alsa-devel] [PATCH] pas2: Added a fix where a sound card failure should not result in reading beyond array memory index.
by Takashi Iwai 03 Jan '13

03 Jan '13

At Fri, 28 Dec 2012 12:28:57 -0600, Asim Kadav wrote: > > Signed-off-by: Asim Kadav <kadav(a)cs.wisc.edu> > --- > sound/oss/pas2_card.c | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c > index dabf8a8..0b8cf64 100644 > --- a/sound/oss/pas2_card.c > +++ b/sound/oss/pas2_card.c > @@ -332,6 +332,10 @@ static void __init attach_pas_card(struct address_info *hw_config) > if ((pas_model = pas_read(0xFF88))) > { > char temp[100]; > + if ((int )pas_model < 0 || (int )pas_model >= 5) { Try to run scripts/checkpatch.pl. You'll get a few errors there. But, you can simply remove the cast to int instead of fixing the coding style issues since the cast is superfluous. Also, use ARRAY_SIZE() instead of a magic number 5. > + printk ("pas2card unrecognized model.\n"); Put a KERN_ERR prefix. thanks, Takashi > + return; > + } > > sprintf(temp, > "%s rev %d", pas_model_names[(int) pas_model], > -- > 1.7.11.3

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[alsa-devel] ASoC: EA3250 V2 machine driver
by Bryan Kaminski 02 Jan '13

02 Jan '13

Hello, I have an Embedded Artists LPC3250 V2 development board, which has the NXP UDA1380 audio codec on-board. I'm wondering if someone could help me get started on making an ASoC driver for it. There already exists a machine driver for a similar Phytec 3250 development board, with the same audio codec. I think it should be possible to modify the existing Phytec driver, but I'm unable to accomplish that. I modified some source and everything compiled, but I can't get the driver to load correctly. After loading the module with 'modprobe', I'm unable to get ALSA to recognize any sound cards. Can anyone give me a jump-start on this? Sorry if this is the wrong list to post on. I would post at the LPCLinux official site but it has been down for multiple weeks now. :| Bryan

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[alsa-devel] [PATCH 0/2] ASoC: palm27x: machine driver fixes and updates
by Mike Dunn 02 Jan '13

02 Jan '13

Hi, A fix and an update to the palm27x machine driver. Happy New Year! Mike Dunn (2): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe sound/soc/pxa/palm27x.c | 36 +++++++++--------------------------- 1 files changed, 9 insertions(+), 27 deletions(-) -- 1.7.8.6

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[alsa-devel] [PATCH 0/8] ASoC: OMAP-twl4030: Add support for more3 boards
by Peter Ujfalusi 02 Jan '13

02 Jan '13

Hello, This series will update the omap-twl4030 machine driver and the twl4030 codec driver to enable support for more boards. With this change we can remove two more machine drivers: zoom2 and sdp3430 since they will be supported by the common omap-twl4030 machine driver. The arch/arm/mach-omap2/ patches will go as a separate series to avoid cross tree issues. Regards, Peter --- Peter Ujfalusi (8): ASoC: twl4030: Correct the support for Voice port ASoC: zoom2: No need to configure the Voice port anymore ASoC: sdp3430: No need to configure the Voice port anymore ASoC: twl4030: Convert MICBIAS to SUPPLY widget ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use ASoC: sdp3430: No need to configure pin mux for extmute ASoC: omap-twl4030: Add support for routing, voice port and jack detect ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 .../devicetree/bindings/sound/omap-twl4030.txt | 46 ++++ sound/soc/codecs/twl4030.c | 71 ++++-- sound/soc/omap/Kconfig | 19 +- sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-twl4030.c | 204 ++++++++++++++- sound/soc/omap/omap3pandora.c | 8 +- sound/soc/omap/sdp3430.c | 278 --------------------- sound/soc/omap/zoom2.c | 207 --------------- 8 files changed, 304 insertions(+), 533 deletions(-) delete mode 100644 sound/soc/omap/sdp3430.c delete mode 100644 sound/soc/omap/zoom2.c -- 1.8.0.2

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[alsa-devel] [PATCH] ASoC: pxa2xx-ac97: fix oops when codec_driver probe runs
by Mike Dunn 02 Jan '13

02 Jan '13

This patch fixes a kernel oops that occurs when the snd_soc_codec_driver probe function runs. The exception occurs when the codec calls soc_ac97_ops.reset(). The pxa2xx-ac97's clocks have not been initialized at this point, but the cpu's reset function (pxa_ac97_cold_pxa27x()) expects them to be, resulting in a NULL pointer dereference in the pxa's clk_enable(). The pxa2xx-ac97's clocks are initialized in pxa2xx_ac97_hw_probe(). To ensure that the clocks are initialized when the codec_driver probe runs, the calls to pxa2xx_ac97_hw_{probe,remove}() are moved from the cpu's dai_driver probe/remove functions to those of its platform_driver. This is necessary because snd_soc_instantiate_card() always calls the codec_driver probe before any of the dai_drivers' probes. This change leaves nothing in the cpu's dai_driver probe/remove functions, so those functions are eliminated. Tested on the pxa27x; palmtreo680 machine using the wm9712 codec. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- I'm new to the ASoC code. Experts, please have a look. Thanks! sound/soc/pxa/pxa2xx-ac97.c | 38 +++++++++++++++++++------------------- 1 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4b0a009..940e9f4 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -104,17 +104,6 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -184,8 +173,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, .resume = pxa2xx_ac97_resume, .playback = { @@ -236,21 +223,34 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int ret; + if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); return -ENXIO; } - /* Punt most of the init to the SoC probe; we may need the machine - * driver to do interesting things with the clocking to get us up - * and running. - */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, - ARRAY_SIZE(pxa_ac97_dai_driver)); + ret = snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, + ARRAY_SIZE(pxa_ac97_dai_driver)); + if (ret < 0) { + dev_err(&pdev->dev, "snd_soc_register_dais failed with %d\n", + ret); + return ret; + } + + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret < 0) { + dev_err(&pdev->dev, "pxa2xx_ac97_hw_probe failed with %d\n", + ret); + snd_soc_unregister_dais(&pdev->dev, + ARRAY_SIZE(pxa_ac97_dai_driver)); + } + return ret; } static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + pxa2xx_ac97_hw_remove(pdev); snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); return 0; } -- 1.7.8.6

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[alsa-devel] Looking for ALSA developer
by Eben Upton 01 Jan '13

01 Jan '13

We're looking for someone to do some contract development on the ALSA driver for the Raspberry Pi. Would anyone here be interested in taking this on? A little background: Our driver is architecturally fairly similar to a USB sound card driver, sitting on top of a stream-like connection to a remote entity implemented in software on an internal DSP. Due to our limited understanding of ALSA internals, there are a number of bugs which lead to underrun conditions in common audio applications, and we'd like to pay someone on and hourly basis to go through the failing cases, root cause them, and propose patches. Eben Upton Executive Director, Raspberry Pi Foundation

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[alsa-devel] [PATCH 0/6] ARM: OMAP3: Audio support update for omap-twl4030
by Peter Ujfalusi 31 Dec '12

31 Dec '12

Hello, omap-twl4030 ASoC machine driver can support more boards with the following series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/058119.h… Rework the omap-twl4030 audio support code in arch/arm/mach-omap2/ to take advantages of this. The same machine driver can support the zoom2 and sdp3430 boards as well. Regards, Peter --- Peter Ujfalusi (6): ARM: OMAP: 3430sdp: Enable extmute functionality for audio ARM: OMAP: zoom: Zoom2 does not have extmute functionality ARM: OMAP2+: twl-common: Add default twl4030 audio configuration ARM: OMAP2+: twl-common: Allow boards to customize the twl4030 audio setup ARM: OMAP: zoom: Audio support via the common omap-twl4030 machine driver ARM: OMAP: sdp3430: Audio support via the common omap-twl4030 machine driver arch/arm/mach-omap2/board-3430sdp.c | 20 +++++++++++++++++++ arch/arm/mach-omap2/board-cm-t35.c | 2 +- arch/arm/mach-omap2/board-devkit8000.c | 2 +- arch/arm/mach-omap2/board-igep0020.c | 2 +- arch/arm/mach-omap2/board-omap3beagle.c | 2 +- arch/arm/mach-omap2/board-omap3evm.c | 2 +- arch/arm/mach-omap2/board-overo.c | 2 +- arch/arm/mach-omap2/board-zoom-peripherals.c | 29 ++++++++++++++++++++-------- arch/arm/mach-omap2/twl-common.c | 17 ++++++++++------ arch/arm/mach-omap2/twl-common.h | 3 ++- 10 files changed, 60 insertions(+), 21 deletions(-) -- 1.8.0.2

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Re: [alsa-devel] Help with adding mic bias to omap-twl4030 audio driver
by Peter Ujfalusi 31 Dec '12

31 Dec '12

Hi Neil, On 12/18/2012 12:18 AM, NeilBrown wrote: > Hi Peter, > I recently discovered your omap-twl4030 driver and thought to use it to > replace the hacked-around driver I'm using for the GTA04 mobile phone > platform. > > It works fine for playing sound out the speaker, but the microphone doesn't > work - I think because there is no mechanism to enable the microphone bias. > > I copied some bits out of the old driver - just enough to make it work. > However this probably makes the code specific to my platform. > > So I'm wondering if you could give me some guidelines on how this "should" > be done. Should these tables be passed in with the platform_data (I haven't > graduated to device-tree yet)? I'm going to send a series soon to update the omap-twl4030 machine driver. This update going to add support for microphones and all other routings as well. With the upcoming series the omap-twl4030 will support zoom2 and sdp3430. I'll CC you with the series so you can take a look if it is going to help with your board. > > My 'hacked-up' driver has a few more entries in this table than just those > few, but I wanted to strip it down to what is needed for the main microphone > to work. > > Thanks for any help you can provide, > NeilBrown > > P.S is there some good reason to use the name > struct omap_tw4030_pdata > rather than > struct omap_twl4030_pdata > > (i.e. with the 'l')?? > > diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c > index 3b97b879..aa4d866 100644 > --- a/sound/soc/omap/omap-twl4030.c > +++ b/sound/soc/omap/omap-twl4030.c > @@ -87,6 +87,32 @@ static struct snd_soc_ops omap_twl4030_ops = { > .hw_params = omap_twl4030_hw_params, > }; > > +static const struct snd_soc_dapm_widget gta04_dapm_widgets[] = { > + SND_SOC_DAPM_MIC("Internal Mic", NULL), > +}; > + > +static const struct snd_soc_dapm_route audio_map[] = { > + {"MAINMIC", NULL, "Mic Bias 1"}, > + {"Mic Bias 1", NULL, "Internal Mic"}, > +}; > + > +static int omap3gta04_init(struct snd_soc_pcm_runtime *runtime) > +{ > + int ret; > + struct snd_soc_codec *codec = runtime->codec; > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + ret = snd_soc_dapm_new_controls(dapm, gta04_dapm_widgets, > + ARRAY_SIZE(gta04_dapm_widgets)); > + if (ret < 0) > + return ret; > + > + snd_soc_dapm_add_routes(dapm, audio_map, > + ARRAY_SIZE(audio_map)); > + > + return snd_soc_dapm_sync(dapm); > +} > + > /* Digital audio interface glue - connects codec <--> CPU */ > static struct snd_soc_dai_link omap_twl4030_dai_links[] = { > { > @@ -97,6 +123,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { > .platform_name = "omap-pcm-audio", > .codec_name = "twl4030-codec", > .ops = &omap_twl4030_ops, > + .init = &omap3gta04_init, > }, > }; > > -- Péter

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[alsa-devel] [PATCH v2 0/2] ASoC: palm27x: machine driver fixes and updates
by Mike Dunn 29 Dec '12

29 Dec '12

Hi, A fix and an update to the palm27x machine driver. Change log: v2: Patch #2: in palm27x_asoc_remove(), the call to platform_device_unregister() is replaced with snd_soc_unregister_card(). This version v2 is unsolicited; I noticed the omission today. Patch preparation error; tripping over my own patches. No new ones for a while! Thanks for your patience. Mike Dunn (2): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe sound/soc/pxa/palm27x.c | 38 ++++++++++---------------------------- 1 files changed, 10 insertions(+), 28 deletions(-) -- 1.7.8.6

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[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6
by Martin Schwenke 29 Dec '12

29 Dec '12

Probing this device currently fails in snd_usb_audio_probe() because the call to snd_usb_create_mixer() fails. This is due to unknown or non-standard interface descriptor subtypes in parse_audio_unit(): usbaudio: unit 51: unexpected type 0x09 snd-usb-audio: probe of 1-8:1.0 failed with error -5 Some people are working around this by recompiling usb-audio with the call to snd_usb_create_mixer() commented out. It would be nice to avoid that. While the best idea would be to look into the mixer creation failure, a reasonable short-term solution is to use quirks to only probe the trouble-free interfaces. This allows audio and MIDI interfaces to be used without any obvious issues. Interface 0 is the main one to ignore. It contains lots of control-fu, including the unexpected interface descriptor subtypes. Interface 5 is for firmware updates and I'm not sure how to get support for this. Interface 3 is some sort of control interface that I don't understand: Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 1 Signed-off-by: Martin Schwenke <martin(a)meltin.net> --- sound/usb/quirks-table.h | 52 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 52 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 88d8ceb..c9dde5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2882,6 +2882,58 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * Focusrite Scarlett 18i6 + * + * Avoid mixer creation, which otherwise fails because some of + * the interface descriptor subtypes for interface 0 are + * unknown. That should be fixed or worked-around but this at + * least allows the device to be used successfully with a DAW + * and an external mixer. See comments below about other + * ignored interfaces. + */ + USB_DEVICE(0x1235, 0x8004), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i6", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ -- 1.7.10.4

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[alsa-devel] EMU-1212m "no Audiodock - no firmware" patch
by oeai 29 Dec '12

29 Dec '12

Aloha! i've been posting about my problem earlier, but i decided to DIY the problem was that when going up from suspend the firmware for additional card (that i don't have) was trying to load, -but there's no card - sorry, i'm gonna check few more thousand times that caused my cpu to load insanely so in the patch i've changed few more strings for debugging but actually the one (but not the only) bug is here *** 728,739 **** ! if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { --- 729,740 ---- ! if (reg & !EMU_HANA_OPTION_DOCK_OFFLINE) { the thing is in the "!"EMU_HANA_OPTION_DOCK_OFFLINE -> then the firmware should be loaded. and to debug some errors if something i've put the one more else state so if no audiodock - it could continue to load config *** 777,784 **** /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } ! snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); return 0; } --- 778,790 ---- /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } + else if ((reg & 0x3f) == 0x15) + { + snd_printk(KERN_INFO "emu1212m: trying to load config emu_hana_fpga_config"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG,0x02); + } } ! snd_printk(KERN_INFO "emu1010: emu dock firmware thread stopping\n"); return 0; } *************** then i've commented the power_down for audiodock - it should be in other place, but as i'm 1212m user - it works for me - you know. *** 841,847 **** outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ ! snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); --- 847,853 ---- outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ ! /*snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0);*/ /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg); *************** and here i've changed that the emu_model was firstly enabled and then ecard - since i don't know what's the ecard i thought that it is a second module of my emu card. *** 2079,2088 **** { if (emu->card_capabilities->ca_cardbus_chip) snd_emu10k1_cardbus_init(emu); ! if (emu->card_capabilities->ecard) ! snd_emu10k1_ecard_init(emu); ! else if (emu->card_capabilities->emu_model) snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); --- 2098,2111 ---- { if (emu->card_capabilities->ca_cardbus_chip) snd_emu10k1_cardbus_init(emu); ! ! if (emu->card_capabilities->emu_model) ! { snd_emu10k1_emu1010_init(emu); + + } + else if (emu->card_capabilities->ecard) + snd_emu10k1_ecard_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); Now i still have the problem of waking up i cannot request_firmware() properly on resume and it is trying to load nothing and other changes in patch - that was trying to debug the situation so i can at least alsa-reload at anytime. also i'd like to ask if i can use the ctxfi modules - CA0110-IBG i got xfi card (it is second of 5) that actually uses as i believe the same logic - maybe a lil modified, but not so much imho. -- RA OEAI

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[alsa-devel] Fwd: Request for HW auto-recognition: HP Pavilion dv7 with STAC92xx
by Nicolas Maître 27 Dec '12

27 Dec '12

Hi, At Wed, 12 Dec 2012 18:08:25 +0100, Tomas Pospisek wrote: > > Hello, > > I have a HP Pavilion dv7 laptop (LS066EA#UUZ), with: > > # cat /proc/asound/card0/codec* | grep Codec > Codec: IDT 92HD81B1X5 > Codec: Intel CougarPoint HDMI > > when play sound without any tweaks, I get a very weak sound output. If I > want to have "beats audio", that is subwoofer and all speakers, I need to > set: > > # cat /etc/modprobe.d/alsa-base.conf > options snd-hda-intel model=hp-dv7-4000 > > Could the snd_hda_intel alsa driver be updated to recognize and setup the > driver accordingly? > > alsa-info.sh output here: > > > http://www.alsa-project.org/db/?f=54ea874f43f12c1572aff3ee3113dc4b0ad0b398 I'm not quite sure that I have the exact same hardware, but I think so. I have 5 speakers (rear left and right, front left and right + subwoofer) Here is my alsa-info: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1056669/+attachm… In my case, model=hp-dv7-4000 doesn't fully solve the problem. The subwoofer works but it is not mapped as such. After some tests, it seems that the alsa's speakers rear left and rear right are mapped to the subwoofer. The front ones are alright. With the model=hp-dv5 that I'm using, the subwoofer isnt recognized at all, but the rear speakers are correctly mapped. Also, with hp-dv7-4000, the internal microphone is still used when a micro jack is plugged in. That's not the case with hp-dv5. If you think there's a configuration issue on my side, please let me know, but I think hp-dv5 is actually a better default choice if I have to choose between both. Kind regards, -- nimai

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Re: [alsa-devel] [PATCH 001/001] ASoC: Replace max98090 Device Driver
by Mark Brown 27 Dec '12

27 Dec '12

On Thu, Dec 13, 2012 at 11:19:29AM -0800, Jerry Wong wrote: For legibility please submit future revisions as a two stage patch series, one removing the old driver and another adding the new one. > +/* codec platform data */ > +struct max98090_pdata { > + > + int irq; This should be passed in using the normal mechanism provided by the bus, not custom implemented. > + int power_over_performance; This doesn't look like platform data to me, it looks like something that should be configured dynamically at runtime. > + /* Equalizers for DAC */ > + struct max98090_eq_cfg *eq_cfg; > + unsigned int eq_cfgcnt; Coefficients should be configured using SND_SOC_BYTES(). Some older drivers use this method but it's deprecated. > + { 0xE0, 0x00 }, /* E0 */ > + { 0xE1, 0x00 }, /* E1 */ > + { 0xE2, 0x00 }, /* E2 */ Do these number-only registers actually exist in the published register map or are they undefined? If they're undefined and not used by software there's no need to include them here. > + switch (reg) { > + case M98090_REG_01_IRQ_STATUS: > + case M98090_REG_02_JACK_STATUS: > + case M98090_REG_FF_REV_ID: Including the register number in the constants seems to rather defeat the point of having symbolic names for the registers; it means people still have to remember the register numbers. > +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) What's the relevance of _tlv here? It looks like these are normal get and sets. > +static const char * max98090_dsts_text[] = > + { "None", "Left", "Right", "Left and Right" }; > + > +static const struct soc_enum max98090_dsts_enum = > + SOC_ENUM_SINGLE(M98090_REG_1A_LVL_SIDETONE, M98090_DSTS_SHIFT, > + ARRAY_SIZE(max98090_dsts_text), max98090_dsts_text); > +static const struct snd_kcontrol_new max98090_snd_controls[] = { > + SOC_SINGLE("DMIC MIC Left Enable", M98090_REG_13_MIC_CFG1, > + M98090_DIGMICL_SHIFT, M98090_DIGMICL_NUM - 1, 0), > + SOC_SINGLE("DMIC MIC Right Enable", M98090_REG_13_MIC_CFG1, > + M98090_DIGMICR_SHIFT, M98090_DIGMICR_NUM - 1, 0), This looks like it should be DAPM. > + SOC_ENUM_EXT("External MIC Mux", max98090_extmic_mux_enum, > + max98090_extmic_mux_get, max98090_extmic_mux_set), Likewise. > + SOC_ENUM("Digital Sidetone Source", max98090_dsts_enum), DAPM. > + SOC_SINGLE_EXT_TLV("Digital Sidetone Level Volume", > + M98090_REG_1A_LVL_SIDETONE, M98090_DVST_SHIFT, > + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, > + max98090_put_enab_tlv, max98090_micboost_tlv), Level Volume? > + SOC_SINGLE_TLV("Digital Gain Volume", M98090_REG_27_DAI_LVL, > + M98090_DAI_LVL_DVG_SHIFT, M98090_DAI_LVL_DVG_NUM - 1, 0, > + max98090_dvg_tlv), Gain Volume? > + SOC_SINGLE("Digital EQ 3 Band Enable", M98090_REG_41_DSP_EQ_EN, > + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), Switch. > + SOC_SINGLE("DAC HP Playback Performance Mode", M98090_REG_43_DAC_CFG, > + M98090_DAC_PERFMODE_SHIFT, M98090_DAC_PERFMODE_NUM - 1, 0), > + SOC_SINGLE("DAC High Performance Mode", M98090_REG_43_DAC_CFG, > + M98090_DACHP_SHIFT, M98090_DACHP_NUM - 1, 0), These should either be enumerations with real names of Switches if it's just an on/off (looks like the former). > + SOC_SINGLE_TLV("Headphone Left Volume", M98090_REG_2C_LVL_HP_LEFT, > + M98090_HPVOLL_SHIFT, M98090_HPVOLL_NUM - 1, 0, > + max98090_hp_tlv), > + SOC_SINGLE_TLV("Headphone Right Volume", M98090_REG_2D_LVL_HP_RIGHT, > + M98090_HPVOLR_SHIFT, M98090_HPVOLR_NUM - 1, 0, > + max98090_hp_tlv), Should be a single stereo control (and similarly for the other output controls). > + SOC_SINGLE("Quick Setup System Clock", M98090_REG_04_QCFG_SYS_CLK, > + M98090_CLK_ALL_SHIFT, M98090_CLK_ALL_NUM - 1, 0), What do these controls do? > + if (val >= 1) { > + if (w->reg == M98090_REG_10_LVL_MIC1) > + max98090->mic1pre = val - 1; /* Update for volatile */ > + else > + max98090->mic2pre = val - 1; /* Update for volatile */ > + } Use braces consistently within an if statemeent - if you use them for the outer one use them for the inner one too. > +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { > + > + {"MIC1 Input", NULL, "MIC1"}, > + {"MIC2 Input", NULL, "MIC2"}, > + {"MIC1 Input", NULL, "MICBIAS"}, > + {"MIC2 Input", NULL, "MICBIAS"}, Unless this chip is *very* unusual I expect the bias is something that's connected externally and therefore these connections should be being made by the machine driver. > + > + /* Check for user calculated MI and NI ratios */ > + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { > + if ((user_pclk_rates[i] == max98090->sysclk) && > + (user_lrclk_rates[i] == max98090->lrclk)) { > + dev_info(codec->dev, > + "Found user supported PCLK to LRCLK rates\n"); > + dev_info(codec->dev, "i %d ni %lld mi %lld\n", > + i, ni_value[i], mi_value[i]); dev_dbg() would be fine but dev_info() is too loud. > +/* > + * This accommodates an inverted logic in the MAX98090 chip > + * for Bit Clock Invert (BCI). The inverted logic is only seen > + * for the case of TDM mode. The remaining cases have normal logic. > + */ > + if (max98090->tdm_slots > 1) { > + regval ^= M98090_DAI_BCI_MASK; > + } Coding style, the indentation is inconsistent here and no braces for single line if () statements. > + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { > + /* > + * Set to normal bias level. > + */ > + snd_soc_update_bits(codec, M98090_REG_12_MIC_BIAS, > + M98090_MBVSEL_MASK, M98090_MBVSEL_2V4); > + > + snd_soc_update_bits(codec, M98090_REG_3E_PWR_EN_IN, > + M98090_PWR_MBEN_MASK, M98090_PWR_MBEN_MASK); > + } This looks suspicous, why does the headset state have any bearing here? > +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + int regval; > + > + dev_info(codec->dev, "max98090_dai_digital_mute\n"); Drop this, it's not conveying any useful information. > +static void max98090_dmic_switch(struct snd_soc_codec *codec, int state) > +{ I have no idea what this function is supposed to do... > +static void max98090_headset_button_event(struct snd_soc_codec *codec) > +{ > + dev_info(codec->dev, "max98090_headset_button_event\n"); > +} Delete this, it doesn't do anything except log. > + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { > + case M98090_LSNS_MASK | M98090_JKSNS_MASK: > + { > + dev_info(codec->dev, "No Headset Detected\n"); > + > + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; > + > + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + > + snd_soc_dapm_disable_pin(&codec->dapm, "HPL"); > + snd_soc_dapm_disable_pin(&codec->dapm, "HPR"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "SPKL"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "SPKR"); > + snd_soc_dapm_disable_pin(&codec->dapm, "MIC1"); > + snd_soc_dapm_disable_pin(&codec->dapm, "MIC2"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "DMIC1"); > + snd_soc_dapm_force_enable_pin(&codec->dapm, "DMIC2"); > + max98090_dmic_switch(codec, 1); > + > + break; > + } This doesn't look obviously sensible... calling set_bias_level() outside of DAPM is going to get overridden in pretty short order, none of the DAPM pin setting looks like a good idea (especially not forcing on the speakers whenever there's no headset) and there's no DAPM sync to make any of this take effect. I'm a bit confused about the intended effect. I also can't see any interaction with soc-jack... What is this code actually intended to do? > + dev_info(codec->dev, "***** max98090_interrupt *****\n"); > + There's lots of these noisy, uninformative prints at info level throughout the driver. > + /* Send work to be scheduled */ > + if (active & M98090_IRQ_CLD_MASK) { > + dev_info(codec->dev, "M98090_IRQ_CLD_MASK\n"); > + } This doesn't actually schedule anything... > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); > + if (ret != 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } SND_SOC_REGMAP. > + if ( (request_threaded_irq(pdata->irq, NULL, > + max98090_interrupt, IRQF_TRIGGER_FALLING, > + "max98090_interrupt", codec)) < 0) { > + dev_info(codec->dev, "request_irq failed\n"); > + } > + /* > + * Clear any old interrupts. Coding style and it's not terribly helpful to hide the return code from users. > + /* Turn on VCM bandgap reference */ > + snd_soc_write(codec, M98090_REG_42_BIAS_CNTL, > + M98090_VCM_MODE_MASK); This shouldn't be managed at runtime? > +/* > + * Driver revision > + */ > +#define MAX98090_REVISION "0.01.00" The kernel already has perfectly good versioning.

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[alsa-devel] [PATCH 1/2] ASoC: wm8985: Refactor set_pll code to avoid gcc warnings
by Fabio Estevam 27 Dec '12

27 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8985.c:852:50: warning: 'pll_div.k' may be used uninitialized in this function sound/soc/codecs/wm8985.c:849:9: warning: 'pll_div.n' may be used uninitialized in this function sound/soc/codecs/wm8985.c:848:23: warning: 'pll_div.div2' may be used uninitialized in this function Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8985.c | 43 ++++++++++++++++++++----------------------- 1 file changed, 20 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index ab37826..dd6ce3b 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -830,33 +830,30 @@ static int wm8985_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, 0); + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - /* disable the PLL before reprogramming it */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, 0); - - if (!freq_in || !freq_out) - return 0; - - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8985_PLL_N, - (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); - /* set the source of the clock to be the PLL */ - snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, - WM8985_CLKSEL_MASK, WM8985_CLKSEL); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, - WM8985_PLLEN_MASK, WM8985_PLLEN); + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8985_PLL_N, + (pll_div.div2 << WM8985_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8985_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8985_PLL_K_1, (pll_div.k >> 18)); + /* set the source of the clock to be the PLL */ + snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, + WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, + WM8985_PLLEN_MASK, WM8985_PLLEN); + } return 0; } -- 1.7.9.5

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Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Padma Venkat 27 Dec '12

27 Dec '12

Hi, On Sat, Dec 22, 2012 at 12:32 AM, Kukjin Kim <kgene.kim(a)samsung.com> wrote: > Padma Venkat wrote: >> >> Hi, >> >> On Wed, Dec 19, 2012 at 10:39 PM, Mark Brown >> <broonie(a)opensource.wolfsonmicro.com> wrote: >> > On Wed, Dec 19, 2012 at 01:24:14PM +0000, Grant Likely wrote: >> >> On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna >> <padma.v(a)samsung.com> wrote: >> > >> >> > +- compatible : "samsung,samsung-i2s" >> > >> >> Isn't that kind of redundant? :-) >> > >> >> The format of the compatible strings should be "<vendor>,<part- >> number>-i2s". >> >> Please be specific about the part number that you're doing the binding >> >> for. For example; use "samsung,exynos4210-i2s" instead of >> "samsung,exynos-i2s". >> > >> > There are actually versioned IPs here (where the versions are used >> > publically in a few places) but it's not clearly documented which is >> > which. It would be reasonable to use the IP versions here I think. >> >> Samsung has three i2s drivers one for s3c24xx, one for s3c2412 and one >> for rest of the platforms. The above mentioned other platforms has >> Version 3/4/5 of i2s controllers. This dt binding is for for the i2s > > Where is the version defined such as 3, 4, 5? So, what is the > "sound/soc/Samsung/s3c-i2s-v2.[ch]"? Versions 3, 4, 5 are defined in dev-audio.c file of corresponding platforms. s3c-i2s-v2 is used in s3c2412 platform. > >> driver that has support for Version 3/4/5 of i2s controller. So >> "samsung,i2s-v5" is okay as compatible name? Please suggest me. >> > I agree with using version here but we need some consensus about that. > > - Kukjin > > -- > To unsubscribe from this list: send the line "unsubscribe linux-samsung-soc" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html Thanks Padma

2 2

[alsa-devel] [PATCH] ASoC: cs42l52: Convert to devm_input_allocate_device()
by Axel Lin 27 Dec '12

27 Dec '12

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..73a8049 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1040,7 +1040,7 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); int ret; - cs42l52->beep = input_allocate_device(); + cs42l52->beep = devm_input_allocate_device(codec->dev); if (!cs42l52->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -1061,7 +1061,6 @@ static void cs42l52_init_beep(struct snd_soc_codec *codec) ret = input_register_device(cs42l52->beep); if (ret != 0) { - input_free_device(cs42l52->beep); cs42l52->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -1078,7 +1077,6 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(cs42l52->beep); cancel_work_sync(&cs42l52->beep_work); cs42l52->beep = NULL; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 1/2] ASoC: core: fix the memory leak in case of device_add() failure
by Mark Brown 27 Dec '12

27 Dec '12

On Wed, Dec 26, 2012 at 12:56:05AM +0800, Chuansheng Liu wrote: > > After called device_initialize(), even device_add() returns > error, we still need use the put_device() to release the reference > to call rtd_release(), which will do the free() action. Applied both, thanks.

1 0

[alsa-devel] [PATCH] ASoC: cs42l52: Catch no-match case in cs42l52_get_clk
by Axel Lin 24 Dec '12

24 Dec '12

In the case of no-match, return -EINVAL instead of 0. Since we assign i to ret in the for loop, ret always less than ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..9811a54 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = { static int cs42l52_get_clk(int mclk, int rate) { - int i, ret = 0; + int i, ret = -EINVAL; u_int mclk1, mclk2 = 0; for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { @@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) - return -EINVAL; return ret; } -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:32AM +0100, Lucas Stach wrote: > This connection is used by the AC97 controller. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Mark Brown 24 Dec '12

24 Dec '12

On Thu, Dec 20, 2012 at 12:17:33AM +0100, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Applied, though the above isn't strictly true - a number of later AC'97 CODECs had PLLs which allowed them to use non-standard AC'97 MCLKs.

1 0

[alsa-devel] [PATCH v2] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the comment sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Remove redundant check
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> The condition "if (!freq_in || !freq_out)" has already been tested previously, so no need to do it again. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8804.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d321a87..1704b1e 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -395,9 +395,6 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, /* power down the PLL before reprogramming it */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); - if (!freq_in || !freq_out) - return 0; - /* set PLLN and PRESCALE */ snd_soc_update_bits(codec, WM8804_PLL4, 0xf | 0x10, pll_div.n | (pll_div.prescale << 4)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: cs4271: add optional soft reset workaround
by Daniel Mack 24 Dec '12

24 Dec '12

The CS4271 requires its LRCLK and MCLK to be stable before its RESET line is de-asserted. That also means that clocks cannot be changed without putting the chip back into hardware reset, which also requires a complete re-initialization of all registers. One (undocumented) workaround is to assert and de-assert the PDN bit in the MODE2 register. This patch adds a new flag to both the DT bindings as well as to the platform data to enable that workaround. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 12 ++++++++ include/sound/cs4271.h | 15 ++++++++++ sound/soc/codecs/cs4271.c | 34 ++++++++++++++++++++++ 3 files changed, 61 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index a850fb9..e2cd1d7 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -20,6 +20,18 @@ Optional properties: !RESET pin - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag is enabled. + - cirrus,enable-soft-reset: + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. Examples: diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index dd8c48d..70f4535 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -20,6 +20,21 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ + + /* + * The CS4271 requires its LRCLK and MCLK to be stable before its RESET + * line is de-asserted. That also means that clocks cannot be changed + * without putting the chip back into hardware reset, which also requires + * a complete re-initialization of all registers. + * + * One (undocumented) workaround is to assert and de-assert the PDN bit + * in the MODE2 register. This workaround can be enabled with the + * following flag. + * + * Note that this is not needed in case the clocks are stable + * throughout the entire runtime of the codec. + */ + bool enable_soft_reset; }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac99238..ffa72dd 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -167,6 +167,8 @@ struct cs4271_private { int gpio_nreset; /* GPIO that disable serial bus, if any */ int gpio_disable; + /* enable soft reset workaround */ + bool enable_soft_reset; }; /* @@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, int i, ret; unsigned int ratio, val; + if (cs4271->enable_soft_reset) { + /* + * Put the codec in soft reset and back again in case it's not + * currently streaming data. This way of bringing the codec in + * sync to the current clocks is not explicitly documented in + * the data sheet, but it seems to work fine, and in contrast + * to a read hardware reset, we don't have to sync back all + * registers every time. + */ + + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !dai->capture_active) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !dai->playback_active)) { + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + } + } + cs4271->rate = params_rate(params); /* Configure DAC */ @@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; + + if (of_get_property(codec->dev->of_node, + "cirrus,enable-soft-reset", NULL)) + cs4271->enable_soft_reset = true; } #endif @@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } if (gpio_nreset >= 0) -- 1.7.11.7

3 2

[alsa-devel] [PATCH] ASoC: soc-core: Remove unused 'ret' variable
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced the following build warning: sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable] Remove the unused 'ret' variable. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e0d4630..29046e7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2996,7 +2996,6 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: atmel-soc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a couple of format type warnings. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi, Build tested on RM9200 and x86_64. ARCH_AT91 depend is pushed down to the machine drivers since they still relies on mach includes. regards Joachim Eastwood sound/soc/atmel/Kconfig | 6 +++--- sound/soc/atmel/atmel-pcm-pdc.c | 4 ++-- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 -- 4 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1b691b..3fdd87f 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 + depends on HAS_IOMEM help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,7 +24,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -34,7 +34,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 6a293c7..054ea4d 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -159,7 +159,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, pr_debug("atmel-pcm: " "hw_params: DMA for %s initialized " - "(dma_bytes=%u, period_size=%u)\n", + "(dma_bytes=%zu, period_size=%zu)\n", prtd->params->name, runtime->dma_bytes, prtd->period_size); @@ -201,7 +201,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, int ret = 0; pr_debug("atmel-pcm:buffer_size = %ld," - "dma_area = %p, dma_bytes = %u\n", + "dma_area = %p, dma_bytes = %zu\n", rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); switch (cmd) { diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index e99f181..3109db7 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -49,7 +49,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%d\n", + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", (void *)buf->area, (void *)buf->addr, size); if (!buf->area) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1c76634..2755750 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -42,8 +42,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <mach/hardware.h> - #include "atmel-pcm.h" #include "atmel_ssc_dai.h" -- 1.8.0

4 4

[alsa-devel] [PATCH] ASoC: atmel-ssc: make it buildable on other architectures
by Joachim Eastwood 24 Dec '12

24 Dec '12

Not very useful on non AT91/AVR32 platforms but it provides more build coverage and prepares for ARM multiplatform. Also fixes a truncated warning that would come when building on a 64-bit arch. Signed-off-by: Joachim Eastwood <manabian(a)gmail.com> --- Hi Mark, Sending it to you since seem like you're the last one who commited changes to the driver and the driver is mostly used on ASoC. Tested on RM9200 and build tested on x86_64. regards Joachim Eastwood drivers/misc/Kconfig | 2 +- drivers/misc/atmel-ssc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/misc/Kconfig b/drivers/misc/Kconfig index b151b7c..8f59d88 100644 --- a/drivers/misc/Kconfig +++ b/drivers/misc/Kconfig @@ -192,7 +192,7 @@ config ICS932S401 config ATMEL_SSC tristate "Device driver for Atmel SSC peripheral" - depends on AVR32 || ARCH_AT91 + depends on HAS_IOMEM ---help--- This option enables device driver support for Atmel Synchronized Serial Communication peripheral (SSC). diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 158da5a..0cee274 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -167,7 +167,7 @@ static int ssc_probe(struct platform_device *pdev) /* disable all interrupts */ clk_enable(ssc->clk); - ssc_writel(ssc->regs, IDR, ~0UL); + ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); clk_disable(ssc->clk); -- 1.8.0

4 3

[alsa-devel] [PATCH v2] ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

Update lm49453_reg_defs values as per LM49453 HW revision-B Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 61 +++++++++++--------------------------------- 1 files changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 99b0a9d..808e50a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, };

2 1

[alsa-devel] [PATCH v2] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 24 Dec '12

24 Dec '12

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

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[alsa-devel] [PATCH] ASoC: wm8962: Add device tree support
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add device tree support. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8962.txt | 16 ++++++++++++++++ sound/soc/codecs/wm8962.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt new file mode 100644 index 0000000..dceb3b1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -0,0 +1,16 @@ +WM8962 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8962" + + - reg : the I2C address of the device. + +Example: + +codec: wm8962@1a { + compatible = "wlf,wm8962"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8fd38cb..1ebeff4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3758,10 +3758,17 @@ static const struct i2c_device_id wm8962_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); +static const struct of_device_id wm8962_of_match[] = { + { .compatible = "wlf,wm8962", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8962_of_match); + static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off
by Mark Brown 24 Dec '12

24 Dec '12

On Fri, Dec 21, 2012 at 06:17:12PM +0800, Chuansheng Liu wrote: > > Just found some cases that some codec drivers set the bias to _STANDBY and > set idle_bias_off to 1 during probing. > It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested, > there is no reason to start from _STANDBY bias with idle_bias_off == 1. Applied, thanks.

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[alsa-devel] [PATCH RFT] ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
by Axel Lin 24 Dec '12

24 Dec '12

The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi M R Swami, I don't have the datasheet. Can you check if this patch is correct? Thanks, Axel sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d..c0d203b 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1218,7 +1218,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Add sgtl5000 into the Subject sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Fix maximum value for microphone gain
by Fabio Estevam 24 Dec '12

24 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. >From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675c..92bbfec 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP -- 1.7.9.5

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[alsa-devel] Does it will effect the precision of DMA position?
by Zhang wei 24 Dec '12

24 Dec '12

Hello guys. In the function snd_pcm_update_hw_ptr0(): pos -= pos % runtime->min_align; after running it, the 'pos' maybe smaller than before.because there is need to aligned.so that means lost or drop some Audio data for ever ,is that true? The data playbacked is a bit of different from we send to? Best Regards zw

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[alsa-devel] [alsa-lib, dmix] Support for S24_3BE
by Vincent Vinel 24 Dec '12

24 Dec '12

Hello, The bugtracker seems to be down, so I'm posting here instead. I was trying to use software mixing with my Native Instruments USB soundcard (snd-usb-caiaq), but all attemps to use the dmix pcm ended up with the rather unhelpful error : ALSA lib pcm_dmix.c:1311:(_snd_pcm_dmix_open) Unsupported format When using the hw pcm, however, aplay was kind enough to display the list of available formats, which in my case was limited to S24_3BE. A quick check confirmed that dmix doesn't support that format, which happens to be the only format available for NI USB soundcards (as defined in snd-usb-caiaq). Is it worth writing a patch for 24bit big-endian support in dmix ? External soundcards like these ones are usually used for realtime effects processing and low-latency audio, things for which dmix proably isn't suited anyway. -- Vincent Vinel

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Re: [alsa-devel] [PATCH V5 2/2] ASoC: SAMSUNG: Add DT support for i2s
by Grant Likely 24 Dec '12

24 Dec '12

On Thu, 13 Dec 2012 16:12:53 +0530, Padmavathi Venna <padma.v(a)samsung.com> wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > .../devicetree/bindings/sound/samsung-i2s.txt | 75 +++++++ > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 233 ++++++++++++++++---- > 4 files changed, 269 insertions(+), 41 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > new file mode 100644 > index 0000000..bff47b4 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > @@ -0,0 +1,75 @@ > +* Samsung I2S controller > + > +Required SoC Specific Properties: > + > +- compatible : "samsung,samsung-i2s" Isn't that kind of redundant? :-) The format of the compatible strings should be "<vendor>,<part-number>-i2s". Please be specific about the part number that you're doing the binding for. For example; use "samsung,exynos4210-i2s" instead of "samsung,exynos-i2s". > +- reg: physical base address of the controller and length of memory mapped > + region. > + > +[PRELIMINARY: the dma channel allocation will change once there are > +official DMA bindings] > + > +- tx-dma-channel-secondary: The dma channel specifier for secondary tx > + operations. The format of the dma specifier depends on the dma > + controller. > +- tx-dma-channel: The dma channel specifier for tx operations. The format of > + the dma specifier depends on the dma controller. > +- rx-dma-channel: The dma channel specifier for rx operations. The format of > + the dma specifier depends on the dma controller. > + > +Optional SoC Specific Properties: > + > +- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel > + support, this flag is enabled. > +- samsung,supports-rstclr: This flag should be set if I2S software reset bit > + control is required. When this flag is set I2S software reset bit will be > + enabled or disabled based on need. > +- samsung,supports-secdai:If I2S block has a secondary FIFO and internal DMA, > + then this flag is enabled. > +- samsung,idma-addr: Internal DMA register base address of the audio > + sub system(used in secondary sound source). > + > +Required Board Specific Properties: > + > +- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK > + interface lines. The format of the gpio specifier depends on the gpio > + controller. > + The syntax of samsung gpio specifier is > + <[phandle of the gpio controller node] > + [pin number within the gpio controller] > + [mux function] > + [flags and pull up/down] > + [drive strength]> > + > +Aliases: > + > +- All the I2S controller nodes are represented in the aliases node using > + the following format 'i2s{n}' where n is a unique number for the alias. Are the aliases actually being used now? If not, then drop this section. I don't see any references in this patch. Otherwise the binding looks good to me. g.

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[alsa-devel] sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 23 Dec '12

23 Dec '12

Hi folks, I have put together full support for Digidesign Mbox 2, but it needs someone to help me merge it into alsa. I haven't seen the latest snd-usb-audio code since last year, it seems to have branched into multiple files. I have worked out how the Digidesign Mbox 2 needs to be configured, it supports SPDIF i/o simultaneously with analogue but can be toggled on and off by using different altsettings. Also the sample clock rate must be configured via a special method. Therefore I have provided for this in a device_setup parameter with 4 settings: 48Khz Spdif+analog = 0 44.1Khz Spdif+analog = 1 48Khz Analog only = 2 44.1Khz Analog only = 3 ###### audio /* Digidesign Mbox 2 workaround: * supports S24_3BE BIG ENDIAN in an unusual way */ if (chip->usb_id == USB_ID(0x0dba, 0x3000)) pcm_format = SNDRV_PCM_FORMAT_S24_3BE; static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { ..... ..... /* Digidesign Mbox 2: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0dba, 0x3000) && mbox2_skip_setting_quirk(chip, iface_no, altno)) continue; ..... } #define MBOX2_SET_48K_SPDIF 0 #define MBOX2_SET_44K_SPDIF 1 #define MBOX2_SET_48K_ANALOG 2 #define MBOX2_SET_44K_ANALOG 3 static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { u8 srate[3]; /* Reset ifaces 2-5 to 0 altsetting. */ usb_set_interface(chip->dev, iface, 0); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF || device_setup[chip->index] == MBOX2_SET_48K_ANALOG) { //setup 48k srate[0]=0x80; srate[1]=0xbb; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48000Hz"); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } else { //setup 44k srate[0]=0x44; srate[1]=0xac; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 44100Hz"); if (device_setup[chip->index] == MBOX2_SET_44K_SPDIF || device_setup[chip->index] == MBOX2_SET_44K_ANALOG) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } return 0; } #define MBOX2_FIRMWARE_SIZE 646 #define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ #define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) { struct usb_host_config *config = dev->actconfig; int err; u8 enablemagic[3]; u8 bootresponse; int fwsize; if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); /* From USB Snoop, * * SetupPacket = RT RQ VHVL INDX SIZE * RT = 0xRT Request Type * RQ = 0xRQ Request * VHVL = 0xVLVH Value in reverse byte order * INDX = 0xDXIN Index in reverse byte order * SIZE = 0xZESI Size in reverse byte order * * Magic boot code setup packet: c0 85 01 00 00 00 12 00 * RQ RT VLVH DXIN ZESI * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT * for snd_usb_ctl_msg() */ mbox2_reboot: snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); if (bootresponse == MBOX2_BOOT_LOADING) { snd_printdd("device not ready, resending boot sequence...\n"); goto mbox2_reboot; } if (bootresponse == MBOX2_BOOT_READY) { snd_printdd("device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); snd_printdd("mbox2_boot: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) * but now we will try sending capture/playback enable magic * * 80 bb 00 = 24bit mode - S24_3BE * 44 ac 00 = 16bit mode? * */ /* We want 48khz mode for now */ enablemagic[0]=0x80; enablemagic[1]=0xbb; enablemagic[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); return 0; /* Succesful boot */ } snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); return -ENODEV; } snd_printk("Invalid firmware size: %d\n",fwsize); return -ENODEV; } ####### quirk: { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 6, .type = QUIRK_MIDI_MBOX2, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, .in_interval = 0x01, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ##### midi case QUIRK_MIDI_MBOX2: /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; /* * We have to make sure that the USB core looks * again at interface 6 by calling usb_set_interface() on it. */ usb_set_interface(umidi->chip->dev, 6, 0); memcpy(&endpoints[0], quirk->data, sizeof(struct snd_usb_midi_endpoint_info)); err = 0; break; Damien

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[alsa-devel] [PATCH] ALSA: usb-audio: support delay calculation on capture streams
by Pierre-Louis Bossart 21 Dec '12

21 Dec '12

Enable delay report on capture path. The delay is reset when an URB is retired and increment at each call to .pointer based on frame counter changes. The precision of the delay information is limited to 1ms as in the playback case. This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..3a38447 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -59,7 +59,12 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, /* Approximation based on number of samples per USB frame (ms), some truncation for 44.1 but the estimate is good enough */ - est_delay = subs->last_delay - (frame_diff * rate / 1000); + est_delay = frame_diff * rate / 1000; + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) + est_delay = subs->last_delay - est_delay; + else + est_delay = subs->last_delay + est_delay; + if (est_delay < 0) est_delay = 0; return est_delay; @@ -78,8 +83,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); @@ -1147,6 +1151,10 @@ static void retire_capture_urb(struct snd_usb_substream *subs, int i, period_elapsed = 0; unsigned long flags; unsigned char *cp; + int current_frame_number; + + /* read frame number here, update pointer in critical section */ + current_frame_number = usb_get_current_frame_number(subs->dev); stride = runtime->frame_bits >> 3; @@ -1180,6 +1188,15 @@ static void retire_capture_urb(struct snd_usb_substream *subs, subs->transfer_done -= runtime->period_size; period_elapsed = 1; } + /* capture delay is by construction limited to one URB, + * reset delays here + */ + runtime->delay = subs->last_delay = 0; + + /* realign last_frame_number */ + subs->last_frame_number = current_frame_number; + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ if (oldptr + bytes > runtime->buffer_size * stride) { -- 1.7.9.5

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[alsa-devel] [PATCH RFT] ASoC: wm2200: Fix setting dai format in wm2200_set_fmt
by Axel Lin 21 Dec '12

21 Dec '12

According to the defines in wm2200.h: /* * R1284 (0x504) - Audio IF 1_5 */ #define WM2200_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ #define WM2200_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ We should not left shift 1 bit for fmt_val when setting dai format. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Mark, wm2200.h defines WM2200_AIF1_FMT_SHIFT to be 0, so current code looks wrong to me. I don't have the datasheet, so can you check if this patch is correct? Axel sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31d..a12fc2f 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1626,7 +1626,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ASoC: fsi: cleanup FSI driver
by Kuninori Morimoto 21 Dec '12

21 Dec '12

Hi Mark Cc Simon These patches changed platform setting to use FSI new style, and cleanup FSI driver. Kuninori Morimoto (3): ASoC: fsi: remove platform depended .set_rate() callback support ASoC: fsi: remove SH_FSI_xxx_INV flags ASoC: fsi: cleanup sh_fsi.h arch/arm/mach-shmobile/board-ap4evb.c | 11 +-- arch/arm/mach-shmobile/board-mackerel.c | 11 +-- arch/sh/boards/mach-ecovec24/setup.c | 12 +-- arch/sh/boards/mach-se/7724/setup.c | 12 +-- include/sound/sh_fsi.h | 70 +------------- sound/soc/sh/fsi.c | 156 +++---------------------------- 6 files changed, 27 insertions(+), 245 deletions(-) >> Simon #2 patch modifies both FSI driver and platform. So, it needs your Acked-by Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCHv2 0/9] ALSA: CA0132: DSP Update
by Ian Minett 21 Dec '12

21 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Thanks for the comments. This resubmit should help to correct many of the points raised. For example, use of C99-style init for structures, the previous large patch is now split into several subpatches, and there are more descriptions to the code and individual patchfiles. Patch Summary: 1: Add new definitions and structures that will be used by the DSP routines 2: Init chip registers, DSP parameters and mixer settings 3: Add the new DSP mixer kcontrols and helpers. 4: Update and add PCM callbacks and enhancements. 5: Add unsol handler for DSP response and jack detection reporting 6: Add tuning controls 7/8: Code shuffle and tidy, split into two for a cleaner diff. 9: Updates to the mic sample rate and module description, and fix potential errors on init. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

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Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the driver for the Tegra 2x AC97 host controller. The AC'97 DT binding file should really be added as part of this patch. I'm not at all familiar with AC'97, but this mostly looks fine. I have a bunch of comments below though, mostly for my own understanding. > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA20_AC97 > + tristate > + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC This should also select SND_SOC_TEGRA20_DAS, just like the I2S driver. > diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c > +static struct tegra20_ac97 *workdata; Why not put that into the device's drvdata? Oh, is there no way to stash driver-private data into the struct snd_ac97? > +static void tegra20_ac97_codec_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* reset line is not driven by DAC pad group, have to toggle GPIO */ > + gpio_set_value(workdata->reset_gpio, 0); > + udelay(2); I'm not quite sure what that implies. Might some (Tegra) HW designs use a dedicated signal from the AC'97 HW to reset the CODEC and others not? That would make the GPIO optional. Or, does the comment mean that Tegra doesn't ever have a dedicated CODEC reset signal, so we always have to use a GPIO? If so, this comment might be more appropriate inside probe() where I asked my related questions. > +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) > +{ > + u32 readback; > + unsigned long timeout; > + > + /* > + * although sync line is driven by the DAC pad group warm reset using > + * the controller cmd is not working, have to toggle sync line > + * manually. > + */ > + gpio_request(workdata->sync_gpio, "codec-sync"); Hmm. There's an AC'97 command to reset the CODEC and we don't implement it? Uggh. > +static const struct snd_soc_dai_ops tegra20_ac97_dai_ops = { > + .trigger = tegra20_ac97_trigger, > +}; No .set_fmt or .hw_params? Does the AC'97 ASoC core code handle that somehow? I guess that's what soc_ac97_ops is for? > +static struct snd_soc_dai_driver tegra20_ac97_dai = { > + .name = "tegra-ac97-pcm", > + .ac97_control = 1, > + .probe = tegra20_ac97_probe, > + .playback = { > + .stream_name = "PCM Playback", Out of curiosity, why "PCM Playback" not just "Playback"? > +static bool tegra20_ac97_volatile_reg(struct device *dev, unsigned int reg) > +{ > + switch (reg) { > + case TEGRA20_AC97_CMD: ... > + return true; CMD is volatile; did we put status bits in there? > +static int tegra20_ac97_platform_probe(struct platform_device *pdev) > + ac97->reset_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-reset-gpio", 0); > + if (gpio_is_valid(ac97->reset_gpio)) { > + ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, > + GPIOF_OUT_INIT_HIGH, "codec-reset"); Shouldn't this get the flags from the GPIO specifier, and deal with active-high/active-low resets, or is the polarity set by the AC'97 spec? > + if (ret) { > + dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); > + goto err_clk_put; > + } > + } else { > + dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); > + goto err_clk_put; > + } Is a reset GPIO necessarily required? What if a board arranged to reset the CODEC during power-on-reset, and there was no software control over the reset? > + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, > + "nvidia,codec-sync-gpio", 0); > + if (!gpio_is_valid(ac97->sync_gpio)) { > + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); > + goto err_clk_put; > + } I don't know what this is, so I'll ask if it's strictly required too. > + ret = snd_soc_register_dais(&pdev->dev, &tegra20_ac97_dai, 1); snd_soc_register_dai() would be marginally simpler. > + ret = clk_prepare_enable(ac97->clk_ac97); > + if (ret) { > + dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); > + goto err_asoc_utils_fini; > + } Can you do runtime PM instead? See the I2S driver for an example. > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 5/7] ASoC: tegra: add ac97 host driver
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/20/2012 01:10 PM, Lucas Stach wrote: > Am Donnerstag, den 20.12.2012, 12:44 -0700 schrieb Stephen Warren: >> On 12/19/2012 04:17 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> +static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) >>> +{ >>> + u32 readback; >>> + unsigned long timeout; >>> + >>> + /* >>> + * although sync line is driven by the DAC pad group warm reset using >>> + * the controller cmd is not working, have to toggle sync line >>> + * manually. >>> + */ >>> + gpio_request(workdata->sync_gpio, "codec-sync"); >> >> Hmm. There's an AC'97 command to reset the CODEC and we don't implement >> it? Uggh. > > As far as I could figure from the downstream register doc there is a > command implemented in the host controller that's supposed to do the > warm reset, but in my testing it didn't work. Sadly I have not received > any documentation about the host controller from NVIDIA up until now, so > I don't now if we could make this work somehow. Bug Submission ID#: > 196075 What system was that bug submitted into? It's not a regular NVIDIA internal bug number. I'll see who I can ask about this feature... >>> + ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, >>> + "nvidia,codec-sync-gpio", 0); >>> + if (!gpio_is_valid(ac97->sync_gpio)) { >>> + dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); >>> + goto err_clk_put; >>> + } >> >> I don't know what this is, so I'll ask if it's strictly required too. > > Until we manage to get the host controller warm reset command working > this is fixed to be the _FS line of the DAP connected to the codec, so > strictly required. I'd like to investigate whether the warm reset command can be made to work before actually committing to a DT binding that requires this GPIO to be specified; the DT binding is supposed to be basically static.

1 0

Re: [alsa-devel] [PATCH 7/7] ASoC: tegra: add machine driver using wm9712 codec
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds a simple machine driver using the Wolfson wm9712 AC97 > codec. If the AC'97 code is going to be reworked to probe the CODEC rather than manually specify it in DT, then will this machine driver become generic for any AC'97 system, and hence you could rename the DT binding to nvidia,tegra-audio-ac97? > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt > +- nvidia,audio-codec : The phandle of the WM9712 audio codec I wonder how you'll get that if you dynamically probe the AC'97 bus... > diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig > +config SND_SOC_TEGRA_WM9712 > + tristate "SoC Audio support for Tegra boards using a WM9712 codec" > + depends on SND_SOC_TEGRA I'd suggest making this depend on the Tegra AC97 module, or ARCH_TEGRA_2x_SOC instead, since it can only run on Tegra20. The AC'97 module has been removed on Tegra30 and replaced with an HD Audio controller instead. > + select SND_SOC_TEGRA20_AC97 if ARCH_TEGRA_2x_SOC That select (or a dependency as mentioned above) is always required. > + select SND_SOC_AC97_BUS Shouldn'y the AC'97 driver select that? > diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c > +static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) > + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); Hmm. I guess that potentially ties this driver to whichever CODECs actually expose a widget with that exact name... > +static int tegra_wm9712_remove(struct snd_soc_card *card) > +{ > + return 0; > +} Do you need to implement this at all if it does nothing? > +MODULE_LICENSE("GPL"); "GPL v2"

2 1

Re: [alsa-devel] [PATCH 6/7] ASoC: tegra: add ac97 host controller to device tree
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This adds the DT bindings and a default entry for the Tegra 2x > AC97 host controller. Everything here except the .dtsi file should be part of the previous patch. > diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt > +- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number > + of the GPIO corresponding with the AC97 DAP _FS line Hmm, for I2S at least, FS (frame sync I assume) definitely is something that the audio controller itself drives. Are you absolutely sure that this should be represented as a GPIO here? > diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi > + tegra_ac97: ac97@70002000 { There's only 1 node named ac97, so you can leave out the unit address since it's not required to generate a unique node name. In other words just "ac97".

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Re: [alsa-devel] [PATCH 1/7] ARM: tegra: add ac97 clock
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > Add clock used by the AC97 host controller. This is going to conflict with Prashant's clock driver rework. Can you rebase it on top of that once it's checked in. That might be a couple weeks out though.

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Re: [alsa-devel] [PATCH 4/7] ASoC: tegra: add function to set ac97 rate
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > AC97 uses a fixed rate, unrelated to the sample rate. Add a function to > make the setup more trivial. Acked-by: Stephen Warren <swarren(a)nvidia.com>

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Re: [alsa-devel] [PATCH 3/7] ASoC: tegra: setup DAP3<->DAC3 connection by default
by Stephen Warren 20 Dec '12

20 Dec '12

On 12/19/2012 04:17 PM, Lucas Stach wrote: > This connection is used by the AC97 controller. Acked-by: Stephen Warren <swarren(a)nvidia.com>

1 0

[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:07 PM, prasannatsmkumar <prasannatsmkumar(a)gmail.com> wrote: > Hi ALSA people, > > I am having a Realtek ALC 883 chip. The sound volume is very low > compared to what I hear in windows - I maintain same speaker volume > and play same audio. If I keep the volume as 10% in windows I have to > keep a volume of at least 25+% in Linux. Is there any way to improve > it? I have been seeing such a situation for past 4 or 5 years (from > Fedora 9 period till today). I have tried several distributions > (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the > same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. > > $cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.25. > > Should I file a bug? If so in which component (kernel or pulseaudio)? > > Thanks in advance, > PrasannaKumar Resending this as this mail was bounced by alsa devel list. Please feel free to ask for information necessary to solve this issue. I have copied output of some commands that may be useful for understanding or digging this issue. dmesg | grep hda output: [ 30.648663] snd_hda_intel 0000:00:1b.0: >irq 43 for MSI/MSI-X [ 30.716741] hda_codec: ALC883: SKU not ready 0x411111f0 lspci -vvv output (only info related audio device is in mail all others are stripped): 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) Subsystem: Intel Corporation Device d605 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 43 Region 0: Memory at 501c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee0300c Data: 4189 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=1 ArbSelect=Fixed TC/VC=80 Status: NegoPending- InProgress- Capabilities: [130 v1] Root Complex Link Desc: PortNumber=0f ComponentID=02 EltType=Config Link0: Desc: TargetPort=00 TargetComponent=02 AssocRCRB- LinkType=MemMapped LinkValid+ Addr: 00000000fed1c000 Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel amixer output: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 13 [42%] [-27.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] [30.00dB] Front Right: 3 [100%] [30.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 8 [26%] [0.00dB] [on] Front Right: Capture 8 [26%] [0.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [off] Front Right: Capture 0 [0%] [-12.00dB] [off] Simple mixer control '2ch-Upmix Master',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 255 Front Left: 255 [100%] Front Right: 255 [100%] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Rear Mic' 'Front Mic' 'Line' Item0: 'Rear Mic' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Thanks and Regards, PrasannaKumar

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[alsa-devel] Fwd: Low volume with Realtek ALC 883.
by prasannatsmkumar 20 Dec '12

20 Dec '12

Hi ALSA / pulseaudio people, Resending this as this mail was bounced by alsa devel list. I am having a Realtek ALC 883 chip. The sound volume is very low compared to what I hear in windows - I maintain same speaker volume and play same audio. If I keep the volume as 10% in windows I have to keep a volume of at least 25+% in Linux. Is there any way to improve it? I have been seeing such a situation for past 4 or 5 years (from Fedora 9 period till today). I have tried several distributions (Fedora, Vector Linux, PC Linux OS, Ubuntu, Mint) and all have the same behaviour. Currently I am running Ubuntu 11.10 with 3.5 kernel. $cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. Should I file a bug? If so in which component (kernel or pulseaudio)? Thanks in advance, PrasannaKumar

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[alsa-devel] [PATCH RFT] ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
by Axel Lin 20 Dec '12

20 Dec '12

Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Rajeev, I don't have this hardware, can you test this patch? Thanks, Axel sound/soc/codecs/sta529.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4..40c07be 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 20 Dec '12

20 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,6 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- 1.7.8.3 Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] [PATCH RFT] ASoC: arizona: Do proper shift for setting aif1 rate
by Axel Lin 20 Dec '12

20 Dec '12

ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/arizona.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b..38248a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -714,7 +714,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 06:37:26AM +0000, Liu, Chuansheng wrote: > Meanwhile, is it useful to add one warning there for that case? > After all, in probing, set the bias to _STANDBY even idle_bias_off == 1, and calling get_runtime_sync(), it > will let the code more obscure. So giving a warning there to indicate the driver: > it is not suggested that in probing, set the bias to _STANDBY even idle_bias_off == 1. Probably, send a patch please. Like I say it is possible to start off in _STANDBY providing the driver grabs the runtime PM reference too but I can't think of any reason for doing that so the warning seems sensible.

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Re: [alsa-devel] [PATCH 2/7] ASoC: allow wolfson wm9712 codec to be instantiated using device tree
by Mark Brown 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 12:17:31AM +0100, Lucas Stach wrote: > static struct platform_driver wm9712_codec_driver = { > .driver = { > .name = "wm9712-codec", > .owner = THIS_MODULE, > + .of_match_table = wm9712_of_match, > }, No, this is really not a sane way of handling AC'97 at all. They're not platform devices, there is no sane reason why they should appear in the device tree as such. Really we ought to be just hotplugging them, AC'97 is perfectly enumerable.

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[alsa-devel] Adding wm8962 support to mx6qsabresd
by Fabio Estevam 20 Dec '12

20 Dec '12

Hi, I am working on adding audio support to mx6qsabresd board, which contains a wm8962 codec. Here is the log with debug information turned on that I am getting so far: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 22 => 9 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 3a => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 39 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 64 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 65 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 69 => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 6a => 0 wm8962 0-001a: read 200 => ffffffff wm8962 0-001a: read 201 => 0 wm8962 0-001a: read 202 => 0 wm8962 0-001a: read 203 => ffffffff wm8962 0-001a: read 204 => 8100 wm8962 0-001a: read 205 => 8100 wm8962 0-001a: DMIC not in use, disabling input: WM8962 Beep Generator as /devices/soc.0/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input0 wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 31 => ffffffff wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 63 => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 1a => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 19 => 0 wm8962 0-001a: read 26 => 8 wm8962 0-001a: read 25 => 8 wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 30 => ffffffff wm8962 0-001a: read 300 => 1c00 wm8962 0-001a: read 17 => 160 wm8962 0-001a: read 48 => 0 wm8962 0-001a: read 8 => ffffffff wm8962 0-001a: read 52 => 4 wm8962 0-001a: read 19 => 0 wm8962 0-001a: write a0 = 149 wm8962 0-001a: write a1 = 271 wm8962 0-001a: write a2 = 7 wm8962 0-001a: FLL configured for 24000000Hz->11289600Hz wm8962 0-001a: No audio clocks configured ... imx-wm8962 sound.14: wm8962 <-> 2028000.ssi mapping ok root@freescale /home$ ./aplay Front_Left.wav wm8962 0-001a: hw_params set BCLK 1411200Hz LRCLK 44100Hz wm8962 0-001a: wm8962->sysclk_rate = 11289600 wm8962 0-001a: wm8962->lrclk = 44100 wm8962 0-001a: Selected sysclk ratio 256 wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 3 => 100 wm8962 0-001a: write 3 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -5 wm8962 0-001a: read 29 => 100 wm8962 0-001a: write 29 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -5 wm8962 0-001a: read 2 => 100 wm8962 0-001a: write 2 = 100 wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -5 wm8962 0-001a: read 28 => 100 wm8962 0-001a: write 28 = 100 wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -5 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 wm8962 0-001a: DC servo complete after 1ms Playing WAVE 'Front_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Any idea of why register 4 and 42 read fails? wm8962 0-001a: read 4 => ffffffff wm8962 0-001a: Failed to read DSPCLK: -1 wm8962 0-001a: read 42 => ffffffff wm8962 0-001a: Failed to read DCS status: -1 Any suggestion is appreciated. Thanks, Fabio Estevam

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[alsa-devel] how to check the CODEC registers vaule
by Zhang wei 20 Dec '12

20 Dec '12

Hello everyone,I am confused again : ( I read the wm8985.pdf and found the description: If the address is not recognised or the R/W bit is '1' when operating in write only mode,the WM8985 returns to the idle conditon and waits for a new start condition and valid address. Does it mean the wm8985 registers write only, can not be read? I found other kinds of codec datasheet same as it. Is it necessary to read back the value from the register ? Thanks a lot. Best Regards zw

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: core: Fix SOC_DOUBLE_RANGE() macros
by Mark Brown 19 Dec '12

19 Dec '12

Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 10 ++++++---- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++-- 2 files changed, 36 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4a27822..1e4ea78 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -58,8 +58,9 @@ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -88,8 +89,9 @@ .info = snd_soc_info_volsw_range, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b1738d5..d4732ac 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2922,7 +2922,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2946,12 +2946,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2959,7 +2961,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2979,11 +2995,13 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + int ret; ucontrol->value.integer.value[0] = (snd_soc_read(codec, reg) >> shift) & mask; @@ -2993,6 +3011,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); -- 1.7.10.4

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[alsa-devel] Newbie help with WM8962 and tinyalsa
by Randi Eckstein 19 Dec '12

19 Dec '12

If anyone has an explanation on how to use tinymix to play on a wm8962, I'd really appreciate it. Thanks, Randi Eckstein

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Re: [alsa-devel] [PATCH] ASoC: dapm: Fix the unpaired runtime_get/put cases
by Mark Brown 19 Dec '12

19 Dec '12

On Wed, Dec 19, 2012 at 06:36:37PM +0800, Chuansheng Liu wrote: > But some devices has been set to STANDY bias directly during device probing, > such as cs42l73_probe(): > cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > Then it will cause runtime_get() not be called but laterly runtime_put() will > be called. Also found some other uppaired cases. This is just a bug in the driver, if it's idle_bias_off then it really should be starting in _OFF or at the very least starting actually in _STANDBY (including taking the runtime reference) rather than partially in _STANDBY. > So here add new flag get_runtime, and the logic will be: > 1/ when device is from off to non-off bias, runtime_get() will be called if not yet; > 2/ When device is off bias, runtime_put() will be called if runtime_get() has > been called; This is really not a good idea at all, it's just adding new special cases and making the code more obscure.

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[alsa-devel] Speaker not working; muted?
by Martin Michlmayr 19 Dec '12

19 Dec '12

I've a problem with audio on my HP 2560p laptop (using snd_hda_intel): the speaker no longer works. However, I successfully get sound on a headphone. I've attached the output of alsa-info. It would be great if someone could take a look at let me know if this is a bug in the driver or wrong settings. Background: I used ALSA successfully in the past to play audio with the in-built speaker. At some point, I used aumix (which requires OSS modules) to turn the volume to zero. I believe I turned the volume back up again. I also rebooted my laptop at some point. I don't know if it's related to using aumix or something else, but at some point the in-built speaker stopped working... I can still listen to audio using a headphone. Another observation is that my laptop has a speaker mute button. When I reboot my laptop, and the BIOS or grub is running, I see that the mute button is activated. However, pressing the button doesn't change anything... when the Linux kernel loads, the mute button is still on, but when ALSA starts the button changes colour: it seems to me that ALSA thinks that mute is off while in reality it is on. -- Martin Michlmayr http://www.cyrius.com/

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[alsa-devel] [PATCH 1/2] Revert "ALSA: usb-audio: ignore delay calculation for capture stream"
by Pierre-Louis Bossart 19 Dec '12

19 Dec '12

This reverts commit 3f94fad09538ec988919ec3f371841182df71d04. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..b09b895 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -78,8 +78,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add noise gate hold time enumeration
by Mark Brown 18 Dec '12

18 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++++++ sound/soc/codecs/arizona.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a85a68f..89c0433 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -323,6 +323,15 @@ const struct soc_enum arizona_lhpf4_mode = arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); +static const char *arizona_ng_hold_text[] = { + "30ms", "120ms", "250ms", "500ms", +}; + +const struct soc_enum arizona_ng_hold = + SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, + 4, arizona_ng_hold_text); +EXPORT_SYMBOL_GPL(arizona_ng_hold); + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index db0c8dd..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -176,6 +176,8 @@ extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; +extern const struct soc_enum arizona_ng_hold; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: arizona: Correct FLL source definitions
by Mark Brown 18 Dec '12

18 Dec '12

The FLL source constants were numbered as a simple enumeration but were being used in the code as direct values to be written to the registers. Renumber the constants to reflect the usage. Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 7f22b4f..68dfae5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 -- 1.7.10.4

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[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of more Haswell HDMI codec pins
by mengdong.lin＠intel.com 18 Dec '12

18 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec may report invalid connection list entries on all three pins. But our previous patch only fixes the HDMI pin, not the DP pin, so audio playbak via DP can still fail. This patch adds fixup for Haswell to set proper connection list entries for the other 2 pins including the DP pin. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 42fdf14..73d298a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1710,8 +1710,14 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, if (vendor_param == -1 || !(vendor_param & 0x02)) return; + snd_hda_override_conn_list(codec, 0x05, 3, list); + hdmi_read_pin_conn(codec, 1); + snd_hda_override_conn_list(codec, 0x06, 3, list); hdmi_read_pin_conn(codec, 1); + + snd_hda_override_conn_list(codec, 0x07, 3, list); + hdmi_read_pin_conn(codec, 1); } -- 1.7.9.5

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Re: [alsa-devel] [pulseaudio-discuss] Front mic not working properly
by Raymond Yau 18 Dec '12

18 Dec '12

>> > >> > I'm totally stuck with a sound problem I fail to debug on my own. >> > >> > My setup: >> > Asus P5Q Motherboard with Realtec ALC1200 soundchip. Stereo speakers plugged in to the rear output and headphone speaker/mic combo plugged in to the front jacks. System: Ubuntu 12.10 >> > Desired behaviour: Sound is played back on headphones AND rear speaker. Front mic should "just-work" Asus P5Q specification Realtek ALC1200, 8-channel High Definition Audio CODEC - Support Jack-Detection, Multi-streaming, and Front Panel Jack-Retasking Alt_playback is still not enabled to support multi streaming playback on realtec codecs >> > Problem 2: >> > This is the biggest problem. >> > As soon as I plug in the front mic, Inside gnome-alsamixer the volume for master, surround,center,lfe start to jump between on and off. Why is the Front mic jack controlling these values? Do you mean BIOS did not setup pin default correctly after asus-p5q model was removed? http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=6f… > How can I debug this behaviour? Post the output of alsa-info.sh > I just want them to stay the same, if the mic is plugged in or not. >> > Inside the gnome volume settings, the output port starts to jump between "Analog Output" and "Analog Headphones" >> >> This look like problem in Jack detection of the headphone and Mic Jack of front audio panel >> >> alsamixer -c0 >> >> Does the playback switch change rapidly on and off in alsamixer ? >> >> Do your computer chassis have a hda front audio panel and you connect them to the front audio connector of the motherboard according to the user manual ? >> >> Does it help if you disabled the automute switch ? >> > Thanks a lot. Your input helped me to at least work around the problem. My chassis has a HDA and an AC97 connector. The HDA connector was plugged in and correctly configured in BIOS. > I now use the AC97 connector (and changed the BIOS settings) and everything is working properly. Checking the front audio panel specification of your computer chassis Setting front audio panel to AC97 in BIOS of Asus motherboard change the miscellaneous bit of pin default of front HP and Mic Jack （i.e. misc = NO_PRESENCE) The alsa driver won't enable the unsolicited event on these nodes > > Regarding your questions: Disabling automute didn't help. And yes, the playback switch toggled rapidly. http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Try disable trigger sense by codec->no_trigger_sense = 1; > > One thing I didn't test: I mistakenly plugged the rear speaker into the front channel jack (plugging the green connector into a green jack seemed so obvious). I don't know is this is the same channel as the headphone connector on the front panel... > > In AC97 mode it doesn't matter if I use the front channel jack or the rear channel jack (both at the rear of the computer) to connect my main speakers.

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[alsa-devel] [PATCH 0/2] A few fixes for Reatek codecs
by Takashi Iwai 17 Dec '12

17 Dec '12

Hi, here is a patch series for fixing some obvious errors I've overseen for Realtek codecs. It'll be queued for 3.8-rc1. Takashi

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[alsa-devel] [sound:topic/hda-ca0132-dsp 4/11] sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)'
by Dan Carpenter 17 Dec '12

17 Dec '12

Hi Ian, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-ca0132-dsp head: cdc83c59e59d57b250be46c4b9d31e3b2b5ae589 commit: 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 [4/11] ALSA: hda - Update CA0132 codec to load DSP firmware binary + sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)' sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not propagate 'status' from dsp_dma_stop() instead of (-5)? sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1857) git remote add sound git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git remote update sound git checkout 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 vim +714 sound/pci/hda/patch_ca0132.c 01ef7dbf Ian Minett 2012-09-20 698 VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); 01ef7dbf Ian Minett 2012-09-20 699 mutex_unlock(&spec->chipio_mutex); 01ef7dbf Ian Minett 2012-09-20 700 } 01ef7dbf Ian Minett 2012-09-20 701 01ef7dbf Ian Minett 2012-09-20 702 /* 01ef7dbf Ian Minett 2012-09-20 703 * CA0132 DSP IO stuffs 01ef7dbf Ian Minett 2012-09-20 704 */ 01ef7dbf Ian Minett 2012-09-20 705 static int dspio_send(struct hda_codec *codec, unsigned int reg, 01ef7dbf Ian Minett 2012-09-20 706 unsigned int data) 01ef7dbf Ian Minett 2012-09-20 707 { 01ef7dbf Ian Minett 2012-09-20 708 unsigned int res; 01ef7dbf Ian Minett 2012-09-20 709 int retry = 50; 01ef7dbf Ian Minett 2012-09-20 710 01ef7dbf Ian Minett 2012-09-20 711 /* send bits of data specified by reg to dsp */ 01ef7dbf Ian Minett 2012-09-20 712 do { 01ef7dbf Ian Minett 2012-09-20 713 res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); 01ef7dbf Ian Minett 2012-09-20 @714 if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) 01ef7dbf Ian Minett 2012-09-20 715 return res; 01ef7dbf Ian Minett 2012-09-20 716 } while (--retry); 01ef7dbf Ian Minett 2012-09-20 717 01ef7dbf Ian Minett 2012-09-20 718 return -EIO; 01ef7dbf Ian Minett 2012-09-20 719 } 01ef7dbf Ian Minett 2012-09-20 720 01ef7dbf Ian Minett 2012-09-20 721 static void dspio_write_wait(struct hda_codec *codec) 01ef7dbf Ian Minett 2012-09-20 722 { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] The new alsa-driver repository
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Hello all, the ALSA server has new alsa-driver repository which contains several branches for now: master - just empty one with README build - the previous master branch (but all was moved to the alsa/ subtree) mirror - mirrored code from upstream (ALSA parts only) release - merging branch for mirror/build trees to create a relationship between them All branches are ophan. It means that they have no common parent. I put some handy configuration hints to the ALSA WIKI: http://www.alsa-project.org/main/index.php/GIT_Server#Syncing_the_branch_re… The reason to move the old master branch to the alsa/ subtree is quite simple - you can merge/pull the branch to the full linux-2.6 kernel tree now and do your development there. Also, cherry picks between mirror and the linux-2.6 tree should work. Another handy extension - in the current repo - might be to create the linux-2.6 branch with the whole upstream and create another branch for merging the build branch with the whole tree like 'release26' or so. In this way, we can manage another releases of the out-of-kernel build code for the ALSA tree in stable linux kernel versions. Comments, other ideas are welcome. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ALSA: hda - Fix clicking noise on Thinkpad T61/R61i
by David Henningsson 17 Dec '12

17 Dec '12

The bug reporter reports that setting the speaker pin to D3 before turning off its pinctl fixes the clicking noise on powersave for Thinkpad T61. Thanks to c4pp4 for doing most of the work with this bug, including reporting and testing, and writing preliminary patches. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 36 ++++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) Note: c4pp4 pointed out that in his case, only the speaker needed to go to D3 before pinctl, but noticed no regressions from the below implementation, so I made the fixup more generic (easier to reuse later). diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a3a2263..27e6ca7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -144,6 +144,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; + unsigned int d3_before_shutup:1; /* Needs speaker to go to D3 before pinctl */ unsigned int adc_switching:1; @@ -558,13 +559,22 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM static int conexant_suspend(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + if (spec->d3_before_shutup) + snd_hda_codec_set_power_to_all(codec, codec->afg, AC_PWRST_D3, + false); snd_hda_shutup_pins(codec); return 0; } -#endif + +static void conexant_reboot_notify(struct hda_codec *codec) +{ + conexant_suspend(codec); +} + static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, @@ -575,7 +585,7 @@ static const struct hda_codec_ops conexant_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4408,7 +4418,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; /* @@ -4421,6 +4431,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_D3_BEFORE_SHUTUP, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4430,6 +4441,13 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt_fixup_d3_before_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + spec->d3_before_shutup = 1; +} + static void cxt5066_increase_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4499,6 +4517,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_D3_BEFORE_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_d3_before_shutup, + }, +}; + +static const struct snd_pci_quirk cxt5045_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20ac, "Lenovo T61/R61i", CXT_FIXUP_D3_BEFORE_SHUTUP), + {} }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4553,6 +4580,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; + snd_hda_pick_fixup(codec, NULL, cxt5045_fixups, cxt_fixups); break; case 0x14f15051: add_cx5051_fake_mutes(codec); -- 1.7.9.5

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[alsa-devel] ASoC updates for v3.8
by Mark Brown 17 Dec '12

17 Dec '12

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[alsa-devel] [PATCH 0/3] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 17 Dec '12

17 Dec '12

Hi Mark, This series goes on top of current Linus' git tree and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. I have also included a typo patch that you have already taken as 315656b (ASoC: atmel-ssc: change disable to disable in dts node) in your topic/atmel branch but that was not described properly. If you find it easier, I can provide you a git tree for this material: just tell me... Thanks for your help, best regards, Bo Shen (3): ARM: at91/dts: fix typo in SSC status property ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 +++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9g45.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9n12.dtsi | 26 +++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 153 insertions(+), 11 deletions(-) -- 1.8.0

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Re: [alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Nicolas Ferre 17 Dec '12

17 Dec '12

On 12/17/2012 10:39 AM, Jean-Christophe PLAGNIOL-VILLARD : > On 11:24 Wed 21 Nov , Mark Brown wrote: >> On Wed, Nov 21, 2012 at 10:17:09AM +0800, Bo Shen wrote: >>> Add pinctrl support for atmel ssc peripheral >>> >>> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> >> >> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > you handle this one with scc Hold on Jean-Christophe, I made a little experiment this morning, and this patch is not applying correctly on: - linus' tree - linux-next - neither Mark's for-next branch So, *I* will manage this patch and try to make sure that Mark is able to seamlessly include it in his tree on top of a known source for a future ASoC "fixes" pull-request. Mark, what about rebasing this one on top of current Linus' tree (toguether with the one that you have just applied on the topic/atmel branch: (ASoC: atmel-ssc: change disable to disable in dts node)? I hope that once reworked, they will be able to reach Linus' tree soon because the lack of this patch is causing DTB compilation errors on all AT91... Best regards, -- Nicolas Ferre

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[alsa-devel] [PATCH] ASoC: atmel-ssc: change disable to disable in dts node
by Bo Shen 17 Dec '12

17 Dec '12

Change the value of status to disabled to keep the consistent Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- arch/arm/boot/dts/at91sam9260.dtsi | 2 +- arch/arm/boot/dts/at91sam9263.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9g45.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9x5.dtsi | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index aaa42d8..03c56cb 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,7 +217,7 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffe0000 { diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 3b721ee..15f12dd 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -179,14 +179,14 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fff9c000 { compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; macb0: ethernet@fffbc000 { diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index acfa207..44a38d0 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -232,14 +232,14 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fffa0000 { compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffb0000 { diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 69667d0..0beff72 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -92,7 +92,7 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xf0010000 0x4000>; interrupts = <28 4 5>; - status = "disable"; + status = "disabled"; }; tcb0: timer@f8008000 { -- 1.7.9.5

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[alsa-devel] "hda-intel: azx_get_response timeout"
by Mantas M. 17 Dec '12

17 Dec '12

After upgrading to Linux 3.7 (on Arch Linux, x86_64), I've been getting the following messages in dmesg: > [Dec11 21:52] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x001f0500 > [ +1.006709] hda-intel: No response from codec, disabling MSI: last cmd=0x001f0500 > [ +1.006773] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f0500 This seems to happen when resuming from suspend-to-memory, and is new to v3.7 – I haven't seen any such messages on any kernels I've used earlier (between v3.0 and v3.6.9). The device is: > # lspci -v > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) > Subsystem: ASUSTeK Computer Inc. Device 1643 > Flags: bus master, fast devsel, latency 0, IRQ 22 > Memory at d5200000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > Capabilities: [100] Virtual Channel > Capabilities: [130] Root Complex Link > Kernel driver in use: snd_hda_intel Is there anything I can/should do about this? -- Mantas Mikulėnas <grawity(a)gmail.com>

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Re: [alsa-devel] Could you help me about the ALSA?
by Zhang wei 17 Dec '12

17 Dec '12

> Yes,when buffer_size wraped the 'pos' is smallar than last one.If we > want to confirm the hw_base need to adjust there must be consider > another confidition: > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) > > how to connect this code with checking unnormal interrupt(lost or delayed)? This condition is for the double interrupt acknowledge problems (see comment). It makes sure, that the DMA buffer position was wrapped using another external timing source. Indeed, it's mostly a lowlevel driver fault to call this pointer update routine with the in_interrupt flag when the period was not really elapsed. -------------------------------------- I am sorry. I forget signed my info and this mail sent by me. Yes,I saw the comment and known the target of the pices of code and known alsa use jiffies as another timing source to detect the wrap situation,but I do not understand how to detect or check that case.In other word,why does alsa use the value runtime->hw_ptr_buffer_jiffies/2 not runtime->hw_ptr_buffer_jiffies/3 or runtime->hw_ptr_buffer_jiffies/4 ...? Best Regards zw

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Re: [alsa-devel] Could you help me about the ALSA?
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Date 14.12.2012 09:07, Zhang wei wrote: > Good afternoon . > > Excuse me if I disturb you, I found your contact information from the > alsa-devel > http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030605.html > . > > Recently I am studying the alsa-driver and the code blocked me which > from the function snd_pcm_update_hw_ptr0inthe pcm_lib.c. > > =====================start==============================è > > if (in_interrupt) { > > /* we know that one period was processed */ > > /* delta = "expected next hw_ptr" for in_interrupt != > 0 */ > > delta = runtime->hw_ptr_interrupt + runtime->period_size; > > if (delta > new_hw_ptr) { > > /* check for double acknowledged interrupts */ > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { > > hw_base += runtime->buffer_size; > > if (hw_base >= runtime->boundary) > > hw_base = 0; > > new_hw_ptr = hw_base + pos; > > goto __delta; > > } > > } > > } > > ç================end================================= > > I’ve already read your mail but I still do not understand your intend. I > know the target of the pices of code is prevent the variable hw_ptr and > hw_base delayed update from some kind of case .so my question is > > (1) how could happen on delta>new_hw_ptr? And why? > > (2) Why we can conclue the hw_ptr crossed the boundary of > buffer_size when hdelta>runitme->hw_ptr_jiffies/2? What principle we > based on? The interrupts can be lost or delayed in some cases (small period sizes etc.). So you need to check for these situations. The driver returns only position in the ring buffer without count of "cross" points, so we need to trace these situations using another clock source (like system clock). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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