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[alsa-devel] [PATCH] Fix oops caused by recent commit "Fix internal mic for Lenovo Ideapad U300s"
by David Henningsson 10 Apr '12

10 Apr '12

Make sure we don't dereference the "quirk" pointer when it is null. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cbe115b..abb59f4 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4441,7 +4441,9 @@ static void apply_fixup(struct hda_codec *codec, struct conexant_spec *spec = codec->spec; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk && table[quirk->value]) { + if (!quirk) + return; + if (table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); -- 1.7.9.1

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Re: [alsa-devel] [PATCH] alsa: hide HDMI/ELD printks unless in debug kernels
by Takashi Iwai 10 Apr '12

10 Apr '12

At Mon, 09 Apr 2012 09:12:44 -0600, Stephen Warren wrote: > > On 04/08/2012 08:15 PM, Fengguang Wu wrote: > > These trivial messages will show up repeatedly on hot plug as well as > > video mode changes, which could be annoying. > > > > Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> > > Those messages are very useful for end-users to diagnose problems with > HDMI audio. Is there a way to make the messages dynamic, so users can > get them if they want, without recompiling their kernel? We can define a new macro like below, add call like _snd_printd(2, ....) instead of snd_printd(). (Yes, the number 2 can be replaced better in an enum, too.) In future, we should move to the standard dynamic_printk, but a patch like this would be a good compromise for merging to 3.4 kernel, IMO. Takashi --- diff --git a/include/sound/core.h b/include/sound/core.h index b6e0f57..442f822 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -354,6 +354,8 @@ void __snd_printk(unsigned int level, const char *file, int line, */ #define snd_printd(fmt, args...) \ __snd_printk(1, __FILE__, __LINE__, fmt, ##args) +#define _snd_printd(level, fmt, args...) \ + __snd_printk(level, __FILE__, __LINE__, fmt, ##args) /** * snd_BUG - give a BUG warning message and stack trace @@ -383,6 +385,7 @@ void __snd_printk(unsigned int level, const char *file, int line, #else /* !CONFIG_SND_DEBUG */ #define snd_printd(fmt, args...) do { } while (0) +#define _snd_printd(level, fmt, args...) do { } while (0) #define snd_BUG() do { } while (0) static inline int __snd_bug_on(int cond) {

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[alsa-devel] [PATCH 24/26] ASoC: wm8994: Don't test for NULL before release_firmware()
by Jesper Juhl 10 Apr '12

10 Apr '12

release_firmware() does its own NULL ptr testing, it's redundant to also test before calling it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/wm8994.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642..cf7e5d2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3972,7 +3972,7 @@ err_irq: return ret; } -static int wm8994_codec_remove(struct snd_soc_codec *codec) +static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -4013,14 +4013,10 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } - if (wm8994->mbc) - release_firmware(wm8994->mbc); - if (wm8994->mbc_vss) - release_firmware(wm8994->mbc_vss); - if (wm8994->enh_eq) - release_firmware(wm8994->enh_eq); + release_firmware(wm8994->mbc); + release_firmware(wm8994->mbc_vss); + release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); - return 0; } -- 1.7.10 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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Re: [alsa-devel] ALSA: hda - Fix internal mic for Lenovo Ideapad U300s
by Dan Carpenter 10 Apr '12

10 Apr '12

Hello David Henningsson, This is a semi-automatic email about new static checker warnings. The patch 18dcd3044e4c: "ALSA: hda - Fix internal mic for Lenovo Ideapad U300s" from Apr 2, 2012, leads to the following Smatch complaint: sound/pci/hda/patch_conexant.c:4449 apply_fixup() error: we previously assumed 'quirk' could be null (see line 4444) sound/pci/hda/patch_conexant.c 4443 quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); 4444 if (quirk && table[quirk->value]) { ^^^^^ Old check. 4445 snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", 4446 quirk->name); 4447 apply_pincfg(codec, table[quirk->value]); 4448 } 4449 if (quirk->value == CXT_FIXUP_STEREO_DMIC) { ^^^^^^^^^^^^ New unchecked dereference. 4450 snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", 4451 quirk->name); regards, dan carpenter

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Re: [alsa-devel] [PATCH][Trivial] Delete trailing whitespace from sound/soc/codecs/wm8994.c
by Jesper Juhl 10 Apr '12

10 Apr '12

On Mon, 9 Apr 2012, Mark Brown wrote: > On Mon, Apr 09, 2012 at 12:40:32AM +0200, Jesper Juhl wrote: > > While reading through sound/soc/codecs/wm8994.c I noticed a fair > > amount of trailing whitespace. This patch gets rid of it. > > Please try to make sure your changelog lines up with the subsysetem - > this is especially important with trivial patches. > Arrgh, sorry about that. I always forget (although not intentionally) to check if there's some special convention for a given subsystem :-( I guess this would have been better: ASoC: wm8994: Delete trailing whitespace -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [PATCH 23/26] ALSA: riptide: remove redundant NULL test before release_firmware()
by Jesper Juhl 10 Apr '12

10 Apr '12

release_firmware() deals gracefully with NULL pointers, no need to check first. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/pci/riptide/riptide.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 0481d94..cbeb3f7 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1837,8 +1837,7 @@ static int snd_riptide_free(struct snd_riptide *chip) } if (chip->irq >= 0) free_irq(chip->irq, chip); - if (chip->fw_entry) - release_firmware(chip->fw_entry); + release_firmware(chip->fw_entry); release_and_free_resource(chip->res_port); kfree(chip); return 0; -- 1.7.10 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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Re: [alsa-devel] [PATCH] sound/isa/sscape.c: add missing resource-release code
by Takashi Iwai 10 Apr '12

10 Apr '12

At Mon, 9 Apr 2012 10:16:35 +0200, Julia Lawall wrote: > > From: Julia Lawall <Julia.Lawall(a)lip6.fr> > > At the point of this error-handling code, both regions and the dma have > been allocated, so free it as done in previous and subsequent > error-handling code. > > Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Applied now. Thanks. Takashi > > --- > sound/isa/sscape.c | 6 ++++-- > 1 file changed, 4 insertions(+), 2 deletions(-) > > diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c > index b4a6aa9..131dd98 100644 > --- a/sound/isa/sscape.c > +++ b/sound/isa/sscape.c > @@ -1019,13 +1019,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) > irq_cfg = get_irq_config(sscape->type, irq[dev]); > if (irq_cfg == INVALID_IRQ) { > snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); > - return -ENXIO; > + err = -ENXIO; > + goto _release_dma; > } > > mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); > if (mpu_irq_cfg == INVALID_IRQ) { > snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); > - return -ENXIO; > + err = -ENXIO; > + goto _release_dma; > } > > /* >

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Re: [alsa-devel] [PATCH] sound/oss/msnd_pinnacle.c: add vfrees
by Takashi Iwai 10 Apr '12

10 Apr '12

At Mon, 9 Apr 2012 10:16:32 +0200, Julia Lawall wrote: > > From: Julia Lawall <Julia.Lawall(a)lip6.fr> > > At the point of this error-handling code, HAVE_DSPCODEH may be undefined, > so free INITCODE and PERMCODE as done elsewhere. A jump and label are > introduced to avoid code duplication. > > Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> Thanks, applied now. Takashi > > --- > sound/oss/msnd_pinnacle.c | 8 ++++++-- > 1 file changed, 6 insertions(+), 2 deletions(-) > > diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c > index 2c79d60..536c4c0 100644 > --- a/sound/oss/msnd_pinnacle.c > +++ b/sound/oss/msnd_pinnacle.c > @@ -1294,6 +1294,8 @@ static int __init calibrate_adc(WORD srate) > > static int upload_dsp_code(void) > { > + int ret = 0; > + > msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); > #ifndef HAVE_DSPCODEH > INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE); > @@ -1312,7 +1314,8 @@ static int upload_dsp_code(void) > memcpy_toio(dev.base, PERMCODE, PERMCODESIZE); > if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) { > printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); > - return -ENODEV; > + ret = -ENODEV; > + goto out; > } > #ifdef HAVE_DSPCODEH > printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); > @@ -1320,12 +1323,13 @@ static int upload_dsp_code(void) > printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); > #endif > > +out: > #ifndef HAVE_DSPCODEH > vfree(INITCODE); > vfree(PERMCODE); > #endif > > - return 0; > + return ret; > } > > #ifdef MSND_CLASSIC >

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[alsa-devel] [PATCH][Trivial] Delete trailing whitespace from sound/soc/codecs/wm8994.c
by Jesper Juhl 09 Apr '12

09 Apr '12

While reading through sound/soc/codecs/wm8994.c I noticed a fair amount of trailing whitespace. This patch gets rid of it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/wm8994.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642..c6c2cb2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -398,7 +398,7 @@ static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block) wm8994->dac_rates[iface]); /* The EQ will be disabled while reconfiguring it, remember the - * current configuration. + * current configuration. */ save = snd_soc_read(codec, base); save &= WM8994_AIF1DAC1_EQ_ENA; @@ -991,7 +991,7 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) WM8994_CP_DYN_SRC_SEL_MASK, source | WM8994_CP_DYN_PWR); wm8994->hubs.class_w = true; - + } else { dev_dbg(codec->dev, "Class W disabled\n"); snd_soc_update_bits(codec, WM8994_CLASS_W_1, @@ -1358,7 +1358,7 @@ static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); static const char *mono_pcm_out_text[] = { - "None", "AIF2ADCL", "AIF2ADCR", + "None", "AIF2ADCL", "AIF2ADCR", }; static const struct soc_enum mono_pcm_out_enum = @@ -2122,7 +2122,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, case WM8994_SYSCLK_OPCLK: /* Special case - a division (times 10) is given and - * no effect on main clocking. + * no effect on main clocking. */ if (freq) { for (i = 0; i < ARRAY_SIZE(opclk_divs); i++) @@ -2958,14 +2958,14 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) /* Expand the array... */ t = krealloc(wm8994->retune_mobile_texts, - sizeof(char *) * + sizeof(char *) * (wm8994->num_retune_mobile_texts + 1), GFP_KERNEL); if (t == NULL) continue; /* ...store the new entry... */ - t[wm8994->num_retune_mobile_texts] = + t[wm8994->num_retune_mobile_texts] = pdata->retune_mobile_cfgs[i].name; /* ...and remember the new version. */ @@ -3907,7 +3907,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8994_dac_widgets)); break; } - wm_hubs_add_analogue_routes(codec, 0, 0); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); -- 1.7.10 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [RFC 1/2] snd-usb-audio: Skip un-parseable mixer units instead of erroring
by Mark Hills 09 Apr '12

09 Apr '12

Some interfaces reference endpoints which do not exists. To accomodate these, do not fail completely, but skip over them. This allows the Electrix Ebox-44 with earlier firmware to be detected and used for audio. Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/usb/mixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ab23869..c374c72 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1388,7 +1388,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r for (pin = 0; pin < input_pins; pin++) { err = parse_audio_unit(state, desc->baSourceID[pin]); if (err < 0) - return err; + continue; err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) return err; -- 1.7.4.4

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[alsa-devel] [PATCH] alsa: hide HDMI/ELD printks unless in debug kernels
by Fengguang Wu 09 Apr '12

09 Apr '12

These trivial messages will show up repeatedly on hot plug as well as video mode changes, which could be annoying. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_eld.c | 6 +++--- sound/pci/hda/patch_hdmi.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) --- linux.orig/sound/pci/hda/hda_eld.c 2011-11-11 19:08:21.000000000 +0800 +++ linux/sound/pci/hda/hda_eld.c 2011-11-11 19:09:35.000000000 +0800 @@ -410,7 +410,7 @@ static void hdmi_show_short_audio_desc(s else buf2[0] = '\0'; - printk(KERN_INFO "HDMI: supports coding type %s:" + snd_printdd("HDMI: supports coding type %s:" " channels = %d, rates =%s%s\n", cea_audio_coding_type_names[a->format], a->channels, @@ -434,14 +434,14 @@ void snd_hdmi_show_eld(struct hdmi_eld * { int i; - printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + snd_printdd("HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + snd_printdd("HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) --- linux.orig/sound/pci/hda/patch_hdmi.c 2011-11-11 19:07:15.000000000 +0800 +++ linux/sound/pci/hda/patch_hdmi.c 2011-11-11 19:08:16.000000000 +0800 @@ -757,7 +757,7 @@ static void hdmi_intrinsic_event(struct int pin_idx; struct hdmi_eld *eld; - printk(KERN_INFO + snd_printdd( "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, pd, eldv); @@ -989,7 +989,7 @@ static void hdmi_present_sense(struct hd else eld->eld_valid = 0; - printk(KERN_INFO + snd_printdd( "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);

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[alsa-devel] [PATCH][Trivial] sound/pci/riptide/riptide.c: remove trailing whitespace
by Jesper Juhl 09 Apr '12

09 Apr '12

There's a bunch of trailing whitespace at the top of the sound/pci/riptide/riptide.c file - this patch gets rid of it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/pci/riptide/riptide.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 0481d94..a8b07be 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -21,29 +21,29 @@ /* History: - 02/15/2004 first release - + This Driver is based on the OSS Driver version from Linuxant (riptide-0.6lnxtbeta03111100) credits from the original files: - - MODULE NAME: cnxt_rt.h + + MODULE NAME: cnxt_rt.h AUTHOR: K. Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 02/1/2000 KNL - MODULE NAME: int_mdl.c + MODULE NAME: int_mdl.c AUTHOR: Konstantin Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/01/99 KNL - - MODULE NAME: riptide.h + + MODULE NAME: riptide.h AUTHOR: O. Druzhinin (Transcribed by OLD) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/16/97 OLD - MODULE NAME: Rp_Cmdif.cpp + MODULE NAME: Rp_Cmdif.cpp AUTHOR: O. Druzhinin (Transcribed by OLD) K. Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By @@ -51,7 +51,7 @@ Adopted from NT4 driver 6/22/99 OLD Ported to Linux 9/01/99 KNL - MODULE NAME: rt_hw.c + MODULE NAME: rt_hw.c AUTHOR: O. Druzhinin (Transcribed by OLD) C. Lazarev (Transcribed by CNL) HISTORY: Major Revision Date By @@ -73,13 +73,13 @@ ----------------------------- -------- ----- Created 11/18/97 CNL - MODULE NAME: rt_mdl.c + MODULE NAME: rt_mdl.c AUTHOR: Konstantin Lazarev (Transcribed by KNL) HISTORY: Major Revision Date By ----------------------------- -------- ----- Created 10/01/99 KNL - MODULE NAME: mixer.h + MODULE NAME: mixer.h AUTHOR: K. Kenney HISTORY: Major Revision Date By ----------------------------- -------- ----- -- 1.7.10 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [RFC] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 07 Apr '12

07 Apr '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, use a separate mixer control for the inverted channel. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Inverted Internal Mic, take 2. I refined my idea a little. By calling the left channel "Internal Mic" and the right channel "Inverted Internal Mic", we're giving the user a choice whether to activate it or keeping it muted. For PulseAudio, the quick fix is just to make sure the "Inverted Internal Mic" control is muted. If we in the long run want to actually handle this and swap the phase around, we now also have a reasonable indicator if we should do this, by checking if the "Inverted Internal Mic" control exists. For the ALSA init DB (i e, should the "Inverted Internal Mic" be initially muted?) that is no big deal for me, since this will be overwritten by PulseAudio later anyway. I'm also open for better names than "Inverted Internal Mic" if you have any. As with the previous version, this is a draft version of the patch, and could be done slightly more elegant I guess. What do you say? sound/pci/hda/patch_conexant.c | 45 +++++++++++++++++++++++++++++---------- 1 files changed, 33 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e6eafb1..cd3859b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4107,9 +4108,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4119,7 +4120,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4138,7 +4139,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4216,14 +4217,24 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + int chs = 3; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (spec->fixup_stereo_dmic && (strcmp("Internal Mic", label) == 0)) { + /* Make two independent kcontrols */ + int err = cx_auto_add_volume_idx(codec, "Inverted Internal Mic", pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + if (err < 0) + return err; + chs = 1; + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, chs); } return 0; } @@ -4405,22 +4416,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4431,10 +4450,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4477,7 +4498,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds -- 1.7.9.1

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Re: [alsa-devel] [PATCH 0/6] ALSA: hda - CX20549 cleanup/fixes
by Takashi Iwai 07 Apr '12

07 Apr '12

At Fri, 6 Apr 2012 15:34:14 +0200, Michael Karcher wrote: > > This is the revised patch series for the CX20549 (and related stuff) > > Changes: > Patch 1/6: > - spec->single_adc_amp has been moved to codec->single_adc_amp > to fix printing in /proc. > Patch 2/6: > - "Line In" has been changed to "Line". > - This change has also been performed on the Mixer Labels. > Patch 4/6: > - .num_items has been adjusted, too. > - A comment about physical testing has been added to the commit message. > Patch 5/6: > - pin_amp_workaround is not set for CX20549 in auto-setup, too Thanks, applied all patches now. (BTW, the first patch caused a few obvious warnings from checkpatch.pl, so I fixed manually. At the next time, please run checkpatch.pl once before submission.) Takashi

1 0

Re: [alsa-devel] [PATCH 0/6] sound/pci/hda: CX20549 cleanup/fixes
by Takashi Iwai 07 Apr '12

07 Apr '12

At Fri, 06 Apr 2012 15:01:47 +0200, Michael Karcher wrote: > > On Thu, 2012-04-05 at 16:28 +0200, Takashi Iwai wrote: > > BTW, did you try model=auto option to enable the auto-parser? > > If it doesn't work well for you, let me know. > > I tried the model=auto option, too. It works mostly, but the mixer node > seems to not be utilized correctly/at all. The capture source list does > not contain "mixer" anymore, and there are no playback controls for > Internal Mic/Mic/Line. Right, the analog loopback is dropped in the auto-parser for simplicity. > The Capture setup seems to be built correctly > (except for the missing mixer). The auto setup adds two extra input > devices for jack sense. > > > In anyway, give alsa-info.sh output (run with --no-upload option) for > > better checks. > > alsa-info for the revised patch-set (mailed soon) with and without > model=auto is attached. Information I consider irrelevant (non-sound > modules and dmesg context not relating to sound) has been edited out. Thanks. > BTW: I seem to have encountered an ALSA user-space bug: The codec > provides a "master capture switch", i.e. the "mute" of the adc in amp. > It also provides a "capture source selection enum", i.e. the "Connection > Select" of the adc widget. The enum provides mutually exclusive capture > sources that have no volume control, whilst the master capture switch is > coupled with the capture volume control, and should not be related to > the source enumeration. But if I play around with capture sources in > alsamixer, the "Capture Switch" is turned off as soon as I switch the > source from the first item in the Mux list to something else. This sounds like a bug in alsa-lib mixer abstraction. There is a known shortcomings in the "capture source" enum, e.g. it can't take more than one ADCs. > This happens with both the patched hard-coded "BenQ" configuration and > the auto configuration. OK, I'll check it in the next week. thanks, Takashi

1 0

[alsa-devel] ASoC updates for 3.4
by Mark Brown 07 Apr '12

07 Apr '12

The following changes since commit dd775ae2549217d3ae09363e3edb305d0fa19928: Linux 3.4-rc1 (2012-03-31 16:24:09 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 8abe05c6eb358967f16bce8a02c88d57c82cfbd6: ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS (2012-04-06 12:28:49 +0100) ---------------------------------------------------------------- ASoC: fixes for 3.4 A bunch of driver-specific fixes and one generic fix for the new support for platform DAPM contexts - we were picking the wrong default for the idle_bias_off setting which was meaning we weren't actually achieving any useful runtime PM on platform devices. ---------------------------------------------------------------- Fabio Estevam (2): ASoC: imx-audmux: Fix ssi port numbers in sysfs ASoC: imx-audmux: Check for NULL pointer Jarkko Nikula (1): MAINTAINERS: Add missing ASoC OMAP co-maintainer Kuninori Morimoto (1): ASoC: ak4642: fixup: mute needs +1 step Mark Brown (1): MAINTAINERS: Don't list everyone working on Wolfson drivers Martin Jansa (1): ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro Stephen Warren (3): ASoC: tegra: ensure clocks are enabled when touching registers ASoC: set idle_bias_off=1 for all platform DAPM contexts ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS Zeng Zhaoming (1): ASoC: sgtl5000: Enable VAG when DAC/ADC up MAINTAINERS | 4 ++-- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/sgtl5000.c | 25 +++++++++++++------------ sound/soc/imx/imx-audmux.c | 5 ++++- sound/soc/pxa/pxa2xx-i2s.c | 1 + sound/soc/soc-core.c | 2 ++ sound/soc/tegra/tegra_i2s.c | 6 +++++- sound/soc/tegra/tegra_spdif.c | 4 ++++ 8 files changed, 32 insertions(+), 17 deletions(-)

2 1

[alsa-devel] sgtl5000 module install fails during instantiation for i.MX53 QSB
by madhukesh s 07 Apr '12

07 Apr '12

Hi, On insmod'ing alsa modules for i.MX53 QSB, I get the following error: root@freescale-qsb:/lib/modules/2.6.38.8+# modprobe snd-soc-sgtl5000 snd_soc_instantiate_card:Entry snd_soc_instantiate_card:Entry soc_probe_dai_link:Entry soc_probe_codec: Probing for card sgtl5000-audio sgtl5000_enable_regulators:Entry _regulator_get: devname = 1-000a sgtl5000 1-000a: regulator_bulk_get [for (null)]: Failed to get supply 'VDDA': -19 sgtl5000_enable_regulators: regulator_bulk_get err!!! print_constraints: 1-000a: 1200 mV normal _regulator_get: devname = 1-000a sgtl5000 1-000a: regulator_bulk_get [for (null)]: Failed to get supply 'VDDA': -19 sgtl5000 1-000a: Failed to request supplies:1 -19 sgtl5000 1-000a: asoc: failed to probe CODEC sgtl5000.1-000a: -19 asoc: failed to probe codec sgtl5000.1-000a asoc: failed to instantiate card sgtl5000-audio: -19 On adding more printks, I realize that "_regulator_get" function is failing(not able to find supply VDDA). The statically built module gives no such errors. Can someone give some info on this? Regards, maas

1 0

[alsa-devel] Alsa support for conexant
by Sql Sql 06 Apr '12

06 Apr '12

I have a Lenovo Thinkpad T400 with a mini dock which has SPDIF output. It is running Debian Squeeze. When I run alsamixer, I can see the soundcard detected is: Conexant CX20561 When I run aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: Conexant Digital [Conexant Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 However, I can't get it to output through SPDIF. I tried: aplay -Dplug:iec958 /usr/share/sounds/alsa/Front_Center.wav = no sound I tried: pcm.!default { type plug slave { pcm "front:CARD=Intel" } } (restarted alsautils), it still doesn't work. I tried adding: options snd-hda-intel model=lenovo-x200 options snd-hda-intel model=lenovo options snd-hda-intel model=laptop options snd-hda-intel model=thinkpad and then alsa force reload None of these work. Is there anyway I can get it to output through SPDIF? My other computer that has SPDIF outputs fine using the custom asound.conf so why isn't it working for the lenovo? BTW, I confirmed the SPDIF is not defective by booting into Windows and testing it.

2 2

[alsa-devel] [PATCH V2] ASoC: tegra: utils: add support for Tegra30 devices
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Tegra30 has some additional clocks that need to be manipulated, names some clocks differently, runs PLLs at different base rates, etc. The utility code needs to handle this. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Determine Tegra version internally to the utility code, rather than requiring each ASoC machine driver be updated to pass in this information. sound/soc/tegra/tegra_asoc_utils.c | 29 ++++++++++++++++++++++++----- sound/soc/tegra/tegra_asoc_utils.h | 9 +++++++-- 2 files changed, 31 insertions(+), 7 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f8428e4..f59e8b5 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -2,7 +2,7 @@ * tegra_asoc_utils.c - Harmony machine ASoC driver * * Author: Stephen Warren <swarren(a)nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -25,6 +25,7 @@ #include <linux/err.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/of.h> #include "tegra_asoc_utils.h" @@ -40,7 +41,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 22050: case 44100: case 88200: - new_baseclock = 56448000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 56448000; + else + new_baseclock = 564480000; break; case 8000: case 16000: @@ -48,7 +52,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, case 48000: case 64000: case 96000: - new_baseclock = 73728000; + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + new_baseclock = 73728000; + else + new_baseclock = 552960000; break; default: return -EINVAL; @@ -78,7 +85,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return err; } - /* Don't set cdev1 rate; its locked to pll_a_out0 */ + /* Don't set cdev1/extern1 rate; it's locked to pll_a_out0 */ err = clk_enable(data->clk_pll_a); if (err) { @@ -112,6 +119,15 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->dev = dev; + if (!of_have_populated_dt()) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra20")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; + else if (of_machine_is_compatible("nvidia,tegra30")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; + else + return -EINVAL; + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); @@ -126,7 +142,10 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a; } - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) + data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + else + data->clk_cdev1 = clk_get_sys("extern1", NULL); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 4818195..44db1db 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -2,7 +2,7 @@ * tegra_asoc_utils.h - Definitions for Tegra DAS driver * * Author: Stephen Warren <swarren(a)nvidia.com> - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010,2012 - NVIDIA, Inc. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -26,8 +26,14 @@ struct clk; struct device; +enum tegra_asoc_utils_soc { + TEGRA_ASOC_UTILS_SOC_TEGRA20, + TEGRA_ASOC_UTILS_SOC_TEGRA30, +}; + struct tegra_asoc_utils_data { struct device *dev; + enum tegra_asoc_utils_soc soc; struct clk *clk_pll_a; struct clk *clk_pll_a_out0; struct clk *clk_cdev1; @@ -42,4 +48,3 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); #endif - -- 1.7.0.4

2 1

[alsa-devel] [PATCH V2] ASoC: tegra: set a sensible initial clock rate
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Initialize the audio clock tree appropriately for some reasonable rate. This makes sure the PLLs etc. are actually programmed to something reasonable when the audio driver is loaded. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Use 44100 as the default rate, not 48000, since it's more common, and consistent with the clock tables in board-dt-tegra*.c. This patch is ordered after the patch I just sent to add Tegra30 support to the utils file. However, the dependency is just for context; this should be independent from that patch. sound/soc/tegra/tegra_asoc_utils.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index f59e8b5..266189d 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -152,8 +152,14 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, goto err_put_pll_a_out0; } + ret = tegra_asoc_utils_set_rate(data, 44100, 256 * 44100); + if (ret) + goto err_put_cdev1; + return 0; +err_put_cdev1: + clk_put(data->clk_cdev1); err_put_pll_a_out0: clk_put(data->clk_pll_a_out0); err_put_pll_a: -- 1.7.0.4

2 1

Re: [alsa-devel] [PATCH] ASoC: JZ4740: Replaced comma operators with semicolons.
by Mark Brown 06 Apr '12

06 Apr '12

On Fri, Apr 06, 2012 at 07:16:27PM +0200, Maarten ter Huurne wrote: > They were harmless but also unnecessary, probably a leftover from earlier code. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: tegra: use devm_ APIs in SPDIF driver
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The devm_ APIs remove the need to manually clean up allocations, thus removing some code. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This patch is ordered after the header re-order/cleanup patch that I just posted. This only affects context (well, actually just line numbers). sound/soc/tegra/tegra20_spdif.c | 33 ++++++++++----------------------- 1 files changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 00a2d33..ef5d49e 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -245,11 +245,12 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) struct resource *mem, *memregion, *dmareq; int ret; - spdif = kzalloc(sizeof(struct tegra20_spdif), GFP_KERNEL); + spdif = devm_kzalloc(&pdev->dev, sizeof(struct tegra20_spdif), + GFP_KERNEL); if (!spdif) { dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); ret = -ENOMEM; - goto exit; + goto err; } dev_set_drvdata(&pdev->dev, spdif); @@ -257,7 +258,7 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) if (IS_ERR(spdif->clk_spdif_out)) { pr_err("Can't retrieve spdif clock\n"); ret = PTR_ERR(spdif->clk_spdif_out); - goto err_free; + goto err; } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -274,19 +275,19 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_clk_put; } - memregion = request_mem_region(mem->start, resource_size(mem), - DRV_NAME); + memregion = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), DRV_NAME); if (!memregion) { dev_err(&pdev->dev, "Memory region already claimed\n"); ret = -EBUSY; goto err_clk_put; } - spdif->regs = ioremap(mem->start, resource_size(mem)); + spdif->regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!spdif->regs) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENOMEM; - goto err_release; + goto err_clk_put; } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; @@ -298,7 +299,7 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; - goto err_unmap; + goto err_clk_put; } ret = tegra_pcm_platform_register(&pdev->dev); @@ -313,37 +314,23 @@ static __devinit int tegra20_spdif_platform_probe(struct platform_device *pdev) err_unregister_dai: snd_soc_unregister_dai(&pdev->dev); -err_unmap: - iounmap(spdif->regs); -err_release: - release_mem_region(mem->start, resource_size(mem)); err_clk_put: clk_put(spdif->clk_spdif_out); -err_free: - kfree(spdif); -exit: +err: return ret; } static int __devexit tegra20_spdif_platform_remove(struct platform_device *pdev) { struct tegra20_spdif *spdif = dev_get_drvdata(&pdev->dev); - struct resource *res; tegra_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); tegra20_spdif_debug_remove(spdif); - iounmap(spdif->regs); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, resource_size(res)); - clk_put(spdif->clk_spdif_out); - kfree(spdif); - return 0; } -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: tegra: sort includes, remove mach/iomap.h
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra20_das.c | 5 ++--- sound/soc/tegra/tegra20_i2s.c | 7 +++---- sound/soc/tegra/tegra20_spdif.c | 5 ++--- sound/soc/tegra/tegra_pcm.c | 2 +- 4 files changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index 486d7b2..812696d 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -20,14 +20,13 @@ * */ -#include <linux/module.h> #include <linux/debugfs.h> #include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> #include <linux/platform_device.h> #include <linux/seq_file.h> #include <linux/slab.h> -#include <linux/io.h> -#include <mach/iomap.h> #include <sound/soc.h> #include "tegra20_das.h" diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 878798c..9427f36 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -29,15 +29,14 @@ */ #include <linux/clk.h> -#include <linux/module.h> #include <linux/debugfs.h> #include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/seq_file.h> #include <linux/slab.h> -#include <linux/io.h> -#include <linux/of.h> -#include <mach/iomap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 052bff8..00a2d33 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -21,14 +21,13 @@ */ #include <linux/clk.h> -#include <linux/module.h> #include <linux/debugfs.h> #include <linux/device.h> +#include <linux/io.h> +#include <linux/module.h> #include <linux/platform_device.h> #include <linux/seq_file.h> #include <linux/slab.h> -#include <linux/io.h> -#include <mach/iomap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 476b8ac..127348d 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -29,8 +29,8 @@ * */ -#include <linux/module.h> #include <linux/dma-mapping.h> +#include <linux/module.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> -- 1.7.0.4

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Re: [alsa-devel] [PATCH REPOST] ASoC: tegra: complete Tegra->Tegra20 renaming
by Mark Brown 06 Apr '12

06 Apr '12

On Fri, Apr 06, 2012 at 10:30:52AM -0600, Stephen Warren wrote: > From: Stephen Warren <swarren(a)nvidia.com> > > Rename Tegra20-specific Kconfig variables, module filenames, all internal > symbol names, clocks, and platform devices, to reflect the fact the DAS > and I2S drivers are for a specific HW version. Applied, thanks.

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Re: [alsa-devel] [PATCH 2/2] ASoC: tegra: complete Tegra->Tegra20 renaming
by Mark Brown 06 Apr '12

06 Apr '12

On Thu, Apr 05, 2012 at 03:54:54PM -0600, Stephen Warren wrote: > From: Stephen Warren <swarren(a)nvidia.com> > > Rename Tegra20-specific Kconfig variables, module filenames, all internal > symbol names, clocks, and platform devices, to reflect the fact the DAS > and I2S drivers are for a specific HW version. Oops, sorry - replied to the wrong one - this one doesn't seem to apply against for-3.5, I applied 1/2 though. I believe this is due to 3.4 fixes not having propagated over 3.5 yet. Can you regenerate against for-3.5 please, or perhaps it's best to wait until things have merged back up?

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Re: [alsa-devel] [PATCH 2/2] ASoC: tegra: complete Tegra->Tegra20 renaming
by Mark Brown 06 Apr '12

06 Apr '12

On Thu, Apr 05, 2012 at 03:54:54PM -0600, Stephen Warren wrote: > From: Stephen Warren <swarren(a)nvidia.com> > > Rename Tegra20-specific Kconfig variables, module filenames, all internal > symbol names, clocks, and platform devices, to reflect the fact the DAS > and I2S drivers are for a specific HW version. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
by Stephen Warren 06 Apr '12

06 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed the prototype of tegra_i2s_debug_add, but didn't update the dummy inline used when !CONFIG_DEBUG_FS. Fix that. Cc: <stable(a)vger.kernel.org> # 3.3 Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 86f5ca4..64fd12e 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -116,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) { } -- 1.7.5.4

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[alsa-devel] [PATCH 1/2] ASoC: tegra: rename Tegra20-specific driver files
by Stephen Warren 05 Apr '12

05 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Rename these files so they include a specific hardware version in their filenames. The contents is only touched minimally so that git's rename tracking operates correctly; renaming all symbols in the files results in a diff so large that the rename detection fails. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Makefile | 12 ++++++------ sound/soc/tegra/{tegra_das.c => tegra20_das.c} | 4 ++-- sound/soc/tegra/{tegra_das.h => tegra20_das.h} | 2 +- sound/soc/tegra/{tegra_i2s.c => tegra20_i2s.c} | 4 ++-- sound/soc/tegra/{tegra_i2s.h => tegra20_i2s.h} | 2 +- sound/soc/tegra/{tegra_spdif.c => tegra20_spdif.c} | 4 ++-- sound/soc/tegra/{tegra_spdif.h => tegra20_spdif.h} | 2 +- 7 files changed, 15 insertions(+), 15 deletions(-) rename sound/soc/tegra/{tegra_das.c => tegra20_das.c} (98%) rename sound/soc/tegra/{tegra_das.h => tegra20_das.h} (98%) rename sound/soc/tegra/{tegra_i2s.c => tegra20_i2s.c} (99%) rename sound/soc/tegra/{tegra_i2s.h => tegra20_i2s.h} (99%) rename sound/soc/tegra/{tegra_spdif.c => tegra20_spdif.c} (99%) rename sound/soc/tegra/{tegra_spdif.h => tegra20_spdif.h} (99%) diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 714d360..4a33884 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,15 +1,15 @@ # Tegra platform Support snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o -snd-soc-tegra-das-objs := tegra_das.o -snd-soc-tegra-i2s-objs := tegra_i2s.o -snd-soc-tegra-spdif-objs := tegra_spdif.o +snd-soc-tegra20-das-objs := tegra20_das.o +snd-soc-tegra20-i2s-objs := tegra20_i2s.o +snd-soc-tegra20-spdif-objs := tegra20_spdif.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra-das.o -obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o -obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o +obj-$(CONFIG_SND_SOC_TEGRA_DAS) += snd-soc-tegra20-das.o +obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra20-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra20-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra20_das.c similarity index 98% rename from sound/soc/tegra/tegra_das.c rename to sound/soc/tegra/tegra20_das.c index 3b3c1ba..b5e3698 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -1,5 +1,5 @@ /* - * tegra_das.c - Tegra DAS driver + * tegra20_das.c - Tegra20 DAS driver * * Author: Stephen Warren <swarren(a)nvidia.com> * Copyright (C) 2010 - NVIDIA, Inc. @@ -29,7 +29,7 @@ #include <linux/io.h> #include <mach/iomap.h> #include <sound/soc.h> -#include "tegra_das.h" +#include "tegra20_das.h" #define DRV_NAME "tegra-das" diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra20_das.h similarity index 98% rename from sound/soc/tegra/tegra_das.h rename to sound/soc/tegra/tegra20_das.h index 1810cd1..896bf03 100644 --- a/sound/soc/tegra/tegra_das.h +++ b/sound/soc/tegra/tegra20_das.h @@ -1,5 +1,5 @@ /* - * tegra_das.h - Definitions for Tegra DAS driver + * tegra20_das.h - Definitions for Tegra20 DAS driver * * Author: Stephen Warren <swarren(a)nvidia.com> * Copyright (C) 2010 - NVIDIA, Inc. diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra20_i2s.c similarity index 99% rename from sound/soc/tegra/tegra_i2s.c rename to sound/soc/tegra/tegra20_i2s.c index 86f5ca4..fa71af2 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -1,5 +1,5 @@ /* - * tegra_i2s.c - Tegra I2S driver + * tegra20_i2s.c - Tegra20 I2S driver * * Author: Stephen Warren <swarren(a)nvidia.com> * Copyright (C) 2010,2012 - NVIDIA, Inc. @@ -43,7 +43,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include "tegra_i2s.h" +#include "tegra20_i2s.h" #define DRV_NAME "tegra-i2s" diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra20_i2s.h similarity index 99% rename from sound/soc/tegra/tegra_i2s.h rename to sound/soc/tegra/tegra20_i2s.h index c08e019..56f1e0f 100644 --- a/sound/soc/tegra/tegra_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -1,5 +1,5 @@ /* - * tegra_i2s.h - Definitions for Tegra I2S driver + * tegra20_i2s.h - Definitions for Tegra20 I2S driver * * Author: Stephen Warren <swarren(a)nvidia.com> * Copyright (C) 2010 - NVIDIA, Inc. diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra20_spdif.c similarity index 99% rename from sound/soc/tegra/tegra_spdif.c rename to sound/soc/tegra/tegra20_spdif.c index e16c4ff..6b1bbe4 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -1,5 +1,5 @@ /* - * tegra_spdif.c - Tegra SPDIF driver + * tegra20_spdif.c - Tegra20 SPDIF driver * * Author: Stephen Warren <swarren(a)nvidia.com> * Copyright (C) 2011-2012 - NVIDIA, Inc. @@ -34,7 +34,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include "tegra_spdif.h" +#include "tegra20_spdif.h" #define DRV_NAME "tegra-spdif" diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra20_spdif.h similarity index 99% rename from sound/soc/tegra/tegra_spdif.h rename to sound/soc/tegra/tegra20_spdif.h index f5fc712..001f9dc 100644 --- a/sound/soc/tegra/tegra_spdif.h +++ b/sound/soc/tegra/tegra20_spdif.h @@ -1,5 +1,5 @@ /* - * tegra_spdif.h - Definitions for Tegra SPDIF driver + * tegra20_spdif.h - Definitions for Tegra20 SPDIF driver * * Author: Stephen Warren <swarren(a)nvidia.com> * Copyright (C) 2011 - NVIDIA, Inc. -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC:lm49453: fix build warnings
by MR.Swami.Reddy＠ti.com 05 Apr '12

05 Apr '12

From: M R Swami Reddy <mr.swami.reddy(a)ti.com> sound/soc/codecs/lm49453.c: In function 'lm49453_set_dai_fmt': sound/soc/codecs/lm49453.c:1189:4: warning: overflow in implicit constant conversion [-Woverflow] sound/soc/codecs/lm49453.c:1193:4: warning: overflow in implicit constant conversion [-Woverflow] sound/soc/codecs/lm49453.c:1197:4: warning: overflow in implicit constant conversion [-Woverflow] Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 13 +- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 744063d..74409ec 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1178,27 +1178,24 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - int aif_val = 0; + u16 aif_val; int mode = 0; int clk_phase = 0; int clk_shift = 0; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | - ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + aif_val = 0; break; case SND_SOC_DAIFMT_CBS_CFM: - aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | - LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; break; case SND_SOC_DAIFMT_CBM_CFS: - aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | - ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; break; case SND_SOC_DAIFMT_CBM_CFM: aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | - LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; break; default: return -EINVAL;

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[alsa-devel] [PATCH] ASoC: tegra: make Tegra20 drivers depend on Tegra20 support
by Stephen Warren 05 Apr '12

05 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> Without this, the Tegra20 drivers can be built into a kernel that's built only for Tegra30. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This depends on "ASoC: tegra: make Tegra20 drivers depend on Tegra20 support" for context. sound/soc/tegra/Kconfig | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 16632b1..d51945e 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_TEGRA config SND_SOC_TEGRA_DAS tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC help Say Y or M if you want to add support for the Tegra DAS module. You will also need to select the individual machine drivers to @@ -14,7 +14,7 @@ config SND_SOC_TEGRA_DAS config SND_SOC_TEGRA_I2S tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC select SND_SOC_TEGRA_DAS help Say Y or M if you want to add support for codecs attached to the @@ -23,7 +23,7 @@ config SND_SOC_TEGRA_I2S config SND_SOC_TEGRA_SPDIF tristate - depends on SND_SOC_TEGRA + depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC default m help Say Y or M if you want to add support for the SPDIF interface. @@ -41,7 +41,7 @@ config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C depends on MACH_HAS_SND_SOC_TEGRA_WM8903 - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -51,7 +51,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the @@ -60,7 +60,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA_I2S + select SND_SOC_TEGRA_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: tegra: drop Kconfig description for SND_SOC_TEGRA_DAS
by Stephen Warren 05 Apr '12

05 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The DAS, I2S, and SPDIF Kconfig options are intended to be selected by the Kconfig options for ASoC machine drivers. As such, they don't need to be user-visible themselves. Drop the description from the DAS variable to achieve this. I2S and SPDIF already don't have a description. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 23ab00d..16632b1 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -5,7 +5,7 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA_DAS - tristate "Tegra Digital Audio Switch driver" + tristate depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra DAS module. -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: set idle_bias_off=1 for all platform DAPM contexts
by Stephen Warren 05 Apr '12

05 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> The ASoC core currently defaults to using STANDBY rather than OFF for idle ASoC platform devices, which causes a permanent pm_runtime_get() on them. This keeps the device active unnecessarily. This can be especially problematic when the ASoC platform device and DAI device are the same device. The distinction between OFF and STANDBY is likely not relevant for ASoC platform drivers, since they aren't analog devices. So, solve this issue by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this turns out to be a problem, this value could be sourced from the snd_soc_platform_driver, similarly to soc_probe_codec(). Note: Prior to this change, this caused a large (10) runtime_active count for the Tegra I2S controller even when not in use, and a leak in that value as streams were started and stopped. This change probably hides a bug. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8a17048..63e6114 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1072,6 +1072,8 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); } + platform->dapm.idle_bias_off = 1; + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { -- 1.7.0.4

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[alsa-devel] Alsa soc UDA1380 - no duplex
by jarek 05 Apr '12

05 Apr '12

Hello! I've received information from lpclinux.com that uda1380 on lpc313x doesn't support duplex. I've got some information that uda1380 on s3c24xx works with duplex, so I think, that it should work also on lpc313x. I'm ready to write some code, but I have no experience with ALSA, so I'll be happy if someone will give me some hints where to start. The other question is why there are two drivers to the same device. best regards Jarek

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[alsa-devel] [PATCH] Fix internal mic for Lenovo Ideapad U300s
by David Henningsson 05 Apr '12

05 Apr '12

The internal mic input is phase inverted on one channel. To avoid people in userspace summing the channels together and get zero result, use a separate mixer control for the inverted channel. BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 88 ++++++++++++++++++++++++++++++++++------ 1 files changed, 75 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e6eafb1..c18e3b1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int single_adc_amp:1; + unsigned int fixup_stereo_dmic:1; unsigned int adc_switching:1; @@ -4107,9 +4108,9 @@ static int cx_auto_init(struct hda_codec *codec) static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, const char *dir, int cidx, - hda_nid_t nid, int hda_dir, int amp_idx) + hda_nid_t nid, int hda_dir, int amp_idx, int chs) { - static char name[32]; + static char name[44]; static struct snd_kcontrol_new knew[] = { HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), @@ -4119,7 +4120,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; - knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx, + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx, hda_dir); knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; knew[i].index = cidx; @@ -4138,7 +4139,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, } #define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \ - cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0) + cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3) #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) @@ -4208,6 +4209,36 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) return 0; } +/* Returns zero if this is a normal stereo channel, and non-zero if it should + be split in two independent channels. + dest_label must be at least 44 characters. */ +static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label, + char *dest_label, int nid) +{ + struct conexant_spec *spec = codec->spec; + int i; + + if (!spec->fixup_stereo_dmic) + return 0; + + for (i = 0; i < AUTO_CFG_MAX_INS; i++) { + int def_conf; + if (spec->autocfg.inputs[i].pin != nid) + continue; + + if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC) + return 0; + def_conf = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) + return 0; + + /* Finally found the inverted internal mic! */ + snprintf(dest_label, 44, "Inverted %s", label); + return 1; + } + return 0; +} + static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, const char *label, const char *pfx, int cidx) @@ -4216,14 +4247,25 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int i; for (i = 0; i < spec->num_adc_nids; i++) { + char rightch_label[44]; hda_nid_t adc_nid = spec->adc_nids[i]; int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; if (spec->single_adc_amp) idx = 0; + + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + /* Make two independent kcontrols for left and right */ + int err = cx_auto_add_volume_idx(codec, label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, pfx, + cidx, adc_nid, HDA_INPUT, idx, 2); + } return cx_auto_add_volume_idx(codec, label, pfx, - cidx, adc_nid, HDA_INPUT, idx); + cidx, adc_nid, HDA_INPUT, idx, 3); } return 0; } @@ -4236,9 +4278,19 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, int i, con; nid = spec->imux_info[idx].pin; - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { + char rightch_label[44]; + if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) { + int err = cx_auto_add_volume_idx(codec, label, " Boost", + cidx, nid, HDA_INPUT, 0, 1); + if (err < 0) + return err; + return cx_auto_add_volume_idx(codec, rightch_label, " Boost", + cidx, nid, HDA_INPUT, 0, 2); + } return cx_auto_add_volume(codec, label, " Boost", cidx, nid, HDA_INPUT); + } con = __select_input_connection(codec, spec->imux_info[idx].adc, nid, &mux, false, 0); if (con < 0) @@ -4405,22 +4457,30 @@ static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg) } -static void apply_pin_fixup(struct hda_codec *codec, +enum { + CXT_PINCFG_LENOVO_X200, + CXT_FIXUP_STEREO_DMIC, +}; + +static void apply_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, const struct cxt_pincfg **table) { + struct conexant_spec *spec = codec->spec; + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); - if (quirk) { + if (quirk && table[quirk->value]) { snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n", quirk->name); apply_pincfg(codec, table[quirk->value]); } + if (quirk->value == CXT_FIXUP_STEREO_DMIC) { + snd_printdd(KERN_INFO "hda_codec: applying internal mic workaround for %s\n", + quirk->name); + spec->fixup_stereo_dmic = 1; + } } -enum { - CXT_PINCFG_LENOVO_X200, -}; - static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4431,10 +4491,12 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_FIXUP_STEREO_DMIC] = NULL, }; static const struct snd_pci_quirk cxt_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), {} }; @@ -4477,7 +4539,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + apply_fixup(codec, cxt_fixups, cxt_pincfg_tbl); /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds -- 1.7.9.1

2 1

Re: [alsa-devel] [PATCH 5/6] sound/pci/hda: CX20549 doesn't need pin_amp_workaround.
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:35 +0200, Michael Karcher wrote: > > CX20549 (ctx5045) doesn't accept data on index 1 for output pins, > as shown in the following hda-var transaction: > > $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126 > nid = 0x10, verb = 0x300, param = 0xb126 > value = 0x0 > $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001 > nid = 0x10, verb = 0xb00, param = 0x8001 > value = 0x0 OK, it's good to know. Could you fix it in the corresponding part in patch_conexant_auto(), too? Takashi > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 8d7a977..3ab7b25 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -1087,7 +1087,6 @@ static int patch_cxt5045(struct hda_codec *codec) > if (!spec) > return -ENOMEM; > codec->spec = spec; > - codec->pin_amp_workaround = 1; > > spec->multiout.max_channels = 2; > spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); > -- > 1.7.9.1 > > >

1 0

Re: [alsa-devel] [PATCH 4/6] sound/pci/hda: Remove CD control from model=benq for CX20549
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:25 +0200, Michael Karcher wrote: > > The ID used for detection of the BenQ R55E actually identifies the > Quana TW3 ODM design, which is also used for the Gigabyte W551 laptop > series. Schematics on the internet clearly indicate that the "Port C" > (analog input connected to record source #4 and mixer input #4) is > unconnected. Well, such an action needs to be done carefully since the documentation often lies. But I also doubt that there is any analog CD line. If this is confirmed that actually it's not present on the real machine, fine to remove it indeed. Takashi > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index bd7e7ce..8d7a977 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -695,7 +695,6 @@ static const struct hda_input_mux cxt5045_capture_source = { > static const struct hda_input_mux cxt5045_capture_source_benq = { > .num_items = 5, > .items = { > - { "CD", 0x4 }, > { "Internal Mic", 0x1 }, > { "Mic", 0x2 }, > { "Line In", 0x3 }, > @@ -820,9 +819,6 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { > - HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), > - HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), > - > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > -- > 1.7.9.1 > > >

1 0

Re: [alsa-devel] [PATCH 3/6] sound/pci/hda: fix record volume controls of CX20459 ("Venice")
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:16 +0200, Michael Karcher wrote: > > The "input converter" widget of the CX20459 has only one input amplifier, > expose that one as "Capture Volume/Capture Switch". The actual record > source selection is already exposed through the separately installed > input mux. > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 7d92aac..bd7e7ce 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -798,10 +798,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, > } > > static const struct snd_kcontrol_new cxt5045_mixers[] = { > - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), > - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), > - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), > - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), > + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), > + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), > @@ -822,27 +820,18 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { > - HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT), > - HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT), > HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), > HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), > > - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), > - HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > - HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), > - HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), > - > {} > }; > > static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = { > - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), > - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), > - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), > - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), > + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), > + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), > HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT), > @@ -978,16 +967,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { > .put = conexant_mux_enum_put, > }, > /* Audio input controls */ > - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), > - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), > - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), > - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), > - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), > - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), > - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), > + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), > { } /* end */ > }; Looks good. Takashi

1 0

Re: [alsa-devel] [PATCH 2/6] sound/pci/hda: Rename capture sources of CX20549 to match common conventions
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:30:06 +0200, Michael Karcher wrote: > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 8c6523b..7d92aac 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -687,18 +687,18 @@ static const struct hda_channel_mode cxt5045_modes[1] = { > static const struct hda_input_mux cxt5045_capture_source = { > .num_items = 2, > .items = { > - { "IntMic", 0x1 }, > - { "ExtMic", 0x2 }, > + { "Internal Mic", 0x1 }, > + { "Mic", 0x2 }, > } > }; > > static const struct hda_input_mux cxt5045_capture_source_benq = { > .num_items = 5, > .items = { > - { "IntMic", 0x1 }, > - { "ExtMic", 0x2 }, > - { "LineIn", 0x3 }, > { "CD", 0x4 }, > + { "Internal Mic", 0x1 }, > + { "Mic", 0x2 }, > + { "Line In", 0x3 }, This could be simply "Line". Other than that, looks good. Takashi

1 0

Re: [alsa-devel] [PATCH 1/6] sound/pci/hda: Fix Amp-In count for widgets with implied selector
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:29:56 +0200, Michael Karcher wrote: > > A selector widget with AC_WCAP_IN_AMP has multiple input amplifiers (if > the selector widget would have only one common amp, it would be labeled > as "output amplifier" instead). This is just like mixer widgets. > > If a non-mixer/selector widget has an input amplifier, it is only one > input amplifier, even if it has a connection list with more than one > entry. The implied selector is placed before the single input amplifier. > > At least that is how I read the HDA specs, and what Conexant implements in > their CX20549 codec. It's rather cx20549-specific behavior. If you take a look at the BIOS auto-parser, spec->single_adc_amp is set as a workaround. Yes, the proc file output should be fixed as well, but you can't change it globally for all codecs. Takashi > diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c > index 254ab52..9b942f0 100644 > --- a/sound/pci/hda/hda_proc.c > +++ b/sound/pci/hda/hda_proc.c > @@ -653,7 +653,7 @@ static void print_codec_info(struct snd_info_entry *entry, > snd_iprintf(buffer, " Amp-In vals: "); > print_amp_vals(buffer, codec, nid, HDA_INPUT, > wid_caps & AC_WCAP_STEREO, > - wid_type == AC_WID_PIN ? 1 : conn_len); > + (wid_type == AC_WID_AUD_SEL || wid_type == AC_WID_AUD_MIX) ? conn_len : 1); > } > if (wid_caps & AC_WCAP_OUT_AMP) { > snd_iprintf(buffer, " Amp-Out caps: "); > -- > 1.7.9.1 > > >

1 0

Re: [alsa-devel] [PATCH 0/6] sound/pci/hda: CX20549 cleanup/fixes
by Takashi Iwai 05 Apr '12

05 Apr '12

At Thu, 05 Apr 2012 15:29:37 +0200, Michael Karcher wrote: > > This series of patches fixes recording on a Gigabyte W551N laptop that > seems to be identical to the BenW R55E. The Audio Codec uses the same > subsystem ID, which is assigned to Quanta. The Gigabyte laptop is based > on the Quanta TW3 ODM. Thanks for the patch. I'll review and give comments back for each. > (for the whole series) > Signed-Off-By: Michael Karcher <kernel(a)mkarcher.dialup.fu-berlin.de> Please give your sign-off in each patch changelog, so that I can apply the patches directly via git-am. Also, the subject lines should be with ALSA: prefix. Take a look at other git commits to that file as an example. Takashi

1 1

[alsa-devel] [PATCH 1/2] ASoC: imx-audmux: Fix ssi port numbers in sysfs
by Fabio Estevam 05 Apr '12

05 Apr '12

Doing a 'cat /sys/kernel/debug/audmux/ssi7' causes the following oops to be printed by the kernel: Uhandled fault: external abort on non-linefetch (0x008) at 0xf53b003c Internal error: : 8 [#1] PREEMPT Modules linked in: CPU: 0 Not tainted (3.3.0-00033-gecc726e-dirty #307) PC is at audmux_read_file+0x68/0x2f4 LR is at clk_enable+0x3c/0x48 pc : [<c001b8c8>] lr : [<c00190a0>] psr: a0000013 sp : c3ad3f38 ip : c30a4000 fp : 00000003 r10: 00001000 r9 : be83fb00 r8 : c3ad3f80 r7 : c3ad3f80 r6 : 00000007 r5 : 00031010 r4 : c30a5000 r3 : f53b0000 r2 : 0000003c r1 : 380fa100 r0 : c068dda0 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 0005317f Table: 83034000 DAC: 00000015 Process cat (pid: 1042, stack limit = 0xc3ad2270) Stack: (0xc3ad3f38 to 0xc3ad4000) 3f20: c3139180 00000000 3f40: c3bc6500 00001000 be83fb00 c3ad3f80 00001000 c3ad2000 00000000 c0095f3c 3f60: 00000003 c3bc6508 c3bc6500 be83fb00 00000000 00000000 00001000 c0096010 3f80: 00000000 00000000 b6fe2050 00000000 00001000 be83fb00 00000003 00000003 3fa0: c000eb88 c000e9e0 00001000 be83fb00 00000003 be83fb00 00001000 00000000 3fc0: 00001000 be83fb00 00000003 00000003 00000001 00000001 00000000 00000003 3fe0: 000bec8c be83fae0 0000f808 b6ea8d5c 60000010 00000003 7dff7ede 749bedf1 [<c001b8c8>] (audmux_read_file+0x68/0x2f4) from [<c0095f3c>] (vfs_read+0xb0/0x144) [<c0095f3c>] (vfs_read+0xb0/0x144) from [<c0096010>] (sys_read+0x40/0x70) [<c0096010>] (sys_read+0x40/0x70) from [<c000e9e0>] (ret_fast_syscall+0x0/0x2c) Code: e1a02186 e2822004 e3500000 e7935186 (e7937002) ---[ end trace 4d046e31309023de ]--- Fix the ssi port numbers in sysfs to fix this problem. Reported-by: Joan Carles <joancarles(a)fqingenieria.es> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 1765a19..d467a1b 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -152,7 +152,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 1; i < 8; i++) { + for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) -- 1.7.1

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[alsa-devel] [PATCH] ASoC: wm8994: Implement FLL bypass support
by Mark Brown 05 Apr '12

05 Apr '12

Later WM8994 class devices can bypass the FLL from BCLK. Do this automatically when the FLL input and output frequencies match up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 19 +++++++++++++++++++ sound/soc/codecs/wm8994.h | 1 + 2 files changed, 20 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e94a84d..1dee30d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2112,6 +2112,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && + freq_in == freq_out) { + dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP, WM8958_FLL1_BYP); + goto out; + } + reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) | (fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset, @@ -2126,6 +2134,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, + WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | @@ -2188,6 +2197,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } +out: wm8994->fll[id].in = freq_in; wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; @@ -3714,6 +3724,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + + switch (wm8994->revision) { + case 0: + break; + default: + wm8994->fll_byp = true; + break; + } break; case WM1811: @@ -3721,6 +3739,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.no_cache_class_w = true; + wm8994->fll_byp = true; switch (wm8994->revision) { case 0: diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c724112..91650bb 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; + bool fll_byp; int vmid_refcount; int active_refcount; -- 1.7.9.1

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: imx-audmux: Check for NULL pointer
by Mark Brown 05 Apr '12

05 Apr '12

On Thu, Apr 05, 2012 at 02:49:16PM +0200, Marc Kleine-Budde wrote: > On 04/05/2012 02:45 PM, Fabio Estevam wrote: > > if (audmux_clk) > > clk_prepare_enable(audmux_clk); > > + if (!audmux_base) > > + return -ENOSYS; > What about "clk_prepare_enable"? Do you need to undo it in case of an error? Or alternatively just reverse the order of the two tests so you don't need to worry about that.

1 0

[alsa-devel] [PATCH] ASoC: ak4642: fixup: mute needs +1 step
by Kuninori Morimoto 05 Apr '12

05 Apr '12

ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute. But current settings didn't care +1 step for mute. This patch adds it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 278c0a0..45a087b 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -140,7 +140,7 @@ * min : 0xFE : -115.0 dB * mute: 0xFF */ -static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { -- 1.7.5.4

2 1

[alsa-devel] UHCI dropped audio samples / DMA starvation fix
by Monty Montgomery 04 Apr '12

04 Apr '12

Hello Clemens and Takashi (and list), I've been living with USB audio dropping partial frames on UHCI controllers for quite a few years. I've finally found the cause (DMA starvation) and have a lightweight fix. problem summary: Playback (but not capture) on UHCI controllers randomly loses samples; they are skipped completely during playback. This happens anywhere from a few times a second to once an hour or so. These are not ALSA underruns; the UHCI host controller itself is reporting hard DMA starvation errors. ALSA ignores these error and continues as if nothing happened, and the UHCI linux driver does not log this error anywhere. UHCI overlaps sending on the USB bus with DMAing the data from memory. It does not have all the data it needs to transmit a complete packet before it begins doing so. If the data it needs is in a dirty cache line on a CPU, the flush triggers after DMA is already started; the UHCI host controller has to wait for the flush before accessing it, and if the flush happens too late, UHCI aborts the packet. This is happening regularly on all the UHCI controllers I've tried thusfar. My guess is that this issue never attracted much attention because a) every one knows isoch transfers are 'unreliable' and b) other transfer types automatically retry so they eventually succeed without any obvious sign that something went wrong (and they're usually smaller than one cacheline and so unaffected anyway). Fix: The fix for usbaudio is dead easy. Call: clflush_cache_range(urb->transfer_buffer, urb->transfer_buffer_length); as the last step of prepare_playback_urb() and prepare_nodata_playback_urb(). This completely corrects the DMA starvation behavior on my machines here. Note that the clflush_cache_range() has nothing to do with maintaining coherency. The call is only to force the flush ahead of time to avoid the fatal latency spikes from on-demand flushes when DMA is already running. The drawback to this fix is that it's in one high level driver (ALSA USB audio) instead of inside UHCI where it really belongs. Unfortunately, the URB abstraction gives us no reliable way to do it in the HCD, because other drivers don't necessarily pass a valid CPU-side buffer address when using URB_NO_TRANSFER_DMA_MAP. Should I whip up some patches for usb audio for 2.6.x and 3.3.x and leave it at that, or should we discuss a bit more 'why isn't the kernel doing this for us?' Monty

2 2

[alsa-devel] [PATCH] ASoC: codecs: Remove rtd->codec usage from CODEC drivers
by Mark Brown 04 Apr '12

04 Apr '12

In order to support CODEC<->CODEC links remove the assumption that there is only a single CODEC on a DAI link by removing the use of the CODEC pointer in the rtd from the CODEC drivers. They are already being passed their DAI whenever they are passed an rtd and can get the CODEC from there. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ac97.c | 4 +--- sound/soc/codecs/ad1836.c | 4 +--- sound/soc/codecs/ad193x.c | 4 +--- sound/soc/codecs/adau1701.c | 3 +-- sound/soc/codecs/ak4104.c | 3 +-- sound/soc/codecs/ak4535.c | 3 +-- sound/soc/codecs/ak4641.c | 3 +-- sound/soc/codecs/alc5623.c | 3 +-- sound/soc/codecs/alc5632.c | 3 +-- sound/soc/codecs/cs4270.c | 3 +-- sound/soc/codecs/cs4271.c | 3 +-- sound/soc/codecs/cs42l51.c | 3 +-- sound/soc/codecs/cs42l73.c | 3 +-- sound/soc/codecs/da7210.c | 3 +-- sound/soc/codecs/jz4740.c | 3 +-- sound/soc/codecs/lm49453.c | 3 +-- sound/soc/codecs/rt5631.c | 3 +-- sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/ssm2602.c | 9 +++------ sound/soc/codecs/sta32x.c | 3 +-- sound/soc/codecs/tlv320aic23.c | 9 +++------ sound/soc/codecs/tlv320aic26.c | 3 +-- sound/soc/codecs/tlv320aic3x.c | 3 +-- sound/soc/codecs/tlv320dac33.c | 35 +++++++++++++++-------------------- sound/soc/codecs/twl4030.c | 18 ++++++------------ sound/soc/codecs/twl6040.c | 9 +++------ sound/soc/codecs/uda134x.c | 6 ++---- sound/soc/codecs/uda1380.c | 6 ++---- sound/soc/codecs/wl1273.c | 6 ++---- sound/soc/codecs/wm8400.c | 3 +-- sound/soc/codecs/wm8510.c | 3 +-- sound/soc/codecs/wm8523.c | 3 +-- sound/soc/codecs/wm8728.c | 3 +-- sound/soc/codecs/wm8737.c | 3 +-- sound/soc/codecs/wm8741.c | 3 +-- sound/soc/codecs/wm8750.c | 3 +-- sound/soc/codecs/wm8753.c | 6 ++---- sound/soc/codecs/wm8900.c | 3 +-- sound/soc/codecs/wm8903.c | 3 +-- sound/soc/codecs/wm8940.c | 3 +-- sound/soc/codecs/wm8960.c | 3 +-- sound/soc/codecs/wm8962.c | 3 +-- sound/soc/codecs/wm8971.c | 3 +-- sound/soc/codecs/wm8978.c | 3 +-- sound/soc/codecs/wm8988.c | 3 +-- sound/soc/codecs/wm8990.c | 3 +-- sound/soc/codecs/wm9705.c | 4 +--- sound/soc/codecs/wm9712.c | 8 ++------ 48 files changed, 78 insertions(+), 152 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 1bbad4c..a99a1b3 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -26,9 +26,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 12e3b41..c67b50d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -162,9 +162,7 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index a4a6bef..13e62be 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -245,9 +245,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { int word_len = 0, master_rate = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); /* bit size */ diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 78e9ce4..3d50fc8 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -258,8 +258,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, static int adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; snd_pcm_format_t format; unsigned int val; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index ceb96ec..31d4483 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -88,8 +88,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 838ae8b..618fdc3 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -262,8 +262,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7f42d4a..543a12f 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -296,8 +296,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); int rate = params_rate(params), fs = 256; u8 mode2; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d47b62d..3061d35 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -705,8 +705,7 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index e2111e0..93a5909 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -861,8 +861,7 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int coeff, rate; u16 iface; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index df8fe38..047917f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -307,8 +307,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index bf71412..9eb01d7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -318,8 +318,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int i, ret; unsigned int ratio, val; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 85d6069..091d019 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -356,8 +356,7 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret; unsigned int i; diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index d39b3e7..2cc50b3 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1089,8 +1089,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); int id = dai->id; int mclk_coeff; diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 8e45aa9..5dfdf6e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -716,8 +716,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u32 dai_cfg1; u32 fs, bypass; diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 4624e75..85d9cab 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -164,8 +164,7 @@ static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { uint32_t val; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; switch (params_rate(params)) { case 8000: diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 744063d..89bbe03 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1140,8 +1140,7 @@ static int lm49453_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); u16 clk_div = 0; diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 20c324c..95147e4 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1361,8 +1361,7 @@ static int get_coeff(int mclk, int rate, int timesofbclk) static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); int timesofbclk = 32, coeff; unsigned int iface = 0; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ce1d5ce..8af6a524 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -665,8 +665,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); int channels = params_channels(params); int i2s_ctl = 0; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index de2b2054..4c94fd2 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -254,8 +254,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); @@ -291,8 +290,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -328,8 +326,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); if (ssm2602->master_substream == substream) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 7db6fa5..8d717f4 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -609,8 +609,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int rate; int i, mcs = -1, ir = -1; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0cafd3..8c758b21 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -323,8 +323,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface_reg; int ret; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); @@ -369,8 +368,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* set active */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -381,8 +379,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 802064b..85944e9 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -126,8 +126,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8d20f6e..bde2d1a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -802,8 +802,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4587ddd..0dd41077 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -62,8 +62,10 @@ #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) -static void dac33_calculate_times(struct snd_pcm_substream *substream); -static int dac33_prepare_chip(struct snd_pcm_substream *substream); +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec); enum dac33_state { DAC33_IDLE = 0, @@ -427,8 +429,8 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (likely(dac33->substream)) { - dac33_calculate_times(dac33->substream); - dac33_prepare_chip(dac33->substream); + dac33_calculate_times(dac33->substream, w->codec); + dac33_prepare_chip(dac33->substream, w->codec); } break; case SND_SOC_DAPM_POST_PMD: @@ -799,8 +801,7 @@ static void dac33_oscwait(struct snd_soc_codec *codec) static int dac33_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Stream started, save the substream pointer */ @@ -812,8 +813,7 @@ static int dac33_startup(struct snd_pcm_substream *substream, static void dac33_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); dac33->substream = NULL; @@ -825,8 +825,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); /* Check parameters for validity */ @@ -868,10 +867,9 @@ static int dac33_hw_params(struct snd_pcm_substream *substream, * writes happens in different order, than dac33 might end up in unknown state. * Use the known, working sequence of register writes to initialize the dac33. */ -static int dac33_prepare_chip(struct snd_pcm_substream *substream) +static int dac33_prepare_chip(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; u8 aictrl_a, aictrl_b, fifoctrl_a; @@ -1067,10 +1065,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) return 0; } -static void dac33_calculate_times(struct snd_pcm_substream *substream) +static void dac33_calculate_times(struct snd_pcm_substream *substream, + struct snd_soc_codec *codec) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned int period_size = substream->runtime->period_size; unsigned int rate = substream->runtime->rate; @@ -1128,8 +1125,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -1161,8 +1157,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; unsigned int time_delta, uthr; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 170cf9a..391fcfc 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1685,8 +1685,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream) { @@ -1715,8 +1714,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream, static void twl4030_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (twl4030->master_substream == substream) @@ -1740,8 +1738,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode, old_mode, format, old_format; @@ -1974,8 +1971,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 mode; @@ -2007,8 +2003,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; /* Enable voice digital filters */ twl4030_voice_enable(codec, substream->stream, 0); @@ -2017,8 +2012,7 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); u8 old_mode, mode; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b8..ccbc88a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1340,8 +1340,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -1355,8 +1354,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int rate; @@ -1392,8 +1390,7 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, static int twl6040_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 797b0dd..6c3d43b 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -159,8 +159,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) static int uda134x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); struct snd_pcm_runtime *master_runtime; @@ -191,8 +190,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, static void uda134x_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); if (uda134x->master_substream == substream) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f1b23d..2502214 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -502,8 +502,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec); int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); @@ -528,8 +527,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 3d868dc..7b24d6d 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -293,8 +293,7 @@ static const struct snd_kcontrol_new wl1273_controls[] = { static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); switch (wl1273->mode) { @@ -329,8 +328,7 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(dai->codec); struct wl1273_core *core = wl1273->core; unsigned int rate, width, r; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 898979d..fcc0cac 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1145,8 +1145,7 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); audio1 &= ~WM8400_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9166126..56a0495 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -392,8 +392,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3..1c3ffb2 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -145,8 +145,7 @@ static int wm8523_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); int i; u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index fc3d59e..1467f97 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -88,8 +88,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4fe9d19..d052012 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -329,8 +329,7 @@ static int wm8737_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int i; u16 clocking = 0; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 3941f50..6e849cb 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -203,8 +203,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; int i; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e4c50ce..89151ca 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -547,8 +547,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e27e7b6..a26482c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -931,8 +931,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01f3; u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x017f; @@ -1161,8 +1160,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 srate = snd_soc_read(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01f3; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f18c554..077c962 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -610,8 +610,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg; reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 302c76a..8365b43 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1453,8 +1453,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int fs = params_rate(params); int bclk; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index d2883af..481a3d9 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -371,8 +371,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; u16 companding = snd_soc_read(codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 840d720..8bc659d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -505,8 +505,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; int i; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7dba006..28a0fe2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2534,8 +2534,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; int aif0 = 0; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 28fe59e..eef783f 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -478,8 +478,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 72d5fdc..a5be3ad 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -723,8 +723,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Word length mask = 0x60 */ u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 6cdf6a2b..1d4c5cf 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -668,8 +668,7 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9d24235..db63c97 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1112,8 +1112,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cacc6a8..7c09593 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,9 +236,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b342ae5..2603863 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -467,9 +467,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec =rtd->codec; + struct snd_soc_codec *codec = dai->codec; int reg; u16 vra; @@ -487,9 +485,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic23: Remove driver-specific version number
by Mark Brown 04 Apr '12

04 Apr '12

It's never been updated since the driver was merged. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic23.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f9..b0cafd3 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -34,8 +34,6 @@ #include "tlv320aic23.h" -#define AIC23_VERSION "0.1" - /* * AIC23 register cache */ @@ -548,8 +546,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- 1.7.9.1

1 0

Re: [alsa-devel] trying to get an eleven rack to work
by David Alexander 04 Apr '12

04 Apr '12

Grant, I had an entry using the USB_DEVICE which is very similar to the one you suggested. I changed it over to the USB_DEVICE_VENDOR_SPEC you had listed. I am getting: [ 60.952067] usb 2-4: new high speed USB device number 2 using ehci_hcd [ 61.468413] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 [ 61.468433] snd-usb-audio: probe of 2-4:1.1 failed with error -5 [ 61.469257] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 [ 61.469266] snd-usb-audio: probe of 2-4:1.3 failed with error -5 [ 61.469280] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 [ 61.469285] snd-usb-audio: probe of 2-4:1.4 failed with error -5 [ 61.469305] usbcore: registered new interface driver snd-usb-audio and I have entries in /proc/asound as before: ls -l total 0 -r--r--r-- 1 root root 0 2012-04-02 13:47 id -r--r--r-- 1 root root 0 2012-04-02 13:47 midi0 -r--r--r-- 1 root root 0 2012-04-02 13:47 usbbus -r--r--r-- 1 root root 0 2012-04-02 13:47 usbid Those entries had disappeared after making the changed to mixer.c. It didn't seem to be the magic bullet though, as I need some sort of pcm entries to be created? What is the difference between the usb_device and usb_device_vendor_spec? Any other ideas? thanks, David On Fri, Mar 30, 2012 at 5:02 PM, Grant Diffey <gdiffey(a)gmail.com> wrote: > > On Sat, Mar 31, 2012 at 4:27 AM, David Alexander <lidxv0n(a)gmail.com> wrote: >> >> Grant, >> >> You had mentioned something about the end point for the mixer. I have >> found this information from the trace I had performed: >> >> From a dispatch urb_function_get_descriptor_from_device message under >> the fid_USBPORT_Endpoint_Descriptor: >> >> fid_bLength 7 >> fid_bDescriptortype 5 >> fid_bEndpointAddress 0 >> fid_bmAttributes 0 >> fid_wMaxPacketSize 64 >> fid_bInterval 0 >> >> Is this what you were asking about, and does it help us? >> >> thanks, >> David >> >> >> > > What does the quirk look like that you have now? because what I wrote was > wrong. > > it seems I can't count to 5 :) > > so based on what Clemens said and me re-reading the interface descriptor > > Interface 0 is dfu upload > Interface 1 is a mixer > > Interface 2 is the midi > Interface 3 is audio output > Interface 4 is audio Input and provides clock data for output > > 0 and 2 were previously recognised so do not need to be quirked. > > so I think this is what the quirk should look like: > > > USB_DEVICE_VENDOR_SPEC(0x0dba, 0xb011), > > .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { > .vendor_name = "Digidesign", > .product_name = "Eleven Rack", > .ifnum = QUIRK_ANY_INTERFACE, > .type = QUIRK_COMPOSITE, > .data = & (const struct snd_usb_audio_quirk[]) { > > { > .ifnum = 1, > .type = QUIRK_AUDIO_STANDARD_MIXER > }, > { > .ifnum = 3, > .type = QUIRK_AUDIO_STANDARD_INTERFACE > > }, > { > .ifnum = 4, > .type = QUIRK_AUDIO_STANDARD_INTERFACE > }, > { > .ifnum = -1 > } > } > } > } > > is that similar to what you have? > > Grant. >

3 5

[alsa-devel] [PATCH 1/2] ASoC: fsl: remove redundant Kconfig option SND_SOC_FSL_SSI
by Shawn Guo 04 Apr '12

04 Apr '12

While commit 606d620 (ASoC: imx: merge sound/soc/imx into sound/soc/fsl) adds SND_SOC_FSL_SSI outside "menuconfig SND_POWERPC_SOC" to make it visible for both PowerPC and ARM/IMX, it forgot removing the one inside "menuconfig SND_POWERPC_SOC". Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/Kconfig | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa14fc9..d23d612 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -16,10 +16,6 @@ if SND_POWERPC_SOC config SND_MPC52xx_DMA tristate -config SND_SOC_FSL_SSI - tristate - depends on FSL_SOC - config SND_SOC_POWERPC_DMA tristate depends on FSL_SOC -- 1.7.5.4

2 2

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix warning due to the lack of REGULATOR_CHANGE_VOLTAGE
by Fabio Estevam 04 Apr '12

04 Apr '12

Fix the following warning during kernel boot: 0-000a: 850 <--> 1600 mV at 1200 mV normal 0-000a: Voltage range but no REGULATOR_CHANGE_VOLTAGE Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: Keep the voltage constant sound/soc/codecs/sgtl5000.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8e92fb8..431c9c1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -84,8 +84,8 @@ static struct regulator_consumer_supply ldo_consumer[] = { static struct regulator_init_data ldo_init_data = { .constraints = { - .min_uV = 850000, - .max_uV = 1600000, + .min_uV = 1200000, + .max_uV = 1200000, .valid_modes_mask = REGULATOR_MODE_NORMAL, .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Convert to module_i2c_driver()
by Mark Brown 03 Apr '12

03 Apr '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 12 +----------- 1 files changed, 1 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 159e3a8..6a7347f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1451,17 +1451,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { .id_table = sgtl5000_id, }; -static int __init sgtl5000_modinit(void) -{ - return i2c_add_driver(&sgtl5000_i2c_driver); -} -module_init(sgtl5000_modinit); - -static void __exit sgtl5000_exit(void) -{ - i2c_del_driver(&sgtl5000_i2c_driver); -} -module_exit(sgtl5000_exit); +module_i2c_driver(sgtl5000_i2c_driver); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel(a)gmail.com>"); -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix warning due to the lack of REGULATOR_CHANGE_VOLTAGE
by Fabio Estevam 03 Apr '12

03 Apr '12

Fix the following warning during kernel boot: 0-000a: 850 <--> 1600 mV at 1200 mV normal 0-000a: Voltage range but no REGULATOR_CHANGE_VOLTAGE Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8e92fb8..bad5ba5 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -87,7 +87,8 @@ static struct regulator_init_data ldo_init_data = { .min_uV = 850000, .max_uV = 1600000, .valid_modes_mask = REGULATOR_MODE_NORMAL, - .valid_ops_mask = REGULATOR_CHANGE_STATUS, + .valid_ops_mask = REGULATOR_CHANGE_VOLTAGE | + REGULATOR_CHANGE_STATUS, }, .num_consumer_supplies = 1, .consumer_supplies = &ldo_consumer[0], -- 1.7.1

2 3

[alsa-devel] [PATCH v2] ASoC: core: Initialize err for snd_soc_put_volsw_sx
by Brian Austin 03 Apr '12

03 Apr '12

sound/soc/soc-core.c: In function ‘snd_soc_put_volsw_sx’: sound/soc/soc-core.c:2600: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7b1a4fd..a6922bc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2586,7 +2586,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int mask = (1 << (fls(min + max) - 1)) - 1; - int err; + int err = 0; unsigned short val, val_mask, val2 = 0; val_mask = mask << shift; -- 1.7.9.1

2 1

[alsa-devel] [PATCH] ASoC: core: Initialize err for snd_soc_put_volsw_sx
by Brian Austin 03 Apr '12

03 Apr '12

sound/soc/soc-core.c: In function ‘snd_soc_put_volsw_sx’: sound/soc/soc-core.c:2600: warning: ‘err’ may be used uninitialized in this function Acked-by:Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7b1a4fd..a6922bc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2586,7 +2586,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int mask = (1 << (fls(min + max) - 1)) - 1; - int err; + int err = 0; unsigned short val, val_mask, val2 = 0; val_mask = mask << shift; -- 1.7.9.1

2 3

[alsa-devel] [PATCH] ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV
by Brian Austin 03 Apr '12

03 Apr '12

Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/soc.h | 48 +++++++----- sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l73.c | 26 +++--- sound/soc/soc-core.c | 174 ++++++++++++++++++++----------------------- 4 files changed, 125 insertions(+), 129 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf787..1e17e04 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -55,6 +55,18 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array),\ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx,\ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #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, \ @@ -85,6 +97,18 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_sx, \ + .put = snd_soc_put_volsw_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xrreg, \ + .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -171,20 +195,6 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } -#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\ - xmin, xmax, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_2r_sx, \ - .get = snd_soc_get_volsw_2r_sx, \ - .put = snd_soc_put_volsw_2r_sx, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg_left, \ - .rreg = xreg_right, .shift = xshift, \ - .min = xmin, .max = xmax} } - #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ @@ -413,6 +423,10 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #define snd_soc_get_volsw_2r snd_soc_get_volsw #define snd_soc_put_volsw_2r snd_soc_put_volsw +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, @@ -421,12 +435,6 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index a8bf588..85d6069 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -141,15 +141,15 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, - 8, 0xffffff19, 0x18, aout_tlv), + 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 7, 0xffffff99, 0x18, adc_pcm_tlv), + 6, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3..d39b3e7 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -402,37 +402,37 @@ static const struct snd_kcontrol_new ear_amp_ctl = static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", - CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, - 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 0, + 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, - CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv), SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, - CS42L73_MICBPREPGABVOL, 5, 0xffffff35, - 0x34, micpga_tlv), + CS42L73_MICBPREPGABVOL, 5, 0x34, + 0x24, micpga_tlv), SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, CS42L73_MICBPREPGABVOL, 6, 1, 1), SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, - CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + CS42L73_IPBDVOL, 0, 0xA0, 0x6C, ipd_tlv), SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", - CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, - 0xE4, hl_tlv), + CS42L73_HLADVOL, CS42L73_HLBDVOL, + 0, 0x34, 0xE4, hl_tlv), SOC_SINGLE_TLV("ADC A Boost Volume", CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), SOC_SINGLE_TLV("ADC B Boost Volume", - CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), - SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", - CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", + CS42L73_SPKDVOL, 0, 0x34, 0xE4, hl_tlv), - SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", - CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", + CS42L73_ESLDVOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, 1, 1), diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4deebc..548fb05 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2531,6 +2531,87 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw); /** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + + ucontrol->value.integer.value[0] = + ((snd_soc_read(codec, reg) >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) + ucontrol->value.integer.value[1] = + ((snd_soc_read(codec, reg2) >> rshift) - min) & mask; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err; + unsigned short val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + if (snd_soc_update_bits_locked(codec, reg2, val_mask, val2)) + return err; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** * snd_soc_info_volsw_s8 - signed mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -2651,99 +2732,6 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_limit_volume); -/** - * snd_soc_info_volsw_2r_sx - double with tlv and variable data size - * mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r_sx); - -/** - * snd_soc_get_volsw_2r_sx - double with tlv and variable data size - * mixer get callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<<mc->shift)-1; - int min = mc->min; - int val = snd_soc_read(codec, mc->reg) & mask; - int valr = snd_soc_read(codec, mc->rreg) & mask; - - ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask; - ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); - -/** - * snd_soc_put_volsw_2r_sx - double with tlv and variable data size - * mixer put callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int mask = (1<<mc->shift)-1; - int min = mc->min; - int ret; - unsigned int val, valr, oval, ovalr; - - val = ((ucontrol->value.integer.value[0]+min) & 0xff); - val &= mask; - valr = ((ucontrol->value.integer.value[1]+min) & 0xff); - valr &= mask; - - oval = snd_soc_read(codec, mc->reg) & mask; - ovalr = snd_soc_read(codec, mc->rreg) & mask; - - ret = 0; - if (oval != val) { - ret = snd_soc_write(codec, mc->reg, val); - if (ret < 0) - return ret; - } - if (ovalr != valr) { - ret = snd_soc_write(codec, mc->rreg, valr); - if (ret < 0) - return ret; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); - int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { -- 1.7.9.1

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[alsa-devel] [PATCH] MAINTAINERS: Add missing ASoC OMAP co-maintainer
by Jarkko Nikula 03 Apr '12

03 Apr '12

Peter Ujfalusi has been co-maintaining sound/soc/omap/ for years but was missing from this MAINTAINERS entry. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- MAINTAINERS | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index eecf344..85c599b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4807,6 +4807,7 @@ F: arch/arm/mach-omap2/clockdomain2xxx_3xxx.c F: arch/arm/mach-omap2/clockdomain44xx.c OMAP AUDIO SUPPORT +M: Peter Ujfalusi <peter.ujfalusi(a)ti.com> M: Jarkko Nikula <jarkko.nikula(a)bitmer.com> L: alsa-devel(a)alsa-project.org (subscribers-only) L: linux-omap(a)vger.kernel.org -- 1.7.9.1

2 1

[alsa-devel] [PATCH] ASoC: MAINTAINERS: Add maintainer for TI LM49453 Audio codec driver
by MR.Swami.Reddy＠ti.com 03 Apr '12

03 Apr '12

From: M R Swami Reddy <mr.swami.reddy(a)ti.com> I will be supporting the TI LM49453 audio driver and also a few new TI codec drivers, that are currently in development and will be upstreamed shortly. Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- MAINTAINERS | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index eecf344..cc9e035 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6664,6 +6664,12 @@ +TI LM49xxx FAMILY ASoC CODEC DRIVERS +M: M R Swami Reddy <mr.swami.reddy(a)ti.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/lm49453* + TI TWL4030 SERIES SOC CODEC DRIVER M: Peter Ujfalusi <peter.ujfalusi(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers)

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[alsa-devel] [PATCH 00/17] ASoC: tegra: Add Tegra30 support
by Stephen Warren 03 Apr '12

03 Apr '12

From: Stephen Warren <swarren(a)nvidia.com> This series adds basic support for 16-bit stereo on Tegra30. This will be enabled for the Cardhu board in a separate series to arch/arm/mach-tegra. I don't believe this depends on anything that won't be in 3.4. The related arch/arm/mach-tegra series will have some dependencies. There are a couple patches in here which also touch arch/arm/mach-tegra. I believe it should be fine to apply these to the ASoC tree without issue (I don't anticpate conflicts) but as you (Mark) mentioned, putting them into a separate branch just in case might be a good idea. Stephen Warren (17): ASoC: tegra: remove open-coded clk reference counting ASoC: tegra: ensure clocks are enabled when touching registers ASoC: tegra: remove unnecessary includes ASoC: tegra: add description to a couple Kconfig options ASoC: tegra: fix Kconfig SND_SOC_TEGRA_ALC5632 indentation ASoC: tegra: fix some checkpatch warnings ASoC: tegra: introduce separate Kconfig variable for DAS driver ASoC: tegra: make Tegra20 drivers depend on Tegra20 support ASoC: tegra: sort Makefile into common and per-SoC files ASoC: tegra: rename Tegra20-specific driver files ASoC: tegra: complete Tegra->Tegra20 renaming ASoC: tegra: utils: add support for Tegra30 devices ASoC: tegra: set a sensible initial clock rate ASoC: tegra: add tegra30-ahub driver ASoC: tegra: add tegra30-i2s driver ASoC: tegra: add Kconfig and Makefile support for Tegra30 ASoC: tegra+wm8903 machine: support Tegra30 .../bindings/sound/tegra-audio-wm8903.txt | 8 +- .../devicetree/bindings/sound/tegra30-ahub.txt | 18 + .../devicetree/bindings/sound/tegra30-i2s.txt | 12 + arch/arm/mach-tegra/board-dt-tegra20.c | 6 +- arch/arm/mach-tegra/devices.c | 6 +- arch/arm/mach-tegra/tegra2_clocks.c | 4 +- sound/soc/tegra/Kconfig | 61 ++- sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra20_das.c | 261 +++++++++ sound/soc/tegra/tegra20_das.h | 135 +++++ sound/soc/tegra/tegra20_i2s.c | 462 +++++++++++++++ sound/soc/tegra/tegra20_i2s.h | 165 ++++++ sound/soc/tegra/tegra20_spdif.c | 369 ++++++++++++ sound/soc/tegra/tegra20_spdif.h | 472 ++++++++++++++++ sound/soc/tegra/tegra30_ahub.c | 593 ++++++++++++++++++++ sound/soc/tegra/tegra30_ahub.h | 489 ++++++++++++++++ sound/soc/tegra/tegra30_i2s.c | 538 ++++++++++++++++++ sound/soc/tegra/tegra30_i2s.h | 242 ++++++++ sound/soc/tegra/tegra_alc5632.c | 6 +- sound/soc/tegra/tegra_asoc_utils.c | 35 +- sound/soc/tegra/tegra_asoc_utils.h | 10 +- sound/soc/tegra/tegra_das.c | 261 --------- sound/soc/tegra/tegra_das.h | 135 ----- sound/soc/tegra/tegra_i2s.c | 464 --------------- sound/soc/tegra/tegra_i2s.h | 166 ------ sound/soc/tegra/tegra_spdif.c | 369 ------------ sound/soc/tegra/tegra_spdif.h | 473 ---------------- sound/soc/tegra/tegra_wm8903.c | 39 +- sound/soc/tegra/trimslice.c | 10 +- 29 files changed, 3896 insertions(+), 1931 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra30-ahub.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra30-i2s.txt create mode 100644 sound/soc/tegra/tegra20_das.c create mode 100644 sound/soc/tegra/tegra20_das.h create mode 100644 sound/soc/tegra/tegra20_i2s.c create mode 100644 sound/soc/tegra/tegra20_i2s.h create mode 100644 sound/soc/tegra/tegra20_spdif.c create mode 100644 sound/soc/tegra/tegra20_spdif.h create mode 100644 sound/soc/tegra/tegra30_ahub.c create mode 100644 sound/soc/tegra/tegra30_ahub.h create mode 100644 sound/soc/tegra/tegra30_i2s.c create mode 100644 sound/soc/tegra/tegra30_i2s.h delete mode 100644 sound/soc/tegra/tegra_das.c delete mode 100644 sound/soc/tegra/tegra_das.h delete mode 100644 sound/soc/tegra/tegra_i2s.c delete mode 100644 sound/soc/tegra/tegra_i2s.h delete mode 100644 sound/soc/tegra/tegra_spdif.c delete mode 100644 sound/soc/tegra/tegra_spdif.h

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Re: [alsa-devel] [PATCH] ARM: pxa: pxa2xx-i2s: add io.h for IOMEM macro
by Mark Brown 02 Apr '12

02 Apr '12

On Mon, Apr 02, 2012 at 10:24:08AM +0200, Martin Jansa wrote: > * fixes *Please* make some effort to use subject lines appropriate to your patches. I just about deleted this unread because it doesn't mention anything to do with audio.

1 0

[alsa-devel] [PATCH] ASoC: max98095: Fix build failure
by Mark Brown 02 Apr '12

02 Apr '12

sound/soc/codecs/max98095.c: In function 'max98095_jack_detect_enable': sound/soc/codecs/max98095.c:2229:14: error: 'struct max98095_priv' has no member named 'jack_detect_delay' sound/soc/codecs/max98095.c:2230:18: error: 'struct max98095_priv' has no member named 'jack_detect_delay' Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/max98095.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0752840..35179e2 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2226,8 +2226,8 @@ int max98095_jack_detect_enable(struct snd_soc_codec *codec) if (max98095->pdata->jack_detect_pin5en) detect_enable |= M98095_PIN5EN; - if (max98095->jack_detect_delay) - slew = max98095->jack_detect_delay; + if (max98095->pdata->jack_detect_delay) + slew = max98095->pdata->jack_detect_delay; ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); if (ret < 0) { -- 1.7.9.1

2 1

Re: [alsa-devel] Rawmidi vs. /dev/snd/*
by Paul D. DeRocco 02 Apr '12

02 Apr '12

> From: Gabriel M. Beddingfield [mailto:gabrbedd@gmail.com] > > It would be a good idea for you to check out the alsa-lib sources and > snoop around in the src/rawmidi/ folder. > > In reality, only one file descriptor is returned by > snd_rawmidi_poll_descriptors(). However, the API returns an array so > that it can expand in the future without having to recompile programs > linked to the library. > > Yes, the ALSA documentation sucks. But I assure you that if > you try to > use the raw kernel interface... the documentation is worse. :-) Since ALSA is big an complicated, and since my embedded system will probably be based on SALSA, I downloaded that source, which is comparatively simple (hence the S, I guess). It makes it clear what's going on: the RawMidi handle already contains a pollfd for its internal use, and snd_rawmidi_poll_descriptors makes that available to the app. Thanks for your time. -- Ciao, Paul D. DeRocco Paul mailto:pderocco@ix.netcom.com

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Re: [alsa-devel] loading mixer control values from a different file when opening an device
by Clemens Ladisch 02 Apr '12

02 Apr '12

Ziv Haziz wrote: > I have tried the method described in the link (using <> for loading > other files) but I got an error - maybe I did not construct the small > configuration file correctly. Maybe, maybe not. My crystal ball broke yesterday, so I do not know what exactly you tried. > Is the <> method works like the c preprocessor #include directive? Yes. I put this into my ~/.asoundrc: pcm.test { type plug slave { pcm "hw:0" channels 2 } </home/cladisch/my-ttable.conf> } and this into ~/my-ttable.conf: ttable [ [ 0 1 ] [ 1 0 ] ] and it works. Regards, Clemens

1 0

[alsa-devel] lsusb -v for AKAI's new EIE pro interface
by Grant Diffey 02 Apr '12

02 Apr '12

So device is 4 in 4 out with inserts and midi Composite with 2 interfaces. I can't tell if they're independent midi and audio or more essoteric anyone want to take a punt? 1. Bus 001 Device 012: ID 09e8:0010 AKAI Professional M.I. Corp. 2. Device Descriptor: 3. bLength 18 4. bDescriptorType 1 5. bcdUSB 2.00 6. bDeviceClass 255 Vendor Specific Class 7. bDeviceSubClass 255 Vendor Specific Subclass 8. bDeviceProtocol 255 Vendor Specific Protocol 9. bMaxPacketSize0 64 10. idVendor 0x09e8 AKAI Professional M.I. Corp. 11. idProduct 0x0010 12. bcdDevice 2.00 13. iManufacturer 1 14. iProduct 2 15. iSerial 3 16. bNumConfigurations 1 17. Configuration Descriptor: 18. bLength 9 19. bDescriptorType 2 20. wTotalLength 80 21. bNumInterfaces 2 22. bConfigurationValue 1 23. iConfiguration 0 24. bmAttributes 0x40 25. (Missing must-be-set bit!) 26. Self Powered 27. MaxPower 0mA 28. Interface Descriptor: 29. bLength 9 30. bDescriptorType 4 31. bInterfaceNumber 0 32. bAlternateSetting 0 33. bNumEndpoints 0 34. bInterfaceClass 255 Vendor Specific Class 35. bInterfaceSubClass 0 36. bInterfaceProtocol 0 37. iInterface 0 38. Interface Descriptor: 39. bLength 9 40. bDescriptorType 4 41. bInterfaceNumber 0 42. bAlternateSetting 1 43. bNumEndpoints 3 44. bInterfaceClass 255 Vendor Specific Class 45. bInterfaceSubClass 0 46. bInterfaceProtocol 0 47. iInterface 0 48. Endpoint Descriptor: 49. bLength 7 50. bDescriptorType 5 51. bEndpointAddress 0x02 EP 2 OUT 52. bmAttributes 5 53. Transfer Type Isochronous 54. Synch Type Asynchronous 55. Usage Type Data 56. wMaxPacketSize 0x009c 1x 156 bytes 57. bInterval 1 58. Endpoint Descriptor: 59. bLength 7 60. bDescriptorType 5 61. bEndpointAddress 0x83 EP 3 IN 62. bmAttributes 2 63. Transfer Type Bulk 64. Synch Type None 65. Usage Type Data 66. wMaxPacketSize 0x0200 1x 512 bytes 67. bInterval 4 68. Endpoint Descriptor: 69. bLength 7 70. bDescriptorType 5 71. bEndpointAddress 0x04 EP 4 OUT 72. bmAttributes 2 73. Transfer Type Bulk 74. Synch Type None 75. Usage Type Data 76. wMaxPacketSize 0x0200 1x 512 bytes 77. bInterval 4 78. Interface Descriptor: 79. bLength 9 80. bDescriptorType 4 81. bInterfaceNumber 1 82. bAlternateSetting 0 83. bNumEndpoints 0 84. bInterfaceClass 255 Vendor Specific Class 85. bInterfaceSubClass 0 86. bInterfaceProtocol 0 87. iInterface 0 88. Interface Descriptor: 89. bLength 9 90. bDescriptorType 4 91. bInterfaceNumber 1 92. bAlternateSetting 1 93. bNumEndpoints 2 94. bInterfaceClass 255 Vendor Specific Class 95. bInterfaceSubClass 0 96. bInterfaceProtocol 0 97. iInterface 0 98. Endpoint Descriptor: 99. bLength 7 100. bDescriptorType 5 101. bEndpointAddress 0x81 EP 1 IN 102. bmAttributes 5 103. Transfer Type Isochronous 104. Synch Type Asynchronous 105. Usage Type Data 106. wMaxPacketSize 0x0040 1x 64 bytes 107. bInterval 4 108. Endpoint Descriptor: 109. bLength 7 110. bDescriptorType 5 111. bEndpointAddress 0x86 EP 6 IN 112. bmAttributes 2 113. Transfer Type Bulk 114. Synch Type None 115. Usage Type Data 116. wMaxPacketSize 0x0200 1x 512 bytes 117. bInterval 1 118. Device Status: 0x9000 119. (Bus Powered)

2 1

[alsa-devel] loading mixer control values from a different file when opening an device
by Ziv Haziz 02 Apr '12

02 Apr '12

Hi All, i'm using asound.conf file to define several devices where each one is configuring my sound device differently changing route, setting different volumes and etc) i would have like to load setting from a different file only when accessing the a specific device - something like this: pcm.MyDevice { type hooks slave.pcm { type hw device 0 } <i would like to place here something which will "func load" from another file> hooks.0 { type ctl_elems hook_args [ ] } so when someone will playback MyDevice it will call also hooks from the other file but i can't make it work, tried to use the alsa.conf as example - any example will be highly appreciated

2 1

[alsa-devel] Rawmidi vs. /dev/snd/*
by Paul D. DeRocco 02 Apr '12

02 Apr '12

In writing an embedded app that will be the only thing talking to MIDI, and in which I don't need any timing or routing capability, is there any reason to use ALSA Rawmidi, as opposed to merely reading and writing the device via the file system? Which is the lower-level, and therefore most efficient, interface? I'm leaning toward the latter, because I already know how to use poll() to wait for incoming data on multiple input devices, but I don't see how to do that via Rawmidi. -- Ciao, Paul D. DeRocco Paul mailto:pderocco@ix.netcom.com

2 1

[alsa-devel] [PATCH 1/1] ASoC: soc-dmaengine-pcm: fix compile error since dma api is changed
by Dong Aisheng 02 Apr '12

02 Apr '12

From: Dong Aisheng <dong.aisheng(a)linaro.org> The following compiling error is caused by the API device_prep_dma_cyclic is changed. Using dmaengine_prep_dma_cyclic to fix it. sound/soc/soc-dmaengine-pcm.c: In function 'dmaengine_pcm_prepare_and_submit': sound/soc/soc-dmaengine-pcm.c:149: error: too few arguments to function 'chan->device->device_prep_dma_cyclic' make[2]: *** [sound/soc/soc-dmaengine-pcm.o] Error 1 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Dong Aisheng <dong.aisheng(a)linaro.org> --- sound/soc/soc-dmaengine-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 4420b70..4756952 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -143,7 +143,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) direction = snd_pcm_substream_to_dma_direction(substream); prtd->pos = 0; - desc = chan->device->device_prep_dma_cyclic(chan, + desc = dmaengine_prep_dma_cyclic(chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), direction); -- 1.7.0.4

4 3

[alsa-devel] Invitation to connect on LinkedIn
by Paulo Cavalcanti via LinkedIn 01 Apr '12

01 Apr '12

LinkedIn ------------ Paulo Cavalcanti requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Paulo Accept invitation from Paulo Cavalcanti http://www.linkedin.com/e/jpxkk3-h0iijeoe-71/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View invitation from Paulo Cavalcanti http://www.linkedin.com/e/jpxkk3-h0iijeoe-71/uii-rtptP1H4ViCg9YiBUOvto1r4_5… ------------------------------------------ Why might connecting with Paulo Cavalcanti be a good idea? Paulo Cavalcanti's connections could be useful to you: After accepting Paulo Cavalcanti's invitation, check Paulo Cavalcanti's connections to see who else you may know and who you might want an introduction to. Building these connections can create opportunities in the future. -- (c) 2012, LinkedIn Corporation

1 0

[alsa-devel] [BAD PATCH] ice1712: Add support for Philips PSC724 Ultimate Edge
by Ondrej Zary 01 Apr '12

01 Apr '12

Hello, I'm trying do implement support for Philips PSC724 Ultimate Edge card, which is based on VT1722 + WM8776 + WM8766 chips. I've examined the connections and commented them in the code. The patch is heavily based on se.c. When trying to play a sound, the player runs but no sound comes from the output, no matter what I do with alsamixer. The interesting thing is that recording works - I can record something from e.g. line-in and play it on another machine. Recording source selection in alsamixer works too (if line-in is deselected, only silence is recorded). So WM8776 I2C control works. I'm missing something but don't know what. Any ideas? diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index f7ce33f..6d6ae06 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -5,7 +5,7 @@ snd-ice17xx-ak4xxx-objs := ak4xxx.o snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o +snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o psc724.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9236297..d19db09 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -54,6 +54,7 @@ #include "wtm.h" #include "se.h" #include "quartet.h" +#include "psc724.h" MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)"); @@ -2234,6 +2235,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_se_cards, snd_vt1724_qtet_cards, snd_vt1724_ooaoo_cards, + snd_vt1724_psc724_cards, NULL, }; @@ -2348,7 +2350,7 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, return -EIO; } ice->eeprom.version = snd_vt1724_read_i2c(ice, dev, 0x05); - if (ice->eeprom.version != 2) + if (ice->eeprom.version != 1 && ice->eeprom.version != 2) printk(KERN_WARNING "ice1724: Invalid EEPROM version %i\n", ice->eeprom.version); size = ice->eeprom.size - 6; diff --git a/sound/pci/ice1712/psc724.c b/sound/pci/ice1712/psc724.c new file mode 100644 index 0000000..ac9511b --- /dev/null +++ b/sound/pci/ice1712/psc724.c @@ -0,0 +1,664 @@ +/* + * ALSA driver for ICEnsemble VT1724 (Envy24HT) + * + * Lowlevel functions for Philips PSC724 Ultimate Edge + * + * Copyright (c) 2012 Ondrej Zary <linux(a)rainbow-software.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <asm/io.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/tlv.h> + +#include "ice1712.h" +#include "envy24ht.h" +#include "psc724.h" + +struct psc724_spec { + struct { + unsigned char ch1, ch2; + } vol[6]; +}; + +/****************************************************************************/ +/* PHILIPS PSC724 ULTIMATE EDGE */ +/****************************************************************************/ +/* + * system configuration ICE_EEP2_SYSCONF=0x42 + * XIN1 49.152MHz + * no MPU401 + * one stereo ADC, no S/PDIF receiver + * three stereo DACs (FRONT, REAR, CENTER+LFE) + * + * AC-Link configuration ICE_EEP2_ACLINK=0x80 + * use I2S, not AC97 + * + * I2S converters feature ICE_EEP2_I2S=0x30 + * I2S codec has no volume/mute control feature + * I2S codec does not support 96KHz or 192KHz(bug!) + * I2S codec 24bits + * + * S/PDIF configuration ICE_EEP2_SPDIF=0xc1 + * Enable integrated S/PDIF transmitter + * internal S/PDIF out implemented + * No S/PDIF input + * External S/PDIF out implemented + * + * + * ** connected chips ** + * + * WM8776 + * 2-channel DAC used for main output and stereo ADC (with 10-channel MUX) + * AIN1: LINE IN, AIN2: CD/VIDEO, AIN3: AUX, AIN4: Front MIC, AIN5: Rear MIC + * Controlled by I2C using VT1722 I2C interface: + * MODE (pin16) -- GND + * CE (pin17) -- GND I2C mode (address=0x34) + * DI (pin18) -- SDA (VT1722 pin70) + * CL (pin19) -- SCLK (VT1722 pin71) + * + * WM8766 + * 6-channel DAC used for rear & center/LFE outputs (only 4 channels used) + * Controlled by SPI using VT1722 GPIO pins: + * MODE (pin 1) -- GPIO19 (VT1722 pin99) + * ML/I2S (pin11) -- GPIO18 (VT1722 pin98) + * MC/IWL (pin12) -- GPIO17 (VT1722 pin97) + * MD/DM (pin13) -- GPIO16 (VT1722 pin96) + * MUTE (pin14) -- GPIO20 (VT1722 pin101) + * + * ** output pins and device names ** + * + * 7.1ch name -- output connector color -- device (-D option) + * + * FRONT 2ch -- green -- plughw:0,0 + * REAR 2ch -- black -- plughw:0,2,0 + * CENTER(Lch) SUBWOOFER(Rch) -- orange -- plughw:0,2,1 + * + */ + + +/****************************************************************************/ +/* WM8766 interface */ +/****************************************************************************/ + +static void psc724_WM8766_write(struct snd_ice1712 *ice, + unsigned int addr, unsigned int data) +{ + unsigned int st; + unsigned int bits; + int i; + const unsigned int DATA = 0x010000; + const unsigned int CLOCK = 0x020000; + const unsigned int LOAD = 0x040000; + const unsigned int ALL_MASK = (DATA | CLOCK | LOAD); + + snd_ice1712_save_gpio_status(ice); + + st = ((addr & 0x7f) << 9) | (data & 0x1ff); + snd_ice1712_gpio_set_dir(ice, ice->gpio.direction | ALL_MASK); + snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask & ~ALL_MASK); + bits = snd_ice1712_gpio_read(ice) & ~ALL_MASK; + + snd_ice1712_gpio_write(ice, bits); + for (i = 0; i < 16; i++) { + udelay(1); + bits &= ~CLOCK; + st = (st << 1); + if (st & 0x10000) + bits |= DATA; + else + bits &= ~DATA; + + snd_ice1712_gpio_write(ice, bits); + + udelay(1); + bits |= CLOCK; + snd_ice1712_gpio_write(ice, bits); + } + + udelay(1); + bits |= LOAD; + snd_ice1712_gpio_write(ice, bits); + + udelay(1); + bits |= (DATA | CLOCK); + snd_ice1712_gpio_write(ice, bits); + + snd_ice1712_restore_gpio_status(ice); +} + +static void psc724_WM8766_set_volume(struct snd_ice1712 *ice, int ch, + unsigned int vol1, unsigned int vol2) +{ + switch (ch) { + case 0: + psc724_WM8766_write(ice, 0x000, vol1); + psc724_WM8766_write(ice, 0x001, vol2 | 0x100); + break; + case 1: + psc724_WM8766_write(ice, 0x004, vol1); + psc724_WM8766_write(ice, 0x005, vol2 | 0x100); + break; + case 2: + psc724_WM8766_write(ice, 0x006, vol1); + psc724_WM8766_write(ice, 0x007, vol2 | 0x100); + break; + } +} + +static void __devinit psc724_WM8766_init(struct snd_ice1712 *ice) +{ + psc724_WM8766_write(ice, 0x1f, 0x000); /* RESET ALL */ + udelay(10); + + psc724_WM8766_set_volume(ice, 0, 0, 0); /* volume L=0 R=0 */ + psc724_WM8766_set_volume(ice, 1, 0, 0); /* volume L=0 R=0 */ + psc724_WM8766_set_volume(ice, 2, 0, 0); /* volume L=0 R=0 */ + + psc724_WM8766_write(ice, 0x03, 0x022); /* serial mode I2S-24bits */ + psc724_WM8766_write(ice, 0x0a, 0x080); /* MCLK=256fs */ + psc724_WM8766_write(ice, 0x12, 0x000); /* MDP=0 */ + psc724_WM8766_write(ice, 0x15, 0x000); /* MDP=0 */ + psc724_WM8766_write(ice, 0x09, 0x000); /* demp=off mute=off */ + + psc724_WM8766_write(ice, 0x02, 0x124); /* ch-assign L=L R=R RESET */ + psc724_WM8766_write(ice, 0x02, 0x120); /* ch-assign L=L R=R */ +} + +static void psc724_WM8766_set_pro_rate(struct snd_ice1712 *ice, + unsigned int rate) +{ + if (rate > 96000) + psc724_WM8766_write(ice, 0x0a, 0x000); /* MCLK=128fs */ + else + psc724_WM8766_write(ice, 0x0a, 0x080); /* MCLK=256fs */ +} + + +/****************************************************************************/ +/* WM8776 interface */ +/****************************************************************************/ + +static void psc724_WM8776_write(struct snd_ice1712 *ice, + unsigned int addr, unsigned int data) +{ + unsigned int val; + + val = (addr << 9) | data; + printk("write_i2c 0x%x 0x%x\n", addr, data); + snd_vt1724_write_i2c(ice, 0x34, val >> 8, val & 0xff); +} + + +static void psc724_WM8776_set_output_volume(struct snd_ice1712 *ice, + unsigned int vol1, unsigned int vol2) +{ + printk("%s %d,%d\n", __FUNCTION__, vol1, vol2); + psc724_WM8776_write(ice, 0x03, vol1); + psc724_WM8776_write(ice, 0x04, vol2 | 0x100); +} + +static void psc724_WM8776_set_input_volume(struct snd_ice1712 *ice, + unsigned int vol1, unsigned int vol2) +{ + psc724_WM8776_write(ice, 0x0e, vol1); + psc724_WM8776_write(ice, 0x0f, vol2 | 0x100); +} + +static const char *psc724_sel[] = { + "LINE-IN", "CD-IN", "AUX-IN", "FRONT-MIC-IN", "REAR-MIC-IN", "ALL-MIX", NULL +}; + +static void psc724_WM8776_set_input_selector(struct snd_ice1712 *ice, + unsigned int sel) +{ + static unsigned char vals[] = { + /* LINE, CD, AUX, FRONT-MIC, REAR-MIC, ALL, nothing */ + 0x01, 0x02, 0x04, 0x08, 0x10, 0x1f, 0x00 + }; + if (sel > 6) + sel = 6; + psc724_WM8776_write(ice, 0x15, vals[sel]); +} + +static void psc724_WM8776_set_afl(struct snd_ice1712 *ice, unsigned int afl) +{ + /* AFL -- After Fader Listening */ + if (afl) + psc724_WM8776_write(ice, 0x16, 0x005); + else + psc724_WM8776_write(ice, 0x16, 0x001); +} + +static const char *psc724_agc[] = { + "Off", "LimiterMode", "ALCMode", NULL +}; + +static void psc724_WM8776_set_agc(struct snd_ice1712 *ice, unsigned int agc) +{ + /* AGC -- Auto Gain Control of the input */ + switch (agc) { + case 0: + psc724_WM8776_write(ice, 0x11, 0x000); /* Off */ + break; + case 1: + psc724_WM8776_write(ice, 0x10, 0x07b); + psc724_WM8776_write(ice, 0x11, 0x100); /* LimiterMode */ + break; + case 2: + psc724_WM8776_write(ice, 0x10, 0x1fb); + psc724_WM8776_write(ice, 0x11, 0x100); /* ALCMode */ + break; + } +} + +static void __devinit psc724_WM8776_init(struct snd_ice1712 *ice) +{ + int i; + static unsigned short __devinitdata default_values[] = { + 0x100, 0x100, 0x100, + 0x100, 0x100, 0x100, + 0x000, 0x090, 0x000, 0x000, + 0x022, 0x022, 0x022, + 0x008, 0x0cf, 0x0cf, 0x07b, 0x000, + 0x032, 0x000, 0x0a6, 0x001, 0x001 + }; + + psc724_WM8776_write(ice, 0x17, 0x000); /* reset all */ + /* ADC and DAC interface is I2S 24bits mode */ + /* The sample-rate are automatically changed */ + udelay(10); + /* BUT my board can not do reset all, so I load all by manually. */ + for (i = 0; i < ARRAY_SIZE(default_values); i++) + psc724_WM8776_write(ice, i, default_values[i]); + + psc724_WM8776_set_input_selector(ice, 0); + psc724_WM8776_set_afl(ice, 0); + psc724_WM8776_set_agc(ice, 0); + psc724_WM8776_set_input_volume(ice, 0, 0); + psc724_WM8776_set_output_volume(ice, 0, 0); + + /* head phone mute and power down */ +// psc724_WM8776_write(ice, 0x00, 0); +// psc724_WM8776_write(ice, 0x01, 0); +// psc724_WM8776_write(ice, 0x02, 0x100); +// psc724_WM8776_write(ice, 0x0d, 0x080); +} + +static void psc724_WM8776_set_pro_rate(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* nothing to do */ +} + + +/****************************************************************************/ +/* runtime interface */ +/****************************************************************************/ + +static void psc724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + psc724_WM8766_set_pro_rate(ice, rate); + psc724_WM8776_set_pro_rate(ice, rate); +} + +struct psc724_control { + char *name; + enum { + WM8766, + WM8776in, + WM8776out, + WM8776sel, + WM8776agc, + WM8776afl + } target; + enum { VOLUME1, VOLUME2, BOOLEAN, ENUM } type; + int ch; + const char **member; + const char *comment; +}; + +static const struct psc724_control psc724_cont[] = { + { + .name = "Front Playback Volume", + .target = WM8776out, + .type = VOLUME1, + .comment = "Front(green)" + }, + { + .name = "Surround Playback Volume", + .target = WM8766, + .type = VOLUME1, + .ch = 0, + .comment = "SurroundBack(white)" + }, + { + .name = "CLFE Playback Volume", + .target = WM8766, + .type = VOLUME1, + .ch = 1, + .comment = "Center(Lch)&SubWoofer(Rch)(black)" + }, + { + .name = "Capture Volume", + .target = WM8776in, + .type = VOLUME2 + }, + { + .name = "Capture Select", + .target = WM8776sel, + .type = ENUM, + .member = psc724_sel + }, + { + .name = "AGC Capture Mode", + .target = WM8776agc, + .type = ENUM, + .member = psc724_agc + }, + { + .name = "AFL Bypass Playback Switch", + .target = WM8776afl, + .type = BOOLEAN + } +}; + +static int psc724_get_enum_count(int n) +{ + const char **member; + int c; + + member = psc724_cont[n].member; + if (!member) + return 0; + for (c = 0; member[c]; c++) + ; + return c; +} + +static int psc724_cont_volume_info(struct snd_kcontrol *kc, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; /* mute */ + uinfo->value.integer.max = 0xff; /* 0dB */ + return 0; +} + +#define psc724_cont_boolean_info snd_ctl_boolean_mono_info + +static int psc724_cont_enum_info(struct snd_kcontrol *kc, + struct snd_ctl_elem_info *uinfo) +{ + int n, c; + + n = kc->private_value; + c = psc724_get_enum_count(n); + if (!c) + return -EINVAL; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = c; + if (uinfo->value.enumerated.item >= c) + uinfo->value.enumerated.item = c - 1; + strcpy(uinfo->value.enumerated.name, + psc724_cont[n].member[uinfo->value.enumerated.item]); + return 0; +} + +static int psc724_cont_volume_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct psc724_spec *spec = ice->spec; + int n = kc->private_value; + uc->value.integer.value[0] = spec->vol[n].ch1; + uc->value.integer.value[1] = spec->vol[n].ch2; + return 0; +} + +static int psc724_cont_boolean_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct psc724_spec *spec = ice->spec; + int n = kc->private_value; + uc->value.integer.value[0] = spec->vol[n].ch1; + return 0; +} + +static int psc724_cont_enum_get(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct psc724_spec *spec = ice->spec; + int n = kc->private_value; + uc->value.enumerated.item[0] = spec->vol[n].ch1; + return 0; +} + +static void psc724_cont_update(struct snd_ice1712 *ice, int n) +{ + struct psc724_spec *spec = ice->spec; + switch (psc724_cont[n].target) { + case WM8766: + psc724_WM8766_set_volume(ice, + psc724_cont[n].ch, + spec->vol[n].ch1, + spec->vol[n].ch2); + break; + + case WM8776in: + psc724_WM8776_set_input_volume(ice, + spec->vol[n].ch1, + spec->vol[n].ch2); + break; + + case WM8776out: + psc724_WM8776_set_output_volume(ice, + spec->vol[n].ch1, + spec->vol[n].ch2); + break; + + case WM8776sel: + psc724_WM8776_set_input_selector(ice, + spec->vol[n].ch1); + break; + + case WM8776agc: + psc724_WM8776_set_agc(ice, spec->vol[n].ch1); + break; + + case WM8776afl: + psc724_WM8776_set_afl(ice, spec->vol[n].ch1); + break; + + default: + break; + } +} + +static int psc724_cont_volume_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct psc724_spec *spec = ice->spec; + int n = kc->private_value; + unsigned int vol1, vol2; + int changed; + + changed = 0; + vol1 = uc->value.integer.value[0] & 0xff; + vol2 = uc->value.integer.value[1] & 0xff; + if (spec->vol[n].ch1 != vol1) { + spec->vol[n].ch1 = vol1; + changed = 1; + } + if (spec->vol[n].ch2 != vol2) { + spec->vol[n].ch2 = vol2; + changed = 1; + } + if (changed) + psc724_cont_update(ice, n); + + return changed; +} + +static int psc724_cont_boolean_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct psc724_spec *spec = ice->spec; + int n = kc->private_value; + unsigned int vol1; + + vol1 = !!uc->value.integer.value[0]; + if (spec->vol[n].ch1 != vol1) { + spec->vol[n].ch1 = vol1; + psc724_cont_update(ice, n); + return 1; + } + return 0; +} + +static int psc724_cont_enum_put(struct snd_kcontrol *kc, + struct snd_ctl_elem_value *uc) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kc); + struct psc724_spec *spec = ice->spec; + int n = kc->private_value; + unsigned int vol1; + + vol1 = uc->value.enumerated.item[0]; + if (vol1 >= psc724_get_enum_count(n)) + return -EINVAL; + if (spec->vol[n].ch1 != vol1) { + spec->vol[n].ch1 = vol1; + psc724_cont_update(ice, n); + return 1; + } + return 0; +} + +static const DECLARE_TLV_DB_SCALE(db_scale_gain1, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_gain2, -10350, 50, 1); + +static int __devinit psc724_add_controls(struct snd_ice1712 *ice) +{ + int i; + struct snd_kcontrol_new cont; + int err; + + memset(&cont, 0, sizeof(cont)); + cont.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + for (i = 0; i < ARRAY_SIZE(psc724_cont); i++) { + cont.private_value = i; + cont.name = psc724_cont[i].name; + cont.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + cont.tlv.p = NULL; + switch (psc724_cont[i].type) { + case VOLUME1: + case VOLUME2: + cont.info = psc724_cont_volume_info; + cont.get = psc724_cont_volume_get; + cont.put = psc724_cont_volume_put; + cont.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + if (psc724_cont[i].type == VOLUME1) + cont.tlv.p = db_scale_gain1; + else + cont.tlv.p = db_scale_gain2; + break; + case BOOLEAN: + cont.info = psc724_cont_boolean_info; + cont.get = psc724_cont_boolean_get; + cont.put = psc724_cont_boolean_put; + break; + case ENUM: + cont.info = psc724_cont_enum_info; + cont.get = psc724_cont_enum_get; + cont.put = psc724_cont_enum_put; + break; + default: + snd_BUG(); + return -EINVAL; + } + err = snd_ctl_add(ice->card, snd_ctl_new1(&cont, ice)); + if (err < 0) + return err; + } + + return 0; +} + +/****************************************************************************/ +/* probe/initialize/setup */ +/****************************************************************************/ + +static int __devinit psc724_init(struct snd_ice1712 *ice) +{ + struct psc724_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + + ice->num_total_dacs = 6; + ice->num_total_adcs = 2; + psc724_WM8766_init(ice); + psc724_WM8776_init(ice); + ice->gpio.set_pro_rate = psc724_set_pro_rate; + return 0; +} + +/* PSC724 has buggy EEPROM, so override it here */ +static unsigned char psc724_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x42, /* 49.152MHz, ADC, 3DACs */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0x78, /* 96k-ok, 24bit, 192k-ok */ + [ICE_EEP2_SPDIF] = 0xc1, /* out-en, out-int, no spdif-in */ + + [ICE_EEP2_GPIO_DIR] = 0x00, /* not used */ + [ICE_EEP2_GPIO_DIR1] = 0x00, /* not used */ + [ICE_EEP2_GPIO_DIR2] = 0x1f, /* WM8766 MUTE/MODE/ML/MC/MD 1=output */ + + [ICE_EEP2_GPIO_MASK] = 0x00, /* 0=writable */ + [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0=writable */ + [ICE_EEP2_GPIO_MASK2] = 0x00, /* 0=writable */ + + [ICE_EEP2_GPIO_STATE] = 0x00, /* not used */ + [ICE_EEP2_GPIO_STATE1] = 0x00, /* not used */ + [ICE_EEP2_GPIO_STATE2] = 0x07, /* WM8766 ML/MC/MD */ +}; + +struct snd_ice1712_card_info snd_vt1724_psc724_cards[] __devinitdata = { + { + .subvendor = VT1724_SUBDEVICE_PSC724, + .name = "Philips PSC724 Ultimate Edge", + .model = "psc724", + .chip_init = psc724_init, + .build_controls = psc724_add_controls, + .eeprom_size = sizeof(psc724_eeprom), + .eeprom_data = psc724_eeprom, + }, + {} /*terminator*/ +}; diff --git a/sound/pci/ice1712/psc724.h b/sound/pci/ice1712/psc724.h new file mode 100644 index 0000000..858e5fd --- /dev/null +++ b/sound/pci/ice1712/psc724.h @@ -0,0 +1,13 @@ +#ifndef __SOUND_PSC724_H +#define __SOUND_PSC724_H + +/* ID */ +#define PSC724_DEVICE_DESC \ + "{Philips,PSC724 Ultimate Edge}," + +#define VT1724_SUBDEVICE_PSC724 0xab170619 + +/* entry struct */ +extern struct snd_ice1712_card_info snd_vt1724_psc724_cards[]; + +#endif /* __SOUND_PSC724_H */ -- Ondrej Zary

3 6

Re: [alsa-devel] [PATCH v5] ASoC: Support TI LM49453 Audio driver
by Mark Brown 01 Apr '12

01 Apr '12

On Fri, Mar 30, 2012 at 04:03:43PM +0530, MR.Swami.Reddy(a)ti.com wrote: > From: M R Swami Reddy <mr.swami.reddy(a)ti.com> > > Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> Applied, thanks.

1 0

[alsa-devel] [PATCH 0/5] ARM: tegra: audio-related changes for Tegra30
by Stephen Warren 31 Mar '12

31 Mar '12

From: Stephen Warren <swarren(a)nvidia.com> This is the arch/arm/mach-tegra changes that match the ASoC series I just posted. This depends on the previous GPIO/pinmux work that I posted. We should be able to apply this to the Tegra tree independently from the ASoC driver series. Stephen Warren (5): ARM: tegra: provide clock aliases for AHUB configlink ARM: tegra: set up audio clocks for tegra30 dt ARM: tegra: add AUXDATA required for audio ARM: dt: tegra30.dtsi: Add audio-related nodes ARM: dt: tegra cardhu: basic audio support arch/arm/boot/dts/tegra-cardhu.dts | 61 ++++++++++++++++++++++++++++++++ arch/arm/boot/dts/tegra30.dtsi | 32 +++++++++++++++++ arch/arm/mach-tegra/board-dt-tegra30.c | 15 ++++++++ arch/arm/mach-tegra/tegra30_clocks.c | 9 +++++ 4 files changed, 117 insertions(+), 0 deletions(-)

2 6

Re: [alsa-devel] trying to get an eleven rack to work
by Clemens Ladisch 30 Mar '12

30 Mar '12

David Alexander wrote: > kernel: [ 74.128035] usb 2-4: new high speed USB device number 2 > kernel: [ 74.634592] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 > kernel: [ 74.634606] snd-usb-audio: probe of 2-4:1.0 failed with error -5 0x09 is UAC2's EXTENSION_UNIT, which should not be a reason for failing. Please try the patch below. Regards, Clemens --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1926,7 +1926,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_processing_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); - return -EINVAL; + return 0; } }

2 2

Re: [alsa-devel] [patch] ALSA: asihpi - fix return type of hpios_locked_mem_alloc()
by Takashi Iwai 30 Mar '12

30 Mar '12

At Thu, 29 Mar 2012 10:43:20 +1300, Eliot Blennerhassett wrote: > > On 29/03/12 03:38, Takashi Iwai wrote: > > At Wed, 28 Mar 2012 09:57:02 +0300, > > Dan Carpenter wrote: > >> > >> This function returns zero or -ENOMEM, but because it's type is u16, the > >> -ENOMEM gets changed to 65524. None of the callers care, but lets fix > >> it anyway as a cleanup. > >> > >> Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> > > > > Applied it now. Thanks. > > Hmm. I guess it is too late to NAK this change? Yes, Dan's fix itself is correct, so no big reason to revert. > I'd prefer changing the return value to HPI_ERROR_MEMORY_ALLOC and > leaving the function signature alone. It's fine to change to HPI_ERROR_MEMORY_ALLOC. Just write another patch to change it appropriately and submit. But, if we change in that way, we should think again over the return type of these functions, too. If functions are supposed to return these specific error numbers, they should return rather enum HPI_ERROR_CODES type instead of u16. Otherwise it's misleading and an error like this can happen again. Or, follow the common style, returning int with 0 or a negative error number. thanks, Takashi

3 3

[alsa-devel] [PATCH] ASoC: fsl: enable ssi clock in probe function
by Shawn Guo 30 Mar '12

30 Mar '12

For power saving, most IMX platform initilization code turns off modules' clock, and expects driver turn on clock as needed. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++++++++++++-- 1 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4f76b5d..4ed2afd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -14,6 +14,7 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/interrupt.h> +#include <linux/clk.h> #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> @@ -119,6 +120,7 @@ struct fsl_ssi_private { bool new_binding; bool ssi_on_imx; + struct clk *clk; struct platform_device *imx_pcm_pdev; struct imx_pcm_dma_params dma_params_tx; struct imx_pcm_dma_params dma_params_rx; @@ -722,6 +724,15 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; + + ssi_private->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi_private->clk)) { + ret = PTR_ERR(ssi_private->clk); + dev_err(&pdev->dev, "could not get clock: %d\n", ret); + goto error_irq; + } + clk_prepare_enable(ssi_private->clk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -742,7 +753,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) "fsl,ssi-dma-events", dma_events, 2); if (ret) { dev_err(&pdev->dev, "could not get dma events\n"); - goto error_irq; + goto error_clk; } ssi_private->dma_params_tx.dma = dma_events[0]; ssi_private->dma_params_rx.dma = dma_events[1]; @@ -830,6 +841,12 @@ error_dev: dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); +error_clk: + if (ssi_private->ssi_on_imx) { + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } + error_irq: free_irq(ssi_private->irq, ssi_private); @@ -851,8 +868,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { platform_device_unregister(ssi_private->imx_pcm_pdev); + clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: wm8994: Don't bother lowering clock dividers inside idle AIFs
by Mark Brown 30 Mar '12

30 Mar '12

This increases the chances we'll manage to hit a partially configured state on restart and the power savings are extremely small. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 29 ----------------------------- 1 files changed, 0 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ea68c14..5c83505 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2626,33 +2626,6 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } -static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - int rate_reg = 0; - - switch (dai->id) { - case 1: - rate_reg = WM8994_AIF1_RATE; - break; - case 2: - rate_reg = WM8994_AIF2_RATE; - break; - default: - break; - } - - /* If the DAI is idle then configure the divider tree for the - * lowest output rate to save a little power if the clock is - * still active (eg, because it is system clock). - */ - if (rate_reg && !dai->playback_active && !dai->capture_active) - snd_soc_update_bits(codec, rate_reg, - WM8994_AIF1_SR_MASK | - WM8994_AIF1CLK_RATE_MASK, 0x9); -} - static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2750,7 +2723,6 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2760,7 +2732,6 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, - .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, -- 1.7.9.1

1 0

Re: [alsa-devel] [PATCH v5] ASoC: Support TI LM49453 Audio driver
by Mark Brown 30 Mar '12

30 Mar '12

On Thu, Mar 29, 2012 at 07:21:13AM -0700, Reddy, MR Swami wrote: > > ASoC: Support TI LM49453 Audio driver > > The LM49453 is a high performance audio subsystem that supports both analog and > digital audio functions. It includes a high quality stereo DAC, a high quality > stereo ADC, stereo line level outputs, stereo Class D haptic drivers, stereo > class D loudspeaker amplifiers, class D earpiece amplifier and stereo class G > headphone amplifier with microphone detection capability. This basically looks good and I was going to apply it but unfortunately it looks like your mailer corrupted the patch and replaced all tabs with spaces so the tools don't understand it - can you please check and resend? If you're really stuck try an attachment but ideally not. A personal account might also work (all the Marvell people used to use gmail for submitting patches for similar reasons). Some other TI people can probably help you out here, I know a lot of them have been through similar hassle setting up the mail system there.

2 1

[alsa-devel] How to Tx/Rx data via TDM/I2S interface of Intel HD Audio controller without a CODEC attached?
by Couch, Kelly J 30 Mar '12

30 Mar '12

I'm designing software for a specific test application that will utilize an Intel HD audio controller. I must verify the functionality of several sets of TDM audio signals (CK, FS, TXD, RXD) from the Intel HD Audio controller. In this scenario, the Intel HD Audio controller is NOT connected directly to a CODEC, but to a header. My proposal is to connect these signals directly to an FPGA, using it to loopback the data (in various permutations of loopback to channels) to verify each of the signals. Because the FPGA will be in place of a real CODEC device, I'm concerned with how the software side of this approach will work? I essentially just want to push a known bitstream through one TDM interface and verify it upon reception via an alternate TDM interface. This doesn't necessarily require actual audio data. Lastly, I should point out that the system will be tested with it booted to Linux (using a recent kernel 3.x) so utilizing the standard linux audio interfaces (like ALSA - though this isn't a requirement) is preferable to expedite development. I have spent several hours reading over documentation at the official site and elsewhere, but am left unsure of how to accomplish my goal. I am hopeful it is simply a matter of ALSA configuration, but am not sure if that will suffice, nor how to accomplish it. The test method (hardware configuration) is unlikely to change, so I'm soliciting any suggestions and information that you all have to enable sending and receiving a specific data stream over the TDM interfaces of the Intel HD Audio controller. Thanks, -Kelly

2 1

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Enable VAG when DAC/ADC up
by Shawn Guo 30 Mar '12

30 Mar '12

From: Zeng Zhaoming <zengzm.kernel(a)gmail.com> As manual described, VAG is an internal voltage reference of DAC/ADC, So enabled it before DAC/ADC up. One more thing should care about is VAG fully ramped down requires 400ms, wait it to avoid pop. Signed-off-by: Zeng Zhaoming <zengzm.kernel(a)gmail.com> Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 77beb6d..159e3a8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } /* - * using codec assist to small pop, hp_powerup or lineout_powerup - * should stay setting until vag_powerup is fully ramped down, - * vag fully ramped down require 400ms. + * As manual described, ADC/DAC only works when VAG powerup, + * So enabled VAG before ADC/DAC up. + * In power down case, we need wait 400ms when vag fully ramped down. */ -static int small_pop_event(struct snd_soc_dapm_widget *w, +static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { @@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), @@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), }; @@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ -- 1.7.5.4

2 4

[alsa-devel] submiting an asoc-core patch with codec updates
by Brian Austin 29 Mar '12

29 Mar '12

I would like to submit a patch that adds some new macros for TLV in ASoC. this patch requires changes in 2 codecs (CS42L51,CS42L73). Should I submit just the asoc-core patch then follow up with the codec patches or just submit as a series? Thanks, Brian

2 1

[alsa-devel] ASoC updates for 3.4
by Mark Brown 29 Mar '12

29 Mar '12

The following changes since commit 22f8d055350066b4a87de4adea8c5213cac54534: ASoC: wm8994: Provide VMID mode control and fix default sequence (2012-03-19 19:08:17 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 41ba6b711a40b83a877ae8242a4b3e692de91f97: ASoC: dmaengine_pcm: use dmaengine cyclic wrapper (2012-03-26 18:28:11 +0100) Not sure where your current ASoC branch is - topic/asoc and fix/asoc are both behind Linus' tree (which is what this pull request was generated against). ---------------------------------------------------------------- ASoC: A few more updates for 3.4 The cleanup of the dmaengine parameter messup and a tweak to some callibration values for WM1811. ---------------------------------------------------------------- Fabio Estevam (1): ASoC: Add extra parameter to device_prep_dma_cyclic Mark Brown (1): ASoC: wm8994: Update WM8994 DCS calibration Vinod Koul (1): ASoC: dmaengine_pcm: use dmaengine cyclic wrapper sound/soc/codecs/wm8994.c | 2 +- sound/soc/soc-dmaengine-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)

2 2

[alsa-devel] [PATCH] sound: fix isa/opti9xx module param type
by Randy Dunlap 29 Mar '12

29 Mar '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix module parameter data type to eliminate build warnings. sound/isa/opti9xx/opti92x-ad1848.c:87:1: warning: return from incompatible pointer type sound/isa/opti9xx/opti92x-ad1848.c:87:1: warning: return from incompatible pointer type Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> --- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20120328.orig/sound/isa/opti9xx/opti92x-ad1848.c +++ linux-next-20120328/sound/isa/opti9xx/opti92x-ad1848.c @@ -65,7 +65,7 @@ static int index = SNDRV_DEFAULT_IDX1; / static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ //static bool enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ #ifdef CONFIG_PNP -static int isapnp = 1; /* Enable ISA PnP detection */ +static bool isapnp = true; /* Enable ISA PnP detection */ #endif static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */ static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */

2 1

[alsa-devel] [PATCH] sound: fix oss/msnd_pinnacle module param type
by Randy Dunlap 29 Mar '12

29 Mar '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix module parameter data type to eliminate build warning. sound/oss/msnd_pinnacle.c:1727:1: warning: return from incompatible pointer type Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> --- sound/oss/msnd_pinnacle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20120328.orig/sound/oss/msnd_pinnacle.c +++ linux-next-20120328/sound/oss/msnd_pinnacle.c @@ -1631,7 +1631,7 @@ static int ide_irq __initdata = 0; static int joystick_io __initdata = 0; /* If we have the digital daugherboard... */ -static int digital __initdata = 0; +static bool digital __initdata = false; #endif static int fifosize __initdata = DEFFIFOSIZE;

2 1

[alsa-devel] [PATCH v2] ASoC: da7210: Add support for spi regmap
by Ashish Chavan 29 Mar '12

29 Mar '12

This patch adds support for spi regmap feature to existing da7210 driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - used regmap_register_patch() to write special register sequences - used devm_kzalloc() instead of kzalloc() - used devm_regmap_init_spi() instead of regmap_init_spi() - removed "-codec" from codec name string --- sound/soc/codecs/da7210.c | 154 ++++++++++++++++++++++++++++++++++++++------ 1 files changed, 133 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7843711..8e45aa9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> @@ -27,6 +28,7 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_PAGE_CONTROL 0x00 #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 @@ -631,6 +633,7 @@ struct da7210_priv { }; static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, { 0x01, 0x11 }, { 0x03, 0x00 }, { 0x04, 0x00 }, @@ -928,13 +931,6 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* - * make sure that DA7210 use bypass mode before start up - */ - snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - - /* * ADC settings */ @@ -1025,16 +1021,6 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* As suggested by Dialog */ - /* unlock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); - regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); - regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); - /* re-lock */ - regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); - regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); - /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1055,7 +1041,26 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; -static struct regmap_config da7210_regmap = { +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static struct reg_default da7210_regmap_i2c_patch[] = { + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* make sure that DA7210 use bypass mode before start up */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP }, + + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, +}; + +static const struct regmap_config da7210_regmap_config_i2c = { .reg_bits = 8, .val_bits = 8, @@ -1066,7 +1071,6 @@ static struct regmap_config da7210_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1080,13 +1084,18 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, da7210); - da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap_config_i2c); if (IS_ERR(da7210->regmap)) { ret = PTR_ERR(da7210->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); return ret; } + ret = regmap_register_patch(da7210->regmap, da7210_regmap_i2c_patch, + ARRAY_SIZE(da7210_regmap_i2c_patch)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); if (ret < 0) { @@ -1119,7 +1128,7 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); /* I2C codec control layer */ static struct i2c_driver da7210_i2c_driver = { .driver = { - .name = "da7210-codec", + .name = "da7210", .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, @@ -1128,12 +1137,112 @@ static struct i2c_driver da7210_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) + +static struct reg_default da7210_regmap_spi_patch[] = { + /* Dummy read to give two pulses over nCS for SPI */ + { DA7210_AUX2, 0x00 }, + { DA7210_AUX2, 0x00 }, + + /* System controller master disable */ + { DA7210_STARTUP1, 0x00 }, + /* make sure that DA7210 use bypass mode before start up */ + { DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP }, + + /* to set PAGE1 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x80 }, + /* to unlock */ + { DA7210_A_HID_UNLOCK, 0x8B}, + { DA7210_A_TEST_UNLOCK, 0xB4}, + { DA7210_A_PLL1, 0x01}, + { DA7210_A_CP_MODE, 0x7C}, + /* to re-lock */ + { DA7210_A_HID_UNLOCK, 0x00}, + { DA7210_A_TEST_UNLOCK, 0x00}, + /* to set back PAGE0 of SPI register space */ + { DA7210_PAGE_CONTROL, 0x00 }, +}; + +static const struct regmap_config da7210_regmap_config_spi = { + .reg_bits = 8, + .val_bits = 8, + .read_flag_mask = 0x01, + .write_flag_mask = 0x00, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da7210_spi_probe(struct spi_device *spi) +{ + struct da7210_priv *da7210; + int ret; + + da7210 = devm_kzalloc(&spi->dev, sizeof(struct da7210_priv), + GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + spi_set_drvdata(spi, da7210); + da7210->regmap = devm_regmap_init_spi(spi, &da7210_regmap_config_spi); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + ret = regmap_register_patch(da7210->regmap, da7210_regmap_spi_patch, + ARRAY_SIZE(da7210_regmap_spi_patch)); + if (ret != 0) + dev_warn(&spi->dev, "Failed to apply regmap patch: %d\n", ret); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + + return ret; +} + +static int __devexit da7210_spi_remove(struct spi_device *spi) +{ + struct da7210_priv *da7210 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(da7210->regmap); + return 0; +} + +static struct spi_driver da7210_spi_driver = { + .driver = { + .name = "da7210", + .owner = THIS_MODULE, + }, + .probe = da7210_spi_probe, + .remove = __devexit_p(da7210_spi_remove) +}; +#endif + static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&da7210_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(da7210_modinit); @@ -1143,6 +1252,9 @@ static void __exit da7210_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&da7210_spi_driver); +#endif } module_exit(da7210_exit); -- 1.7.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Convert mic bias to a supply widget
by Mark Brown 29 Mar '12

29 Mar '12

No current users and it's the last user of MICBIAS_E(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d192626..77beb6d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -197,9 +197,9 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT"), - SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, small_pop_event, -- 1.7.9.1

4 7

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: Enable VAG when DAC/ADC up
by Shawn Guo 29 Mar '12

29 Mar '12

Zhaoming, On Thu, Mar 29, 2012 at 08:09:33PM +0800, Zeng Zhaoming wrote: ... > This "Mic Bias" is codec internal controller, Machine code should not touch this. In the end, machine driver will have to use it to specify the map. You even said the below. and platform code + {"Mic Bias", NULL, "MIC"}, - {"MIC_IN", NULL, "MIC"}, -- Regards, Shawn

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Don't allow reconfiguration of FLL when it provides SYSCLK
by Mark Brown 29 Mar '12

29 Mar '12

Rather than trying to work around machine drivers which try to reprogram the FLL while it is providing SYSCLK just return an error if they try. This will avoid audio glitches during FLL reconfiguration, or at least move the introduction of the glitches to the machine driver. Since disabling the source for an active SYSCLK is not supported in the first place systems shouldn't be doing this in the first place. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 35 +++++++++++++++++------------------ 1 files changed, 17 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index cc6023c..f43b2ec 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1903,24 +1903,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, struct wm8994 *control = wm8994->wm8994; int reg_offset, ret; struct fll_div fll; - u16 reg, aif1, aif2; + u16 reg, clk1, aif_reg, aif_src; unsigned long timeout; bool was_enabled; - aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) - & WM8994_AIF1CLK_ENA; - - aif2 = snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) - & WM8994_AIF2CLK_ENA; - switch (id) { case WM8994_FLL1: reg_offset = 0; id = 0; + aif_src = 0x10; break; case WM8994_FLL2: reg_offset = 0x20; id = 1; + aif_src = 0x18; break; default: return -EINVAL; @@ -1962,11 +1958,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, if (ret < 0) return ret; - /* Gate the AIF clocks while we reclock */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, 0); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, 0); + /* Make sure that we're not providing SYSCLK right now */ + clk1 = snd_soc_read(codec, WM8994_CLOCKING_1); + if (clk1 & WM8994_SYSCLK_SRC) + aif_reg = WM8994_AIF2_CLOCKING_1; + else + aif_reg = WM8994_AIF1_CLOCKING_1; + reg = snd_soc_read(codec, aif_reg); + + if ((reg & WM8994_AIF1CLK_ENA) && + (reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) { + dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n", + id + 1); + return -EBUSY; + } /* We always need to disable the FLL while reconfiguring */ snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, @@ -2052,12 +2057,6 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, wm8994->fll[id].out = freq_out; wm8994->fll[id].src = src; - /* Enable any gated AIF clocks */ - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA, aif1); - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA, aif2); - configure_clock(codec); return 0; -- 1.7.9.1

1 0

[alsa-devel] Setting up the alsa configuration from scratch on imx
by madhukesh s 29 Mar '12

29 Mar '12

Hi All, I am in an unfortunate position of developing ALSA kernel modules separately(as a kernel build) and populating the necessary files to a filesystem. I am able to install all the ALSA modules on the board I am working on successfully. Can someone point me to how to start with the configuration part on the board to populate /proc and /dev entries? My config(as per the alsa diagnostic script) is: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Fri Mar 2 01:17:47 UTC 2012 !!Linux Distribution !!------------------ !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: !!Kernel Information !!------------------ Kernel release: 2.6.38.8+ Operating System: GNU/Linux Architecture: armv7l Processor: unknown SMP Enabled: No !!ALSA Version !!------------ Driver version: 1.0.23 Library version: Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- --- no soundcards --- !!PCI Soundcards installed in the system !!-------------------------------------- !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- !!Loaded sound module options !!-------------------------- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 33 Mar 2 00:01 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY aplay: device_list:240: no soundcards found... ARECORD arecord: device_list:240: no soundcards found... !!Amixer output !!------------- !!Alsactl output !!------------- --startcollapse-- --endcollapse-- !!All Loaded Modules !!------------------ Module snd_soc_imx_sgtl5000 snd_soc_core snd_pcm snd_timer snd snd_page_alloc ac97_bus !!ALSA/HDA dmesg !!------------------ udevd (1296): /proc/1296/oom_adj is deprecated, please use /proc/1296/oom_score_adj instead. Cirrus Logic CS42888 ALSA SoC Codec Driver i2c-core: driver [cs42888] using legacy suspend method Thanks, Maas

1 0

Re: [alsa-devel] [PATCH] ASoC: sgtl5000: Enable VAG when DAC/ADC up
by Mark Brown 29 Mar '12

29 Mar '12

On Thu, Mar 29, 2012 at 08:09:33PM +0800, Zeng Zhaoming wrote: > On Thu 2012-03-29 12:16:05, Mark Brown wrote: > > > + {"Mic Bias", NULL, "MIC_IN"}, /* mic_in --> mic bias */ > > > + {"Capture Mux", "MIC_IN", "Mic Bias"}, /* mic bias --> adc_mux */ > > This should be in the machine driver. > This "Mic Bias" is codec internal controller, Machine code should not > touch this. What is it actually doing? Calling it "Mic Bias" if it's not a bias source for microphones is horribly confusing.

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Enable VAG when DAC/ADC up
by Shawn Guo 29 Mar '12

29 Mar '12

From: Zeng Zhaoming <zengzm.kernel(a)gmail.com> As manual described, VAG is an internal voltage reference of DAC/ADC, So enabled it before DAC/ADC up. As the result, the issue that captured wave file has no sound gets fixed. One more thing should care about is VAG fully ramped down requires 400ms, wait it to avoid pop. Signed-off-by: Zeng Zhaoming <zengzm.kernel(a)gmail.com> Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 29 +++++++++++++++-------------- 1 files changed, 15 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d192626..d6f5aaf 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } /* - * using codec assist to small pop, hp_powerup or lineout_powerup - * should stay setting until vag_powerup is fully ramped down, - * vag fully ramped down require 400ms. + * As manual described, ADC/DAC only works when VAG powerup, + * So enabled VAG before ADC/DAC up. + * In power down case, we need wait 400ms when vag fully ramped down. */ -static int small_pop_event(struct snd_soc_dapm_widget *w, +static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { @@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), @@ -221,15 +217,20 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), - - SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + SND_SOC_DAPM_ADC_E("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; /* routes for sgtl5000 */ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"Mic Bias", NULL, "MIC_IN"}, /* mic_in --> mic bias */ + {"Capture Mux", "MIC_IN", "Mic Bias"}, /* mic bias --> adc_mux */ {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ -- 1.7.5.4

3 2

[alsa-devel] ASoC:Question on strict sequence followed for trigger
by gsantosh＠codeaurora.org 29 Mar '12

29 Mar '12

Hi All, I see there is a strict sequence followed for trigger callback to codec dai, platform dai and cpu dai. In the below function of soc-core . int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd); In some scenarios we may require the sequence to be the other case to make sure the underlying DSP / codec is configured and triggered properly. I have a requirement that this sequence has to be 1) Codec_dai 2) Cpu_dai 3) Platform dai Will this call sequence can be changed or there is no room to change this sequence. Regards, Santosh M G.

3 2

Re: [alsa-devel] [PATCH linux-next] dmaengine: add context parameter fixups
by Fabio Estevam 29 Mar '12

29 Mar '12

On Fri, Mar 23, 2012 at 11:00 AM, Alexandre Bounine <alexandre.bounine(a)idt.com> wrote: > This patch fixes drivers that use device_prep_slave_sg() and > device_prep_dma_cyclic() interfaces and have been missed during the initial > release of interface changing patches. > > This patch is based on linux-next-20120322 code tree. > See https://lkml.org/lkml/2012/3/22/339 for build failure report. > > Signed-off-by: Alexandre Bounine <alexandre.bounine(a)idt.com> > Reported-by: Paul Gortmaker <paul.gortmaker(a)windriver.com> > --- > drivers/dma/sa11x0-dma.c | 2 +- > drivers/net/irda/sa1100_ir.c | 2 +- > sound/soc/sh/fsi.c | 7 +++---- > sound/soc/soc-dmaengine-pcm.c | 2 +- I have already taken care of sound/soc/soc-dmaengine-pcm.c: http://permalink.gmane.org/gmane.linux.alsa.devel/96215

4 10

Re: [alsa-devel] How to cross compile the alsa ?
by Dámaso López 29 Mar '12

29 Mar '12

Yeah, I configure with --enable-static=yes and another parameter : #CC=arm-gp2x-linux ./configure –host=i686.linux –target=arm-gp2x-linux –enable-store=no –enable-static=yes And same make and make install... But when I copy all files in corresponding directory on /usr/arm-gp2x-linux/usr/* after then I compile. But I look errors that not defined reference, I don't undertand, What I do bad?

2 14

[alsa-devel] [PATCH] ASoC: sgtl5000: fix power on/off of MIC bias
by Shawn Guo 29 Mar '12

29 Mar '12

From: Lionel Xu <Lionel.Xu(a)freescale.com> There is no sound in the recorded wav, to enable recording, the VAG should be powered up, and the mic bias resistor should be set up with proper value. Signed-off-by: Lionel Xu <lionel.xu(a)freescale.com> Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Zeng Zhaoming <zengzm.kernel(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 14 +++++++++++--- 1 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d192626..a24299c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -132,11 +132,18 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_MASK, SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT); + + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, 0); + SGTL5000_BIAS_R_MASK, + SGTL5000_BIAS_R_off << SGTL5000_BIAS_R_SHIFT); + + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); break; } return 0; @@ -197,7 +204,7 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP_OUT"), SND_SOC_DAPM_OUTPUT("LINE_OUT"), - SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SND_SOC_NOPM, 0, 0, mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -229,7 +236,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { /* routes for sgtl5000 */ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ - {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"Mic Bias", NULL, "MIC_IN"}, /* mic_in --> mic bias */ + {"Capture Mux", "MIC_IN", "Mic Bias"}, /* mic bias --> adc_mux */ {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ -- 1.7.4.1

4 5

Re: [alsa-devel] trying to get an eleven rack to work
by David Alexander 28 Mar '12

28 Mar '12

---------- Forwarded message ---------- From: "David Alexander" <lidxv0n(a)gmail.com> Date: Mar 27, 2012 10:45 PM Subject: Re: [alsa-devel] trying to get an eleven rack to work To: "Clemens Ladisch" <clemens(a)ladisch.de> Grant & Clemens, Sorry for the delay in getting the dev box set up. Here are the errors listed from the syslog with the suggested additions to quirks-table.h implemented: kernel: [ 74.128035] usb 2-4: new high speed USB device number 2 using ehci_hcd mtp-probe: checking bus 2, device 2: "/sys/devices/pci0000:00/0000:00:1d.7/usb2/2-4" mtp-probe: bus: 2, device: 2 was not an MTP device kernel: [ 74.634592] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 kernel: [ 74.634606] snd-usb-audio: probe of 2-4:1.0 failed with error -5 kernel: [ 74.635228] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 kernel: [ 74.635246] snd-usb-audio: probe of 2-4:1.1 failed with error -5 kernel: [ 74.635717] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 kernel: [ 74.635732] snd-usb-audio: probe of 2-4:1.2 failed with error -5 kernel: [ 74.636515] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 kernel: [ 74.636529] snd-usb-audio: probe of 2-4:1.3 failed with error -5 kernel: [ 74.636973] ALSA mixer.c:1938 usbaudio: unit 32: unexpected type 0x09 kernel: [ 74.636986] snd-usb-audio: probe of 2-4:1.4 failed with error -5 kernel: [ 74.637027] usbcore: registered new interface driver snd-usb-audio Thank you, David On Tue, Mar 27, 2012 at 4:57 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: > Grant Diffey wrote: >> my guess at the interface purposes from David's post are: >> >> Interface 1 is really custom (perhaps some sort of custom control/management thing tho I would of thought they'd use midi sysex for that) > > It's an audio control interface, just marked as vendor-specific. > >> Interface 3 looks like maybe a usb mixer without a descriptor? but that's a guess. > > Interface 3 is audio output. > >> Interface 4 is potentially usb2 audio output // looks similar to the maudioi fast track isosyncronous outputs > > Interface 4 is audio input, it has only an input endpoint. > >> Interface 5 is potentially usb2 audio input. // "" > > There is no interface 5. > >> can USB_DEVICE_VENDOR_SPEC deal with out of order compliant/quirked interfaces or only sequencial?? > > The one has nothing to do with the other. > > > Regards, > Clemens

1 0

[alsa-devel] [PATCH alsa-lib 1/1] doc: Fix typo in PCM formats
by Adrian Knoth 28 Mar '12

28 Mar '12

Of course, 24bit samples are not stored in three bits, but bytes. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index ea1afdc..18b43b3 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -209,13 +209,13 @@ The device is physicaly disconnected. It does not accept any I/O calls in this s \section pcm_formats PCM formats The full list of formats present the #snd_pcm_format_t type. -The 24-bit linear samples uses 32-bit physical space, but the sample is -stored in low three bits. Some hardware does not support processing of full +The 24-bit linear samples use 32-bit physical space, but the sample is +stored in the lower three bytes. Some hardware does not support processing of full range, thus you may get the significant bits for linear samples via #snd_pcm_hw_params_get_sbits() function. The example: ICE1712 chips support 32-bit sample processing, but low byte is ignored (playback) or zero (capture). The function snd_pcm_hw_params_get_sbits() -returns 24 in the case. +returns 24 in this case. \section alsa_transfers ALSA transfers -- 1.7.9.1

2 1

[alsa-devel] [patch] ALSA: asihpi - fix return type of hpios_locked_mem_alloc()
by Dan Carpenter 28 Mar '12

28 Mar '12

This function returns zero or -ENOMEM, but because it's type is u16, the -ENOMEM gets changed to 65524. None of the callers care, but lets fix it anyway as a cleanup. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 4cc315d..8c63200 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned. If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle. */ -u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, +int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, /**< memory handle */ u32 size, /**< Size in bytes to allocate */ struct pci_dev *p_os_reference diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 2d7d1c2..87f4385 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -43,7 +43,7 @@ void hpios_delay_micro_seconds(u32 num_micro_sec) On error, return -ENOMEM, and *pMemArea.size = 0 */ -u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, +int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) { /*?? any benefit in using managed dmam_alloc_coherent? */

2 1

Re: [alsa-devel] [PATCH] ASoC: ep93xx-i2s: use devm_* helpers to cleanup probe
by Mark Brown 28 Mar '12

28 Mar '12

On Mon, Mar 26, 2012 at 04:00:17PM -0700, H Hartley Sweeten wrote: > Use the devm_* helpers to cleanup the probe routine. This also eliminates > having to carry the mem value in the private data for the remove. > Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> > Cc: Mika Westerberg <mika.westerberg(a)iki.fi> > Cc: Liam Girdwood <lrg(a)ti.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > CC: Takashi Iwai <tiwai(a)suse.de> > Cc: Vinod Koul <vinod.koul(a)intel.com> Applied, but again please take care of the above.

1 0

Re: [alsa-devel] [PATCH] ASoC: ep93xx-ac97: use devm_* helpers to cleanup probe
by Mark Brown 28 Mar '12

28 Mar '12

On Mon, Mar 26, 2012 at 04:00:08PM -0700, H Hartley Sweeten wrote: > Use the devm_* helpers to cleanup the probe routine. This also eliminates > having to carry the mem and irq values in the private data for the remove. > Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> > Cc: Mika Westerberg <mika.westerberg(a)iki.fi> > Cc: Liam Girdwood <lrg(a)ti.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > CC: Takashi Iwai <tiwai(a)suse.de> Applied, but please: - Think more about your CC list. - Try to avoid putting it in the body of the e-mail.

1 0

[alsa-devel] [PATCH] ASoC: fsl: assign dma peripheral type according to bus topology
by Shawn Guo 28 Mar '12

28 Mar '12

The dma peripheral_type for SSI should be IMX_DMATYPE_SSI_SP if the SSI is on SPBA bus. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 6 ++++++ sound/soc/fsl/imx-pcm-dma.c | 3 ++- sound/soc/fsl/imx-pcm.h | 1 + 3 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 30ff605..4f76b5d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -746,6 +746,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) } ssi_private->dma_params_tx.dma = dma_events[0]; ssi_private->dma_params_rx.dma = dma_events[1]; + + ssi_private->dma_params_tx.shared_peripheral = + of_device_is_compatible(of_get_parent(np), + "fsl,spba-bus"); + ssi_private->dma_params_rx.shared_peripheral = + ssi_private->dma_params_tx.shared_peripheral; } /* Initialize the the device_attribute structure */ diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index e43c8fa..8c43262 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -106,7 +106,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); - dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->peripheral_type = dma_params->shared_peripheral ? + IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; dma_data->dma_request = dma_params->dma; diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index b5f5c3a..83c0ed7 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -22,6 +22,7 @@ struct imx_pcm_dma_params { int dma; unsigned long dma_addr; int burstsize; + bool shared_peripheral; /* The peripheral is on SPBA bus */ }; int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, -- 1.7.4.1

2 1

[alsa-devel] DMA problem with The GTA02 Machine.
by Denis 'GNUtoo' Carikli 28 Mar '12

28 Mar '12

Hi, There is a problem with DMA with the GTA02 machine: instead of playing a wav file saying "incomming call..." it plays "in-in-in-co-co-co-mi-mi-mi...". I don't know how that problem is called. In any case I've bisected it during hours(the commit that broke it was in an untestable area, so I had to bisect what made the area untestable too(it's already fixed in linux-next)). Then after identifying the commit I've tried to narrow it more. But I don't know how to narrow it even more than what I've in the patch attached. Denis.

3 3

[alsa-devel] [PATCH v5] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 28 Mar '12

28 Mar '12

Changes since v4: * Add new_binding flag for fsl_ssi driver to support both "old binding" and "new binding" --- Shawn Guo (8): ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: rename imx-pcm Kconfig options and filename ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: remove helper fsl_asoc_get_codec_dev_name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 1 + arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 +++++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 141 ++++++++++++---- sound/soc/fsl/fsl_utils.c | 91 ++++++++++ sound/soc/fsl/fsl_utils.h | 26 +++ sound/soc/{imx => fsl}/imx-audmux.c | 0 sound/soc/{imx => fsl}/imx-audmux.h | 0 .../{imx/imx-pcm-dma-mx2.c => fsl/imx-pcm-dma.c} | 0 sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/{imx => fsl}/imx-pcm.c | 0 sound/soc/{imx => fsl}/imx-pcm.h | 0 sound/soc/fsl/imx-sgtl5000.c | 177 ++++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 2 +- sound/soc/{imx => fsl}/imx-ssi.h | 0 sound/soc/fsl/mpc8610_hpcd.c | 166 ++----------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 0 sound/soc/fsl/p1022_ds.c | 158 +---------------- sound/soc/{imx => fsl}/phycore-ac97.c | 0 sound/soc/{imx => fsl}/wm1133-ev1.c | 0 sound/soc/imx/Kconfig | 79 --------- sound/soc/imx/Makefile | 22 --- 28 files changed, 601 insertions(+), 445 deletions(-)

5 21

Re: [alsa-devel] [PATCH] ASoC: ep93xx-i2s: use devm_* helpers to cleanup probe
by Mika Westerberg 27 Mar '12

27 Mar '12

On Mon, Mar 26, 2012 at 04:00:17PM -0700, H Hartley Sweeten wrote: > Use the devm_* helpers to cleanup the probe routine. This also eliminates > having to carry the mem value in the private data for the remove. > > Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> > Cc: Mika Westerberg <mika.westerberg(a)iki.fi> Acked-by: Mika Westerberg <mika.westerberg(a)iki.fi> > Cc: Liam Girdwood <lrg(a)ti.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > CC: Takashi Iwai <tiwai(a)suse.de> > Cc: Vinod Koul <vinod.koul(a)intel.com>

1 0

Re: [alsa-devel] [PATCH] ASoC: ep93xx-ac97: use devm_* helpers to cleanup probe
by Mika Westerberg 27 Mar '12

27 Mar '12

On Mon, Mar 26, 2012 at 04:00:08PM -0700, H Hartley Sweeten wrote: > Use the devm_* helpers to cleanup the probe routine. This also eliminates > having to carry the mem and irq values in the private data for the remove. > > Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> > Cc: Mika Westerberg <mika.westerberg(a)iki.fi> Tested-by: Mika Westerberg <mika.westerberg(a)iki.fi> Acked-by: Mika Westerberg <mika.westerberg(a)iki.fi> > Cc: Liam Girdwood <lrg(a)ti.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > CC: Takashi Iwai <tiwai(a)suse.de>

1 0

[alsa-devel] dsp-soc: question regarding dsp_add_new_paths
by Bensaid, Selma 27 Mar '12

27 Mar '12

Hi, I'm reversing the dsp-soc code and I have a question regarding the function dsp_add_new_paths: - Why do you check if (be->dai_link->dynamic) when looking for the FE AIF_IN/OUT, in my comprehension the for FE the dai_link->dynamic = 1? - For both loops used to find out the FE and BE why do you check the be->dai_link->dynamicsince in the snd_soc_dapm_get_aif loops over all card widget? Thanks in advance for your answer. Selma. --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 1

[alsa-devel] M-audio fasttrack pro, input device index
by David Henningsson 27 Mar '12

27 Mar '12

Hi, M-Audio fasttrack pro has two stereo outputs and one stereo input. It looks like the input sometimes shows up at device 0 and sometimes at device 1. I e, a device string would sometimes be hw:1,0,0 and sometimes hw:1,1,0. (Assuming the card index is 1) Do we know if there is a reason behind this behaviour, or should we just try both device strings? I don't have this device myself, so I don't know if it's consistent given a specific physical device, if it differs between kernel versions, or anything else. I've just seen both. Example alsa-info: https://launchpadlibrarian.net/45386782/alsa-info.txt.5TIcWGnRjL -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

Re: [alsa-devel] ALSA async callback re-enter and DEADLOCK.
by Clemens Ladisch 27 Mar '12

27 Mar '12

Павел Колодин wrote: > So, in multi-threaded appliction, the better solution would be to start separate thread that would talk with ALSA only? Yes. If you already have a poll()-based event loop in some thread, it might make sense to put the ALSA code there, but this depends on how these threads affect each other. Regards, Clemens

1 0

[alsa-devel] [PATCH 0/8] Add ASoC drivers for SPEAr platform
by Rajeev Kumar 27 Mar '12

27 Mar '12

This patch set adds support for ASoC sound drivers on ST's SPEAr platforms. Details of the SPEAr platforms can be seen here: http://www.st.com/internet/mcu/product/250658.jsp This patch set supports the following modules: 1. Audio codec sta529 (connected to I2S controller). 2. SPDIF IN dummy Codec. 3. Synopsys Designware I2S controller. 4. ST SPDIF IN controller. 5. ST SPDIF OUT controller. This patch set has been tested for linux-kernel-version 3.3-rc3 on SPEAr platform. In past, I had circulated the patch for the same and got some review comments. I have incorporate those commnets also. Please find the link below. http://mailman.alsa-project.org/pipermail/alsa-devel/2011-April/038738.html Rajeev Kumar (6): sound:asoc: Add support for STA529 Audio Codec sound:asoc: Add support for synopsys i2s controller as per ASoC framework. sound:asoc: Add support for SPEAr ASoC pcm layer. sound:asoc: Add support for SPEAr ASoC machine driver. sound:asoc: Add Kconfig and Makefile to support SPEAr audio driver sound:asoc: Update Kconfig and Makefile(sound/soc/) to support SPEAr audio Vipin Kumar (2): sound:asoc:spdif_in: Add spdif IN support sound:asoc:spdif_out: Add spdif out support include/linux/designware_i2s.h | 63 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta529.c | 414 ++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 46 +++ sound/soc/spear/Makefile | 16 + sound/soc/spear/designware_i2s.c | 610 ++++++++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_in.c | 304 +++++++++++++++++++ sound/soc/spear/spdif_in_regs.h | 60 ++++ sound/soc/spear/spdif_out.c | 397 +++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 +++++ sound/soc/spear/spear_evb.c | 278 +++++++++++++++++ sound/soc/spear/spear_pcm.c | 460 ++++++++++++++++++++++++++++ sound/soc/spear/spear_pcm.h | 33 ++ 16 files changed, 2769 insertions(+), 0 deletions(-) create mode 100644 include/linux/designware_i2s.h create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/designware_i2s.c create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_evb.c create mode 100644 sound/soc/spear/spear_pcm.c create mode 100644 sound/soc/spear/spear_pcm.h -- 1.7.2.2

3 24

[alsa-devel] How to cross compile the alsa ?
by Dámaso López 27 Mar '12

27 Mar '12

Hi all, I need you help not to understant how cross compile alsa with arm-gp2x-linux. I downloaded alsa-lib-1.0.17 and alsa-utils-1.0.17 last week Step to step, because I don't know if I will forget any step. Once I descrompressed in /home/dam: tar -xvfj alsa-lib-1.0.17.tar.bz2 tar -xvfj alsa-utils-1.0.17.tar.bz2 Then to descompress I went to directory alsa-lib... and I put in directory, the next line: # CC=arm-gp2x-linux ./configure --host=i686.linux --target=arm-linux #make #make install When it finish I copy all include from alsa-lib-1.0.17 in my directory /usr/arm-gp2x-linux/include. After I did: #arm-gp2x-linux -o micro micro.c -lasound But it say that can't find -lasound and ld returned 1 exit status. What I did bad do? Can you put step to step, please?

2 7

[alsa-devel] Question: PCM short and long sync formats
by Omair Mohammed Abdullah 27 Mar '12

27 Mar '12

I was looking for PCM short frame sync and PCM long frame sync, but they are not present in soc-dai.h (FORMAT_MASK). Is there a way to derive them from other formats or should I just add them?

3 10

[alsa-devel] Electronix Ebox 44 quirk?
by Rune 26 Mar '12

26 Mar '12

Any news about the quirk for this bogus device? I read about some one which was very close to the solution.. Thank you all, Rune

1 0

[alsa-devel] [PATCH] ASoC: dmaengine_pcm: use dmaengine cyclic wrapper
by Vinod Koul 26 Mar '12

26 Mar '12

commit 4a163c82 added extra parameter to device_prep_dma_cyclic this is not required if we use the wrapper over cyclic API. This is split from Alexedre's patch Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/soc/soc-dmaengine-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 76c038b..4756952 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -143,10 +143,10 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) direction = snd_pcm_substream_to_dma_direction(substream); prtd->pos = 0; - desc = chan->device->device_prep_dma_cyclic(chan, + desc = dmaengine_prep_dma_cyclic(chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), direction, NULL); + snd_pcm_lib_period_bytes(substream), direction); if (!desc) return -ENOMEM; -- 1.7.0.4

2 1

Re: [alsa-devel] [PATCH v2] ARM: pxa: use common IOMEM definition
by Russell King - ARM Linux 25 Mar '12

25 Mar '12

What's happening about this and your other patch for PXA? PXA is currently broken in arm-soc, and has been for the past week. See http://www.arm.linux.org.uk/developer/build/ On Thu, Mar 22, 2012 at 08:42:41AM -0500, Rob Herring wrote: > From: Rob Herring <rob.herring(a)calxeda.com> > > pxa was missed in the moving of IOMEM to a common definition, so lots of > IOMEM redefined warnings were introduced. So remove pxa IOMEM definition > and fix all the fallout. > > Reported-by: Paul Gortmaker <paul.gortmaker(a)windriver.com> > Signed-off-by: Rob Herring <rob.herring(a)calxeda.com> > Cc: Eric Miao <eric.y.miao(a)gmail.com> > Cc: Russell King <linux(a)arm.linux.org.uk> > Cc: Haojian Zhuang <haojian.zhuang(a)marvell.com> > Cc: Alessandro Zummo <a.zummo(a)towertech.it> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: rtc-linux(a)googlegroups.com > Cc: alsa-devel(a)alsa-project.org > --- > v2: > - fix my commit msg typos > - add fix for sa1100_wdt.c > > I plan to send this fix thru the arm-soc tree. > > Rob > > arch/arm/mach-pxa/clock-pxa2xx.c | 1 + > arch/arm/mach-pxa/corgi_pm.c | 1 + > arch/arm/mach-pxa/cpufreq-pxa3xx.c | 1 + > arch/arm/mach-pxa/include/mach/hardware.h | 2 -- > arch/arm/mach-pxa/mfp-pxa2xx.c | 1 + > arch/arm/mach-pxa/pxa2xx.c | 1 + > arch/arm/mach-pxa/pxa300.c | 1 + > arch/arm/mach-pxa/pxa320.c | 1 + > arch/arm/mach-pxa/sharpsl_pm.c | 1 + > drivers/rtc/rtc-sa1100.c | 1 + > drivers/watchdog/sa1100_wdt.c | 1 + > sound/arm/pxa2xx-ac97-lib.c | 1 + > sound/arm/pxa2xx-ac97.c | 1 + > sound/soc/pxa/pxa2xx-ac97.c | 1 + > 14 files changed, 13 insertions(+), 2 deletions(-) > > diff --git a/arch/arm/mach-pxa/clock-pxa2xx.c b/arch/arm/mach-pxa/clock-pxa2xx.c > index 1d5859d..9ee2ad6 100644 > --- a/arch/arm/mach-pxa/clock-pxa2xx.c > +++ b/arch/arm/mach-pxa/clock-pxa2xx.c > @@ -9,6 +9,7 @@ > #include <linux/module.h> > #include <linux/kernel.h> > #include <linux/init.h> > +#include <linux/io.h> > #include <linux/syscore_ops.h> > > #include <mach/pxa2xx-regs.h> > diff --git a/arch/arm/mach-pxa/corgi_pm.c b/arch/arm/mach-pxa/corgi_pm.c > index 39e265c..048c429 100644 > --- a/arch/arm/mach-pxa/corgi_pm.c > +++ b/arch/arm/mach-pxa/corgi_pm.c > @@ -19,6 +19,7 @@ > #include <linux/interrupt.h> > #include <linux/platform_device.h> > #include <linux/apm-emulation.h> > +#include <linux/io.h> > > #include <asm/irq.h> > #include <asm/mach-types.h> > diff --git a/arch/arm/mach-pxa/cpufreq-pxa3xx.c b/arch/arm/mach-pxa/cpufreq-pxa3xx.c > index 88fbec0..b85b4ab 100644 > --- a/arch/arm/mach-pxa/cpufreq-pxa3xx.c > +++ b/arch/arm/mach-pxa/cpufreq-pxa3xx.c > @@ -15,6 +15,7 @@ > #include <linux/init.h> > #include <linux/cpufreq.h> > #include <linux/slab.h> > +#include <linux/io.h> > > #include <mach/pxa3xx-regs.h> > > diff --git a/arch/arm/mach-pxa/include/mach/hardware.h b/arch/arm/mach-pxa/include/mach/hardware.h > index 54b64ea..56d92e5 100644 > --- a/arch/arm/mach-pxa/include/mach/hardware.h > +++ b/arch/arm/mach-pxa/include/mach/hardware.h > @@ -40,7 +40,6 @@ > #define io_p2v(x) IOMEM(0xf2000000 + ((x) & 0x01ffffff) + (((x) & 0x1c000000) >> 1)) > > #ifndef __ASSEMBLY__ > -# define IOMEM(x) ((void __iomem *)(x)) > # define __REG(x) (*((volatile u32 __iomem *)io_p2v(x))) > > /* With indexed regs we don't want to feed the index through io_p2v() > @@ -52,7 +51,6 @@ > > #else > > -# define IOMEM(x) x > # define __REG(x) io_p2v(x) > # define __PREG(x) io_v2p(x) > > diff --git a/arch/arm/mach-pxa/mfp-pxa2xx.c b/arch/arm/mach-pxa/mfp-pxa2xx.c > index f147755..b029562 100644 > --- a/arch/arm/mach-pxa/mfp-pxa2xx.c > +++ b/arch/arm/mach-pxa/mfp-pxa2xx.c > @@ -17,6 +17,7 @@ > #include <linux/module.h> > #include <linux/kernel.h> > #include <linux/init.h> > +#include <linux/io.h> > #include <linux/syscore_ops.h> > > #include <mach/pxa2xx-regs.h> > diff --git a/arch/arm/mach-pxa/pxa2xx.c b/arch/arm/mach-pxa/pxa2xx.c > index 8682704..f8ec854 100644 > --- a/arch/arm/mach-pxa/pxa2xx.c > +++ b/arch/arm/mach-pxa/pxa2xx.c > @@ -13,6 +13,7 @@ > #include <linux/module.h> > #include <linux/kernel.h> > #include <linux/device.h> > +#include <linux/io.h> > > #include <mach/hardware.h> > #include <mach/pxa2xx-regs.h> > diff --git a/arch/arm/mach-pxa/pxa300.c b/arch/arm/mach-pxa/pxa300.c > index 40bb165..17cbc0c 100644 > --- a/arch/arm/mach-pxa/pxa300.c > +++ b/arch/arm/mach-pxa/pxa300.c > @@ -16,6 +16,7 @@ > #include <linux/module.h> > #include <linux/kernel.h> > #include <linux/platform_device.h> > +#include <linux/io.h> > > #include <mach/pxa300.h> > > diff --git a/arch/arm/mach-pxa/pxa320.c b/arch/arm/mach-pxa/pxa320.c > index 8d614ec..6dc99d4 100644 > --- a/arch/arm/mach-pxa/pxa320.c > +++ b/arch/arm/mach-pxa/pxa320.c > @@ -16,6 +16,7 @@ > #include <linux/module.h> > #include <linux/kernel.h> > #include <linux/platform_device.h> > +#include <linux/io.h> > > #include <mach/pxa320.h> > > diff --git a/arch/arm/mach-pxa/sharpsl_pm.c b/arch/arm/mach-pxa/sharpsl_pm.c > index 8d5168d..6d3b39f 100644 > --- a/arch/arm/mach-pxa/sharpsl_pm.c > +++ b/arch/arm/mach-pxa/sharpsl_pm.c > @@ -24,6 +24,7 @@ > #include <linux/leds.h> > #include <linux/suspend.h> > #include <linux/gpio.h> > +#include <linux/io.h> > > #include <asm/mach-types.h> > #include <mach/pm.h> > diff --git a/drivers/rtc/rtc-sa1100.c b/drivers/rtc/rtc-sa1100.c > index cb9a585..398ab7a 100644 > --- a/drivers/rtc/rtc-sa1100.c > +++ b/drivers/rtc/rtc-sa1100.c > @@ -30,6 +30,7 @@ > #include <linux/string.h> > #include <linux/pm.h> > #include <linux/bitops.h> > +#include <linux/io.h> > > #include <mach/hardware.h> > #include <asm/irq.h> > diff --git a/drivers/watchdog/sa1100_wdt.c b/drivers/watchdog/sa1100_wdt.c > index 0162454..a3ff321 100644 > --- a/drivers/watchdog/sa1100_wdt.c > +++ b/drivers/watchdog/sa1100_wdt.c > @@ -25,6 +25,7 @@ > #include <linux/miscdevice.h> > #include <linux/watchdog.h> > #include <linux/init.h> > +#include <linux/io.h> > #include <linux/bitops.h> > #include <linux/uaccess.h> > #include <linux/timex.h> > diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c > index d1aa421..d760371 100644 > --- a/sound/arm/pxa2xx-ac97-lib.c > +++ b/sound/arm/pxa2xx-ac97-lib.c > @@ -17,6 +17,7 @@ > #include <linux/clk.h> > #include <linux/delay.h> > #include <linux/module.h> > +#include <linux/io.h> > > #include <sound/ac97_codec.h> > #include <sound/pxa2xx-lib.h> > diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c > index 3a39626..afef72c 100644 > --- a/sound/arm/pxa2xx-ac97.c > +++ b/sound/arm/pxa2xx-ac97.c > @@ -11,6 +11,7 @@ > */ > > #include <linux/init.h> > +#include <linux/io.h> > #include <linux/module.h> > #include <linux/platform_device.h> > > diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c > index 837ff34..b4bfd97 100644 > --- a/sound/soc/pxa/pxa2xx-ac97.c > +++ b/sound/soc/pxa/pxa2xx-ac97.c > @@ -11,6 +11,7 @@ > */ > > #include <linux/init.h> > +#include <linux/io.h> > #include <linux/module.h> > #include <linux/platform_device.h> > > -- > 1.7.5.4 >

1 0

[alsa-devel] LV2-C++-Tools
by Aurélien Leblond 24 Mar '12

24 Mar '12

Hello, I have been coding the porting of the AMS internal modules to LV2 plugins using the Lv2-C++-Tools. The more I'm progressing, the more I am wondering if there are any limitations involved in using this library, especially these days where a lot of new features are being added to the Lv2 Extensions. For example, I am coding a plugin that output the position of the mouse cursor as two control ports (handy to develop a kind of theremin synth). Getting the cursor position from the X Library is an expensive operation, and the new Worker Extension seems perfect for the job, but I can't figure out if somehow it is compatible or not with the Lv2-C++-Tools. The most obvious pros of using Lv2-C++-Tools is its simplicity. Porting an AMS module takes only 20 to 30 minutes with it, while doing it without the help of the library takes much more time and is way more tedious. (I am much more confortable coding in C++ than C, but the question deserves to be asked, am I doing something wrong here that I find it way more difficult to code without the help of Lv2-C++-Tools?) So I am putting the question to the community: what path you would you advise to follow? With or Without Lv2-C++-Tools? Aurélien

1 0

Re: [alsa-devel] [PATCH linux-next] dmaengine: add context parameter fixups
by Mark Brown 24 Mar '12

24 Mar '12

On Fri, Mar 23, 2012 at 10:00:42AM -0400, Alexandre Bounine wrote: > This patch fixes drivers that use device_prep_slave_sg() and > device_prep_dma_cyclic() interfaces and have been missed during the initial > release of interface changing patches. Can you please send a separate patch for the ASoC (sound/soc) bits of this?

1 0

[alsa-devel] ALSA async callback re-enter and DEADLOCK.
by Павел Колодин 24 Mar '12

24 Mar '12

Hello. I've implemented a simple application using this example: http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html (see "async_callback()", line 390). My problem is that sometimes (after 1 or 20 mins of playing sound (unpredictable) everything just hangs - similarly to deadlock. I have found that callback is called WHILE another callback function is still executed (not returned yet). My debug code is like: void async_callback(snd_async_handler_t *ahandler) { std::cout << "?"; std::cout.flush(); // some code with snd_async_handler_get_callback_private(), snd_pcm_avail_update(), snd_pcm_writei() and so on // std::cout << "!"; std::cout.flush(); return; } Everything is ok when i see "?!?!?!...", but if "??" occurs than app goes into an eternal deadlock. I work with ALSA literally as in that example - settings (HW and SW) are the same. Soundcard is "Analog Devices AD1981", ALSA version is 1.0.19. I do fill buffer until there is a place for periods and hear 44100/16/2channels sound without any clicks and noises before "??" occurs. Sorry for poor english; from Siberia with love. Thank you!

1 0

[alsa-devel] [PATCH v6] sound/soc/lapis: add platform driver for ML7213
by Tomoya MORINAGA 23 Mar '12

23 Mar '12

This driver is for LAPIS Semiconductor ML7213 IOH I2S. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- v6 Updated for Mark's comments. - Delete global variable - Put request_irq back probe() and add conditon in interrupt handler(.alsa_ready). - Add braces - Use switch statement not if condition --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/lapis/Kconfig | 4 + sound/soc/lapis/Makefile | 4 + sound/soc/lapis/ioh_i2s.h | 39 + sound/soc/lapis/ml7213ioh-plat.c | 1450 ++++++++++++++++++++++++++++++++++++++ sound/soc/lapis/ml7213ioh-plat.h | 386 ++++++++++ 7 files changed, 1885 insertions(+), 0 deletions(-) create mode 100644 sound/soc/lapis/Kconfig create mode 100644 sound/soc/lapis/Makefile create mode 100644 sound/soc/lapis/ioh_i2s.h create mode 100644 sound/soc/lapis/ml7213ioh-plat.c create mode 100644 sound/soc/lapis/ml7213ioh-plat.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 91c9855..97d0d33 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -37,6 +37,7 @@ source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" source "sound/soc/jz4740/Kconfig" +source "sound/soc/lapis/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 2feaf37..1f266ee 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += lapis/ obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ diff --git a/sound/soc/lapis/Kconfig b/sound/soc/lapis/Kconfig new file mode 100644 index 0000000..551e385 --- /dev/null +++ b/sound/soc/lapis/Kconfig @@ -0,0 +1,4 @@ +config SND_SOC_ML7213_PLATFORM + tristate "ML7213 IOH ASoC platform driver" + help + This option enables support for the AC Link Controllers in ML7213 IOH SoC. diff --git a/sound/soc/lapis/Makefile b/sound/soc/lapis/Makefile new file mode 100644 index 0000000..aba1630 --- /dev/null +++ b/sound/soc/lapis/Makefile @@ -0,0 +1,4 @@ +# Platform +snd-soc-ml7213-plat-objs := ml7213ioh-plat.o + +obj-$(CONFIG_SND_SOC_ML7213_PLATFORM) += snd-soc-ml7213-plat.o diff --git a/sound/soc/lapis/ioh_i2s.h b/sound/soc/lapis/ioh_i2s.h new file mode 100644 index 0000000..9f19f70 --- /dev/null +++ b/sound/soc/lapis/ioh_i2s.h @@ -0,0 +1,39 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML7213_IOH_I2S +#define ML7213_IOH_I2S + +enum ioh_bclkfs { + ML7213IOH_BCLKFS0 = 0, + ML7213IOH_BCLKFS1, + ML7213IOH_BCLKFS2, + ML7213IOH_BCLKFS3, + ML7213IOH_BCLKFS4, + ML7213IOH_BCLKFS5, +}; + +enum ioh_mclkfs { + ML7213IOH_MCLKFS0 = 6, + ML7213IOH_MCLKFS1, + ML7213IOH_MCLKFS2, + ML7213IOH_MCLKFS3, + ML7213IOH_MCLKFS4, + ML7213IOH_MCLKFS5, +}; + +#endif diff --git a/sound/soc/lapis/ml7213ioh-plat.c b/sound/soc/lapis/ml7213ioh-plat.c new file mode 100644 index 0000000..ff55b5d --- /dev/null +++ b/sound/soc/lapis/ml7213ioh-plat.c @@ -0,0 +1,1450 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/scatterlist.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/pci.h> +#include <linux/clk.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ioh_i2s.h" +#include "ml7213ioh-plat.h" + +/* I2S HAL (Hardware Abstruction Layer) */ +static void ioh_i2s_reset(int ch, void *iobase) +{ + iowrite32(1 << ch, iobase + I2SSRST_OFFSET); + iowrite32(0, iobase + I2SSRST_OFFSET); +} + +static void ioh_i2s_enable_interrupts(int ch, int dir, + void *iobase) +{ + unsigned int intr_lines = 0; + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + intr_lines = 1 << (I2S_IMASK_RX_BIT_START + ch); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + intr_lines = 1 << (I2S_IMASK_TX_BIT_START + ch); + break; + } + + /* enable interrupts for specified channel */ + iowrite32(intr_lines, iobase + I2SIMASKCLR_OFFSET); +} + +static void ioh_i2s_disable_interrupts(int ch, int dir, void *iobase) +{ + unsigned int intr_lines; + + intr_lines = ioread32(iobase + I2SIMASK_OFFSET); + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + intr_lines |= 1 << (I2S_IMASK_RX_BIT_START + ch); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + intr_lines |= 1 << (I2S_IMASK_TX_BIT_START + ch); + break; + } + + /* Mask the specific interrupt bits */ + iowrite32(intr_lines, iobase + I2SIMASK_OFFSET); +} + +#define IOH_FIFO_CLR 0 +#define IOH_FIFO_RUN 1 + +static void ioh_i2s_ctrl_fifo(int ch, int dir, int ctrl, void *iobase) +{ + int offset = ch * 0x800; + u32 val; + u32 reg_addr = 0; + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + reg_addr = I2SFIFOCRX_OFFSET; + break; + case SNDRV_PCM_STREAM_PLAYBACK: + reg_addr = I2SFIFOCTX_OFFSET; + break; + } + + val = ioread32(iobase + reg_addr + offset); + if (ctrl) + val |= I2S_FIFO_TX_RUN; + else + val |= I2S_FIFO_TX_FCLR; + + iowrite32(val, iobase + reg_addr + offset); +} + +/* Clear interrupt status */ +static void ioh_i2s_clear_sts_ir(int ch, int dir, void *iobase) +{ + int offset = ch * 0x800; + u32 reg_addr = 0; + u32 val = 0; + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + reg_addr = I2SISTRX_OFFSET; + break; + case SNDRV_PCM_STREAM_PLAYBACK: + reg_addr = I2SISTTX_OFFSET; + break; + } + val = ioread32(iobase + reg_addr + offset) & 0xf; + if (val) + iowrite32(val, iobase + reg_addr + offset); +} + +static void ioh_i2s_clear_dma_mask(int ch, int dir, void *iobase) +{ + u32 val = 0; + u32 mask = 0; + u32 reg_addr = 0; + int offset = ch * 0x800; + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + reg_addr = I2SMSKRX_OFFSET; + mask = RX_BIT_DMAMSK; + break; + case SNDRV_PCM_STREAM_PLAYBACK: + reg_addr = I2SMSKTX_OFFSET; + mask = TX_BIT_DMAMSK; + break; + } + + val = ioread32(iobase + reg_addr + offset); + val &= ~mask; /* Enable Tx DMA Request */ + + iowrite32(val, iobase + reg_addr + offset); +} + +#define IOH_DIS_IRQ 0 +#define IOH_EN_IRQ 1 + +static void ioh_i2s_irq_ctrl(int ch, int dir, int ctrl, void *iobase) +{ + u32 val; + int offset = ch * 0x800; + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + val = ioread32(iobase + I2SMSKRX_OFFSET + offset); + if (ctrl) { + val &= ~RX_BIT_AFIMSK; /* Enable Almost empty IR */ + val &= ~RX_BIT_FIMSK; /* Enable Empty IR */ + } else { + val |= RX_BIT_AFIMSK; /* Disble Almost full IR */ + val |= RX_BIT_FIMSK; /* Disble full IR */ + } + iowrite32(val, iobase + I2SMSKRX_OFFSET + offset); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + val = ioread32(iobase + I2SMSKTX_OFFSET + offset); + if (ctrl) { + val &= ~TX_BIT_AEIMSK; /* Enable Almost empty IR */ + val &= ~TX_BIT_EIMSK; /* Enable Empty IR */ + } else { + val |= TX_BIT_AEIMSK; /* Disble Almost empty IR */ + val |= TX_BIT_EIMSK; /* Disble Empty IR */ + } + iowrite32(val, iobase + I2SMSKTX_OFFSET + offset); + break; + } +} + +/* Linux standard DMA functions */ +static bool ioh_dma_filter(struct dma_chan *chan, void *slave) +{ + struct pch_dma_slave *param = slave; + + if ((chan->chan_id == param->chan_id) && (param->dma_dev == + chan->device->dev)) { + chan->private = param; + return true; + } else { + return false; + } +} + +int ioh_request_dma_channel(int ch, struct ioh_i2s_data *i2s_data, + enum dma_data_direction dir) +{ + dma_cap_mask_t mask; + struct dma_chan *chan; + struct pci_dev *dma_dev; + struct ioh_i2s_dma *ioh = &i2s_data->dmadata[ch]; + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dma_dev = pci_get_bus_and_slot(2, PCI_DEVFN(0, 1)); /* Get DMA's dev + information */ + + switch (dir) { + case DMA_FROM_DEVICE: + ioh->param_rx.width = ioh->dma_rx_width; + ioh->param_rx.dma_dev = &dma_dev->dev; + ioh->param_rx.chan_id = ch * 2 + 1; /* ch Rx=1,3,...11 */ + ioh->param_rx.rx_reg = (dma_addr_t)(i2s_data->mapbase + + ch * 0x800 + I2SDRRXMIRROR_OFFSET); + chan = dma_request_channel(mask, ioh_dma_filter, + &ioh->param_rx); + if (chan == NULL) { + dev_err(i2s_data->dev, "Failed dma_request_channel for" + " I2S %d\n", ch); + return -ENOMEM; + } + ioh->chan_rx = chan; + break; + case DMA_TO_DEVICE: + ioh->param_tx.width = ioh->dma_tx_width; + ioh->param_tx.dma_dev = &dma_dev->dev; + ioh->param_tx.chan_id = ch * 2; /* DMA ch Tx=0,2,...10 */ + + ioh->param_tx.tx_reg = (dma_addr_t)(i2s_data->mapbase + + ch * 0x800 + I2SDRTXMIRROR_OFFSET); + + chan = dma_request_channel(mask, ioh_dma_filter, + &ioh->param_tx); + if (chan == NULL) { + dev_err(i2s_data->dev, "Failed dma_request_channel for" + " I2S %d\n", ch); + return -ENOMEM; + } + ioh->chan_tx = chan; + break; + default: + dev_err(i2s_data->dev, "Invalid direction (%d)\n", dir); + return -EINVAL; + } + + return 0; +} + +static void ioh_i2s_configure_i2s_regs(int ch, int dir, void *iobase) +{ + int offset = ch * 0x800; + + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + /* Rx register */ + iowrite32(I2S_AFULL_THRESH / 2, + iobase + I2SAFRX_OFFSET + offset); + iowrite32(ML7213_I2SAERX_DEFAULT, + iobase + I2SAERX_OFFSET + offset); + iowrite32(ML7213_I2SMSKRX_DEFAULT, + iobase + I2SMSKRX_OFFSET + offset); + iowrite32(ML7213_I2SISTRX_DEFAULT, + iobase + I2SISTRX_OFFSET + offset); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + iowrite32(0, iobase + I2SAFTX_OFFSET + offset); + iowrite32(I2S_AEMPTY_THRESH / 2, + iobase + I2SAETX_OFFSET + offset); + iowrite32(ML7213_I2SMSKTX_DEFAULT, + iobase + I2SMSKTX_OFFSET + offset); + iowrite32(ML7213_I2SISTTX_DEFAULT, + iobase + I2SISTTX_OFFSET + offset); + break; + } +} + +static void i2s_dma_rx_complete(void *arg) +{ + struct ioh_i2s_dma *ioh = (struct ioh_i2s_dma *)arg; + struct ml7213i2s_runtime_data *ioh_rtd = + ioh->rx_substream->runtime->private_data; + struct ioh_i2s_data *i2s_data = container_of(ioh, struct ioh_i2s_data, + dmadata[ioh->number]); + + pr_debug("%s in rx_cur_period=%d\n", __func__, ioh->rx_cur_period); + if (ioh_rtd->rx_stop) { + pr_debug("%s stopped. return.\n", __func__); + return; + } + + ioh->rx_cur_period++; + if (ioh->rx_cur_period == ioh->rx_buf_frags) + ioh->rx_cur_period = 0; + + async_tx_ack(ioh->desc_rx); + if (ioh->rx_substream) + snd_pcm_period_elapsed(ioh->rx_substream); + ioh_i2s_irq_ctrl(ioh->number, SNDRV_PCM_STREAM_CAPTURE, IOH_EN_IRQ, + i2s_data->iobase); +} + +static void i2s_dma_tx_complete(void *arg) +{ + struct ioh_i2s_dma *ioh = (struct ioh_i2s_dma *)arg; + struct ml7213i2s_runtime_data *ioh_rtd = + ioh->tx_substream->runtime->private_data; + struct ioh_i2s_data *i2s_data = container_of(ioh, struct ioh_i2s_data, + dmadata[ioh->number]); + + if (ioh_rtd->tx_stop) { + pr_debug("%s stopped. return.\n", __func__); + return; + } + + ioh->tx_cur_period++; + if (ioh->tx_cur_period == ioh->tx_buf_frags) + ioh->tx_cur_period = 0; + + async_tx_ack(ioh->desc_tx); + + if (ioh->tx_substream) + snd_pcm_period_elapsed(ioh->tx_substream); + + ioh_i2s_irq_ctrl(ioh->number, SNDRV_PCM_STREAM_PLAYBACK, IOH_EN_IRQ, + i2s_data->iobase); +} + +static void i2s_dma_rx_start(struct ioh_i2s_data *i2s_data, + struct ioh_i2s_dma *ioh, char *buff) +{ + struct dma_async_tx_descriptor *desc; + int rx_size; + int rx_num; + int i; + struct scatterlist *sg; + + rx_size = I2S_AFULL_THRESH * ioh->dma_rx_unit; + + /* The number of scatter list (Franction area is not used) */ + if (ioh->rx_period_bytes % rx_size) + /* rx_num = The number of scatter list */ + rx_num = ioh->rx_period_bytes / rx_size + 1; + else + rx_num = ioh->rx_period_bytes / rx_size; + + dev_dbg(i2s_data->dev, "%s: rx: scatter_num=%d scatter_size=%d\n", + __func__, rx_num, rx_size); + + ioh->sg_rx_p = + kzalloc(sizeof(struct scatterlist) * rx_num, GFP_ATOMIC); + + sg = ioh->sg_rx_p; + + sg_init_table(sg, rx_num); /* Initialize SG table */ + for (i = 0; i < rx_num; i++, sg++) { + sg_set_page(sg, virt_to_page(buff), rx_size, rx_size * i); + sg_dma_len(sg) = rx_size / ioh->dma_rx_unit; + sg_dma_address(sg) = ioh->rx_physical_addr + sg->offset; + } + ioh->rx_nent = rx_num; + + dma_sync_sg_for_device(i2s_data->dev, ioh->sg_rx_p, ioh->rx_nent, + DMA_FROM_DEVICE); + + sg = ioh->sg_rx_p; + + desc = ioh->chan_rx->device->device_prep_slave_sg(ioh->chan_rx, + sg, rx_num, DMA_FROM_DEVICE, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(i2s_data->dev, "%s:device_prep_slave_sg Failed\n", + __func__); + return; + } + ioh_i2s_irq_ctrl(ioh->number, SNDRV_PCM_STREAM_CAPTURE, IOH_DIS_IRQ, + i2s_data->iobase); + + ioh->desc_rx = desc; + + desc->callback = i2s_dma_rx_complete; + desc->callback_param = ioh; + + desc->tx_submit(desc); +} + +static void i2s_dma_tx_start(struct ioh_i2s_data *i2s_data, + struct ioh_i2s_dma *ioh, char *buff) +{ + struct dma_async_tx_descriptor *desc; + int tx_size; + int tx_num; + int i; + struct scatterlist *sg; + + tx_size = I2S_AEMPTY_THRESH * ioh->dma_tx_unit; + if (ioh->tx_period_bytes % tx_size) + /* tx_num = The number of scatter list */ + tx_num = ioh->tx_period_bytes / tx_size + 1; + else + tx_num = ioh->tx_period_bytes / tx_size; + + dev_dbg(i2s_data->dev, "%s: tx: scatter_num=%d scatter_size=%d\n", + __func__, tx_num, tx_size); + + ioh->sg_tx_p = + kzalloc(sizeof(struct scatterlist) * tx_num, GFP_ATOMIC); + + sg = ioh->sg_tx_p; + sg_init_table(sg, tx_num); /* Initialize SG table */ + + for (i = 0; i < tx_num; i++, sg++) { + sg_set_page(sg, virt_to_page(buff), tx_size, tx_size * i); + sg_dma_len(sg) = tx_size / ioh->dma_tx_unit; + sg_dma_address(sg) = ioh->tx_physical_addr + sg->offset; + } + ioh->tx_nent = tx_num; + + dma_sync_sg_for_device(i2s_data->dev, ioh->sg_tx_p, ioh->tx_nent, + DMA_TO_DEVICE); + + sg = ioh->sg_tx_p; + + desc = ioh->chan_tx->device->device_prep_slave_sg(ioh->chan_tx, + sg, tx_num, DMA_TO_DEVICE, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(i2s_data->dev, "%s:device_prep_slave_sg Failed\n", + __func__); + return; + } + + ioh_i2s_irq_ctrl(ioh->number, SNDRV_PCM_STREAM_PLAYBACK, IOH_DIS_IRQ, + i2s_data->iobase); + + ioh->desc_tx = desc; + + desc->callback = i2s_dma_tx_complete; + desc->callback_param = ioh; + desc->tx_submit(desc); +} + +static void ioh_release_dma_channel(int ch, int dir, struct ioh_i2s_dma *ioh, + struct ioh_i2s_data *i2s_data) +{ + switch (dir) { + case SNDRV_PCM_STREAM_CAPTURE: + dma_sync_sg_for_cpu(i2s_data->dev, ioh->sg_rx_p, ioh->rx_nent, + DMA_FROM_DEVICE); + + if (ioh->chan_rx) { + ioh->chan_rx->device->device_control(ioh->chan_rx, + DMA_TERMINATE_ALL, + 0); + dma_release_channel(ioh->chan_rx); + ioh->chan_rx = NULL; + } + + kfree(ioh->sg_rx_p); + ioh->rx_buf_dma = 0; + break; + case SNDRV_PCM_STREAM_PLAYBACK: + dma_sync_sg_for_cpu(i2s_data->dev, ioh->sg_tx_p, ioh->tx_nent, + DMA_TO_DEVICE); + + if (ioh->chan_tx) { + ioh->chan_tx->device->device_control(ioh->chan_tx, + DMA_TERMINATE_ALL, + 0); + dma_release_channel(ioh->chan_tx); + ioh->chan_tx = NULL; + } + kfree(ioh->sg_tx_p); + ioh->tx_buf_dma = 0; + break; + } +} + +void ioh_i2s_event(u32 idisp, int ch, struct ioh_i2s_data *i2s_data) +{ + unsigned long flags; + unsigned int status; + char *buff; + int offset = ch * 0x800; + struct ioh_i2s_dma *ioh = &i2s_data->dmadata[ch]; + struct ml7213i2s_runtime_data *ioh_rtd; + + spin_lock_irqsave(&i2s_data->tx_lock, flags); + if (idisp & BIT(ch + 16)) { + dev_dbg(i2s_data->dev, "Rx%d interrupt occures\n", ch); + ioh_rtd = ioh->rx_substream->runtime->private_data; + buff = ioh->rx_dma_addr + + ioh->rx_cur_period * ioh->rx_period_bytes; + status = ioread32(i2s_data->iobase + I2SISTRX_OFFSET + offset); + + if (status & I2S_RX_FINT) + dev_dbg(i2s_data->dev, "%s:I2S%d overrun occurs\n", + __func__, ch); + if (status & I2S_RX_AFINT) + if (!ioh_rtd->rx_stop) + i2s_dma_rx_start(i2s_data, ioh, buff); + + /*Clear the interrupt status */ + iowrite32(status, i2s_data->iobase + I2SISTRX_OFFSET + offset); + } + + if (idisp & BIT(ch)) { + dev_dbg(i2s_data->dev, "Tx%d interrupt occures\n", ch); + ioh_rtd = ioh->tx_substream->runtime->private_data; + buff = ioh->tx_dma_addr + + ioh->tx_cur_period * ioh->tx_period_bytes; + + status = ioread32(i2s_data->iobase + I2SISTTX_OFFSET + offset); + if (status & I2S_TX_EINT) + dev_dbg(i2s_data->dev, "%s:I2S%d under flow occurs\n", + __func__, ch); + if (status & I2S_TX_AEINT) + if (!ioh_rtd->tx_stop) + i2s_dma_tx_start(i2s_data, ioh, buff); + + /*Clear the interrupt status */ + iowrite32(status, i2s_data->iobase + I2SISTTX_OFFSET + offset); + } + spin_unlock_irqrestore(&i2s_data->tx_lock, flags); + return; +} + +static irqreturn_t ioh_i2s_irq(int irq, void *data) +{ + int i; + u32 idisp; + struct ioh_i2s_data *i2s_data = (struct ioh_i2s_data *)data; + + if (i2s_data->alsa_ready) { + idisp = ioread32(i2s_data->iobase + I2SIDISP_OFFSET); + for (i = 0; i < MAX_I2S_CH; i++) + ioh_i2s_event(idisp, i, i2s_data); + } + + return IRQ_HANDLED; +} + +static int snd_card_ml7213i2s_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct ml7213i2s_runtime_data *ioh_rtd; + + ioh_rtd = kzalloc(sizeof(*ioh_rtd), GFP_KERNEL); + if (ioh_rtd == NULL) + return -ENOMEM; + + runtime->private_data = ioh_rtd; + runtime->hw = ml7213i2s_pcm_hw; + + /* makes the infrastructure responsible for freeing dma */ + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &ml7213i2s_pcm_hw); + + spin_lock_init(&ioh_rtd->lock); + return 0; +} + +static int snd_card_ml7213i2s_close(struct snd_pcm_substream *substream) +{ + int ch; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ioh_i2s_dma *ioh = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + struct ioh_i2s_data *i2s_data = container_of(ioh, struct ioh_i2s_data, + dmadata[ioh->number]); + + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ch = ioh->number; + ioh_release_dma_channel(ch, SNDRV_PCM_STREAM_CAPTURE, ioh, + i2s_data); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + ch = ioh->number; + ioh_release_dma_channel(ch, SNDRV_PCM_STREAM_PLAYBACK, ioh, + i2s_data); + break; + } + + kfree(runtime->private_data); + + return 0; +} + +static int snd_card_ml7213i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ml7213i2s_runtime_data *ioh_rtd = runtime->private_data; + struct ioh_i2s_dma *ioh = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + struct ioh_i2s_data *i2s_data = container_of(ioh, struct ioh_i2s_data, + dmadata[ioh->number]); + int err; + int ch; + + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ch = ioh->number; + ioh_rtd->rx_dma = &i2s_data->dmadata[ch]; + err = ioh_request_dma_channel(ch, i2s_data, DMA_FROM_DEVICE); + if (err) + return err; + break; + case SNDRV_PCM_STREAM_PLAYBACK: + ch = ioh->number; + ioh_rtd->tx_dma = &i2s_data->dmadata[ch]; + err = ioh_request_dma_channel(ch, i2s_data, DMA_TO_DEVICE); + if (err) + return err; + break; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(hw_params); + i2s_data->alsa_ready = 1; + + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream) +{ + struct ml7213i2s_runtime_data *ioh_rtd; + ioh_rtd = substream->runtime->private_data; + + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ioh_rtd->rx_dma = NULL; + break; + case SNDRV_PCM_STREAM_PLAYBACK: + ioh_rtd->tx_dma = NULL; + break; + } + snd_pcm_set_runtime_buffer(substream, NULL); + + return snd_pcm_lib_free_pages(substream); +} + +static int snd_card_ml7213i2s_pcm_trigger + (struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ml7213i2s_runtime_data *ioh_rtd = runtime->private_data; + struct ioh_i2s_dma *ioh = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + struct ioh_i2s_data *i2s_data = container_of(ioh, struct ioh_i2s_data, + dmadata[ioh->number]); + int err = 0; + unsigned long flags; + int ch; + + spin_lock_irqsave(&ioh_rtd->lock, flags); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ioh_rtd->rx_stop = 0; + ch = ioh->number; + pr_debug("%s Capture start.(ch=%d)\n", __func__, ch); + ioh->rx_substream = substream; + + ioh_i2s_configure_i2s_regs(ch, SNDRV_PCM_STREAM_CAPTURE, + i2s_data->iobase); + ioh_i2s_ctrl_fifo(ch, SNDRV_PCM_STREAM_CAPTURE, + IOH_FIFO_CLR, i2s_data->iobase); + ioh_i2s_clear_sts_ir(ch, SNDRV_PCM_STREAM_CAPTURE, + i2s_data->iobase); + ioh_i2s_irq_ctrl(ch, SNDRV_PCM_STREAM_CAPTURE, + IOH_EN_IRQ, i2s_data->iobase); + ioh_i2s_ctrl_fifo(ch, SNDRV_PCM_STREAM_CAPTURE, + IOH_FIFO_RUN, i2s_data->iobase); + ioh_i2s_clear_dma_mask(ch, SNDRV_PCM_STREAM_CAPTURE, + i2s_data->iobase); + ioh_i2s_enable_interrupts(ch, SNDRV_PCM_STREAM_CAPTURE, + i2s_data->iobase); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + if (!ioh_rtd) + return -1; + if (!substream) + return -1; + ioh_rtd->tx_stop = 0; + ch = ioh->number; + pr_debug("%s Playback start.(ch=%d)\n", __func__, ch); + ioh->tx_substream = substream; + + ioh_i2s_configure_i2s_regs(ch, + SNDRV_PCM_STREAM_PLAYBACK, + i2s_data->iobase); + ioh_i2s_ctrl_fifo(ch, SNDRV_PCM_STREAM_PLAYBACK, + IOH_FIFO_CLR, i2s_data->iobase); + ioh_i2s_clear_sts_ir(ch, SNDRV_PCM_STREAM_PLAYBACK, + i2s_data->iobase); + ioh_i2s_irq_ctrl(ch, SNDRV_PCM_STREAM_PLAYBACK, + IOH_EN_IRQ, i2s_data->iobase); + ioh_i2s_ctrl_fifo(ch, SNDRV_PCM_STREAM_PLAYBACK, + IOH_FIFO_RUN, i2s_data->iobase); + ioh_i2s_clear_dma_mask(ch, SNDRV_PCM_STREAM_PLAYBACK, + i2s_data->iobase); + ioh_i2s_enable_interrupts(ch, SNDRV_PCM_STREAM_PLAYBACK, + i2s_data->iobase); + break; + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ioh_rtd->rx_stop = 1; + ch = ioh->number; + pr_debug("%s Capture stop.(ch=%d)\n", __func__, ch); + ioh_i2s_disable_interrupts(ch, SNDRV_PCM_STREAM_CAPTURE, + i2s_data->iobase); + ioh_i2s_irq_ctrl(ch, SNDRV_PCM_STREAM_CAPTURE, + IOH_DIS_IRQ, i2s_data->iobase); + ioh_i2s_clear_sts_ir(ch, SNDRV_PCM_STREAM_CAPTURE, + i2s_data->iobase); + ioh_i2s_ctrl_fifo(ch, SNDRV_PCM_STREAM_CAPTURE, + IOH_FIFO_CLR, i2s_data->iobase); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + ioh_rtd->tx_stop = 1; + ch = ioh->number; + pr_debug("%s Playback stop.(ch=%d)\n", __func__, ch); + ioh_i2s_disable_interrupts(ch, + SNDRV_PCM_STREAM_PLAYBACK, + i2s_data->iobase); + ioh_i2s_irq_ctrl(ch, SNDRV_PCM_STREAM_PLAYBACK, + IOH_DIS_IRQ, i2s_data->iobase); + ioh_i2s_clear_sts_ir(ch, SNDRV_PCM_STREAM_PLAYBACK, + i2s_data->iobase); + ioh_i2s_ctrl_fifo(ch, SNDRV_PCM_STREAM_PLAYBACK, + IOH_FIFO_CLR, i2s_data->iobase); + break; + } + break; + default: + err = -EINVAL; + break; + } + + spin_unlock_irqrestore(&ioh_rtd->lock, flags); + + return err; +} + +static int snd_card_ml7213i2s_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct ml7213i2s_runtime_data *ioh_rtd = runtime->private_data; + struct ioh_i2s_dma *ioh; + + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ioh = ioh_rtd->rx_dma; + ioh_rtd->rx_dma->rx_cur_period = 0; + ioh_rtd->rx_dma->rx_dma_addr = runtime->dma_area; + ioh_rtd->rx_dma->rx_physical_addr = runtime->dma_addr; + ioh_rtd->rx_dma->rx_buffer_bytes = + snd_pcm_lib_buffer_bytes(substream); + ioh_rtd->rx_dma->rx_period_bytes = + snd_pcm_lib_period_bytes(substream); + ioh_rtd->rx_dma->rx_buf_frags = + ioh_rtd->rx_dma->rx_buffer_bytes / + ioh_rtd->rx_dma->rx_period_bytes; + pr_debug("%s:Capture:buffbytes=%ld periodbytes=%ld frags=%d\n", + __func__, ioh_rtd->rx_dma->rx_buffer_bytes, + ioh_rtd->rx_dma->rx_period_bytes, + ioh_rtd->rx_dma->rx_buf_frags); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + ioh = ioh_rtd->tx_dma; + ioh_rtd->tx_dma->tx_cur_period = 0; + ioh_rtd->tx_dma->tx_dma_addr = runtime->dma_area; + ioh_rtd->tx_dma->tx_physical_addr = runtime->dma_addr; + ioh_rtd->tx_dma->tx_buffer_bytes = + snd_pcm_lib_buffer_bytes(substream); + ioh_rtd->tx_dma->tx_period_bytes = + snd_pcm_lib_period_bytes(substream); + ioh_rtd->tx_dma->tx_buf_frags = + ioh_rtd->tx_dma->tx_buffer_bytes / + ioh_rtd->tx_dma->tx_period_bytes; + pr_debug("%s:Playback:buffbytes=%ld periodbytes=%ld frags=%d\n", + __func__, ioh_rtd->tx_dma->tx_buffer_bytes, + ioh_rtd->tx_dma->tx_period_bytes, + ioh_rtd->tx_dma->tx_buf_frags); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t +snd_card_ml7213i2s_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct ml7213i2s_runtime_data *ioh_rtd = + substream->runtime->private_data; + unsigned long offset = 0; + + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + offset = ioh_rtd->rx_dma->rx_cur_period * + ioh_rtd->rx_dma->rx_period_bytes; + pr_debug("%s:Capture:offset=0x%lx\n", + __func__, offset); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + offset = ioh_rtd->tx_dma->tx_cur_period * + ioh_rtd->tx_dma->tx_period_bytes; + pr_debug("%s:Playback:offset=0x%lx\n", + __func__, offset); + break; + } + + return bytes_to_frames(substream->runtime, offset); +} + +static struct snd_pcm_ops snd_card_ml7213i2s_ops = { + .open = snd_card_ml7213i2s_open, + .close = snd_card_ml7213i2s_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_card_ml7213i2s_hw_params, + .hw_free = snd_card_ml7213i2s_hw_free, + .prepare = snd_card_ml7213i2s_pcm_prepare, + .trigger = snd_card_ml7213i2s_pcm_trigger, + .pointer = snd_card_ml7213i2s_pcm_pointer, +}; + +static u64 idma_mask = DMA_BIT_MASK(32); +static int ml7213ioh_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + snd_pcm_lib_preallocate_pages_for_all( + pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 0, 64 * 128 * 4 * 64); + + return ret; +} + +static void ml7213ioh_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + buf = &substream->dma_buffer; + if (!buf->area) + continue; + buf->area = NULL; + } +} + +static struct snd_soc_platform_driver ml7213ioh_soc_platform = { + .pcm_new = ml7213ioh_pcm_new, + .pcm_free = ml7213ioh_pcm_free, + .ops = &snd_card_ml7213i2s_ops, +}; + +/* DAI functions */ +static int ml7213i2s_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + int ch; + int byte; + unsigned int clk; + u32 i2sclkcnt; + u32 bclkfs = 0; + u32 mclkfs = 0; + u32 dabit; + int offset; + u32 i2scnt; + int dma_size; + int stereo; + struct ioh_i2s_data *i2s_data = snd_soc_dai_get_drvdata(dai); + void *iobase = i2s_data->iobase; + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_U8: + byte = 1; + bclkfs = IOH_BCLKFS_32FS; + dma_size = PCH_DMA_WIDTH_1_BYTE; + break; + case SNDRV_PCM_FORMAT_S16_LE: + byte = 2; + bclkfs = IOH_BCLKFS_32FS; + dma_size = PCH_DMA_WIDTH_2_BYTES; + break; + case SNDRV_PCM_FORMAT_S32_LE: + byte = 4; + bclkfs = IOH_BCLKFS_64FS; + dma_size = PCH_DMA_WIDTH_4_BYTES; + break; + default: + pr_err("%s: Failed not support format\n", __func__); + return -EINVAL; + } + + switch (byte) { + case 1: + dabit = 0; + break; + case 2: + dabit = 2 << I2S_DABIT_START; + break; + case 4: + dabit = 5 << I2S_DABIT_START; /* 24-bit transfer */ + break; + } + + stereo = params_channels(hw_params); + switch (substream->stream) { + case SNDRV_PCM_STREAM_CAPTURE: + ch = i2s_data->dmadata[substream->number].mapped_rx_ch; + i2s_data->dmadata[ch].dma_rx_unit = byte; + i2s_data->dmadata[ch].dma_rx_width = dma_size; + offset = ch * 0x800; + i2scnt = ioread32(iobase + offset + I2SCNTRX_OFFSET); + i2scnt &= ~(BIT(8) | BIT(9) | BIT(10)); + if (dabit) + i2scnt |= dabit; + + if (stereo == 1) + i2scnt |= ML7213I2S_RX_DLY | ML7213I2S_RX_TEL; + else + i2scnt &= ~(ML7213I2S_RX_DLY | ML7213I2S_RX_TEL); + + iowrite32(i2scnt, iobase + offset + I2SCNTRX_OFFSET); + break; + case SNDRV_PCM_STREAM_PLAYBACK: + ch = i2s_data->dmadata[substream->number].mapped_tx_ch; + i2s_data->dmadata[ch].dma_tx_unit = byte; + i2s_data->dmadata[ch].dma_tx_width = dma_size; + offset = ch * 0x800; + i2scnt = ioread32(iobase + offset + I2SCNTTX_OFFSET); + i2scnt &= ~(BIT(8) | BIT(9) | BIT(10)); + if (dabit) + i2scnt |= dabit; + + if (stereo == 1) + i2scnt |= ML7213I2S_TX_DLY | ML7213I2S_TX_TEL; + else + i2scnt &= ~(ML7213I2S_TX_DLY | ML7213I2S_TX_TEL); + + iowrite32(i2scnt, iobase + offset + I2SCNTTX_OFFSET); + break; + default: + return -EINVAL; + } + + switch (params_rate(hw_params)) { + case 16000: + case 32000: + case 48000: + clk = 12288000; + break; + default: + pr_err("%s: Failed not support rate\n", __func__); + return -EINVAL; + } + + switch (clk / params_rate(hw_params)) { + case 64: + mclkfs = 0; + break; + case 128: + mclkfs = 1; + break; + case 192: + mclkfs = 2; + break; + case 256: + mclkfs = 3; + break; + case 384: + mclkfs = 4; + break; + case 512: + mclkfs = 5; + break; + case 768: + mclkfs = 6; + break; + case 1024: + mclkfs = 7; + break; + default: + pr_err("%s: Failed not support mclkfs\n", __func__); + return -EINVAL; + } + + pr_debug("%s: bclkfs=%d mclkfs=%dFS dabit=%d\n", __func__, + bclkfs, clk / params_rate(hw_params), dabit); + i2sclkcnt = ioread32(i2s_data->iobase + I2SCLKCNT0_OFFSET + 0x10 * ch); + i2sclkcnt &= ~I2S_MCLKFS_MASK; + i2sclkcnt &= ~I2S_BCLKFS_MASK; + i2sclkcnt |= bclkfs << I2SCLKCNT_BCLKFS_OFFSET; + i2sclkcnt |= mclkfs << I2SCLKCNT_MCLKFS_OFFSET; + + i2sclkcnt &= ~ML7213I2S_LRCLK_FMT; + if (stereo == 1) + i2sclkcnt |= ML7213I2S_LRCLK_FMT_LONG; + else + i2sclkcnt |= ML7213I2S_LRCLK_FMT_I2S; + + iowrite32(i2sclkcnt, iobase + I2SCLKCNT0_OFFSET + 0x10 * ch); + + snd_soc_dai_set_dma_data(dai, substream, &i2s_data->dmadata[ch]); + + return 0; +} + +static int ml7213i2s_dai_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + u32 cmn_reg[MAX_I2S_CH]; + u32 tx_reg[MAX_I2S_CH]; + u32 rx_reg[MAX_I2S_CH]; + int i; + int offset = 0; + struct ioh_i2s_data *i2s_data = snd_soc_dai_get_drvdata(dai); + void *iobase = i2s_data->iobase; + + /* set master/slave audio interface */ + for (i = 0; i < MAX_I2S_CH; i++, offset = i * 0x800) { + cmn_reg[i] = ioread32(iobase + I2SCLKCNT0_OFFSET + 0x10 * i); + tx_reg[i] = ioread32(iobase + offset + I2SCNTTX_OFFSET); + rx_reg[i] = ioread32(iobase + offset + I2SCNTRX_OFFSET); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cmn_reg[i] |= I2SCLKCNT_MSSEL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cmn_reg[i] &= ~I2SCLKCNT_MSSEL; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + tx_reg[i] &= ~(ML7213I2S_TX_MSB_LSB | ML7213I2S_TX_AFT); + rx_reg[i] &= ~(ML7213I2S_RX_MSB_LSB | ML7213I2S_RX_AFT); + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cmn_reg[i] &= ~ML7213I2S_BCLKPOL; + break; + case SND_SOC_DAIFMT_IB_NF: + cmn_reg[i] |= ML7213I2S_BCLKPOL; + break; + default: + return -EINVAL; + } + + iowrite32(cmn_reg[i], iobase + I2SCLKCNT0_OFFSET + 0x10 * i); + iowrite32(tx_reg[i], iobase + offset + I2SCNTTX_OFFSET); + iowrite32(rx_reg[i], iobase + offset + I2SCNTRX_OFFSET); + } + + return 0; +} + +static int ml7213i2s_dai_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + u32 reg[MAX_I2S_CH]; + struct ioh_i2s_data *i2s_data = snd_soc_dai_get_drvdata(dai); + void *iobase = i2s_data->iobase; + int i; + + for (i = 0; i < MAX_I2S_CH; i++) { + reg[i] = ioread32(iobase + I2SCLKCNT0_OFFSET + 0x10 * i); + switch (clk_id) { + case IOH_MASTERCLKSEL_MCLK: + reg[i] &= ~ML7213I2S_MASTER_CLK_SEL; + break; + case IOH_MASTERCLKSEL_MLBCLK: + reg[i] |= ML7213I2S_MASTER_CLK_SEL; + break; + default: + return -EINVAL; + } + iowrite32(reg[i], iobase + I2SCLKCNT0_OFFSET + 0x10 * i); + } + + return 0; +} + +static int ml7213i2s_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct ioh_i2s_data *i2s_data = snd_soc_dai_get_drvdata(dai); + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > MAX_I2S_CH) || (rx_num > MAX_I2S_CH)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < MAX_I2S_CH) && + (!(tx_mapped & (1 << slot)))) { + i2s_data->dmadata[i].mapped_tx_ch = slot; + tx_mapped |= 1 << slot; + } else { + return -EINVAL; + } + } + + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < MAX_I2S_CH) && + (!(rx_mapped & (1 << slot)))) { + i2s_data->dmadata[i].mapped_rx_ch = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + +#ifdef CONFIG_PM +static int ml7213i2s_soc_suspend(struct snd_soc_dai *cpu_dai) +{ + int i; + void *iobase; + struct ioh_i2s_pm_ch_reg *save; + struct ioh_i2s_pm_ch_reg_cmn *save_cmn; + int offset; + struct ioh_i2s_data *i2s_data = snd_soc_dai_get_drvdata(cpu_dai); + + iobase = i2s_data->iobase; + for (i = 0, offset = 0; i < MAX_I2S_CH; i++, offset = i * 0x800) { + save = &i2s_data->ch_reg_save[i]; + + save->i2sdrtx = ioread32(iobase + offset + I2SDRTX_OFFSET); + save->i2scnttx = ioread32(iobase + offset + I2SCNTTX_OFFSET); + save->i2sfifoctx = + ioread32(iobase + offset + I2SFIFOCTX_OFFSET); + save->i2saftx = ioread32(iobase + offset + I2SAFTX_OFFSET); + save->i2saetx = ioread32(iobase + offset + I2SAETX_OFFSET); + save->i2smsktx = ioread32(iobase + offset + I2SMSKTX_OFFSET); + save->i2sisttx = ioread32(iobase + offset + I2SISTTX_OFFSET); + + save->i2scntrx = ioread32(iobase + offset + I2SCNTRX_OFFSET); + save->i2sfifocrx = + ioread32(iobase + offset + I2SFIFOCRX_OFFSET); + save->i2safrx = ioread32(iobase + offset + I2SAFRX_OFFSET); + save->i2saerx = ioread32(iobase + offset + I2SAERX_OFFSET); + save->i2smskrx = ioread32(iobase + offset + I2SMSKRX_OFFSET); + save->i2sistrx = ioread32(iobase + offset + I2SISTRX_OFFSET); + } + + save_cmn = &i2s_data->cmn_reg_save; + for (i = 0; i < MAX_I2S_CH; i++) { + save_cmn->i2sclkcnt[i] = + ioread32(i2s_data->iobase + I2SCLKCNT0_OFFSET + 0x10 * i); + } + save_cmn->i2simask = ioread32(i2s_data->iobase + I2SIMASK_OFFSET); + + return 0; +} + +static int ml7213i2s_soc_resume(struct snd_soc_dai *cpu_dai) +{ + int i; + void *iobase; + struct ioh_i2s_pm_ch_reg *save; + int offset; + struct ioh_i2s_data *i2s_data = snd_soc_dai_get_drvdata(cpu_dai); + + iobase = i2s_data->iobase; + save = &i2s_data->ch_reg_save[0]; + for (i = 0, offset = 0; i < MAX_I2S_CH; i++, offset = i * 0x800) { + iowrite32(save->i2sdrtx, iobase + offset + I2SDRTX_OFFSET); + iowrite32(save->i2scnttx, iobase + offset + I2SCNTTX_OFFSET); + iowrite32(save->i2sfifoctx, + iobase + offset + I2SFIFOCTX_OFFSET); + iowrite32(save->i2saftx, iobase + offset + I2SAFTX_OFFSET); + iowrite32(save->i2saetx, iobase + offset + I2SAETX_OFFSET); + iowrite32(save->i2smsktx, iobase + offset + I2SMSKTX_OFFSET); + iowrite32(save->i2sisttx, iobase + offset + I2SISTTX_OFFSET); + + iowrite32(save->i2scntrx, iobase + offset + I2SCNTRX_OFFSET); + iowrite32(save->i2sfifocrx, + iobase + offset + I2SFIFOCRX_OFFSET); + iowrite32(save->i2safrx, iobase + offset + I2SAFRX_OFFSET); + iowrite32(save->i2saerx, iobase + offset + I2SAERX_OFFSET); + iowrite32(save->i2smskrx, iobase + offset + I2SMSKRX_OFFSET); + iowrite32(save->i2sistrx, iobase + offset + I2SISTRX_OFFSET); + } + + for (i = 0; i < MAX_I2S_CH; i++) { + iowrite32(i2s_data->cmn_reg_save.i2sclkcnt[i], + i2s_data->iobase + I2SCLKCNT0_OFFSET + 0x10 * i); + } + + iowrite32(i2s_data->cmn_reg_save.i2simask, + i2s_data->iobase + I2SIMASK_OFFSET); + + return 0; +} +#else +#define ml7213i2s_soc_suspend NULL +#define ml7213i2s_soc_resume NULL +#endif + +static int ml7213i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct ioh_i2s_data *i2s_data = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, i2s_data); + + return 0; +} + +static const struct snd_soc_dai_ops ml7213i2s_dai_ops = { + .hw_params = ml7213i2s_dai_hw_params, + .set_fmt = ml7213i2s_dai_set_dai_fmt, + .set_sysclk = ml7213i2s_dai_set_dai_sysclk, + .set_channel_map = ml7213i2s_dai_set_channel_map, +}; + +static struct snd_soc_dai_driver ml7213i2s_dai_data = { + .playback = { + .channels_min = USE_CHANNELS_MIN, + .channels_max = USE_CHANNELS_MAX, + .rates = ML7213_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = USE_CHANNELS_MIN, + .channels_max = USE_CHANNELS_MAX, + .rates = ML7213_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE, + }, + .probe = ml7213i2s_dai_probe, + .ops = &ml7213i2s_dai_ops, + .suspend = ml7213i2s_soc_suspend, + .resume = ml7213i2s_soc_resume, +}; + +/* PCI functions */ +DEFINE_PCI_DEVICE_TABLE(ioh_pci_tbl) = { + { + .vendor = PCI_VENDOR_ID_ROHM, + .device = PCI_DEVICE_ID_ML7213_I2S, + .subvendor = PCI_ANY_ID, + .subdevice = PCI_ANY_ID, + }, + {0,} +}; + +static int ioh_i2s_pci_probe(struct pci_dev *pdev, + const struct pci_device_id *id) +{ + int rv = 0; + void __iomem *tbl; + unsigned int mapbase; + int i; + static struct ioh_i2s_data *i2s_data; + + rv = pci_enable_device(pdev); + if (rv) + goto enable_device; + + tbl = pci_iomap(pdev, 1, 0); + if (!tbl) { + rv = -ENOMEM; + printk(KERN_ERR "pci_iomap failed\n"); + goto out_ipmap; + } + + mapbase = pci_resource_start(pdev, 1); + if (!mapbase) { + rv = -ENOMEM; + printk(KERN_ERR "pci_resource_start failed\n"); + goto out_pci_resource; + } + + i2s_data = devm_kzalloc(&pdev->dev, sizeof(*i2s_data), GFP_KERNEL); + if (!i2s_data) { + dev_err(&pdev->dev, "Can't allocate i2s_data\n"); + rv = -ENOMEM; + goto out_kzalloc_data; + } + + i2s_data->dev = &pdev->dev; + i2s_data->pdev = pdev; + i2s_data->iobase = tbl; + i2s_data->mapbase = mapbase; + + spin_lock_init(&i2s_data->tx_lock); + dev_set_drvdata(&pdev->dev, i2s_data); + dev_set_name(&pdev->dev, "%s", "ml7213ioh"); + rv = snd_soc_register_dai(&pdev->dev, &ml7213i2s_dai_data); + if (rv < 0) { + printk(KERN_ERR "Failed to snd_soc_register_dai\n"); + goto out_register_dai; + } + + dev_set_name(&pdev->dev, "%s", "ml7213-i2s-audio"); + rv = snd_soc_register_platform(&pdev->dev, &ml7213ioh_soc_platform); + if (rv < 0) { + printk(KERN_ERR "Failed to snd_soc_register_platform\n"); + goto out_soc_register_plat; + } + + rv = request_irq(i2s_data->pdev->irq, ioh_i2s_irq, IRQF_SHARED, + "ml7213_ioh", i2s_data); + if (rv != 0) + printk(KERN_ERR "Failed to allocate irq\n"); + + for (i = 0; i < MAX_I2S_CH; i++) + i2s_data->dmadata[i].number = i; + + return 0; + +out_soc_register_plat: + snd_soc_unregister_dai(&pdev->dev); +out_register_dai: +out_kzalloc_data: +out_pci_resource: + pci_iounmap(pdev, i2s_data->iobase); +out_ipmap: + pci_disable_device(pdev); +enable_device: + + return rv; +} + +static void ioh_i2s_pci_remove(struct pci_dev *pdev) +{ + int i; + struct ioh_i2s_data *i2s_data = dev_get_drvdata(&pdev->dev); + + free_irq(pdev->irq, i2s_data); + + for (i = 0; i < MAX_I2S_CH; i++) + ioh_i2s_reset(i, i2s_data->iobase); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_dai(&pdev->dev); + pci_iounmap(pdev, i2s_data->iobase); + pci_disable_device(pdev); +} + +static int ioh_i2s_pci_suspend(struct pci_dev *pdev, pm_message_t state) +{ + int ret; + + ret = pci_save_state(pdev); + if (ret) { + dev_err(&pdev->dev, + " %s -pci_save_state returns %d\n", __func__, ret); + return ret; + } + pci_enable_wake(pdev, PCI_D3hot, 0); + pci_disable_device(pdev); + pci_set_power_state(pdev, pci_choose_state(pdev, state)); + + return 0; +} + +static int ioh_i2s_pci_resume(struct pci_dev *pdev) +{ + int ret; + + pci_set_power_state(pdev, PCI_D0); + pci_restore_state(pdev); + ret = pci_enable_device(pdev); + if (ret) { + dev_err(&pdev->dev, + "%s-pci_enable_device failed(ret=%d) ", __func__, ret); + return ret; + } + + pci_enable_wake(pdev, PCI_D3hot, 0); + + return 0; +} + +static struct pci_driver ioh_i2s_driver = { + .name = DRV_NAME, + .probe = ioh_i2s_pci_probe, + .remove = __devexit_p(ioh_i2s_pci_remove), + .id_table = ioh_pci_tbl, +#ifdef CONFIG_PM + .suspend = ioh_i2s_pci_suspend, + .resume = ioh_i2s_pci_resume, +#endif +}; + +static int __init ioh_plat_init(void) +{ + return pci_register_driver(&ioh_i2s_driver); +} + +static void __exit ioh_i2s_cleanup(void) +{ + pci_unregister_driver(&ioh_i2s_driver); +} + +module_init(ioh_plat_init); +module_exit(ioh_i2s_cleanup); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML7213 IOH ALSA SoC platform driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/lapis/ml7213ioh-plat.h b/sound/soc/lapis/ml7213ioh-plat.h new file mode 100644 index 0000000..dcd393c --- /dev/null +++ b/sound/soc/lapis/ml7213ioh-plat.h @@ -0,0 +1,386 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML7213IOH_PLAT_H +#define ML7213IOH_PLAT_H + +#include <linux/interrupt.h> +#include <linux/pch_dma.h> + +#define DRV_NAME "ml7213ioh-i2s-pci" +#define PCI_VENDOR_ID_ROHM 0X10DB +#define PCI_DEVICE_ID_ML7213_I2S 0X8033 + +#define I2SCLKCNT_MSSEL BIT(0) +#define ML7213I2S_BCLKPOL BIT(1) +#define ML7213I2S_LRCLK_FMT (BIT(4) | BIT(5)) +#define ML7213I2S_LRCLK_FMT_I2S BIT(4) +#define ML7213I2S_LRCLK_FMT_LONG BIT(5) +#define ML7213I2S_LRCLK_FMT_SHORT (BIT(4) | BIT(5)) + +#define ML7213I2S_TX_TEL BIT(0) +#define ML7213I2S_TX_DLY BIT(12) +#define ML7213I2S_TX_MSB_LSB BIT(13) +#define ML7213I2S_TX_LR_POL BIT(14) +#define ML7213I2S_TX_AFT BIT(15) +#define ML7213I2S_RX_TEL BIT(0) +#define ML7213I2S_RX_DLY BIT(12) +#define ML7213I2S_RX_MSB_LSB BIT(13) +#define ML7213I2S_RX_LR_POL BIT(14) +#define ML7213I2S_RX_AFT BIT(15) +#define ML7213I2S_MASTER_CLK_SEL BIT(2) + +#define ML7213_I2S_RATES \ + (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) + +/* ioh_bclkfs_t */ +#define IOH_BCLKFS_8FS 0 +#define IOH_BCLKFS_16FS 1 +#define IOH_BCLKFS_32FS 2 +#define IOH_BCLKFS_64FS 3 + +#define I2SCLKCNT_MCLKFS_OFFSET (8) +#define I2SCLKCNT_BCLKFS_OFFSET (12) + +#define I2SCLKCNT0_OFFSET 0x3000 +#define I2SCLKCNT1_OFFSET 0x3010 +#define I2SCLKCNT2_OFFSET 0x3020 +#define I2SCLKCNT3_OFFSET 0x3030 +#define I2SCLKCNT4_OFFSET 0x3040 +#define I2SCLKCNT5_OFFSET 0x3050 +#define I2SISTATUS_OFFSET 0x3080 +#define I2SIDISP_OFFSET 0x3084 +#define I2SIMASK_OFFSET 0x3088 +#define I2SIMASKCLR_OFFSET 0x308C +#define I2SSRST_OFFSET 0x3FFC +#define I2SDRTX_OFFSET 0x0 +#define I2SCNTTX_OFFSET 0x4 +#define I2SFIFOCTX_OFFSET 0x8 +#define I2SAFTX_OFFSET 0xC +#define I2SAETX_OFFSET 0x10 +#define I2SMSKTX_OFFSET 0x14 +#define I2SISTTX_OFFSET 0x18 +#define I2SMONTX_OFFSET 0x1C +#define I2SDRRX_OFFSET 0x20 +#define I2SCNTRX_OFFSET 0x24 +#define I2SFIFOCRX_OFFSET 0x28 +#define I2SAFRX_OFFSET 0x2C +#define I2SAERX_OFFSET 0x30 +#define I2SMSKRX_OFFSET 0x34 +#define I2SISTRX_OFFSET 0x38 +#define I2SMONRX_OFFSET 0x3C +#define FIRST_TX_OFFSET 0x0 +#define FIRST_RX_OFFSET 0x0 + +#define I2SDRTXMIRROR_OFFSET 0x100 +#define I2SDRRXMIRROR_OFFSET 0x400 + +#define I2S_ALL_INTERRUPT_BITS 0x3F003F +#define I2S_IDISP_BITS 0x3F003F +#define I2S_IDISP_TX_BITS 0x00003F +#define I2S_IDISP_RX_BITS 0x3F0000 +#define TX_BIT_FIMSK 0x1 /*Fifo full interrupt mask bit*/ +#define TX_BIT_AFIMSK 0x2 /*Fifo Almost full interrupt mask bit*/ +#define TX_BIT_EIMSK 0x4 /*Fifo empty interrupt mask bit*/ +#define TX_BIT_AEIMSK 0x8 /*Fifo Almost empty interrupt mask bit*/ +#define TX_BIT_DMAMSK 0x10 /*Masks DMA*/ +#define TX_BIT_DMATC 0x100 +#define I2S_TX_ALL_INTR_MASK_BITS (TX_BIT_FIMSK | TX_BIT_AFIMSK | TX_BIT_EIMSK \ + | TX_BIT_AEIMSK) +#define I2S_TX_NORMAL_INTR_MASK_BITS (TX_BIT_FIMSK | TX_BIT_AFIMSK) +#define RX_BIT_FIMSK 0x1 /*Fifo full interrupt mask bit*/ +#define RX_BIT_AFIMSK 0x2 /*Fifo Almost full interrupt mask bit*/ +#define RX_BIT_EIMSK 0x4 /*Fifo empty interrupt mask bit*/ +#define RX_BIT_AEIMSK 0x8 /*Fifo Almost empty interrupt mask bit*/ +#define RX_BIT_DMAMSK 0x10 /*Masks DMA*/ +#define RX_BIT_DMATC 0x100 +#define I2S_RX_ALL_INTR_MASK_BITS (RX_BIT_FIMSK | RX_BIT_AFIMSK | RX_BIT_EIMSK \ + | RX_BIT_AEIMSK) +#define I2S_RX_NORMAL_INTR_MASK_BITS (RX_BIT_EIMSK | RX_BIT_AEIMSK) +#define I2S_TX_FINT 0x1 /*Full Interrupt*/ +#define I2S_TX_AFINT 0x2 /*Almost full interrupt*/ +#define I2S_TX_EINT 0x4 /*Empty interrupt*/ +#define I2S_TX_AEINT 0x8 /*Almost empty interrupt*/ +#define I2S_RX_FINT 0x1 /*Full Interrupt*/ +#define I2S_RX_AFINT 0x2 /*Almost full interrupt*/ +#define I2S_RX_EINT 0x4 /*Empty interrupt*/ +#define I2S_RX_AEINT 0x8 /*Almost empty interrupt*/ + +#define I2S_FIFO_TX_FCLR BIT(0) +#define I2S_FIFO_TX_RUN BIT(4) +#define I2S_FIFO_RX_FCLR BIT(0) +#define I2S_FIFO_RX_RUN BIT(4) + +#define FIFO_CTRL_BIT_TX_RUN 0x10 +#define FIFO_CTRL_BIT_RX_RUN 0x10 +#define I2S_CNT_BIT_TEL 0x1 +#define I2S_IMASK_TX_BIT_START 0 +#define I2S_IMASK_RX_BIT_START 16 + +#define I2S_DABIT_START 8 + +#define I2S_MCLKFS_MASK (BIT(8) | BIT(9) | BIT(10)) +#define I2S_BCLKFS_MASK (BIT(12) | BIT(13)) + +/* DMA processing */ +#define PERIOD_POS_MAX I2S_DMA_SG_NUM +#define PERIOD_LEN (I2S_AFULL_THRESH * PERIOD_POS_MAX) + +#define SUPPORT_FORMAT (SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) +#define MAX_PERIOD_SIZE (PERIOD_LEN * 4) + +#define USE_CHANNELS_MIN 1 +#define USE_CHANNELS_MAX 2 +#define MAX_I2S_CH 6 /*I2S0 ~ I2S5*/ +#define USE_PERIODS_MIN (I2S_DMA_SG_MAX) +#define USE_PERIODS_MAX (I2S_DMA_SG_MAX) + +#define I2S_AEMPTY_THRESH 64 /* Almost Empty Threshold */ +#define I2S_AFULL_THRESH 64 /* Almost Full Threshold */ + +#define I2S_DMA_SG_NUM (128) +#define I2S_DMA_SG_MAX (64) + +#define IOH_MSSEL_MASTER 1 + +#define ML7213_I2SAERX_DEFAULT 0x1f +#define ML7213_I2SMSKRX_DEFAULT 0x1f +#define ML7213_I2SISTRX_DEFAULT 0xC + +#define ML7213_I2SMSKTX_DEFAULT 0x1f +#define ML7213_I2SISTTX_DEFAULT 0xC + +enum ioh_i2s_fifo_type { + IOH_FIFO_32 = 4, + IOH_FIFO_16 = 2, + IOH_FIFO_8 = 1, +}; + +enum ioh_i2s_status { + IOH_EOK = 0, + IOH_EDONE = 1, + IOH_EUNDERRUN = 2, + IOH_EOVERRUN = 3, + IOH_EFRAMESYNC = 4, +}; + +enum ioh_bclkpol_t { + ioh_BCLKPOL_FALLING = 0, + ioh_BCLKPOL_RISING, +}; + +enum ioh_masterclksel_t { + IOH_MASTERCLKSEL_MCLK = 0, + IOH_MASTERCLKSEL_MLBCLK, +}; + +enum ioh_lrckfmt_t { + IOH_LRCLKFMT_I2S = 1, + IOH_LRCLKFMT_LONGFRAME, + IOH_LRCLKFMT_SHORTFRAME, +}; + +enum ioh_mclkfs_t { + IOH_MCLKFS_64FS = 0, + IOH_MCLKFS_128FS, + IOH_MCLKFS_192FS, + IOH_MCLKFS_256FS, + IOH_MCLKFS_384FS, + IOH_MCLKFS_512FS, + IOH_MCLKFS_768FS, + IOH_MCLKFS_1024FS, +}; + +enum ioh_dlyoff_t { + IOH_DLYOFF_DLY_ON = 0, /* date delat on */ + IOH_DLYOFF_DLY_OFF, /* date delat off */ +}; + +enum ioh_lrpol_t { + IOH_LRPOL_NO_INVERT = 0, /* Low of LRCLK is L data. + High of LRCLK is R data. */ + IOH_LRPOL_INVERT, /* Low of LRCLK is R data. + High of LRCLK is L data. */ +}; + +enum ioh_aft_t { + IOH_AFR_FRONT = 0, + IOH_AFR_BACK, +}; + +struct ml7213i2s_runtime_data { + spinlock_t lock; + int tx_stop; + int rx_stop; + struct ioh_i2s_dma *rx_dma; + struct ioh_i2s_dma *tx_dma; +}; + +struct ioh_i2s_pm_ch_reg { + u32 i2sdrtx; /* Tx: data register */ + u32 i2scnttx; /* Tx: control register */ + u32 i2sfifoctx; /* Tx: FIFO control register */ + u32 i2saftx; /* Tx: almost full threshold setting */ + u32 i2saetx; /* Tx: almost empty threshold setting */ + u32 i2smsktx; /* Tx: interrupt mask settings */ + u32 i2sisttx; /* Tx: for acknowledging interrupts */ + u32 i2scntrx; /* Rx: control register */ + u32 i2sfifocrx; /* Rx: FIFO control register */ + u32 i2safrx; /* Rx: almost full threshold setting */ + u32 i2saerx; /* Rx: almost empty threshold setting */ + u32 i2smskrx; /* Rx: interrupt mask settings */ + u32 i2sistrx; /* Rx: for acknowledging interrupts */ +}; + +struct ioh_i2s_pm_ch_reg_cmn { + u32 i2sclkcnt[MAX_I2S_CH]; /*clock control register(ch0~5) */ + u32 i2simask; /*interrupt mask */ +}; + +struct ioh_i2s_dma { + /* Transmit side DMA */ + struct scatterlist *sg_tx_p; + struct scatterlist *sg_rx_p; + + int tx_num; /* The number of sent sg */ + int rx_num; /* The number of sent sg */ + + struct dma_chan *chan_tx; + struct dma_chan *chan_rx; + + int rx_nent; /* The number of rx scatter list */ + int tx_nent; /* The number of tx scatter list */ + + struct dma_async_tx_descriptor *desc_tx; + struct dma_async_tx_descriptor *desc_rx; + + dma_addr_t tx_buf_dma; + dma_addr_t rx_buf_dma; + + struct pch_dma_slave param_tx; + struct pch_dma_slave param_rx; + + int dma_tx_unit; /* 1Byte of 2Byte or 4Byte */ + int dma_rx_unit; /* 1Byte of 2Byte or 4Byte */ + int dma_tx_width; + int dma_rx_width; + + struct snd_pcm_substream *tx_substream; + struct snd_pcm_substream *rx_substream; + + int number; + + int mapped_tx_ch; + int mapped_rx_ch; + + unsigned char *rx_dma_addr; + dma_addr_t rx_physical_addr; + unsigned long rx_buffer_bytes; + unsigned long rx_period_bytes; + + unsigned char *tx_dma_addr; + dma_addr_t tx_physical_addr; + unsigned long tx_buffer_bytes; + unsigned long tx_period_bytes; + + + int rx_buf_frags; + int tx_buf_frags; + + int tx_cur_period; + int rx_cur_period; +}; + +struct ioh_i2s_data { + struct device *dev; + void *iobase; + unsigned int mapbase; + spinlock_t tx_lock; + struct ioh_i2s_pm_ch_reg_cmn cmn_reg_save; + struct ioh_i2s_pm_ch_reg ch_reg_save[MAX_I2S_CH]; + struct pci_dev *pdev; + struct ioh_i2s_dma dmadata[MAX_I2S_CH]; + int alsa_ready; +}; + +#if 0 +struct ml7213i2s_dai { + struct snd_soc_dai_driver dai; + struct device *dev; + void *iobase; + u32 freq; +}; +#endif +struct ioh_i2s_config_common_reg { + u32 i2sclkcnt; /*clock control register(ch0~5) */ + u32 i2sistatus; /*interrupt status */ + u32 i2sidisp; /*active interrupts */ + u32 i2simask; /*interrupt mask */ + u32 i2simaskclr; /*interrupt mask clear */ +}; + +struct ioh_i2s_config_tx_reg { + u32 i2sdrtx; /*data register */ + u32 i2scnttx; /*control register */ + u32 i2sfifoctx; /*FIFO control register */ + u32 i2saftx; /*almost full threshold setting */ + u32 i2saetx; /*almost empty threshold setting */ + u32 i2smsktx; /*interrupt mask settings */ + u32 i2sisttx; /*for acknowledging interrupts */ + u32 i2smontx; /*monitor register */ +}; + +struct ioh_i2s_config_rx_reg { + u32 i2sdrrx; /* data register */ + u32 i2scntrx; /* control register */ + u32 i2sfifocrx;/* FIFO control register */ + u32 i2safrx; /* almost full threshold setting */ + u32 i2saerx; /* almost empty threshold setting */ + u32 i2smskrx; /* interrupt mask settings */ + u32 i2sistrx; /* for acknowledging interrupts */ + u32 i2smonrx; /* monitor register */ +}; + +struct ioh_i2s_config_reg { + /* The common register settings */ + struct ioh_i2s_config_common_reg cmn; + + /* TX channel settings */ + struct ioh_i2s_config_tx_reg tx; + + /* RX channel settings */ + struct ioh_i2s_config_rx_reg rx; +}; + +static struct snd_pcm_hardware ml7213i2s_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SUPPORT_FORMAT, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 128 * 4 * 64, + .period_bytes_min = 64 * 4, + .period_bytes_max = 4 * 1024, + .periods_min = 64, + .periods_max = 64, + .fifo_size = 0, +}; +#endif -- 1.7.7.6

1 0

[alsa-devel] trying to get an eleven rack to work
by David Alexander 23 Mar '12

23 Mar '12

Hi. I bought and would like to get an eleven rack interface to work. It is a usb gutiar amp emulator/audio interface. It is a avid/digi device that normal uses asio vendor specific drivers. When I plug it under Linux I get the following: ls -l /proc/asound/card1 total 0 -r--r--r-- 1 root root 0 2012-03-17 18:57 id -r--r--r-- 1 root root 0 2012-03-17 18:57 midi0 -r--r--r-- 1 root root 0 2012-03-17 18:57 usbbus -r--r--r-- 1 root root 0 2012-03-17 18:57 usbid lsusb -v for the device returns: Bus 001 Device 003: ID 0dba:b011 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x0dba idProduct 0xb011 bcdDevice 20.01 iManufacturer 1 Digidesign iProduct 2 Eleven Rack iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 419 bNumInterfaces 5 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 254 Application Specific Interface bInterfaceSubClass 1 Device Firmware Update bInterfaceProtocol 0 iInterface 3 Eleven Rack DFU Device Firmware Upgrade Interface Descriptor: bLength 7 bDescriptorType 33 bmAttributes 1 Will Not Detach Manifestation Intolerant Upload Unsupported Download Supported wDetachTimeout 2000 milliseconds wTransferSize 1032 bytes Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 1 bInterfaceCount 4 bFunctionClass 255 Vendor Specific Class bFunctionSubClass 0 bFunctionProtocol 32 iFunction 2 Eleven Rack Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 32 iInterface 2 Eleven Rack ** UNRECOGNIZED: 09 24 01 00 02 0a ae 00 01 ** UNRECOGNIZED: 11 24 02 00 01 01 00 80 06 00 00 00 00 00 00 00 00 ** UNRECOGNIZED: 0c 24 03 03 01 06 00 40 80 00 00 00 ** UNRECOGNIZED: 11 24 02 02 01 06 00 80 08 00 00 00 00 00 00 00 00 ** UNRECOGNIZED: 0c 24 03 01 01 01 00 41 80 00 00 00 ** UNRECOGNIZED: 10 24 09 20 ff ff 01 00 06 00 00 00 00 00 01 00 ** UNRECOGNIZED: 1a 24 04 40 02 20 02 06 00 00 00 00 00 00 00 00 00 0c 30 c3 0c 30 c3 00 00 00 ** UNRECOGNIZED: 16 24 04 41 01 02 08 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ** UNRECOGNIZED: 0a 24 0b 80 03 81 82 83 03 00 ** UNRECOGNIZED: 08 24 0a 81 03 03 00 00 ** UNRECOGNIZED: 08 24 0a 82 00 05 00 00 ** UNRECOGNIZED: 08 24 0a 83 00 05 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 67 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 4 Rig MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 4 Rig MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 5 iJack 5 External MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 6 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 7 bNrInputPins 1 baSourceID( 0) 6 BaSourcePin( 0) 1 iJack 5 External MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 8 bNrInputPins 1 baSourceID( 0) 5 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 1 baAssocJackID( 1) 5 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 3 baAssocJackID( 1) 7 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 06 00 00 00 00 00 ** UNRECOGNIZED: 06 24 02 01 04 20 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x01a0 1x 416 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 ** UNRECOGNIZED: 10 24 01 01 00 01 01 00 00 00 08 00 00 00 00 00 ** UNRECOGNIZED: 06 24 02 01 04 20 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x01a0 1x 416 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered I am completely new to this, but if someone can point me in the right direction...I would appreciate it, I have just started trying to learn about this and my limited understanding is that it would require some additional coding in the quirks code for the snd-usb-audio driver. How can I determine what the quirks are? Does there exist any documentation or how-tos that I can look for on this? Any help or suggestions are appreciated. sincerely, David Alexander

3 3

Re: [alsa-devel] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x01db0000
by Petr Baudis 23 Mar '12

23 Mar '12

Hi! On Thu, Dec 29, 2011 at 12:21:05AM -0600, Jonathan Nieder wrote: > Petr Baudis wrote[1]: > > > Version: 3.1.5-1 > [...] > > this bug is twofold - I'm not sure which of the two problems is the real > > culprit of the issue I'm seeing, and if it's worth fixing both or just > > either one. It boils down to sometimes choppy playback and mainly > > applications getting stuck on some audio operations. > [...] > > This did not use to happen at least back around 2.6.32. > [...] > > I'd be willing to do some bisecting if people believe it would make > > sense. > > I don't see any relevant newer fixes, but could you test the for-next > branch from [2] anyway? Output from the alsa-info.sh script[3] would > also be useful. I'm really sorry for taking so long - this is still an issue for me, but contrary to my expectations, I was not able to reboot this machine yet and it might take some more time (weeks...) before that's convenient. I will try the then-current kernel and for-next branch then. The output from the alsa-info.sh script is at: http://www.alsa-project.org/db/?f=184928832b33b23c68d17460604b9a37af0fac4b -- Petr "Pasky" Baudis The goal of Computer Science is to build something that will last at least until we've finished building it.

2 5

[alsa-devel] [PATCH 01/16] ASoC: core: Add multi reg control struct & macros
by Ola Lilja 22 Mar '12

22 Mar '12

From: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> Add support for controls that exposes a single signed value while spanning multiple codec registers in a MSB/LSB manner. Definition of a generic control struct added. soc_mreg_control Also four generic convenience macros added: SOC_SINGLE_VALUE_S1R One control value spans one register SOC_SINGLE_VALUE_S2R One control value spans two registers SOC_SINGLE_VALUE_S4R One control value spans four registers SOC_SINGLE_VALUE_S8R One control value spans eight registers Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson(a)stericsson.com> --- include/sound/soc.h | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0992dff..dac20e0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -185,6 +185,29 @@ .rreg = xreg_right, .shift = xshift, \ .min = xmin, .max = xmax} } +#define SOC_SINGLE_VALUE_S1R(xreg0, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mreg_control) \ + { .reg = ((unsigned int[]){ xreg0 }), \ + .rcount = 1, .count = xcount, \ + .invert = xinvert, .min = xmin, .max = xmax}) +#define SOC_SINGLE_VALUE_S2R(xreg0, xreg1, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mreg_control) \ + {.reg = ((unsigned int[]){ xreg0, xreg1 }), \ + .rcount = 2, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) +#define SOC_SINGLE_VALUE_S4R(xreg0, xreg1, xreg2, xreg3, \ + xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mreg_control) \ + {.reg = ((unsigned int[]){ xreg0, xreg1, xreg2, xreg3 }), \ + .rcount = 4, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) +#define SOC_SINGLE_VALUE_S8R(xreg0, xreg1, xreg2, xreg3, xreg4, \ + xreg5, xreg6, xreg7, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mreg_control) \ + {.reg = ((unsigned int[]){ xreg0, xreg1, xreg2, xreg3, \ + xreg4, xreg5, xreg6, xreg7 }), \ + .rcount = 8, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) /* * Simplified versions of above macros, declaring a struct and calculating @@ -875,6 +898,13 @@ struct soc_mixer_control { unsigned int reg, rreg, shift, rshift, invert; }; +/* multi register control */ +struct soc_mreg_control { + long min, max; + unsigned int rcount, count, invert; + unsigned int *reg; +}; + /* enumerated kcontrol */ struct soc_enum { unsigned short reg; -- 1.7.8.3

4 37

Re: [alsa-devel] [PATCH] ASoC: Add extra parameter to device_prep_dma_cyclic
by Mark Brown 22 Mar '12

22 Mar '12

On Thu, Mar 22, 2012 at 10:19:03AM -0300, Fabio Estevam wrote: > Since commit 185ecb5 (dmaengine: add context parameter to prep_slave_sg and prep_dma_cyclic) > prep_dma_cyclic() got an extra context parameter. Applied, thanks.

1 0

Re: [alsa-devel] trying to get an eleven rack to work
by Clemens Ladisch 22 Mar '12

22 Mar '12

David Alexander wrote: > didn't seem to make any difference Are there any error messages in the system log when you're loading the driver? Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: da7210: Add support for spi regmap
by Ashish Chavan 21 Mar '12

21 Mar '12

This patch adds support for spi regmap feature to existing da7210 driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 90 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 90 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7843711..0f13fc4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> @@ -905,6 +906,9 @@ static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); int ret; +#if defined(CONFIG_SPI_MASTER) + unsigned int val; +#endif dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); @@ -927,10 +931,23 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx */ +#if defined(CONFIG_SPI_MASTER) + /* Dummy read to give two pulses over nCS */ + regmap_read(da7210->regmap, DA7210_STATUS, &val); + regmap_read(da7210->regmap, DA7210_STATUS, &val); + regmap_read(da7210->regmap, DA7210_STATUS, &val); +#endif + /* * make sure that DA7210 use bypass mode before start up */ snd_soc_write(codec, DA7210_STARTUP1, 0); + +#if defined(CONFIG_SPI_MASTER) + /* Dummy read to complete last write operation */ + regmap_read(da7210->regmap, DA7210_STATUS, &val); +#endif + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); @@ -1025,6 +1042,10 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); +#if defined(CONFIG_SPI_MASTER) + snd_soc_write(codec, 0x00, 0x80); +#endif + /* As suggested by Dialog */ /* unlock */ regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); @@ -1035,6 +1056,10 @@ static int da7210_probe(struct snd_soc_codec *codec) regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); +#if defined(CONFIG_SPI_MASTER) + snd_soc_write(codec, 0x00, 0x00); +#endif + /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1128,12 +1153,74 @@ static struct i2c_driver da7210_i2c_driver = { }; #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit da7210_spi_probe(struct spi_device *spi) +{ + struct da7210_priv *da7210; + int ret; + + da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); + if (!da7210) + return -ENOMEM; + + spi_set_drvdata(spi, da7210); + + da7210_regmap.read_flag_mask = 0x01; + da7210_regmap.write_flag_mask = 0x00; + da7210->regmap = regmap_init_spi(spi, &da7210_regmap); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + goto err_alloc; + } + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(da7210->regmap); +err_alloc: + kfree(da7210); + + return ret; +} + +static int __devexit da7210_spi_remove(struct spi_device *spi) +{ + struct da7210_priv *da7210 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(da7210->regmap); + kfree(da7210); + return 0; +} + +static struct spi_driver da7210_spi_driver = { + .driver = { + .name = "da7210-codec", + .owner = THIS_MODULE, + }, + .probe = da7210_spi_probe, + .remove = __devexit_p(da7210_spi_remove) +}; +#endif + static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&da7210_spi_driver); + if (ret) { + printk(KERN_ERR "Failed to register da7210 SPI driver: %d\n", + ret); + } +#endif return ret; } module_init(da7210_modinit); @@ -1143,6 +1230,9 @@ static void __exit da7210_exit(void) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&da7210_spi_driver); +#endif } module_exit(da7210_exit); -- 1.7.1

2 5

[alsa-devel] [PATCH] [v3] powerpc/8xxx: remove 85xx/86xx restrictions from fsl_guts.h
by Timur Tabi 21 Mar '12

21 Mar '12

Remove the check for CONFIG_PPC_85xx and CONFIG_PPC_86xx from fsl_guts.h. The check was originally intended to allow the same header file to be used on 85xx and 86xx systems, even though the Global Utilities register could be different. It turns out that they're not actually different, and so the check is not necessary. In addition, neither macro is defined for 64-bit e5500 kernels, so that causes a build break. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- The ASoC part of this patch applies cleanly to the ASoC for-3.4 branch, so this patch should not cause any merge conflicts if carried on the PowerPC branch, so I suggest that Kumar pick this up for 3.4 (it fixes a potential 64-bit book-e build break), with an ACK from Mark. arch/powerpc/include/asm/fsl_guts.h | 26 +++++--------------------- arch/powerpc/platforms/85xx/mpc85xx_mds.c | 2 +- arch/powerpc/platforms/85xx/mpc85xx_rdb.c | 2 +- arch/powerpc/platforms/85xx/p1022_ds.c | 4 ++-- arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 8 ++++---- sound/soc/fsl/p1022_ds.c | 10 +++++----- 7 files changed, 19 insertions(+), 35 deletions(-) diff --git a/arch/powerpc/include/asm/fsl_guts.h b/arch/powerpc/include/asm/fsl_guts.h index ce04530..aa4c488 100644 --- a/arch/powerpc/include/asm/fsl_guts.h +++ b/arch/powerpc/include/asm/fsl_guts.h @@ -16,15 +16,6 @@ #define __ASM_POWERPC_FSL_GUTS_H__ #ifdef __KERNEL__ -/* - * These #ifdefs are safe because it's not possible to build a kernel that - * runs on e500 and e600 cores. - */ - -#if !defined(CONFIG_PPC_85xx) && !defined(CONFIG_PPC_86xx) -#error Only 85xx and 86xx SOCs are supported -#endif - /** * Global Utility Registers. * @@ -36,11 +27,7 @@ * different names. In these cases, one name is chosen to avoid extraneous * #ifdefs. */ -#ifdef CONFIG_PPC_85xx -struct ccsr_guts_85xx { -#else -struct ccsr_guts_86xx { -#endif +struct ccsr_guts { __be32 porpllsr; /* 0x.0000 - POR PLL Ratio Status Register */ __be32 porbmsr; /* 0x.0004 - POR Boot Mode Status Register */ __be32 porimpscr; /* 0x.0008 - POR I/O Impedance Status and Control Register */ @@ -77,11 +64,8 @@ struct ccsr_guts_86xx { u8 res0a8[0xb0 - 0xa8]; __be32 rstcr; /* 0x.00b0 - Reset Control Register */ u8 res0b4[0xc0 - 0xb4]; -#ifdef CONFIG_PPC_85xx - __be32 iovselsr; /* 0x.00c0 - I/O voltage select status register */ -#else - __be32 elbcvselcr; /* 0x.00c0 - eLBC Voltage Select Ctrl Reg */ -#endif + __be32 iovselsr; /* 0x.00c0 - I/O voltage select status register + Called 'elbcvselcr' on 86xx SOCs */ u8 res0c4[0x224 - 0xc4]; __be32 iodelay1; /* 0x.0224 - IO delay control register 1 */ __be32 iodelay2; /* 0x.0228 - IO delay control register 2 */ @@ -136,7 +120,7 @@ struct ccsr_guts_86xx { * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the source (CCSR_GUTS_DMACR_DEV_xx) */ -static inline void guts_set_dmacr(struct ccsr_guts_86xx __iomem *guts, +static inline void guts_set_dmacr(struct ccsr_guts __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -172,7 +156,7 @@ static inline void guts_set_dmacr(struct ccsr_guts_86xx __iomem *guts, * ch: The channel on the DMA controller (0, 1, 2, or 3) * value: the new value for the bit (0 or 1) */ -static inline void guts_set_pmuxcr_dma(struct ccsr_guts_86xx __iomem *guts, +static inline void guts_set_pmuxcr_dma(struct ccsr_guts __iomem *guts, unsigned int co, unsigned int ch, unsigned int value) { if ((ch == 0) || (ch == 3)) { diff --git a/arch/powerpc/platforms/85xx/mpc85xx_mds.c b/arch/powerpc/platforms/85xx/mpc85xx_mds.c index f33662b..e82f06f 100644 --- a/arch/powerpc/platforms/85xx/mpc85xx_mds.c +++ b/arch/powerpc/platforms/85xx/mpc85xx_mds.c @@ -271,7 +271,7 @@ static void __init mpc85xx_mds_qe_init(void) if (machine_is(p1021_mds)) { - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; np = of_find_node_by_name(NULL, "global-utilities"); if (np) { diff --git a/arch/powerpc/platforms/85xx/mpc85xx_rdb.c b/arch/powerpc/platforms/85xx/mpc85xx_rdb.c index db214cd..1a66c3d 100644 --- a/arch/powerpc/platforms/85xx/mpc85xx_rdb.c +++ b/arch/powerpc/platforms/85xx/mpc85xx_rdb.c @@ -128,7 +128,7 @@ static void __init mpc85xx_rdb_setup_arch(void) #if defined(CONFIG_UCC_GETH) || defined(CONFIG_SERIAL_QE) if (machine_is(p1025_rdb)) { - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; np = of_find_node_by_name(NULL, "global-utilities"); if (np) { diff --git a/arch/powerpc/platforms/85xx/p1022_ds.c b/arch/powerpc/platforms/85xx/p1022_ds.c index 0fe88e3..e74b7cd 100644 --- a/arch/powerpc/platforms/85xx/p1022_ds.c +++ b/arch/powerpc/platforms/85xx/p1022_ds.c @@ -150,7 +150,7 @@ static void p1022ds_set_monitor_port(enum fsl_diu_monitor_port port) { struct device_node *guts_node; struct device_node *indirect_node = NULL; - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; u8 __iomem *lbc_lcs0_ba = NULL; u8 __iomem *lbc_lcs1_ba = NULL; u8 b; @@ -269,7 +269,7 @@ exit: void p1022ds_set_pixel_clock(unsigned int pixclock) { struct device_node *guts_np = NULL; - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; unsigned long freq; u64 temp; u32 pxclk; diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 13fa9a6..b8b1f33 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -226,7 +226,7 @@ void mpc8610hpcd_set_monitor_port(enum fsl_diu_monitor_port port) void mpc8610hpcd_set_pixel_clock(unsigned int pixclock) { struct device_node *guts_np = NULL; - struct ccsr_guts_86xx __iomem *guts; + struct ccsr_guts __iomem *guts; unsigned long freq; u64 temp; u32 pxclk; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 0ea4a5a..1f4c3b7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) { struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts_86xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -142,9 +142,9 @@ static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) { struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts_86xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index a5d4e80..d0dee59 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -46,7 +46,7 @@ * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx) */ -static inline void guts_set_dmuxcr(struct ccsr_guts_85xx __iomem *guts, +static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -90,9 +90,9 @@ static int p1022_ds_machine_probe(struct snd_soc_card *card) { struct machine_data *mdata = container_of(card, struct machine_data, card); - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -164,9 +164,9 @@ static int p1022_ds_machine_remove(struct snd_soc_card *card) { struct machine_data *mdata = container_of(card, struct machine_data, card); - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; -- 1.7.10.rc1.8.ga99c5.dirty

3 2

[alsa-devel] [PATCH v8] sound/soc/lapis: add machine driver for ML7213 Carrier Board
by Tomoya MORINAGA 21 Mar '12

21 Mar '12

This machine driver is for LAPIS Semiconductor ML7213 Carrier Board + ML26124 Reference Board Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- v8 - Delete redundant break - Use module_platform_driver - Delete redundant 'space' - Delete snd_soc_dapm_enable_pin()s --- sound/soc/lapis/Kconfig | 7 ++ sound/soc/lapis/Makefile | 2 + sound/soc/lapis/ml7213ioh-machine.c | 169 +++++++++++++++++++++++++++++++++++ 3 files changed, 178 insertions(+), 0 deletions(-) create mode 100644 sound/soc/lapis/ml7213ioh-machine.c diff --git a/sound/soc/lapis/Kconfig b/sound/soc/lapis/Kconfig index 551e385..9bc1b4d 100644 --- a/sound/soc/lapis/Kconfig +++ b/sound/soc/lapis/Kconfig @@ -2,3 +2,10 @@ config SND_SOC_ML7213_PLATFORM tristate "ML7213 IOH ASoC platform driver" help This option enables support for the AC Link Controllers in ML7213 IOH SoC. + +config SND_SOC_ML7213_MACHINE + tristate "ML7213 IOH ASoC machine driver" + select SND_SOC_ML7213_PLATFORM + select SND_SOC_ML26124 + help + This is ASoC machine driver for ML7213 IOH diff --git a/sound/soc/lapis/Makefile b/sound/soc/lapis/Makefile index aba1630..7ec4bea 100644 --- a/sound/soc/lapis/Makefile +++ b/sound/soc/lapis/Makefile @@ -1,4 +1,6 @@ # Platform +snd-soc-ml7213-machine-objs := ml7213ioh-machine.o snd-soc-ml7213-plat-objs := ml7213ioh-plat.o obj-$(CONFIG_SND_SOC_ML7213_PLATFORM) += snd-soc-ml7213-plat.o +obj-$(CONFIG_SND_SOC_ML7213_MACHINE) += snd-soc-ml7213-machine.o diff --git a/sound/soc/lapis/ml7213ioh-machine.c b/sound/soc/lapis/ml7213ioh-machine.c new file mode 100644 index 0000000..14f83d2 --- /dev/null +++ b/sound/soc/lapis/ml7213ioh-machine.c @@ -0,0 +1,169 @@ +/* + * ml7213ioh-machine.c -- SoC Audio for LAPIS Semiconductor ML7213 IOH CRB + * + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <linux/i2c.h> +#include "../codecs/ml26124.h" +#include "ioh_i2s.h" + +static const struct snd_soc_dapm_widget ml7213_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_LINE("LINEOUT", NULL), + SND_SOC_DAPM_LINE("LINEIN", NULL), + SND_SOC_DAPM_MIC("AnalogMIC", NULL), + SND_SOC_DAPM_MIC("DigitalMIC", NULL), +}; + +static const struct snd_soc_dapm_route ml7213_routes[] = { + {"Speaker", NULL, "SPOUT"}, + {"LINEOUT", NULL, "LOUT"}, + {"AnalogMIC", NULL, "MICBIAS"}, + {"MIN", NULL, "AnalogMIC"}, + {"MDIN", NULL, "DigitalMIC"}, + {"MIN", NULL, "LINEIN"}, +}; + +static int ml7213_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int bclkfs; + int mclkfs; + unsigned int clk; + int ret; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int channel_map[] = {5, 0, 1, 2, 3, 4}; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + + switch (params_rate(hw_params)) { + case 16000: + case 32000: + case 48000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + mclkfs = clk / params_rate(hw_params); + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S16_LE: + bclkfs = 32; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bclkfs = 64; + break; + default: + pr_err("%s: Failed not support format\n", __func__); + return -EINVAL; + } + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, ML26124_USE_PLLOUT, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops ml7213_ops = { + .hw_params = ml7213_hw_params, +}; + +static struct snd_soc_dai_link ml7213crb_dai = { + .name = "ml26124", + .stream_name = "ML26124", + .cpu_dai_name = "ml7213ioh", + .codec_dai_name = "ml26124-hifi", + .platform_name = "ml7213-i2s-audio", + .codec_name = "ml26124.1-001a", + .ops = &ml7213_ops, +}; + +static struct snd_soc_card ml7213crb_card = { + .name = "LAPIS Semiconductor ML7213CRB", + .dai_link = &ml7213crb_dai, + .num_links = 1, + .dapm_widgets = ml7213_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml7213_dapm_widgets), + .dapm_routes = ml7213_routes, + .num_dapm_routes = ARRAY_SIZE(ml7213_routes), +}; + +static int __devinit lapis_ml7213_ml26124_probe(struct platform_device *pdev) +{ + int ret; + + ml7213crb_card.dev = &pdev->dev; + ret = snd_soc_register_card(&ml7213crb_card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + return 0; +} + +static int __devexit lapis_ml7213_ml26124_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&ml7213crb_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_ml7213crb_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "lapis-ml7213ioh-CRB", + }, + .probe = lapis_ml7213_ml26124_probe, + .remove = __devexit_p(lapis_ml7213_ml26124_remove), +}; + +module_platform_driver(snd_ml7213crb_driver); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML7213 IOH ALSA SoC machine driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:lapis-ml7213ioh-CRB"); -- 1.7.7.6

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Update WM8994 DCS calibration
by Mark Brown 21 Mar '12

21 Mar '12

Based on latest production information. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8efc42b..058f183 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3623,7 +3623,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case 2: case 3: wm8994->hubs.dcs_codes_l = -9; - wm8994->hubs.dcs_codes_r = -5; + wm8994->hubs.dcs_codes_r = -7; break; default: break; -- 1.7.9.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: omap-mcbsp: Constraint handling changes
by Peter Ujfalusi 21 Mar '12

21 Mar '12

Hello, This series will add the following changes: 1. We do not need to place buffer size constraint in case of a capture stream. 2. Grazvydas Ignotas <notasas(a)gmail.com> reported that on Pandora they have issues with legacy applications (and OSS emulation) when they open the playback with small period size (smaller than the FIFO size). Since these legacy applications can not be fixed (and it is not feasible), this series will provide a solution to overcome with the underruns at the stream start. With the introduction of the new sysfs file (period_protection) the user can request the driver to place the constraint to the period size instead of the buffer size. With the period_protection one can specify the number of samples they want to have as protection in period size constraint over the McBSP FIFO size. As an example: OMAP3, McBSP2, stereo stream. the FIFO is 640 samples long. period_protection = 1 will ensure that the period size is minimum of 641 samples long. With the test code from Grazvydas Ignotas: http://notaz.gp2x.de/misc/alsa_period_test.c I don't see underrun issues with period_protection = 1 on BeagleBoard. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcbsp: buffer size constraint only applies to playback stream ASoC: omap-mcbsp: Restructure omap_mcbsp_dai_startup code ASoC: omap-mcbsp: Add period size protection mode sound/soc/omap/mcbsp.c | 2 + sound/soc/omap/mcbsp.h | 1 + sound/soc/omap/omap-mcbsp.c | 71 ++++++++++++++++++++++++++++++++++-------- 3 files changed, 60 insertions(+), 14 deletions(-) -- 1.7.8.5

5 27

[alsa-devel] Question about sound channel
by Kuninori Morimoto 21 Mar '12

21 Mar '12

Hi ALSA ML I have question about sound channel. If input sound was MONO (= 1ch), but codec supports I2S (= 2ch) only, How to output sound ? What should the driver do in this case ? Best regards --- Kuninori Morimoto

2 2

[alsa-devel] Last minute ASoC updates for 3.4
by Mark Brown 21 Mar '12

21 Mar '12

The following changes since commit 5ec65ee589fdaca7298b6303fd74ad6c121a8f38: ASoC: mx27vis-aic32x4: Convert it to platform driver (2012-03-13 23:09:59 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 22f8d055350066b4a87de4adea8c5213cac54534: ASoC: wm8994: Provide VMID mode control and fix default sequence (2012-03-19 19:08:17 +0000) ---------------------------------------------------------------- Last minute ASoC updates for 3.4 There's a couple of small features here that were added late on but have been in -next in my tree and some bug fixes. The wm_hubs stuff is actually bug fixes - the stuff that's currently in 3.4 is a half way house between the two solutions that the latest change allows the machine to select between. ---------------------------------------------------------------- Daniel Mack (1): ASoC: pxa-ssp: atomically set stream active masks Mark Brown (5): ASoC: wm8996: Add 44.1kHz support ASoC: core: Fix obscure leak of runtime array ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF ASoC: wm8994: Add missing break in resume ASoC: wm8994: Provide VMID mode control and fix default sequence Sangsu Park (1): ASoC: Samsung: Added to support mono recording Timur Tabi (1): ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master sound/soc/codecs/wm8994.c | 187 ++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8994.h | 8 ++ sound/soc/codecs/wm8996.c | 19 +++-- sound/soc/codecs/wm_hubs.c | 16 +--- sound/soc/fsl/p1022_ds.c | 24 ++++-- sound/soc/pxa/pxa-ssp.c | 61 +++++++++------ sound/soc/samsung/i2s.c | 13 +++- sound/soc/soc-core.c | 8 +- 8 files changed, 239 insertions(+), 97 deletions(-)

2 1

Re: [alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 21 Mar '12

21 Mar '12

On Tue, Mar 06, 2012 at 07:49:16PM +0900, Tomoya MORINAGA wrote: > On Tue, Mar 06, 2012　19:00 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > This is not what the above prototype says, the above prototype has only > > one register in it. You need to make this *much* clearer for the > > registers it's affecting. > Sorry, I couldn't understand your saying / concern. > Do you mean I shouldn't use ml26124_update_bits() but snd_soc_update_bits() ? Ideally you should be using the framework features. If there is some reason why you can't you need to make it much clearer what the reason is so people reading the code can understand.

3 15

Re: [alsa-devel] [PATCH v5] sound/soc/lapis: add machine driver for ML7213 Carrier Board
by Mark Brown 21 Mar '12

21 Mar '12

On Tue, Mar 06, 2012 at 02:49:24PM +0900, Tomoya MORINAGA wrote: > 2012年3月2日22:05 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > You're also missing a blank line between this and the previous stanza. > Sorry, I can't understand your saying. Your file should look like other similar files in the kernel. > > Use snd_soc_register_card() and a real device. > OK. > As a matter of interest, why do you recommend me to use > snd_soc_register_card() ? It's been the standard since 2.6.38.

2 11

[alsa-devel] [PATCH 1/3] ASoC: tegra: fix comment indentation in ALC5632 machine
by Stephen Warren 21 Mar '12

21 Mar '12

Fix comment indentation to clear checkpatch errors in a later patch. Signed-off-by: Stephen Warren <swarren(a)wwwdotorg.org> --- This series is for 3.5, and based on a merge of next-20120320 plus ASoC's for-3.5 branch, plus a couple minor patches to get recent next- compiling for Tegra, that don't affect the context of this series. Applying all the patches to the ASoC tree should be fine; I don't anticpate any conflicts with the likely Tegra tree for 3.5. sound/soc/tegra/tegra_alc5632.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e45ccd8..03ae095 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,16 +1,16 @@ /* -* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver -* -* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> -* -* Authors: Leon Romanovsky <leon(a)leon.nu> -* Andrey Danin <danindrey(a)mail.ru> -* Marc Dietrich <marvin24(a)gmx.de> -* -* 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. -*/ + * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver + * + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * + * Authors: Leon Romanovsky <leon(a)leon.nu> + * Andrey Danin <danindrey(a)mail.ru> + * Marc Dietrich <marvin24(a)gmx.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ #include <asm/mach-types.h> -- 1.7.0.4

2 3

[alsa-devel] HDA intel, wired headset w/mic, jack mic detect
by Kevin Hilman 20 Mar '12

20 Mar '12

Hello, I'm trying to get a wired headset with a mic working on MacBookAir running v3.0 (3.0.0-13-generic from Ubuntu to be precise.) It has Intel HDA audio[1] and everything works fine, including the built-in mic. The only think I can't seem to get working is a headset w/mic. I know the HW for dynamic detect works because the mic is detected dynamically under OSX. Is there some setting I need to tweak in order to enable the automatic mic detection? Thanks for any pointers, Kevin [1] $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xa0600000 irq 46

6 25

[alsa-devel] [PATCH 1/3] ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
by Mark Brown 19 Mar '12

19 Mar '12

It can just be enabled all the time with no impact. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 16 +++++----------- 1 files changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c08d1c2..f13f288 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -643,8 +643,6 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0), - SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, @@ -869,11 +867,9 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { - { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, @@ -890,11 +886,9 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { - { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, @@ -996,6 +990,11 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); + if (!lineout1_diff && !lineout2_diff) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, + WM8993_LINEOUT_VMID_BUF_ENA); + if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); @@ -1068,11 +1067,6 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, WM8993_LINEOUT2P_ENA, val); - if (!hubs->lineout1n_ena && !hubs->lineout1p_ena && - !hubs->lineout2n_ena && !hubs->lineout2p_ena) - snd_soc_update_bits(codec, WM8993_ANTIPOP1, - WM8993_LINEOUT_VMID_BUF_ENA, 0); - /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, WM8993_INPUTS_CLAMP, 0); -- 1.7.9.1

1 2

[alsa-devel] [PATCH] ASoC: cs4270: Check that we can enable regulators on resume
by Mark Brown 19 Mar '12

19 Mar '12

It's possible that the regulator enable will fail and if it does we may as well just give up with trying to bring the rest of the device up and report the original error. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 1d672f5..15df950 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -602,8 +602,10 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int reg; - regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret != 0) + return ret; /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ -- 1.7.9.1

2 4

[alsa-devel] [PATCH] ASoC: dapm: Remove bodges for no-widget CODECs
by Mark Brown 19 Mar '12

19 Mar '12

Now that we're creating widgets for all DAIs there should be no more need for the bodges we've been carrying for non-DAPM CODEC drivers so remove them. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I don't actually have any no widget CODECs so can't really test - Timur, would it be possible for you to give it a spin? sound/soc/soc-dapm.c | 36 ++++-------------------------------- 1 files changed, 4 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8ce1e9c..6e5869d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1441,12 +1441,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->n_widgets || d->codec == NULL) { - if (d->idle_bias_off) - d->target_bias_level = SND_SOC_BIAS_OFF; - else - d->target_bias_level = SND_SOC_BIAS_STANDBY; - } + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; } dapm_reset(card); @@ -1491,32 +1489,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } - /* If there are no DAPM widgets then try to figure out power from the - * event type. - */ - if (!dapm->n_widgets) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - case SND_SOC_DAPM_STREAM_RESUME: - dapm->target_bias_level = SND_SOC_BIAS_ON; - break; - case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec && dapm->codec->active) - dapm->target_bias_level = SND_SOC_BIAS_ON; - else - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_NOP: - dapm->target_bias_level = dapm->bias_level; - break; - default: - break; - } - } - /* Force all contexts in the card to the same bias state if * they're not ground referenced. */ -- 1.7.9.1

4 4

[alsa-devel] ASoC:Question rate constraint between the dais
by Ramesh Babu 19 Mar '12

19 Mar '12

Hi I would like to understand how to handle the different sampling rates between dais. I need to make sure that all simultaneous playback and capture (across the all dai_links) needs to be of same frequency family. If the sampling frequency of a new request is different from ongoing playback/capture, then it needs to be rejected. How do I handle it in machine driver? Thanks Ramesh

6 8

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
by Peter Ujfalusi 19 Mar '12

19 Mar '12

With the common irq the driver only needs to use one interrupt line, and it provides better debugging possibilites compared to the legacy TX/RX interrupt lines. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, with chages going via linux-omap: http://marc.info/?l=linux-omap&m=133217004811408&w=2 http://marc.info/?l=linux-omap&m=133217006911415&w=2 the McBSP driver will use the common interrupt line where it is supported. Regards, Peter sound/soc/omap/mcbsp.c | 113 +++++++++++++++++++++++++++++++++++++---------- sound/soc/omap/mcbsp.h | 7 +++- 2 files changed, 95 insertions(+), 25 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index ec6832c..34835e8 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -109,6 +109,47 @@ static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) dev_dbg(mcbsp->dev, "***********************\n"); } +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) +{ + struct omap_mcbsp *mcbsp = dev_id; + u16 irqst; + + irqst = MCBSP_READ(mcbsp, IRQST); + dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); + + if (irqst & RSYNCERREN) + dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); + if (irqst & RFSREN) + dev_dbg(mcbsp->dev, "RX Frame Sync\n"); + if (irqst & REOFEN) + dev_dbg(mcbsp->dev, "RX End Of Frame\n"); + if (irqst & RRDYEN) + dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); + if (irqst & RUNDFLEN) + dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); + if (irqst & ROVFLEN) + dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); + + if (irqst & XSYNCERREN) + dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); + if (irqst & XFSXEN) + dev_dbg(mcbsp->dev, "TX Frame Sync\n"); + if (irqst & XEOFEN) + dev_dbg(mcbsp->dev, "TX End Of Frame\n"); + if (irqst & XRDYEN) + dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); + if (irqst & XUNDFLEN) + dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); + if (irqst & XOVFLEN) + dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); + if (irqst & XEMPTYEOFEN) + dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); + + MCBSP_WRITE(mcbsp, IRQST, irqst); + + return IRQ_HANDLED; +} + static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) { struct omap_mcbsp *mcbsp_tx = dev_id; @@ -176,6 +217,10 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, /* Enable wakeup behavior */ if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); + + /* Enable TX/RX sync error interrupts by default */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN); } /** @@ -489,23 +534,25 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) MCBSP_WRITE(mcbsp, SPCR1, 0); MCBSP_WRITE(mcbsp, SPCR2, 0); - err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, - 0, "McBSP", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request TX IRQ %d " - "for McBSP%d\n", mcbsp->tx_irq, - mcbsp->id); - goto err_clk_disable; - } + if (mcbsp->irq) { + err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, + "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request IRQ\n"); + goto err_clk_disable; + } + } else { + err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, + "McBSP TX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); + goto err_clk_disable; + } - if (mcbsp->rx_irq) { - err = request_irq(mcbsp->rx_irq, - omap_mcbsp_rx_irq_handler, - 0, "McBSP", (void *)mcbsp); + err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, + "McBSP RX", (void *)mcbsp); if (err != 0) { - dev_err(mcbsp->dev, "Unable to request RX IRQ %d " - "for McBSP%d\n", mcbsp->rx_irq, - mcbsp->id); + dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); goto err_free_irq; } } @@ -542,9 +589,16 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - if (mcbsp->rx_irq) + /* Disable interrupt requests */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, 0); + + if (mcbsp->irq) { + free_irq(mcbsp->irq, (void *)mcbsp); + } else { free_irq(mcbsp->rx_irq, (void *)mcbsp); - free_irq(mcbsp->tx_irq, (void *)mcbsp); + free_irq(mcbsp->tx_irq, (void *)mcbsp); + } reg_cache = mcbsp->reg_cache; @@ -949,13 +1003,24 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) else mcbsp->phys_dma_base = res->start; - mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); - mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); - - /* From OMAP4 there will be a single irq line */ - if (mcbsp->tx_irq == -ENXIO) { - mcbsp->tx_irq = platform_get_irq(pdev, 0); - mcbsp->rx_irq = 0; + /* + * OMAP1, 2 uses two interrupt lines: TX, RX + * OMAP2430, OMAP3 SoC have combined IRQ line as well. + * OMAP4 and newer SoC only have the combined IRQ line. + * Use the combined IRQ if available since it gives better debugging + * possibilities. + */ + mcbsp->irq = platform_get_irq_byname(pdev, "common"); + if (mcbsp->irq == -ENXIO) { + mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); + + if (mcbsp->tx_irq == -ENXIO) { + mcbsp->irq = platform_get_irq(pdev, 0); + mcbsp->tx_irq = 0; + } else { + mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); + mcbsp->irq = 0; + } } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 179a822..262a615 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -218,15 +218,19 @@ enum { #define MCBSP_DMA_MODE_ELEMENT 0 #define MCBSP_DMA_MODE_THRESHOLD 1 -/********************** McBSP WAKEUPEN bit definitions *********************/ +/********************** McBSP WAKEUPEN/IRQST/IRQEN bit definitions *********/ #define RSYNCERREN BIT(0) #define RFSREN BIT(1) #define REOFEN BIT(2) #define RRDYEN BIT(3) +#define RUNDFLEN BIT(4) +#define ROVFLEN BIT(5) #define XSYNCERREN BIT(7) #define XFSXEN BIT(8) #define XEOFEN BIT(9) #define XRDYEN BIT(10) +#define XUNDFLEN BIT(11) +#define XOVFLEN BIT(12) #define XEMPTYEOFEN BIT(14) /* Clock signal muxing options */ @@ -294,6 +298,7 @@ struct omap_mcbsp { int configured; u8 free; + int irq; int rx_irq; int tx_irq; -- 1.7.8.5

2 1

Re: [alsa-devel] [PATCH 13/16] ASoC: codecs: Add AB8500 codec-driver
by Mark Brown 19 Mar '12

19 Mar '12

On Tue, Mar 13, 2012 at 04:11:40PM +0100, Ola Lilja wrote: > +++ b/sound/soc/codecs/Kconfig > @@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS > tristate "Build all ASoC CODEC drivers" > select SND_SOC_88PM860X if MFD_88PM860X > select SND_SOC_L3 > + select SND_SOC_AB8500 > select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS > select SND_SOC_AD1836 if SPI_MASTER > select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI This driver really has no dependencies? That seems *extremely* surprising - have you tried building on x86? > + > +ifdef CONFIG_SND_SOC_UX500_DEBUG > +CFLAGS_ab8500_audio.o := -DDEBUG > +endif Once again if you're adding random custom stuff like this for your driver you shouldn't be. > diff --git a/sound/soc/codecs/ab8500_audio.c b/sound/soc/codecs/ab8500_audio.c Everything else uses -codec. > +/* AB8500 register cache */ > +static const u8 ab8500_reg_cache[AB8500_CACHEREGNUM]; > +static enum sid_state sid_status = SID_UNCONFIGURED; No static data or custom cache implementations. > +/* Read a register from the audio-bank of AB8500 */ > +static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, unsigned int reg) > +{ > + int status; > + unsigned int value = 0; > + > + if (reg < VIRT_REG_BASE) { Why are you doing this virtual register stuff? It's making the code a lot more complex and doesn't look at all driver specific. > + } else { > + dev_dbg(codec->dev, "%s: Read 0x%02x from register 0x%02x:0x%02x\n", > + __func__, value8, (u8)AB8500_AUDIO, (u8)reg); There's already *plenty* of trace for register I/O in the framework. > +static const char * const enum_ena_dis[] = {"Enabled", "Disabled"}; > +static const char * const enum_dis_ena[] = {"Disabled", "Enabled"}; Why are the controls using these enums and not Switch controls? UIs know how to display switches. > +/* Earpiece - Mute */ > +static const struct snd_kcontrol_new dapm_ear_mute[] = { > + SOC_DAPM_SINGLE("Playback Switch", REG_MUTECONF, REG_MUTECONF_MUTEAR, 1, 1), > +}; This looks like it's just a mute rather than a mixer input enable (there's only one control...) so should be a regular sound control; unless there's a very good reason mutes other than mixer inputs normally don't affect the audio path as you might get pops and you probably don't want to do things like stop clocking just because the output is silenced. Similar thing applies in several other places. > +/* Earpiece source selector */ > +static const char * const enum_ear_lineout_source[] = {"Headset Left", "IHF Left"}; > +static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, REG_DMICFILTCONF, > + REG_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source); > +static const struct snd_kcontrol_new dapm_ear_lineout_source = > + SOC_DAPM_ENUM("Earpiece or LineOut Mono Source", dapm_enum_ear_lineout_source); This control can probably have a much better name... > +/* IHF - Enable/Disable */ > +static const struct soc_enum enum_ihf_left = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); 80 columns please. > +/* from -31 to 31 dB in 1 dB steps (mute instead of -32 dB) */ > +static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1); I'm really not a big fan of this sort of comment, adds greatly to the verbosity but just duplicates the code. > +static const char * const enum_earselcm[] = {"0.95V", "1.10V", "1.27V", "1.58V"}; > +static SOC_ENUM_SINGLE_DECL(soc_enum_earselcm, > + REG_ANACONF1, REG_ANACONF1_EARSELCM, enum_earselcm); Should this really be configured by the application layer and not platform data? > +/* Digital interface - DA from slot mapping */ > +static const char * const enum_da_from_slot_map[] = {"SLOT0", > + "SLOT1", Indentation. Also are you sure that this isn't DAPM stuff? > + SOC_ENUM("Earpiece DAC Low Power Playback Switch", > + soc_enum_eardaclowpow), These aren't switches, they're enums. Switches are strictly boolean things with values 0 and 1. > + snd_soc_update_bits(codec, > + REG_DIGIFCONF3, > + BMASK(REG_DIGIFCONF3_IF1MASTER), > + BMASK(REG_DIGIFCONF3_IF1MASTER)); You're unconditionally setting these bits? > + snd_soc_update_bits(codec, REG_DIGIFCONF4, clr_mask, set_mask); Either clr_mask and set_mask are misnamed or this doesn't do what you think it does. The bits specified by the third argument are set to the values given in the fourth argument. It looks like the code thinks that the bits in the third argument will be cleared and the bits in the fourth argument will be set. > +static int ab8500_codec_set_word_length_if1(struct snd_soc_codec *codec, unsigned int wl) > +{ Just inline stuff like this. > +static int ab8500_codec_set_bit_delay_if1(struct snd_soc_codec *codec, unsigned int delay) This too. > +/* Configures audio macrocell into the AB8500 Chip */ > +static int ab8500_codec_configure_audio_macrocell(struct snd_soc_codec *codec) > +{ > + u8 value8; > + unsigned int value; > + int status; > + > + status = ab8500_sysctrl_write(AB8500_STW4500CTRL3, > + AB8500_STW4500CTRL3_CLK32KOUT2DIS | AB8500_STW4500CTRL3_RESETAUDN, > + AB8500_STW4500CTRL3_RESETAUDN); > + if (status < 0) > + return status; You're just unconditionally setting these values - shouldn't they be platform data? > +static int ab8500_codec_add_widgets(struct snd_soc_codec *codec) > +{ > + int status; > + > + status = snd_soc_dapm_new_controls(&codec->dapm, ab8500_dapm_widgets, > + ARRAY_SIZE(ab8500_dapm_widgets)); Just point at the tables from the snd_soc_codec and remove all this code. > +static int ab8500_codec_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) > +{ > + dev_dbg(dai->codec->dev, "%s Enter.\n", __func__); > + > + return 0; > +} Remove all empty functions. > + > +struct snd_soc_dai_driver ab8500_codec_dai[] = { > + { > + .name = "ab8500-codec-dai.0", > + .id = 0, > + .playback = { > + .stream_name = "ab8500_0p", You should hook up the links to the widgets using DAPM routes, currently you're using the implicitly created routes from the stream name which is still supported but isn't best practice. > +static int ab8500_codec_remove(struct snd_soc_codec *codec) > +{ > + dev_dbg(codec->dev, "%s Enter.\n", __func__); > + > + snd_soc_dapm_free(&codec->dapm); Why are you doing this? > + /* Create register cache */ > + ab8500_codec_driver.reg_cache_size = AB8500_LAST_REG - AB8500_FIRST_REG; > + ab8500_codec_driver.reg_cache_default = > + kzalloc(ab8500_codec_driver.reg_cache_size * sizeof(u8), GFP_KERNEL); The driver struct should be constant, the stuff you're initialising it with here is compile time constant - why are you doing this dynamically at runtime/ > + > + /* Create driver private-data struct */ > + drvdata = kzalloc(sizeof(struct ab8500_codec_drvdata), GFP_KERNEL); Anything you do alloc should be devm_kzalloc(). > + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(&pdev->dev); > + > + dev_info(&pdev->dev, "%s Enter.\n", __func__); > + > + kfree(ab8500_codec_driver.reg_cache_default); > + > + kfree(drvdata->virt_reg_cache); > + kfree(drvdata); > + > + snd_soc_unregister_codec(&pdev->dev); This would avoid issues like this where you free the data the device uses before you free the device. > +static int __devinit ab8500_codec_platform_driver_init(void) module_platform_driver. > +/* Extended interface for codec-driver */ > + > +int ab8500_audio_power_control(struct snd_soc_codec *codec, bool power_on); > +int ab8500_audio_set_word_length(struct snd_soc_dai *dai, unsigned int wl); > +int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, unsigned int delay); > +int ab8500_audio_setup_if1(struct snd_soc_codec *codec, > + unsigned int fmt, > + unsigned int wl, > + unsigned int delay); > +void ab8500_audio_anc_configure(struct snd_soc_codec *codec, > + bool apply_fir, bool apply_iir); No, you shouldn't be exporting this stuff - it all looks like basic framework stuff. > +/* Virtual register base > + * This address should be located above any physical register used by ASoC. > + * It is used to map virtual registers needed by the codec-driver. > + */ > +#define VIRT_REG_BASE 0x3000 Why are you doing this virtual register stuff? > +#define REG_DAPATHCONF_ENDACHSL 5 All these constants need to be driver local or namespaced (preferrably namespaced whatever).

4 11

[alsa-devel] [PATCH] aplay/arecord: option to treat any xrun as fatal
by Ben Gardiner 19 Mar '12

19 Mar '12

Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- aplay/aplay.1 | 4 ++++ aplay/aplay.c | 17 +++++++++++++++-- 2 files changed, 19 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 0195322..bf9b53e 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -191,6 +191,10 @@ lists capabilities of the selected device such as supported formats, sampling rates, numbers of channels, period and buffer bytes/sizes/times. For raw device hw:X this option basically lists hardware capabilities of the soundcard. +.TP +\fI\-\-fatal\-errors\fP +Disables recovery attempts when errors (e.g. xrun) are encountered; the +aplay process instead aborts immediately. .SH SIGNALS When recording, SIGINT, SIGTERM and SIGABRT will close the output diff --git a/aplay/aplay.c b/aplay/aplay.c index 0633cbd..1ce34c4 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -115,6 +115,7 @@ static int stop_delay = 0; static int monotonic = 0; static int interactive = 0; static int can_pause = 0; +static int fatal_errors = 0; static int verbose = 0; static int vumeter = VUMETER_NONE; static int buffer_pos = 0; @@ -225,7 +226,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " for this many seconds\n" " --process-id-file write the process ID here\n" " --use-strftime apply the strftime facility to the output file name\n" -" --dump-hw-params dump hw_params of the device\n") +" --dump-hw-params dump hw_params of the device\n" +" --fatal-errors treat all errors as fatal\n") , command); printf(_("Recognized sample formats are:")); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { @@ -419,7 +421,8 @@ enum { OPT_MAX_FILE_TIME, OPT_PROCESS_ID_FILE, OPT_USE_STRFTIME, - OPT_DUMP_HWPARAMS + OPT_DUMP_HWPARAMS, + OPT_FATAL_ERRORS, }; int main(int argc, char *argv[]) @@ -465,6 +468,7 @@ int main(int argc, char *argv[]) {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, + {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, {0, 0, 0, 0} }; char *pcm_name = "default"; @@ -669,6 +673,9 @@ int main(int argc, char *argv[]) case OPT_DUMP_HWPARAMS: dump_hw_params = 1; break; + case OPT_FATAL_ERRORS: + fatal_errors = 1; + break; default: fprintf(stderr, _("Try `%s --help' for more information.\n"), command); return 1; @@ -1350,6 +1357,12 @@ static void xrun(void) prg_exit(EXIT_FAILURE); } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) { + if (fatal_errors) { + error(_("fatal %s: %s"), + stream == SND_PCM_STREAM_PLAYBACK ? _("underrun") : _("overrun"), + snd_strerror(res)); + prg_exit(EXIT_FAILURE); + } if (monotonic) { #ifdef HAVE_CLOCK_GETTIME struct timespec now, diff, tstamp; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ALSA: ASoC: pxa-ssp: atomically set stream active masks
by Daniel Mack 19 Mar '12

19 Mar '12

PXA's SSP engine fails to take its current channel phase into account when enabling a stream while the engine is already running. This results in randomly swapped left/right channels on either the record or the playback side, depending on which one was enabled first. The following patch fixes this by factoring out the bit field modifications in question to a separate function that pauses the engine temporarily, modifies the bits and kicks it off again afterwards. Appearantly, a transition of SSCR0_SSE syncs both directions properly. The patch has been rolled out to quite a number of devices over the last weeks and seems to fix the issue reliably. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Sven Neumann <s.neumann(a)raumfeld.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)marvell.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Johannes Stezenbach <js(a)sig21.net> Cc: stable(a)kernel.org --- sound/soc/pxa/pxa-ssp.c | 61 +++++++++++++++++++++++++++------------------- 1 files changed, 36 insertions(+), 25 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a16df0f..fd04ce1 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -668,6 +668,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, return 0; } +static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream, + struct ssp_device *ssp, int value) +{ + uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1); + uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP); + uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR); + + if (value && (sscr0 & SSCR0_SSE)) + pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (value) + sscr1 |= SSCR1_TSRE; + else + sscr1 &= ~SSCR1_TSRE; + } else { + if (value) + sscr1 |= SSCR1_RSRE; + else + sscr1 &= ~SSCR1_RSRE; + } + + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + + if (value) { + pxa_ssp_write_reg(ssp, SSSR, sssr); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE); + } +} + static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -681,42 +713,21 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= SSCR1_TSRE; - else - val |= SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 1); val = pxa_ssp_read_reg(ssp, SSSR); pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= SSCR1_TSRE; - else - val |= SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); - pxa_ssp_enable(ssp); + pxa_ssp_set_running_bit(substream, ssp, 1); break; case SNDRV_PCM_TRIGGER_STOP: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val &= ~SSCR1_TSRE; - else - val &= ~SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 0); break; case SNDRV_PCM_TRIGGER_SUSPEND: pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val &= ~SSCR1_TSRE; - else - val &= ~SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 0); break; default: -- 1.7.7.6

2 1

[alsa-devel] static /dev configuration
by Bob Tracy 19 Mar '12

19 Mar '12

Does ALSA now require dynamic devices (udev) for certain soundcards (specifically, cards handled by the snd-hda-intel driver)? My problem is that I moved a Slackware installation to new hardware, and the old devices under /dev/snd I originally created back in 2003 don't seem to work correctly with snd-hda-intel. "lspci" shows my new system having two audio devices as follows: 00:1b.0 Audio device: Intel Corporation Cougar Point High Definition Audio Controller (rev 04) 01:00.1 Audio device: ATI Technologies Inc Device aa60 (The ATI entry corresponds to a DisplayPort output on a Radeon graphics card that isn't presently of interest to me: REDWOOD firmware.) I had an opportunity to boot a modern Linux distro with udev support that accesses the Intel audio device just fine, so I took note of the device entries under /dev/snd and replicated those on the Slackware setup as follows (ignore the "root" group and 666 permissions rather than the more correct "audio" group and 660 permissions): crw-rw-rw- 1 root root 116, 9 Mar 18 00:24 controlC0 crw-rw-rw- 1 root root 116, 12 Mar 18 00:26 controlC1 crw-rw-rw- 1 root root 116, 8 Mar 18 00:27 hwC0D2 crw-rw-rw- 1 root root 116, 7 Jun 18 2003 hwC0D3 crw-rw-rw- 1 root root 116, 11 Mar 18 00:30 hwC1D0 crw-rw-rw- 1 root root 116, 6 Mar 18 00:34 pcmC0D0c crw-rw-rw- 1 root root 116, 5 Mar 18 20:44 pcmC0D0p crw-rw-rw- 1 root root 116, 4 Mar 18 00:36 pcmC0D1p crw-rw-rw- 1 root root 116, 3 Mar 18 00:39 pcmC0D2c crw-rw-rw- 1 root root 116, 2 Mar 18 00:44 pcmC0D3p crw-rw-rw- 1 root root 116, 10 Mar 18 00:42 pcmC1D3p crw-rw-rw- 1 root root 116, 1 Jun 18 2003 seq crw-rw-rw- 1 root root 116, 33 Jun 18 2003 timer Not all that surprisingly, the above entries work: "alsamixer" and "aplay" do what I expect and seem to work correctly. Note that the minor device numbers are NOT the same as what the "snddevices" script in "alsa-driver-1.0.25" creates (and that script creates pretty much exactly what I had in my static /dev tree back in 2003). According to "snddevices", the "controlCn" devices should have minor numbers 0, 32, 64, etc. The "hwC0Dn" minors are 4, 5, etc., "hwC1Dn" minors are 36, 37, etc., "hwC2Dn" minors are 68, 69, etc. Note that "hwC0D3" is at minor 7 where it should be under the old minor device numbering scheme, but "hwC0D2" is at minor 8 where 6 was expected. The various "pcm*" devices were expected to be as follows: crw-rw-rw- 1 root root 116, 24 Jun 18 2003 pcmC0D0c crw-rw-rw- 1 root root 116, 16 Mar 3 20:50 pcmC0D0p crw-rw-rw- 1 root root 116, 25 Jun 18 2003 pcmC0D1c crw-rw-rw- 1 root root 116, 17 Jun 18 2003 pcmC0D1p crw-rw-rw- 1 root root 116, 26 Jun 18 2003 pcmC0D2c crw-rw-rw- 1 root root 116, 18 Jun 18 2003 pcmC0D2p crw-rw-rw- 1 root root 116, 27 Jun 18 2003 pcmC0D3c crw-rw-rw- 1 root root 116, 19 Jun 18 2003 pcmC0D3p crw-rw-rw- 1 root root 116, 56 Jun 18 2003 pcmC1D0c crw-rw-rw- 1 root root 116, 48 Jun 18 2003 pcmC1D0p crw-rw-rw- 1 root root 116, 57 Jun 18 2003 pcmC1D1c crw-rw-rw- 1 root root 116, 49 Jun 18 2003 pcmC1D1p crw-rw-rw- 1 root root 116, 58 Jun 18 2003 pcmC1D2c crw-rw-rw- 1 root root 116, 50 Jun 18 2003 pcmC1D2p crw-rw-rw- 1 root root 116, 59 Jun 18 2003 pcmC1D3c crw-rw-rw- 1 root root 116, 51 Jun 18 2003 pcmC1D3p I'm thinking the only reason I'm getting away with using any kind of static device scheme is that my sound devices are getting detected and initialized in a consistent order each time I boot, and the /dev/snd entries I copied from the udev-enabled system just happen to agree with what "udev" would consistently create if I had "udev" installed. So, just to restate the original question for clarity, is "udev" required for certain soundcards? Thanks! --Bob

2 2

[alsa-devel] [PATCH] ASoC: Add MICBIAS routing information for WM8904
by Hong Xu 19 Mar '12

19 Mar '12

Add MICBIAS routing entries for WM8904. Tested on ATMEL's AT91SAM9N12-EK with a electret condenser microphone connected to IN1L. Signed-off-by: Hong Xu <hong.xu(a)atmel.com> --- sound/soc/codecs/wm8904.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 285ef87..2158f30 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1318,6 +1318,14 @@ static const struct snd_soc_dapm_route core_intercon[] = { }; static const struct snd_soc_dapm_route adc_intercon[] = { + { "MICBIAS", NULL, "IN1L" }, + { "MICBIAS", NULL, "IN1R" }, + + { "Left Capture Mux", NULL, "MICBIAS" }, + { "Right Capture Mux", NULL, "MICBIAS" }, + { "Left Capture Inverting Mux", NULL, "MICBIAS" }, + { "Right Capture Inverting Mux", NULL, "MICBIAS" }, + { "Left Capture Mux", "IN1L", "IN1L" }, { "Left Capture Mux", "IN2L", "IN2L" }, { "Left Capture Mux", "IN3L", "IN3L" }, -- 1.7.6

3 2

Re: [alsa-devel] ASoC:Question rate constraint between the dais
by Omair Mohammed Abdullah 18 Mar '12

18 Mar '12

>> I need to make sure that all simultaneous playback and capture (across >> the all dai_links) needs to be of same frequency family. >> If the sampling frequency of a new request is different from ongoing >> playback/capture, then it needs to be rejected. > >> How do I handle it in machine driver? > > Why do you need to do this in the machine driver? Such a constraint sounds like it's coming from one of the chips in the system rather than from the machine itself so it'd seem sensible for the chip itself to impose the constraint so that other machines don't need to replicate the code. > > See wm8988 for an example of adding sample rate constraints, obviously you can change the criteria dynamically. You are right to say that it is not a machine constraint but a codec constraint. The problem is that if a DAI (say the "SN95031 Headset DAI") is opened with a 44.1 khz rate, then another DAI (say "SN95031 Voice") is opened with a 48 khz rate, then we need to reject the second open because the codec cannot support both rates at the same time. Is there any way to do this with constraint_lists? The way it is done in the wm8988 driver does not apply here because there it is a single DAI.

4 7

[alsa-devel] [PATCH 1/1] a52: set time_base to avoid crash in avcodec
by Alex Hornung 18 Mar '12

18 Mar '12

* Not setting the avctx->time_base properly results in a crash in avcodec_encode_audio() in more recent versions of it. * Fix it by setting the time_base according to the sampling rate. Signed-off-by: Alex Hornung <alex(a)alexhornung.com> --- a52/pcm_a52.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 00c7c59..560071d 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -436,6 +436,8 @@ static int a52_prepare(snd_pcm_ioplug_t *io) rec->avctx->bit_rate = rec->bitrate * 1000; rec->avctx->sample_rate = io->rate; rec->avctx->channels = io->channels; + rec->avctx->time_base.num = 1; + rec->avctx->time_base.den = io->rate; #if LIBAVCODEC_VERSION_MAJOR > 52 || (LIBAVCODEC_VERSION_MAJOR == 52 && LIBAVCODEC_VERSION_MINOR >= 95) rec->avctx->sample_fmt = AV_SAMPLE_FMT_S16; #else -- 1.7.9.1

1 0

[alsa-devel] [PATCH v2] ASoC: max98095: add jack detection
by Rhyland Klein 17 Mar '12

17 Mar '12

This change adds the logic to support using the jack detect mechanism built in to the codec to detect both when a jack was inserted and what type of jack is present. This change also supports the use of an external mechanism for headphone detection. If this mechanism exists, when the max98095_jack_detect function is called, the hp_jack is simply passed NULL. This change supports both simple headphones, powered headphones, microphones and headsets with both headphones and a mic. v2: - Removed useless function for irq_handler - removed enable/disable irq - Added pdata entry for jack detections slew rate - get irq directly from dev Signed-off-by: Rhyland Klein <rklein(a)nvidia.com> --- include/sound/max98095.h | 12 +++ sound/soc/codecs/max98095.c | 158 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/max98095.h | 22 ++++++ 3 files changed, 191 insertions(+), 1 deletions(-) diff --git a/include/sound/max98095.h b/include/sound/max98095.h index 7513a42..65b1c90 100644 --- a/include/sound/max98095.h +++ b/include/sound/max98095.h @@ -49,6 +49,18 @@ struct max98095_pdata { */ unsigned int digmic_left_mode:1; unsigned int digmic_right_mode:1; + + /* Pin5 is the mechanical method of sensing jack insertion + * but it is something that might not be supported. + * 0 = PIN5 not supported + * 1 = PIN5 supported + */ + int jack_detect_pin5en:1; + + /* Slew amount for jack detection. Calculated as 4 * (delay + 1). + * Default delay is 24 to get a time of 100ms. + */ + unsigned int jack_detect_delay; }; #endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a..0752840 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <asm/div64.h> #include <sound/max98095.h> +#include <sound/jack.h> #include "max98095.h" enum max98095_type { @@ -51,6 +52,8 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2176,125 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static irqreturn_t max98095_report_jack(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return IRQ_NONE; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) { + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + } else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return IRQ_HANDLED; +} + +int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + unsigned int slew = M98095_DEFAULT_SLEW_DELAY; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + if (max98095->jack_detect_delay) + slew = max98095->jack_detect_delay; + + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, slew); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(client->irq, codec); + return 0; +} + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2302,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(client->irq, codec); + } + return 0; } #else @@ -2227,6 +2354,7 @@ static int max98095_probe(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct max98095_cdata *cdata; + struct i2c_client *client; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); @@ -2238,6 +2366,8 @@ static int max98095_probe(struct snd_soc_codec *codec) /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); + client = to_i2c_client(codec->dev); + /* initialize private data */ max98095->sysclk = (unsigned)-1; @@ -2266,11 +2396,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (client->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(client->irq, NULL, + max98095_report_jack, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2448,28 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (client->irq) + free_irq(client->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = to_i2c_client(codec->dev); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (client->irq) + free_irq(client->irq, codec); + return 0; } diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a..2ebbe4e 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: pxa2xx-i2s: Add clk_prepare/clk_unprepare calls
by Philipp Zabel 17 Mar '12

17 Mar '12

This patch adds clk_prepare/clk_unprepare calls to the pxa2xx-i2s driver by using the helper functions clk_prepare_enable and clk_disable_unprepare. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)marvell.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd5..453b092 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -165,7 +165,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); - clk_enable(clk_i2s); + clk_prepare_enable(clk_i2s); clk_ena = 1; pxa_i2s_wait(); @@ -258,7 +258,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); if (clk_ena) { - clk_disable(clk_i2s); + clk_disable_unprepare(clk_i2s); clk_ena = 0; } } -- 1.7.9.1

2 1

[alsa-devel] [PATCH v3 00/11] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 17 Mar '12

17 Mar '12

I believe I have addressed all the outstanding comments and requests reviewers put on previous versions. So, hopefully this will be the last iteration of the series. The current sound/for-next branch seems broken for me, and I have not tracking the problem down. But the series works good on SHA below. 3030763 (Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc) Timur, I did not pick up you Ack-by tag, because there are some changes since v2 you will need to take another look and have the testing, I guess. Regards, Shawn Shawn Guo (11): ASoC: core: missing set_fmt should not be complaint ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: rename imx-pcm Kconfig options and filename ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: remove helper fsl_asoc_get_codec_dev_name ASoC: fsl: check property 'compatible' for the machine name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: remove the fatal error checking on codec-handle ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 1 + arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 ++++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 146 ++++++++++++---- sound/soc/fsl/fsl_utils.c | 91 ++++++++++ sound/soc/fsl/fsl_utils.h | 26 +++ sound/soc/{imx => fsl}/imx-audmux.c | 0 sound/soc/{imx => fsl}/imx-audmux.h | 0 .../{imx/imx-pcm-dma-mx2.c => fsl/imx-pcm-dma.c} | 0 sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/{imx => fsl}/imx-pcm.c | 0 sound/soc/{imx => fsl}/imx-pcm.h | 0 sound/soc/fsl/imx-sgtl5000.c | 177 ++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 2 +- sound/soc/{imx => fsl}/imx-ssi.h | 0 sound/soc/fsl/mpc8610_hpcd.c | 168 ++---------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 0 sound/soc/fsl/p1022_ds.c | 190 ++------------------ sound/soc/{imx => fsl}/phycore-ac97.c | 0 sound/soc/{imx => fsl}/wm1133-ev1.c | 0 sound/soc/imx/Kconfig | 79 -------- sound/soc/imx/Makefile | 22 --- sound/soc/soc-core.c | 11 +- 29 files changed, 616 insertions(+), 480 deletions(-)

7 67

[alsa-devel] [PATCH] ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
by Timur Tabi 17 Mar '12

17 Mar '12

The WM8776 codec driver requires the machine driver to set one of the SND_SOC_DAIFMT_CBx_xxx values. The P1022DS machine driver should be setting SND_SOC_DAIFMT_CBM_CFM, but since that value was zero, no one noticed. Commit 75d9ac46 ("ASoC: Allow DAI formats to be specified in the dai_link"), however, changed the value of SND_SOC_DAIFMT_CBM_CFM from zero to a non-zero value, which means that it now needs to be specifically set by the machine driver. We also set SND_SOC_DAIFMT_NB_NF, for the same reason. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch should be applied on top of Shawn's patchset "[PATCH v5] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi", but it may also need to be back-ported to kernel 3.3. sound/soc/fsl/p1022_ds.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 0f42f31..2022985 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -255,7 +255,8 @@ static int p1022_ds_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -272,31 +273,38 @@ static int p1022_ds_probe(struct platform_device *pdev) } mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { -- 1.7.10.rc1.8.ga99c5.dirty

2 1

Re: [alsa-devel] [PATCH 2/2] ARM: S3C64XX: Hook up supply for WM8994 LDOs on Littlemill
by 김국진 16 Mar '12

16 Mar '12

Samuel Ortiz wrote: > Hi Mark, Kukjin > > On Thu, Mar 08, 2012 at 11:55:07AM +0000, Mark Brown wrote: >> Signed-off-by: Mark Brown >> --- >> >> Kukjin, are you OK with this going via the MFD tree? It depends on the >> previous change. > I'm taking this one for now, unless Kukjin objects to it. Yeah, looks ok to me, if you need my ack on this, feel free to add it. Acked-by: Kukjin Kim <kgene.kim(a)samsung.com> Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

1 0

[alsa-devel] amixer shows capability "penum" on every Simple mixer control
by Christian Esken (KDE) 16 Mar '12

16 Mar '12

Hello, when I run amixer it shows the capability "penum" on nearly every Simple mixer control. I posted the bug report https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5496 , but Raymond asked me to report it on this mailing list ("as it is related to this patch 6d511eefa05878abee9f8b9145697230de605200 Implements support for capture/playback enums."). I posted details for the following two examples in https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5496 : 1) "Creative X-Fi" does not have any Enum's. alsamixer does not show Enums's. amixer lists "penum" for ALL Simple mixer controls. 2) "HDA ATI SB" has no penum's, only 3 cenum's. alsamixer shows it correctly, but amixer disagrees and shows penum for NEARLY ALL Simple mixer controls (except for those with cpability "cenum"). I found this as I received several bug reports for KMix with "absurd" amixer output. I could perceive from those bug report that the bug is affecting a lot (or all?) sound cards. Christian

3 7

[alsa-devel] [PATCH] ASoC: Samsung: Added to support mono recording
by Sangbeom Kim 16 Mar '12

16 Mar '12

From: Sangsu Park <sangsu4u.park(a)samsung.com> The dma size will be changed by requested number of channel(mono/stereo) from platform. For mono recording, channels_min value should be 1. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/i2s.c | 13 ++++++++++++- 1 files changed, 12 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 6553b19..6ac7b82 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -559,6 +559,17 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, mod |= MOD_DC1_EN; break; case 2: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s->dma_playback.dma_size = 4; + else + i2s->dma_capture.dma_size = 4; + break; + case 1: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s->dma_playback.dma_size = 2; + else + i2s->dma_capture.dma_size = 2; + break; default: dev_err(&i2s->pdev->dev, "%d channels not supported\n", @@ -963,7 +974,7 @@ struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; if (!sec) { - i2s->i2s_dai_drv.capture.channels_min = 2; + i2s->i2s_dai_drv.capture.channels_min = 1; i2s->i2s_dai_drv.capture.channels_max = 2; i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; -- 1.7.1

2 1

[alsa-devel] ASoC: Samsung: Update Kconfig for Exynos5250 and Exynos4412
by Sangbeom Kim 16 Mar '12

16 Mar '12

Update Kconfig file to support the Exynos5250 and Exynos4412 Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/Kconfig | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f3417f2..2e790c0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on PLAT_SAMSUNG select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C2410 help @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212) + depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212 || MACH_SMDK4412 || MACH_SMDK5250) depends on I2C=y && GENERIC_HARDIRQS select MFD_WM8994 select SND_SOC_WM8994 @@ -162,7 +162,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212 || MACH_SMDK4412 || MACH_SMDK5250) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -177,7 +177,7 @@ config SND_SOC_SMDK_WM8580_PCM config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212 || MACH_SMDK4412 || MACH_SMDK5250) depends on I2C=y && GENERIC_HARDIRQS select MFD_WM8994 select SND_SOC_WM8994 -- 1.7.1

2 1

Re: [alsa-devel] linux-next: Tree for Mar 16 (sound)
by Takashi Iwai 16 Mar '12

16 Mar '12

At Fri, 16 Mar 2012 08:01:06 -0700, Randy Dunlap wrote: > > [1 <text/plain; UTF-8 (7bit)>] > On 03/16/2012 12:39 AM, Stephen Rothwell wrote: > > > Hi all, > > > > Changes since 20120315: > > > > The sound tree lost its build failure. > > > > > on i386: > > sound/pci/hda/patch_sigmatel.c: In function 'patch_stac92hd83xxx': > sound/pci/hda/patch_sigmatel.c:5589:6: error: 'stac92xx_set_power_state' undeclared (first use in this function) > sound/pci/hda/patch_sigmatel.c:5589:6: note: each undeclared identifier is reported only once for each function it appears in > sound/pci/hda/patch_sigmatel.c:5591:19: error: 'struct hda_codec_ops' has no member named 'pre_resume' > sound/pci/hda/patch_sigmatel.c:5591:33: error: 'stac92xx_pre_resume' undeclared (first use in this function) > /home/rddunlap/lnx/next/linux-next-20120316/sound/pci/hda/patch_sigmatel.c: In function 'patch_stac92hd71bxx': > sound/pci/hda/patch_sigmatel.c:5896:6: error: 'stac92xx_set_power_state' undeclared (first use in this function) > sound/pci/hda/patch_sigmatel.c:5898:19: error: 'struct hda_codec_ops' has no member named 'pre_resume' > sound/pci/hda/patch_sigmatel.c:5898:33: error: 'stac92xx_pre_resume' undeclared (first use in this function) I seem to have overlooked the case CONFIG_PM=n. Fixed now. thanks, Takashi

1 0

[alsa-devel] [PATCH 0/2] ASoC: omap-mcbsp: dma_op_mode changes
by Peter Ujfalusi 16 Mar '12

16 Mar '12

Hello, Removing the frame dm_op_mode, and using packet mode in element mode on OMAP3+ to transfer 1 sample at the time between the McBSP FIFO and memory. Regards, Peter --- Peter Ujfalusi (2): ASoC: omap-mcbsp: Use DMA packet mode for non mono streams on OMAP3+ ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode sound/soc/omap/mcbsp.c | 2 +- sound/soc/omap/mcbsp.h | 1 - sound/soc/omap/omap-mcbsp.c | 31 +++++++++++++++++-------------- 3 files changed, 18 insertions(+), 16 deletions(-) -- 1.7.8.5

3 6

[alsa-devel] [ANNOUNCE] tinycompress library
by Vinod Koul 16 Mar '12

16 Mar '12

The Userland library for ALSA compressed API "tinycompress" is available in dual BSD and LGPL license. This development version 0.0.1 contains support for playback of compressed formats. This library also includes a command line player which support mp3 files only. The library is hosted at: git://git.alsa-project.org/tinycompress.git Currently it includes the Makefile for Android only. Comments, question, criticism and patches are welcome! Thanks -- ~Vinod

1 0

Re: [alsa-devel] snd_vortex_dev_free() BUG: unable to handle kernel NULL pointer
by Raymond Yau 16 Mar '12

16 Mar '12

2012/3/15, Andrew Vagin <avagin(a)gmail.com>: > Hello, > > In this case I try to reload modules and get this bug. > # killall pulseaudio; rmmod snd_au8830 snd_hda_intel snd_seq snd_ac97_codec > ac97_bus snd_hda_codec snd_hwdep snd_mpu401_uart snd_pcm snd_rawmidi > snd_seq_device snd_pcm snd_timer snd_page_alloc snd > # modprobe snd_au8830 > [ 12.657863] snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > [ 12.659223] snd_hda_intel 0000:00:1b.0: irq 48 for MSI/MSI-X > [ 12.660521] snd_hda_intel 0000:00:1b.0: setting latency timer to 64 > [ 12.670024] ALSA sound/pci/hda/hda_intel.c:2837 no codecs found! > [ 12.671307] snd_hda_intel 0000:00:1b.0: PCI INT A disabled Does your hda-intel work properly? speaker-test -c 2 -t wav -Dhw:Intel any error message in system log before you unload the module, your output only show the error after reload ? does "aplay -l" alway show two cards for every reboot ? if your on-board hda-intel can only be loaded randomly, pulseaudio server may behave differently since default is on-board sound card >>> https://bugzilla.redhat.com/show_bug.cgi?id=788978 >> >> Sometimes my sound card hangs up (It do not produce sounds. Applications >> which try to play something hangs). As your au8830 have stac9708 Did you select surround40 or stereo analog output profile ? > APLAY > **** List of PLAYBACK Hardware Devices **** > card 0: au8830 [Aureal Vortex au8830], device 0: Aureal Vortex 2 ADB [au8830 adb] > Subdevices: 32/32 http://git.alsa-project.org/?p=alsa-kernel.git&a=search&h=HEAD&st=commit&s=… since you still have 32 playback subdevices did you have commit fb65c2dfe60d38be6b9193d0b85e66e780cd4373 without this patch, the front and rear can be played correct once and swapped afterwards on my au8830 speaker-test -c 4 -t wav -Dsurround40:au8830 Are you sure that pulseaudio server is still running ? if pulseaudio server crash but you still have an application using hw:au8830 it won't respwan because is_card_busy() in commit d6fb8d10819bebc1cee203de7330cceeafde9fed udev: check busy status of alsa cards before loading alsa modules and hence initiating profile probing Try speaker-test -c 2 -t wav -D hw:au8830 speaker-test -c 4 -t wav -D hw:au8830 Does you enable power saving or the computer enter suspend/resume cycle ? https://bugzilla.redhat.com/show_bug.cgi?id=789156 Sound au8830 does not work after standby mode >> Can you try Taskashi 's patch and reply back to alsa-developer mailing >> list > > The kernel doesn't panic with this patches, but a sound card doesn't work. > > The origin problem is that sometimes my sound card stops to play > anything. I try to reload the module to fix that, but it doesn't help. I > can provide any info which you need to investigate this problem. > > [root@dhcp-10-30-19-46 avagin]# rmmod snd_au8830 > [root@dhcp-10-30-19-46 avagin]# modprobe snd_au8830 > [root@dhcp-10-30-19-46 avagin]# dmesg > [ 3669.934145] snd_au8830 0000:06:00.0: PCI INT A -> GSI 21 (level, low) > -> IRQ 21 > [ 3669.934310] Vortex: init.... > [ 3670.251191] bad adb fifo reset! > [ 3670.251221] bad adb fifo reset! > [ 3670.251239] bad adb fifo reset! > [ 3670.251257] bad adb fifo reset! > [ 3670.251276] bad adb fifo reset! > [ 3670.251294] bad adb fifo reset! > [ 3670.251313] bad adb fifo reset! > [ 3670.251330] bad adb fifo reset! > [ 3670.251348] bad adb fifo reset! > [ 3670.251366] bad adb fifo reset! > [ 3670.251383] bad adb fifo reset! > [ 3670.251401] bad adb fifo reset! > [ 3670.251419] bad adb fifo reset! > [ 3670.251437] bad adb fifo reset! > [ 3670.251456] bad adb fifo reset! > [ 3670.251473] bad adb fifo reset! > [ 3670.251491] bad adb fifo reset! > [ 3670.251509] bad adb fifo reset! > [ 3670.251527] bad adb fifo reset! > [ 3670.251545] bad adb fifo reset! > [ 3670.251563] bad adb fifo reset! > [ 3670.251581] bad adb fifo reset! > [ 3670.251599] bad adb fifo reset! > [ 3670.251617] bad adb fifo reset! > [ 3670.251634] bad adb fifo reset! > [ 3670.251653] bad adb fifo reset! > [ 3670.251670] bad adb fifo reset! > [ 3670.251688] bad adb fifo reset! > [ 3670.251706] bad adb fifo reset! > [ 3670.251723] bad adb fifo reset! > [ 3670.251741] bad adb fifo reset! > [ 3670.251758] bad adb fifo reset! > [ 3670.251777] bad wt fifo reset (0x000160fc, 0xffffffff)! > [ 3670.251797] bad wt fifo reset (0x000160f8, 0xffffffff)! > [ 3670.251814] bad wt fifo reset (0x000160f4, 0xffffffff)! > [ 3670.251832] bad wt fifo reset (0x000160f0, 0xffffffff)! > [ 3670.251850] bad wt fifo reset (0x000160ec, 0xffffffff)! > [ 3670.251868] bad wt fifo reset (0x000160e8, 0xffffffff)! > [ 3670.251886] bad wt fifo reset (0x000160e4, 0xffffffff)! > [ 3670.251904] bad wt fifo reset (0x000160e0, 0xffffffff)! > [ 3670.251922] bad wt fifo reset (0x000160dc, 0xffffffff)! > [ 3670.251941] bad wt fifo reset (0x000160d8, 0xffffffff)! > [ 3670.251959] bad wt fifo reset (0x000160d4, 0xffffffff)! > [ 3670.251977] bad wt fifo reset (0x000160d0, 0xffffffff)! > [ 3670.251995] bad wt fifo reset (0x000160cc, 0xffffffff)! > [ 3670.252024] bad wt fifo reset (0x000160c8, 0xffffffff)! > [ 3670.252043] bad wt fifo reset (0x000160c4, 0xffffffff)! > [ 3670.252062] bad wt fifo reset (0x000160c0, 0xffffffff)! > [ 3670.252081] bad wt fifo reset (0x000160bc, 0xffffffff)! > [ 3670.252100] bad wt fifo reset (0x000160b8, 0xffffffff)! > [ 3670.252120] bad wt fifo reset (0x000160b4, 0xffffffff)! > [ 3670.252139] bad wt fifo reset (0x000160b0, 0xffffffff)! > [ 3670.252157] bad wt fifo reset (0x000160ac, 0xffffffff)! > [ 3670.252175] bad wt fifo reset (0x000160a8, 0xffffffff)! > [ 3670.252194] bad wt fifo reset (0x000160a4, 0xffffffff)! > [ 3670.252212] bad wt fifo reset (0x000160a0, 0xffffffff)! > [ 3670.252231] bad wt fifo reset (0x0001609c, 0xffffffff)! > [ 3670.252249] bad wt fifo reset (0x00016098, 0xffffffff)! > [ 3670.252267] bad wt fifo reset (0x00016094, 0xffffffff)! > [ 3670.252285] bad wt fifo reset (0x00016090, 0xffffffff)! > [ 3670.252303] bad wt fifo reset (0x0001608c, 0xffffffff)! > [ 3670.252322] bad wt fifo reset (0x00016088, 0xffffffff)! > [ 3670.252342] bad wt fifo reset (0x00016084, 0xffffffff)! > [ 3670.252361] bad wt fifo reset (0x00016080, 0xffffffff)! > [ 3670.252380] bad wt fifo reset (0x0001607c, 0xffffffff)! > [ 3670.252398] bad wt fifo reset (0x00016078, 0xffffffff)! > [ 3670.252416] bad wt fifo reset (0x00016074, 0xffffffff)! > [ 3670.252434] bad wt fifo reset (0x00016070, 0xffffffff)! > [ 3670.252452] bad wt fifo reset (0x0001606c, 0xffffffff)! > [ 3670.252470] bad wt fifo reset (0x00016068, 0xffffffff)! > [ 3670.252488] bad wt fifo reset (0x00016064, 0xffffffff)! > [ 3670.252506] bad wt fifo reset (0x00016060, 0xffffffff)! > [ 3670.252524] bad wt fifo reset (0x0001605c, 0xffffffff)! > [ 3670.252542] bad wt fifo reset (0x00016058, 0xffffffff)! > [ 3670.252560] bad wt fifo reset (0x00016054, 0xffffffff)! > [ 3670.252578] bad wt fifo reset (0x00016050, 0xffffffff)! > [ 3670.252596] bad wt fifo reset (0x0001604c, 0xffffffff)! > [ 3670.252614] bad wt fifo reset (0x00016048, 0xffffffff)! > [ 3670.252631] bad wt fifo reset (0x00016044, 0xffffffff)! > [ 3670.252649] bad wt fifo reset (0x00016040, 0xffffffff)! > [ 3670.252667] bad wt fifo reset (0x0001603c, 0xffffffff)! > [ 3670.252686] bad wt fifo reset (0x00016038, 0xffffffff)! > [ 3670.252704] bad wt fifo reset (0x00016034, 0xffffffff)! > [ 3670.252722] bad wt fifo reset (0x00016030, 0xffffffff)! > [ 3670.252740] bad wt fifo reset (0x0001602c, 0xffffffff)! > [ 3670.252758] bad wt fifo reset (0x00016028, 0xffffffff)! > [ 3670.252776] bad wt fifo reset (0x00016024, 0xffffffff)! > [ 3670.252794] bad wt fifo reset (0x00016020, 0xffffffff)! > [ 3670.252813] bad wt fifo reset (0x0001601c, 0xffffffff)! > [ 3670.252831] bad wt fifo reset (0x00016018, 0xffffffff)! > [ 3670.252851] bad wt fifo reset (0x00016014, 0xffffffff)! > [ 3670.252868] bad wt fifo reset (0x00016010, 0xffffffff)! > [ 3670.252886] bad wt fifo reset (0x0001600c, 0xffffffff)! > [ 3670.252904] bad wt fifo reset (0x00016008, 0xffffffff)! > [ 3670.252922] bad wt fifo reset (0x00016004, 0xffffffff)! > [ 3670.252940] bad wt fifo reset (0x00016000, 0xffffffff)! > [ 3670.254466] ALSA sound/core/control.c:344 control 3:0:0:Playback PCM > advanced processing:0 is already present > [ 3670.254580] done. > [ 3670.356981] vortex: ac97 address never arrived > [ 3670.457979] vortex: ac97 address never arrived > [ 3670.559026] vortex: ac97 address never arrived > [ 3670.660022] vortex: ac97 address never arrived > [ 3670.761005] vortex: ac97 address never arrived > [ 3670.861991] vortex: ac97 address never arrived > [ 3670.962974] vortex: ac97 address never arrived > [ 3671.063931] vortex: ac97 address never arrived > [ 3671.164882] vortex: ac97 address never arrived > [ 3671.265897] vortex: ac97 address never arrived > [ 3671.367023] vortex: ac97 address never arrived > [ 3671.468052] vortex: ac97 address never arrived > [ 3671.569092] vortex: ac97 address never arrived > [ 3671.670173] vortex: ac97 address never arrived > [ 3671.771195] vortex: ac97 address never arrived > [ 3671.772029] ALSA sound/pci/ac97/ac97_codec.c:2089 AC'97 0 does not > respond - RESET > [ 3671.872965] vortex: ac97 address never arrived > [ 3671.973942] vortex: ac97 address never arrived > [ 3671.973948] ALSA sound/pci/ac97/ac97_codec.c:2098 AC'97 0 access is > not valid [0xffffffff], removing mixer. > [ 3671.974774] vortex: IRQ fatal error > [ 3671.974780] vortex: IRQ parity error > [ 3671.974783] vortex: IRQ reg error > [ 3671.974785] vortex: IRQ fifo error > [ 3671.974787] vortex: IRQ dma error > [ 3671.974792] Vortex: shutdown... > [ 3671.974828] vortex: FATAL: ResManager: resource type 3 exhausted. > [ 3671.974832] vortex: FATAL: ResManager: resource type 3 exhausted. > [ 3671.974838] vortex_adb_delroutes: route not found! 0x24a0 > [ 3671.974854] vortex: srcdelWTD: error, lifeboat overflow > [ 3671.974859] vortex_adb_delroutes: route not found! 0x24a1 > [ 3671.974874] vortex: srcdelWTD: error, lifeboat overflow > [ 3671.974881] vortex_adb_delroutes: route not found! 0x7e88 > [ 3671.974885] vortex_adb_delroutes: route not found! 0x7f89 > [ 3671.974888] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974891] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974895] vortex_adb_delroutes: route not found! 0x7044 > [ 3671.974899] vortex_adb_delroutes: route not found! 0x7144 > [ 3671.974902] vortex: FATAL: ResManager: resource type 3 exhausted. > [ 3671.974905] vortex: FATAL: ResManager: resource type 3 exhausted. > [ 3671.974909] vortex_adb_delroutes: route not found! 0x2492 > [ 3671.974924] vortex: srcdelWTD: error, lifeboat overflow > [ 3671.974930] vortex_adb_delroutes: route not found! 0x2493 > [ 3671.974945] vortex: srcdelWTD: error, lifeboat overflow > [ 3671.974949] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974954] vortex_adb_delroutes: route not found! 0x6044 > [ 3671.974959] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974963] vortex_adb_delroutes: route not found! 0x6144 > [ 3671.974968] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974972] vortex_adb_delroutes: route not found! 0x6244 > [ 3671.974977] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974981] vortex_adb_delroutes: route not found! 0x6344 > [ 3671.974985] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974990] vortex_adb_delroutes: route not found! 0x6444 > [ 3671.974994] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.974999] vortex_adb_delroutes: route not found! 0x6544 > [ 3671.975001] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975001] vortex_adb_delroutes: route not found! 0xa044 > [ 3671.975024] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975030] vortex_adb_delroutes: route not found! 0xa144 > [ 3671.975035] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975039] vortex_adb_delroutes: route not found! 0xa244 > [ 3671.975044] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975050] vortex_adb_delroutes: route not found! 0xa344 > [ 3671.975055] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975060] vortex_adb_delroutes: route not found! 0xa444 > [ 3671.975066] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975071] vortex_adb_delroutes: route not found! 0xa544 > [ 3671.975090] vortex: FATAL: ResManager: resource type 2 exhausted. > [ 3671.975094] vortex: vortex_Vort3D: ERROR: not enough free mixer > resources. > [ 3671.975099] bad adb fifo reset! > [ 3671.975117] bad adb fifo reset! > [ 3671.975135] bad adb fifo reset! > [ 3671.975153] bad adb fifo reset! > [ 3671.975172] bad adb fifo reset! > [ 3671.975190] bad adb fifo reset! > [ 3671.975207] bad adb fifo reset! > [ 3671.975226] bad adb fifo reset! > [ 3671.975246] bad adb fifo reset! > [ 3671.975264] bad adb fifo reset! > [ 3671.975283] bad adb fifo reset! > [ 3671.975301] bad adb fifo reset! > [ 3671.975319] bad adb fifo reset! > [ 3671.975337] bad adb fifo reset! > [ 3671.975355] bad adb fifo reset! > [ 3671.975373] bad adb fifo reset! > [ 3671.975390] bad adb fifo reset! > [ 3671.975408] bad adb fifo reset! > [ 3671.975428] bad adb fifo reset! > [ 3671.975446] bad adb fifo reset! > [ 3671.975464] bad adb fifo reset! > [ 3671.975482] bad adb fifo reset! > [ 3671.975500] bad adb fifo reset! > [ 3671.975518] bad adb fifo reset! > [ 3671.975536] bad adb fifo reset! > [ 3671.975554] bad adb fifo reset! > [ 3671.975571] bad adb fifo reset! > [ 3671.975588] bad adb fifo reset! > [ 3671.975607] bad adb fifo reset! > [ 3671.975625] bad adb fifo reset! > [ 3671.975643] bad adb fifo reset! > [ 3671.975662] bad adb fifo reset! > [ 3671.975681] bad wt fifo reset (0x000160fc, 0xffffffff)! > [ 3671.975700] bad wt fifo reset (0x000160f8, 0xffffffff)! > [ 3671.975718] bad wt fifo reset (0x000160f4, 0xffffffff)! > [ 3671.975736] bad wt fifo reset (0x000160f0, 0xffffffff)! > [ 3671.975754] bad wt fifo reset (0x000160ec, 0xffffffff)! > [ 3671.975772] bad wt fifo reset (0x000160e8, 0xffffffff)! > [ 3671.975790] bad wt fifo reset (0x000160e4, 0xffffffff)! > [ 3671.975808] bad wt fifo reset (0x000160e0, 0xffffffff)! > [ 3671.975826] bad wt fifo reset (0x000160dc, 0xffffffff)! > [ 3671.975844] bad wt fifo reset (0x000160d8, 0xffffffff)! > [ 3671.975861] bad wt fifo reset (0x000160d4, 0xffffffff)! > [ 3671.975879] bad wt fifo reset (0x000160d0, 0xffffffff)! > [ 3671.975897] bad wt fifo reset (0x000160cc, 0xffffffff)! > [ 3671.975915] bad wt fifo reset (0x000160c8, 0xffffffff)! > [ 3671.975933] bad wt fifo reset (0x000160c4, 0xffffffff)! > [ 3671.975950] bad wt fifo reset (0x000160c0, 0xffffffff)! > [ 3671.975968] bad wt fifo reset (0x000160bc, 0xffffffff)! > [ 3671.975987] bad wt fifo reset (0x000160b8, 0xffffffff)! > [ 3671.976012] bad wt fifo reset (0x000160b4, 0xffffffff)! > [ 3671.976032] bad wt fifo reset (0x000160b0, 0xffffffff)! > [ 3671.976051] bad wt fifo reset (0x000160ac, 0xffffffff)! > [ 3671.976070] bad wt fifo reset (0x000160a8, 0xffffffff)! > [ 3671.976089] bad wt fifo reset (0x000160a4, 0xffffffff)! > [ 3671.976107] bad wt fifo reset (0x000160a0, 0xffffffff)! > [ 3671.976127] bad wt fifo reset (0x0001609c, 0xffffffff)! > [ 3671.976144] bad wt fifo reset (0x00016098, 0xffffffff)! > [ 3671.976163] bad wt fifo reset (0x00016094, 0xffffffff)! > [ 3671.976181] bad wt fifo reset (0x00016090, 0xffffffff)! > [ 3671.976199] bad wt fifo reset (0x0001608c, 0xffffffff)! > [ 3671.976217] bad wt fifo reset (0x00016088, 0xffffffff)! > [ 3671.976236] bad wt fifo reset (0x00016084, 0xffffffff)! > [ 3671.976255] bad wt fifo reset (0x00016080, 0xffffffff)! > [ 3671.976274] bad wt fifo reset (0x0001607c, 0xffffffff)! > [ 3671.976292] bad wt fifo reset (0x00016078, 0xffffffff)! > [ 3671.976310] bad wt fifo reset (0x00016074, 0xffffffff)! > [ 3671.976328] bad wt fifo reset (0x00016070, 0xffffffff)! > [ 3671.976346] bad wt fifo reset (0x0001606c, 0xffffffff)! > [ 3671.976364] bad wt fifo reset (0x00016068, 0xffffffff)! > [ 3671.976382] bad wt fifo reset (0x00016064, 0xffffffff)! > [ 3671.976400] bad wt fifo reset (0x00016060, 0xffffffff)! > [ 3671.976418] bad wt fifo reset (0x0001605c, 0xffffffff)! > [ 3671.976436] bad wt fifo reset (0x00016058, 0xffffffff)! > [ 3671.976454] bad wt fifo reset (0x00016054, 0xffffffff)! > [ 3671.976472] bad wt fifo reset (0x00016050, 0xffffffff)! > [ 3671.976490] bad wt fifo reset (0x0001604c, 0xffffffff)! > [ 3671.976508] bad wt fifo reset (0x00016048, 0xffffffff)! > [ 3671.976526] bad wt fifo reset (0x00016044, 0xffffffff)! > [ 3671.976546] bad wt fifo reset (0x00016040, 0xffffffff)! > [ 3671.976564] bad wt fifo reset (0x0001603c, 0xffffffff)! > [ 3671.976582] bad wt fifo reset (0x00016038, 0xffffffff)! > [ 3671.976600] bad wt fifo reset (0x00016034, 0xffffffff)! > [ 3671.976618] bad wt fifo reset (0x00016030, 0xffffffff)! > [ 3671.976636] bad wt fifo reset (0x0001602c, 0xffffffff)! > [ 3671.976654] bad wt fifo reset (0x00016028, 0xffffffff)! > [ 3671.976672] bad wt fifo reset (0x00016024, 0xffffffff)! > [ 3671.976690] bad wt fifo reset (0x00016020, 0xffffffff)! > [ 3671.976708] bad wt fifo reset (0x0001601c, 0xffffffff)! > [ 3671.976727] bad wt fifo reset (0x00016018, 0xffffffff)! > [ 3671.976745] bad wt fifo reset (0x00016014, 0xffffffff)! > [ 3671.976763] bad wt fifo reset (0x00016010, 0xffffffff)! > [ 3671.976781] bad wt fifo reset (0x0001600c, 0xffffffff)! > [ 3671.976799] bad wt fifo reset (0x00016008, 0xffffffff)! > [ 3671.976817] bad wt fifo reset (0x00016004, 0xffffffff)! > [ 3671.976835] bad wt fifo reset (0x00016000, 0xffffffff)! > [ 3671.983010] done. > [ 3671.983097] snd_au8830 0000:06:00.0: PCI INT A disabled > [ 3671.983110] snd_au8830: probe of 0000:06:00.0 failed with error -5 > > > On 02/28/2012 12:36 PM, Raymond Yau wrote: >> >> >> At Tue, 28 Feb 2012 11:04:07 +0800, >> Raymond Yau wrote: >>> 2012/1/16, Takashi Iwai<tiwai(a)suse.de>: >>>> At Fri, 13 Jan 2012 18:47:45 +0800, >>>> Raymond Yau wrote: >>>>> [PATCH - ALSA 1/2] Au88x0 - Support 4 channels playback when AC97 >>>>> codecs has SDAC bit >>>>> >>>>> - Check SDAC bit of AC97 codec for supporting 4 channels playback. >>>>> >>>>> >>>>> [PATCH - ALSA 2/2] Au88x0 - Reduce the number of playback subdevices >>>>> of au8830 from 32 to 16 >>>>> >>>>> - The maximum number of playback streams depend on the number of sample >>>>> rate conveters (16) and the number of DMA channels (32). >>>> Applied both patches. Thanks. >>>> >>>> >>> Does alsa require the driver to provide private_free for ac97_bus and >>> ac97 ? >> No, and it must be irrelevant. >> >>> There is a bug report >>> >>> snd_vortex_dev_free() BUG: unable to handle kernel NULL pointer >>> >>> https://bugzilla.redhat.com/show_bug.cgi?id=788978 >>> >>> It seem that snd_ac97_bus() fail >> Or something else. au88x0 is built in a strange way, sourcing all >> *.c files into one, so we can't see the exact point. >> >> In anyway, the Oops isn't because the probing fails but due to >> concurrent accesses while removing irq, etc. >> Maybe a patch like below can help. >> when there is more than one sound card and I need to specify index to all sound cards in order to let them loaded in the specified order especially when I have my usb webcam plugged just specify index=0 in one sound card is not enough and some of them may failed at snd_card_create() without any error mesasge in system log >> >>> However I cannot reproduce the bug in 2.6 kernel, >>> >>> Do 3.x kernel require mixer created after pcm devices ? >> No. >> >> >> Takashi >> >>> :Call Trace: >>> : [<ffffffff810dec3d>] __free_irq+0x17d/0x220 >>> : [<ffffffff810dee65>] free_irq+0x55/0xd0 >>> : [<ffffffffa04e2d64>] snd_vortex_dev_free+0x44/0x80 [snd_au8830] >>> : [<ffffffffa01c0aa1>] snd_device_free+0x71/0x170 [snd] >>> : [<ffffffffa01c0f51>] snd_device_free_all+0x61/0xb0 [snd] >>> : [<ffffffffa01b9908>] snd_card_do_free+0x58/0x140 [snd] >>> : [<ffffffffa01badf3>] snd_card_free+0xa3/0xc0 [snd] >>> : [<ffffffffa0261b8f>] ? snd_ac97_bus+0xaf/0x130 [snd_ac97_codec] >>> : [<ffffffffa04e6a9f>] snd_vortex_probe+0x1c40/0x1d09 [snd_au8830] >>> : [<ffffffff812db37c>] local_pci_probe+0x5c/0xd0 >>> : [<ffffffff812dcc19>] pci_device_probe+0x109/0x130 >>> : [<ffffffff813932cc>] driver_probe_device+0x9c/0x300 >>> : [<ffffffff813935db>] __driver_attach+0xab/0xb0 >>> : [<ffffffff81393530>] ? driver_probe_device+0x300/0x300 >>> : [<ffffffff81393530>] ? driver_probe_device+0x300/0x300 >>> : [<ffffffff8139214c>] bus_for_each_dev+0x5c/0x90 >>> : [<ffffffff81392ede>] driver_attach+0x1e/0x20 >>> : [<ffffffff81392ae0>] bus_add_driver+0x1b0/0x2a0 >>> : [<ffffffffa04ed000>] ? 0xffffffffa04ecfff >>> : [<ffffffff81393b36>] driver_register+0x76/0x140 >>> : [<ffffffff815e554d>] ? notifier_call_chain+0x4d/0x70 >>> : [<ffffffffa04ed000>] ? 0xffffffffa04ecfff >>> : [<ffffffff812dc8f6>] __pci_register_driver+0x56/0xd0 >>> : [<ffffffff81041673>] ? set_memory_nx+0x43/0x50 >>> : [<ffffffffa04ed01e>] alsa_card_vortex_init+0x1e/0x1000 [snd_au8830] >>> : [<ffffffff81002042>] do_one_initcall+0x42/0x180 >>> : [<ffffffff810ac970>] sys_init_module+0xc0/0x230 >>> : [<ffffffff815e9782>] system_call_fastpath+0x16/0x1b >> --- >> diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c >> index f13ad53..506d2fc 100644 >> --- a/sound/pci/au88x0/au88x0.c >> +++ b/sound/pci/au88x0/au88x0.c >> @@ -123,10 +123,10 @@ static int snd_vortex_dev_free(struct snd_device >> *device) >> { >> vortex_t *vortex = device->device_data; >> >> + free_irq(vortex->irq, vortex); >> vortex_gameport_unregister(vortex); >> vortex_core_shutdown(vortex); >> // Take down PCI interface. >> - free_irq(vortex->irq, vortex); >> iounmap(vortex->mmio); >> pci_release_regions(vortex->pci_dev); >> pci_disable_device(vortex->pci_dev); >> diff --git a/sound/pci/au88x0/au88x0_core.c >> b/sound/pci/au88x0/au88x0_core.c >> index 1181c5e..525f881 100644 >> --- a/sound/pci/au88x0/au88x0_core.c >> +++ b/sound/pci/au88x0/au88x0_core.c >> @@ -2477,7 +2477,7 @@ static irqreturn_t vortex_interrupt(int irq, void >> *dev_id) >> hwread(vortex->mmio, VORTEX_IRQ_STAT); >> handled = 1; >> } >> - if (source& IRQ_MIDI) { >> + if ((source& IRQ_MIDI)&& vortex->rmidi) { >> snd_mpu401_uart_interrupt(vortex->irq, >> vortex->rmidi->private_data); >> handled = 1; > >

1 0

Re: [alsa-devel] sound: BUG: unable to handle kernel irq event stamp: 2013419
by Takashi Iwai 15 Mar '12

15 Mar '12

At Thu, 15 Mar 2012 17:50:55 +0200, Sasha Levin wrote: > > On Thu, Mar 15, 2012 at 5:46 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 15 Mar 2012 17:36:44 +0200, > > Sasha Levin wrote: > >> > >> On Thu, Mar 15, 2012 at 5:34 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > The bug seems coming from the old OSS driver. Do you enable any OSS > >> > drivers? > >> > >> They're most likely compiled in (kernel is pretty close to a 'make > >> allyesconfig'), but I haven't done anything in particular to start > >> them. > > > > What does trinity at all? > > It's a syscall fuzzer which can be found here: > codemonkey.org.uk/projects/trinity/trinity.git OK. Then, you might have hit a virtual midi device or such. It's present even if you have no real device. Takashi

1 0

Re: [alsa-devel] sound: BUG: unable to handle kernel irq event stamp: 2013419
by Takashi Iwai 15 Mar '12

15 Mar '12

At Thu, 15 Mar 2012 17:36:44 +0200, Sasha Levin wrote: > > On Thu, Mar 15, 2012 at 5:34 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > The bug seems coming from the old OSS driver. Do you enable any OSS > > drivers? > > They're most likely compiled in (kernel is pretty close to a 'make > allyesconfig'), but I haven't done anything in particular to start > them. What does trinity at all? Takashi

1 0

Re: [alsa-devel] sound: BUG: unable to handle kernel irq event stamp: 2013419
by Takashi Iwai 15 Mar '12

15 Mar '12

At Thu, 15 Mar 2012 17:25:06 +0200, Sasha Levin wrote: > > Hi all, > > I got the following kernel panic when running the trinity fuzzer in a > KVM tool guest, using today's linux-next. > > Please note that the guest doesn't have anything which even remotely > reminds a multimedia device of any sort, although most of the drivers > are compiled into the kernel. > > [ 145.747790] BUG: unable to handle kernel irq event stamp: 2013419 > [ 145.744076] hardirqs last enabled at (2013419): > [<ffffffff811582d8>] __call_rcu+0x128/0x2c0 > [ 145.744076] hardirqs last disabled at (2013418): > [<ffffffff81158247>] __call_rcu+0x97/0x2c0 > [ 145.744076] softirqs last enabled at (2013156): > [<ffffffff810b9a93>] __do_softirq+0x133/0x180 > [ 145.744076] softirqs last disabled at (2013139): > [<ffffffff8270c96c>] call_softirq+0x1c/0x30 > [ 145.748761] paging request at ffffc9000279e000 > [ 145.748761] IP: [<ffffffff821474cb>] midi_poll+0x6b/0x180 > [ 145.748761] PGD 3c9bc067 PUD 3c9bd067 PMD 36b5c067 PTE 0 > [ 145.748761] Oops: 0000 [#1] PREEMPT SMP DEBUG_PAGEALLOC > [ 145.748761] CPU 0 > [ 145.748761] Pid: 3129, comm: trinity Tainted: G W > 3.3.0-rc7-next-20120315-sasha-00002-g91cfd71 #51 > [ 145.748761] RIP: 0010:[<ffffffff821474cb>] [<ffffffff821474cb>] > midi_poll+0x6b/0x180 > [ 145.784811] RSP: 0018:ffff88003d603dc0 EFLAGS: 00010086 > [ 145.784811] RAX: 0000000000000000 RBX: 0000000000000282 RCX: ffffc9000279e000 > [ 145.792008] RDX: ffff880034ad0000 RSI: 0000000000000000 RDI: 0000000000000001 > [ 145.792008] RBP: ffff88003d603df0 R08: 0000000000000001 R09: 0000000000000000 > [ 145.792008] R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000001 > [ 145.792008] R13: 0000000010624dd3 R14: 0000000000000001 R15: 0000000000000000 > [ 145.792008] FS: 00007f2af88d0700(0000) GS:ffff88003d600000(0000) > knlGS:0000000000000000 > [ 145.792008] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > [ 145.792008] CR2: ffffc9000279e000 CR3: 0000000035195000 CR4: 00000000000406f0 > [ 145.792008] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 > [ 145.792008] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 > [ 145.792008] Process trinity (pid: 3129, threadinfo > ffff880034aca000, task ffff880034ad0000) > [ 145.792008] Stack: > [ 145.792008] ffff880034ad0000 ffff880034acbfd8 ffff88003d603e10 > 0000000000000102 > [ 145.792008] ffffffff82147460 0000000000000000 ffff88003d603e70 > ffffffff810c0c2a > [ 145.792008] ffffffff810c0ba0 ffff88003d603ea0 ffffffff8300abec > ffffffff83d2e370 > [ 145.792008] Call Trace: > [ 145.792008] <IRQ> > [ 145.792008] [<ffffffff82147460>] ? midi_input_intr+0xb0/0xb0 > [ 145.792008] [<ffffffff810c0c2a>] call_timer_fn.clone.11+0x8a/0x110 > [ 145.792008] [<ffffffff810c0ba0>] ? process_timeout+0x10/0x10 > [ 145.792008] [<ffffffff82147460>] ? midi_input_intr+0xb0/0xb0 > [ 145.792008] [<ffffffff810c0efa>] run_timer_softirq+0x13a/0x220 > [ 145.792008] [<ffffffff810b9a28>] __do_softirq+0xc8/0x180 > [ 145.792008] [<ffffffff811fce9d>] ? dput+0x13d/0x1d0 > [ 145.792008] [<ffffffff8270c96c>] call_softirq+0x1c/0x30 > [ 145.792008] [<ffffffff8104fc05>] do_softirq+0xc5/0x100 > [ 145.792008] [<ffffffff810b9e9e>] irq_exit+0xae/0xe0 > [ 145.792008] [<ffffffff810719e8>] smp_apic_timer_interrupt+0x68/0xa0 > [ 145.792008] [<ffffffff8270bf6f>] apic_timer_interrupt+0x6f/0x80 > [ 145.792008] <EOI> > [ 145.792008] [<ffffffff8270a9f4>] ? retint_restore_args+0x13/0x13 > [ 145.792008] [<ffffffff81119528>] ? lock_release+0x98/0xc0 > [ 145.792008] [<ffffffff82709b6e>] _raw_spin_unlock+0x1e/0x60 > [ 145.792008] [<ffffffff811fce9d>] dput+0x13d/0x1d0 > [ 145.792008] [<ffffffff811f35e1>] do_last+0x461/0x790 > [ 145.792008] [<ffffffff811f39e7>] path_openat+0xd7/0x500 > [ 145.792008] [<ffffffff810ec458>] ? sched_clock_cpu+0xc8/0x140 > [ 145.792008] [<ffffffff811f3e54>] do_filp_open+0x44/0xa0 > [ 145.792008] [<ffffffff82709b80>] ? _raw_spin_unlock+0x30/0x60 > [ 145.792008] [<ffffffff81203a36>] ? alloc_fd+0x176/0x240 > [ 145.792008] [<ffffffff811e2c8d>] do_sys_open+0xfd/0x1d0 > [ 145.792008] [<ffffffff811e2d7c>] sys_open+0x1c/0x20 > [ 145.792008] [<ffffffff8270b2bd>] system_call_fastpath+0x1a/0x1f > [ 145.792008] Code: bd d3 4d 62 10 0f 1f 00 4d 63 f4 4a 83 3c f5 e0 > 32 90 84 00 0f 84 ad 00 00 00 4a 8b 0c f5 a0 5b 90 84 48 85 c9 0f 84 > 9c 00 00 00 <8b> 01 85 c0 0f 84 e3 00 00 00 8b 51 04 eb 3d 66 0f 1f 44 > 00 00 > [ 145.792008] RIP [<ffffffff821474cb>] midi_poll+0x6b/0x180 > [ 145.792008] RSP <ffff88003d603dc0> > [ 145.792008] CR2: ffffc9000279e000 > [ 145.792008] ---[ end trace 31f7a567e17c1567 ]--- > [ 145.792008] Kernel panic - not syncing: Fatal exception in interrupt > [ 145.792008] Rebooting in 1 seconds..\ The bug seems coming from the old OSS driver. Do you enable any OSS drivers? Takashi

1 0

[alsa-devel] [PATCH v4] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 15 Mar '12

15 Mar '12

Changes since v3: * Change imx compatible to "fsl,imx21-ssi" * Leave locking issue as a FIXME for Timur to sort out separately --- Shawn Guo (10): ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: rename imx-pcm Kconfig options and filename ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: remove helper fsl_asoc_get_codec_dev_name ASoC: fsl: check property 'compatible' for the machine name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: remove the fatal error checking on codec-handle ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 1 + arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 ++++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 140 +++++++++++---- sound/soc/fsl/fsl_utils.c | 91 ++++++++++ sound/soc/fsl/fsl_utils.h | 26 +++ sound/soc/{imx => fsl}/imx-audmux.c | 0 sound/soc/{imx => fsl}/imx-audmux.h | 0 .../{imx/imx-pcm-dma-mx2.c => fsl/imx-pcm-dma.c} | 0 sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/{imx => fsl}/imx-pcm.c | 0 sound/soc/{imx => fsl}/imx-pcm.h | 0 sound/soc/fsl/imx-sgtl5000.c | 177 ++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 2 +- sound/soc/{imx => fsl}/imx-ssi.h | 0 sound/soc/fsl/mpc8610_hpcd.c | 168 ++---------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 0 sound/soc/fsl/p1022_ds.c | 190 ++------------------ sound/soc/{imx => fsl}/phycore-ac97.c | 0 sound/soc/{imx => fsl}/wm1133-ev1.c | 0 sound/soc/imx/Kconfig | 79 -------- sound/soc/imx/Makefile | 22 --- 28 files changed, 604 insertions(+), 475 deletions(-)

4 18

[alsa-devel] [PATCH 1/2] ASoC: core: Fix obscure leak of runtime array
by Mark Brown 15 Mar '12

15 Mar '12

We're currently not freeing card->rtd in cases where the card is unregistered before being registered - convert it to devm_kzalloc() to make sure that happens. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9a7ca0e..b0f8666 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1694,7 +1694,6 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_free(&card->dapm); - kfree(card->rtd); snd_card_free(card->snd_card); return 0; @@ -3120,9 +3119,10 @@ int snd_soc_register_card(struct snd_soc_card *card) soc_init_card_debugfs(card); - card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * - (card->num_links + card->num_aux_devs), - GFP_KERNEL); + card->rtd = devm_kzalloc(card->dev, + sizeof(struct snd_soc_pcm_runtime) * + (card->num_links + card->num_aux_devs), + GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; card->rtd_aux = &card->rtd[card->num_links]; -- 1.7.9.1

2 2

[alsa-devel] [PATCH] ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
by Mark Brown 15 Mar '12

15 Mar '12

Allows the constraint lists to be declared const by drivers which seems reasonable; there's plenty of other constification we could do if we were being complete but this was easy and quick. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 5 +++-- sound/core/pcm_lib.c | 3 ++- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d58d79..0d11128 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -264,7 +264,7 @@ struct snd_pcm_hw_constraint_ratdens { struct snd_pcm_hw_constraint_list { unsigned int count; - unsigned int *list; + const unsigned int *list; unsigned int mask; }; @@ -785,7 +785,8 @@ void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interva unsigned int k, struct snd_interval *c); void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, const struct snd_interval *b, struct snd_interval *c); -int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask); +int snd_interval_list(struct snd_interval *i, unsigned int count, + const unsigned int *list, unsigned int mask); int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3420bd3..4d18941 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1029,7 +1029,8 @@ static int snd_interval_ratden(struct snd_interval *i, * * Returns non-zero if the value is changed, zero if not changed. */ -int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask) +int snd_interval_list(struct snd_interval *i, unsigned int count, + const unsigned int *list, unsigned int mask) { unsigned int k; struct snd_interval list_range; -- 1.7.9.1

2 1

[alsa-devel] ASoC updates for 3.4
by Mark Brown 15 Mar '12

15 Mar '12

The following changes since commit 303076342b8a30aa98f8a53b539155d0b81ff0c3: Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc (2012-03-06 14:04:16 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 5ec65ee589fdaca7298b6303fd74ad6c121a8f38: ASoC: mx27vis-aic32x4: Convert it to platform driver (2012-03-13 23:09:59 +0000) ---------------------------------------------------------------- Linus decided to go for another week so here's a few more updates - a mixed bag here, a few minor diagnostic tweaks, some driver enhancements and the dmaengine conversion for ep93xx drivers which was tested a while ago and just waiting for a signoff. ---------------------------------------------------------------- Ashish Chavan (2): ASoC: da7210: Update for using I2C regmap ASoC: da7210: Remove extra registers from defaults list Fabio Estevam (2): ASoC: imx: imx-audmux: Fix section mismatch ASoC: mx27vis-aic32x4: Convert it to platform driver Lars-Peter Clausen (1): ASoC: ep93xx-pcm: Use dmaengine PCM helper functions Liam Girdwood (3): ASoC: dapm: Use dev_warn for debugfs warning message ASoC: core: Add platform component mutex ASoC: DAPM: Make sure DAPM widget IO ops hold the component mutex Mark Brown (5): ASoC: dapm: Only lock CODEC for I/O if not using regmap ASoC: wm8996: Remove some volatile regisers from the defaults table ASoC: wm8996: Remove separate output stage enable step ASoC: Revert widget I/O locking for 3.4 ASoC: wm8994: Prevent ABBA deadlock with CODEC and accdet mutexes Mika Westerberg (1): ASoC: dmaengine_pcm: Reset pointer position when starting a stream Qiao Zhou (1): ASoC: add more sample rate for pxa-ssp Shawn Guo (2): ASoC: core: missing set_fmt should not be complaint ASoC: fsl: check property 'compatible' for the machine name Stephen Warren (1): ASoC: tegra: Remove unused variable arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 1 + include/sound/soc.h | 1 + sound/soc/codecs/da7210.c | 146 +++++++++++++++++++------- sound/soc/codecs/wm8994.c | 69 ++++++------- sound/soc/codecs/wm8996.c | 34 ++---- sound/soc/ep93xx/Kconfig | 1 + sound/soc/ep93xx/ep93xx-pcm.c | 148 ++++----------------------- sound/soc/fsl/fsl_ssi.c | 6 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 32 +----- sound/soc/imx/imx-audmux.c | 8 +- sound/soc/imx/mx27vis-aic32x4.c | 41 ++++---- sound/soc/pxa/pxa-ssp.c | 3 +- sound/soc/soc-core.c | 12 ++- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-dmaengine-pcm.c | 1 + sound/soc/tegra/tegra_alc5632.c | 1 - 17 files changed, 225 insertions(+), 283 deletions(-)

2 1

[alsa-devel] [RFC] ymfpci - re-enable legacy audio after suspend resume
by Raymond Yau 15 Mar '12

15 Mar '12

Since the Legacy Audio Control register is reset to default after suspend resume cycle need to enable the Legacy Audio Control Bit in order to use OPL3 FM Synth however it is still need to use sbiload instrument loader to reload the instruments

2 8

[alsa-devel] sending a sequencer event to a delayed queue
by Henning Thielemann 14 Mar '12

14 Mar '12

Unfortunately the ALSA doc does not say, what happens if I send an event to a queue that is not running. I expected that the event is being delivered once the queue is started. But it seems that instead the event is dropped. What is the reason for that behaviour? What I want to do is the following: I have a program that computes a sequence of events in real-time. Since computation of an event requires varying amount of time, I want to add a latency in order to get a correctly timed output. To this end I set up two queues: One queue triggers computation of events via Echo messages and the other queue delivers the computed events. I want to use the same time stamps for both queues and just start the queues with a delay between them. Why is it not possible to use one queue and add latency manually to the timestamps? Because the user of the program shall be able to halt the computation. Thus the computation must be halted first and the delivery of events must be halted later. I cannot achieve this with one queue and a queue_stop command. Now the problem is: When my program starts it computes the first event and sends it to the second (delayed) queue. But since this queue is started only after a delay, it is not yet running and thus my event is not delayed, but lost. :-( How to solve that problem? Is there a proposed way to manage latency in ALSA sequencing? Regards, Henning

3 12

[alsa-devel] [PATCH 1/2] ASoC: wm8996: Add 44.1kHz support
by Mark Brown 14 Mar '12

14 Mar '12

The WM8996 specification has been updated to specify 44.1kHz as a supported sample rate. Update the driver to accept this configuration. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 19 ++++++++++--------- 1 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 40a124c..1fd6354 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1914,7 +1914,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - int bits, i, bclk_rate; + int bits, i, bclk_rate, best; int aifdata = 0; int lrclk = 0; int dsp = 0; @@ -1963,14 +1963,11 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, return bits; aifdata |= (bits << WM8996_AIF1TX_WL_SHIFT) | bits; + best = 0; for (i = 0; i < ARRAY_SIZE(dsp_divs); i++) { - if (dsp_divs[i] == params_rate(params)) - break; - } - if (i == ARRAY_SIZE(dsp_divs)) { - dev_err(codec->dev, "Unsupported sample rate %dHz\n", - params_rate(params)); - return -EINVAL; + if (abs(dsp_divs[i] - params_rate(params)) < + abs(dsp_divs[best] - params_rate(params))) + best = i; } dsp |= i << dsp_shift; @@ -2030,13 +2027,16 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, } switch (wm8996->sysclk) { + case 5644800: case 6144000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, 0); break; + case 22579200: case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; + case 11289600: case 12288000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE); @@ -3060,7 +3060,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { }; #define WM8996_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) #define WM8996_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -- 1.7.9.1

1 1

[alsa-devel] snd_seq_parse_address matching prefix
by Henning Thielemann 14 Mar '12

14 Mar '12

snd_seq_parse_address(seq, addr, "foo") does not look for a client named "foo" but for a client with prefix "foo". I always found this to be a feature, not a bug, although it is not documented. Is this behavior intended? If yes, how to cope with the case where one client's name is the prefix of another one? E.g. let the clients be "foobar" "foo" (in this order) then parsing "foo" will return the client address of "foobar", not that of "foo". Even more there is no prefix that will select "foo", but not "foobar". Sorting the clients with respect to increasing name lengths and then searching for matching prefixes in the sorted list would solve that problem. Currently it's only a theoretical problem but if that behavior is intended it should at least be documented. Using snd_seq_query_next_client everyone can write his own parse_address. We should only know about the theoretical problem.

2 3

[alsa-devel] Clean up of PCM spinlocks
by Takashi Iwai 14 Mar '12

14 Mar '12

Hi, below is a patch I posted to LKML as the follow up for the lockdep report from Dave Jones. It's a resurrected patch from what I worked on quite ago. Although this fix is suboptimal, I see no better way. If this looks OK, I'm going to queue it as a 3.4 material. thanks, Takashi === From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: pcm - Simplify linked PCM substream spinlocks The spinlocks held for PCM substreams have been always messy. For the better concurrent accessibility, we took always the substream's lock first. When substreams are linked, we need a group-wide lock, but this should be applied outside substream's lock. So, we unlock the substream's lock first, then do locks twice. This scheme is known to be problematic with lockdep. Maybe because the nested lock isn't marked properly, and partly because the lock and unlock sequences are different (A/B -> A/B). This patch tries to simplify the scheme. Instead of holding the substream's lock, we take the group lock always at first. A drawback of this is that the access to the individual substream in a same group won't be allowed any longer. But, the code looks much easier, and likely less buggy. Note that the group lock is identical with the substream's lock initially before the substream is linked with others. So, in the case of non-linked PCM, the new code behaves exactly same as before. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 12 ++++++------ sound/core/pcm_native.c | 36 ++++++++++++++---------------------- 2 files changed, 20 insertions(+), 28 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0cf91b2..d634dcc 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -531,36 +531,36 @@ static inline int snd_pcm_stream_linked(struct snd_pcm_substream *substream) static inline void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { read_lock(&snd_pcm_link_rwlock); - spin_lock(&substream->self_group.lock); + spin_lock(&substream->group->lock); } static inline void snd_pcm_stream_unlock(struct snd_pcm_substream *substream) { - spin_unlock(&substream->self_group.lock); + spin_unlock(&substream->group->lock); read_unlock(&snd_pcm_link_rwlock); } static inline void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) { read_lock_irq(&snd_pcm_link_rwlock); - spin_lock(&substream->self_group.lock); + spin_lock(&substream->group->lock); } static inline void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream) { - spin_unlock(&substream->self_group.lock); + spin_unlock(&substream->group->lock); read_unlock_irq(&snd_pcm_link_rwlock); } #define snd_pcm_stream_lock_irqsave(substream, flags) \ do { \ read_lock_irqsave(&snd_pcm_link_rwlock, (flags)); \ - spin_lock(&substream->self_group.lock); \ + spin_lock(&substream->group->lock); \ } while (0) #define snd_pcm_stream_unlock_irqrestore(substream, flags) \ do { \ - spin_unlock(&substream->self_group.lock); \ + spin_unlock(&substream->group->lock); \ read_unlock_irqrestore(&snd_pcm_link_rwlock, (flags)); \ } while (0) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 25ed9fe..c5f28ed 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -712,7 +712,7 @@ static int snd_pcm_action_group(struct action_ops *ops, int res = 0; snd_pcm_group_for_each_entry(s, substream) { - if (do_lock && s != substream) + if (do_lock) spin_lock_nested(&s->self_group.lock, SINGLE_DEPTH_NESTING); res = ops->pre_action(s, state); @@ -740,8 +740,7 @@ static int snd_pcm_action_group(struct action_ops *ops, if (do_lock) { /* unlock streams */ snd_pcm_group_for_each_entry(s1, substream) { - if (s1 != substream) - spin_unlock(&s1->self_group.lock); + spin_unlock(&s1->self_group.lock); if (s1 == s) /* end */ break; } @@ -779,13 +778,7 @@ static int snd_pcm_action(struct action_ops *ops, int res; if (snd_pcm_stream_linked(substream)) { - if (!spin_trylock(&substream->group->lock)) { - spin_unlock(&substream->self_group.lock); - spin_lock(&substream->group->lock); - spin_lock(&substream->self_group.lock); - } res = snd_pcm_action_group(ops, substream, state, 1); - spin_unlock(&substream->group->lock); } else { res = snd_pcm_action_single(ops, substream, state); } @@ -801,19 +794,13 @@ static int snd_pcm_action_lock_irq(struct action_ops *ops, { int res; - read_lock_irq(&snd_pcm_link_rwlock); + snd_pcm_stream_lock_irq(substream); if (snd_pcm_stream_linked(substream)) { - spin_lock(&substream->group->lock); - spin_lock(&substream->self_group.lock); res = snd_pcm_action_group(ops, substream, state, 1); - spin_unlock(&substream->self_group.lock); - spin_unlock(&substream->group->lock); } else { - spin_lock(&substream->self_group.lock); res = snd_pcm_action_single(ops, substream, state); - spin_unlock(&substream->self_group.lock); } - read_unlock_irq(&snd_pcm_link_rwlock); + snd_pcm_stream_unlock_irq(substream); return res; } @@ -1586,12 +1573,18 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) struct file *file; struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream1; + struct snd_pcm_group *group; file = snd_pcm_file_fd(fd); if (!file) return -EBADFD; pcm_file = file->private_data; substream1 = pcm_file->substream; + group = kmalloc(sizeof(*group), GFP_KERNEL); + if (!group) { + res = -ENOMEM; + goto _nolock; + } down_write(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || @@ -1604,11 +1597,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) goto _end; } if (!snd_pcm_stream_linked(substream)) { - substream->group = kmalloc(sizeof(struct snd_pcm_group), GFP_ATOMIC); - if (substream->group == NULL) { - res = -ENOMEM; - goto _end; - } + substream->group = group; spin_lock_init(&substream->group->lock); INIT_LIST_HEAD(&substream->group->substreams); list_add_tail(&substream->link_list, &substream->group->substreams); @@ -1620,7 +1609,10 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) _end: write_unlock_irq(&snd_pcm_link_rwlock); up_write(&snd_pcm_link_rwsem); + _nolock: fput(file); + if (res < 0) + kfree(group); return res; } -- 1.7.9.2

2 14

Re: [alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Clemens Ladisch 14 Mar '12

14 Mar '12

Jonathan Andrews wrote: > On Thu, 2012-03-08 at 10:55 +0100, Clemens Ladisch wrote: >> Jonathan Andrews wrote: >>> I'm using this script to build both static and dynamic versions of the >>> alsalib. Can you check the configure arguments are correct. >> >> They look correct. > > Tried to compile my application, still fails asking for pthreads. The configure.in file contains the patch, but the generated configure script does not have it (although its time stamp is newer). Please ensure that you've run autoconf (although make should do this automatically). Regards, Clemens

3 10

[alsa-devel] [PATCH] ASoC: max98095: add jack detection
by Rhyland Klein 14 Mar '12

14 Mar '12

This change adds the logic to support using the jack detect mechanism built in to the codec to detect both when a jack was inserted and what type of jack is present. This change also supports the use of an external mechanism for headphone detection. If this mechanism exists, when the max98095_jack_detect function is called, the hp_jack is simply passed NULL. This change supports both simple headphones, powered headphones, microphones and headsets with both headphones and a mic. Signed-off-by: Rhyland Klein <rklein(a)nvidia.com> --- include/sound/max98095.h | 7 ++ sound/soc/codecs/max98095.c | 166 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/max98095.h | 22 ++++++ 3 files changed, 194 insertions(+), 1 deletions(-) diff --git a/include/sound/max98095.h b/include/sound/max98095.h index 7513a42..7d0fb09 100644 --- a/include/sound/max98095.h +++ b/include/sound/max98095.h @@ -49,6 +49,13 @@ struct max98095_pdata { */ unsigned int digmic_left_mode:1; unsigned int digmic_right_mode:1; + + /* Pin5 is the mechanical method of sensing jack insertion + * but it is something that might not be supported. + * 0 = PIN5 not supported + * 1 = PIN5 supported + */ + int jack_detect_pin5en:1; }; #endif diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0bb511a..fc7a335 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <asm/div64.h> #include <sound/max98095.h> +#include <sound/jack.h> #include "max98095.h" enum max98095_type { @@ -51,6 +52,9 @@ struct max98095_priv { u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; + int irq; + struct snd_soc_jack *headphone_jack; + struct snd_soc_jack *mic_jack; }; static const u8 max98095_reg_def[M98095_REG_CNT] = { @@ -2173,9 +2177,135 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) max98095_handle_bq_pdata(codec); } +static int max98095_report_jack(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + int hp_report = 0; + int mic_report = 0; + + /* Read the Jack Status Register */ + value = snd_soc_read(codec, M98095_007_JACK_AUTO_STS); + + /* If ddone is not set, then detection isn't finished yet */ + if ((value & M98095_DDONE) == 0) + return -EINVAL; + + /* if hp, check its bit, and if set, clear it */ + if ((value & M98095_HP_IN || value & M98095_LO_IN) && + max98095->headphone_jack) + hp_report |= SND_JACK_HEADPHONE; + + /* if mic, check its bit, and if set, clear it */ + if ((value & M98095_MIC_IN) && max98095->mic_jack) + mic_report |= SND_JACK_MICROPHONE; + + if (max98095->headphone_jack == max98095->mic_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report | mic_report, + SND_JACK_HEADSET); + else { + if (max98095->headphone_jack) + snd_soc_jack_report(max98095->headphone_jack, + hp_report, SND_JACK_HEADPHONE); + if (max98095->mic_jack) + snd_soc_jack_report(max98095->mic_jack, + mic_report, SND_JACK_MICROPHONE); + } + + return 0; +} + +static irqreturn_t max98095_jack_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + int ret; + + ret = max98095_report_jack(codec); + + return ret ? IRQ_NONE : IRQ_HANDLED; + +} + +int max98095_jack_detect_enable(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int detect_enable = M98095_JDEN; + + if (max98095->pdata->jack_detect_pin5en) + detect_enable |= M98095_PIN5EN; + + /* configure jack detection: slew is calculated as 4 * (delay + 1) + * The default is 24 (0x18) to get 100ms delay. + */ + ret = snd_soc_write(codec, M98095_08E_JACK_DC_SLEW, + M98095_DEFAULT_SLEW_DELAY); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + /* configure auto detection to be enabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, detect_enable); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect_disable(struct snd_soc_codec *codec) +{ + int ret = 0; + + /* configure auto detection to be disabled */ + ret = snd_soc_write(codec, M98095_089_JACK_DET_AUTO, 0x0); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg auto detect %d\n", ret); + return ret; + } + + return ret; +} + +int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + max98095->headphone_jack = hp_jack; + max98095->mic_jack = mic_jack; + + /* only progress if we have at least 1 jack pointer */ + if (!hp_jack && !mic_jack) + return -EINVAL; + + max98095_jack_detect_enable(codec); + + /* enable interrupts for headphone jack detection */ + ret = snd_soc_update_bits(codec, M98095_013_JACK_INT_EN, + M98095_IDDONE, M98095_IDDONE); + if (ret < 0) { + dev_err(codec->dev, "Failed to cfg jack irqs %d\n", ret); + return ret; + } + + max98095_report_jack(codec); + return 0; +} + #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + disable_irq(max98095->irq); max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -2183,8 +2313,16 @@ static int max98095_suspend(struct snd_soc_codec *codec) static int max98095_resume(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (max98095->headphone_jack || max98095->mic_jack) { + max98095_jack_detect_enable(codec); + max98095_report_jack(codec); + } + enable_irq(max98095->irq); + return 0; } #else @@ -2266,11 +2404,23 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095->mic1pre = 0; max98095->mic2pre = 0; + if (max98095->irq) { + /* register an audio interrupt */ + ret = request_threaded_irq(max98095->irq, NULL, + max98095_jack_handler, + IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING, + "max98095", codec); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + goto err_access; + } + } + ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); - goto err_access; + goto err_irq; } dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); @@ -2306,14 +2456,27 @@ static int max98095_probe(struct snd_soc_codec *codec) max98095_add_widgets(codec); + return 0; + +err_irq: + if (max98095->irq) + free_irq(max98095->irq, codec); err_access: return ret; } static int max98095_remove(struct snd_soc_codec *codec) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (max98095->headphone_jack || max98095->mic_jack) + max98095_jack_detect_disable(codec); + + if (max98095->irq) + free_irq(max98095->irq, codec); + return 0; } @@ -2348,6 +2511,7 @@ static int max98095_i2c_probe(struct i2c_client *i2c, max98095->devtype = id->driver_data; i2c_set_clientdata(i2c, max98095); max98095->pdata = i2c->dev.platform_data; + max98095->irq = i2c->irq; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095, max98095_dai, ARRAY_SIZE(max98095_dai)); diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 891584a..2ebbe4e 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -175,11 +175,23 @@ /* MAX98095 Registers Bit Fields */ +/* M98095_007_JACK_AUTO_STS */ + #define M98095_MIC_IN (1<<3) + #define M98095_LO_IN (1<<5) + #define M98095_HP_IN (1<<6) + #define M98095_DDONE (1<<7) + /* M98095_00F_HOST_CFG */ #define M98095_SEG (1<<0) #define M98095_XTEN (1<<1) #define M98095_MDLLEN (1<<2) +/* M98095_013_JACK_INT_EN */ + #define M98095_IMIC_IN (1<<3) + #define M98095_ILO_IN (1<<5) + #define M98095_IHP_IN (1<<6) + #define M98095_IDDONE (1<<7) + /* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ #define M98095_CLKMODE_MASK 0xFF @@ -255,6 +267,10 @@ #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) +/* M98095_089_JACK_DET_AUTO */ + #define M98095_PIN5EN (1<<2) + #define M98095_JDEN (1<<7) + /* M98095_090_PWR_EN_IN */ #define M98095_INEN (1<<7) #define M98095_MB2EN (1<<3) @@ -296,4 +312,10 @@ #define M98095_174_DAI1_BQ_BASE 0x74 #define M98095_17E_DAI2_BQ_BASE 0x7E +/* Default Delay used in Slew Rate Calculation for Jack detection */ +#define M98095_DEFAULT_SLEW_DELAY 0x18 + +extern int max98095_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); + #endif -- 1.7.3.4

2 4

[alsa-devel] [PATCH v4] ASoC: mx27vis-aic32x4: Convert it to platform driver
by Fabio Estevam 14 Mar '12

14 Mar '12

Convert mx27vis-aic32x4 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Tested-by: Javier Martin <javier.martin(a)vista-silicon.com> --- Changes since v3: - Rebased against asoc for-3.4 Changes since v2: - Add missing initialization of mx27vis_aic32x4.dev - Add Tested-by line Changes since v1: - Fix MODULE_ALIAS - Rename it to mx27vis arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 41 ++++++++++++++------------ 2 files changed, 23 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c index c2766ae..428459f 100644 --- a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c +++ b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c @@ -263,6 +263,7 @@ static void __init visstrim_m10_board_init(void) imx27_add_fec(NULL); imx_add_gpio_keys(&visstrim_gpio_keys_platform_data); platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); + imx_add_platform_device("mx27vis", 0, NULL, 0, NULL, 0); } static void __init visstrim_m10_timer_init(void) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..f6d04ad 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -188,22 +188,16 @@ static struct snd_soc_card mx27vis_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; -static struct platform_device *mx27vis_aic32x4_snd_device; - -static int __init mx27vis_aic32x4_init(void) +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { int ret; - mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_aic32x4_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); - ret = platform_device_add(mx27vis_aic32x4_snd_device); - + mx27vis_aic32x4.dev = &pdev->dev; + ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_aic32x4_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; } /* Connect SSI0 as clock slave to SSI1 external pins */ @@ -221,22 +215,31 @@ static int __init mx27vis_aic32x4_init(void) ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) { + if (ret) printk(KERN_ERR "ASoC: unable to setup gpios\n"); - platform_device_put(mx27vis_aic32x4_snd_device); - } return ret; } -static void __exit mx27vis_aic32x4_exit(void) +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) { - platform_device_unregister(mx27vis_aic32x4_snd_device); + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; } -module_init(mx27vis_aic32x4_init); -module_exit(mx27vis_aic32x4_exit); +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mx27vis"); -- 1.7.1

2 1

Re: [alsa-devel] [PATCH 12/16] ASoC: Ux500: Add MSP I2S-driver
by Mark Brown 13 Mar '12

13 Mar '12

On Tue, Mar 13, 2012 at 04:11:39PM +0100, Ola Lilja wrote: > +menuconfig SND_SOC_UX500 > + bool "SoC Audio support for Ux500 platform" > + depends on SND_SOC > + default n This is redundant, the default default is n. > +config SND_SOC_UX500_MSP_I2S > + bool "Platform - MSP (I2S)" > + depends on SND_SOC_UX500 > + default n > + help > + > + Say Y if you want to enable the Ux500 MSP I2S-driver. Don't do this, the driver is only usable with a machine driver so should be selected by the machine driver. > +config SND_SOC_UX500_DEBUG > + bool "Activate Ux500 platform debug-mode (pr_debug)" > + depends on SND_SOC_UX500 > + default n > + help > + Say Y if you want to enable debug-level prints for Ux500 code-files. No. > +ifdef CONFIG_SND_SOC_UX500_MSP_I2S > +snd-soc-ux500-platform-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o > +obj-$(CONFIG_SND_SOC_UX500_MSP_I2S) += snd-soc-ux500-platform-i2s.o > +endif Do this idiomatically please. If you need to do use this odd method the code should explain why. > +#define to_i2s_controller(d) container_of(d, struct i2s_controller, dev) Use a function please for type safety (static inline should work out about the same in terms of the generated code). > +/*** Protocols ***/ > +enum msp_protocol { > + MSP_I2S_PROTOCOL, > + MSP_PCM_PROTOCOL, > + MSP_PCM_COMPAND_PROTOCOL, > + MSP_AC97_PROTOCOL, > + MSP_MASTER_SPI_PROTOCOL, > + MSP_SLAVE_SPI_PROTOCOL, > + MSP_INVALID_PROTOCOL > +}; Does some of this code want to be in a library somewhere so the hardware support can be shared with the SPI subsystem? In general there's a pretty enormous abstraction layer in here which probably isn't doing all that much if this stuff is internal to the audio subsystem - my overall feeling reading the code is that the abstraction layer should be removed as it's internal to this driver. > +static struct ux500_platform_drvdata platform_drvdata[UX500_NBR_OF_DAI] = { > + { > + .msp_i2s_drvdata = NULL, > + .fmt = 0, > + .slots = 1, > + .tx_mask = 0x01, > + .rx_mask = 0x01, > + .slot_width = 16, > + .playback_active = false, > + .capture_active = false, > + .configured = 0, > + .master_clk = UX500_MSP_INTERNAL_CLOCK_FREQ, > + }, This looks like it should be dynamically created by the platform device, not a static struct. > +static const char *stream_str(struct snd_pcm_substream *substream) > +{ > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > + return "Playback"; > + else > + return "Capture"; > +} Don't add generic code in drivers. > + if (mode_playback) > + drvdata->playback_active = true; > + else > + drvdata->capture_active = true; All this mode_playback stuff is non-idiomatic. > +static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; > + bool mode_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); > + > + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, > + stream_str(substream)); > + > + if (drvdata == NULL) > + return; Why might that happen? > + prot_desc->total_clocks_for_one_frame = > + prot_desc->frame_period+1; > + > + dev_dbg(dai->dev, "%s: Total clocks per frame: %u\n", > + __func__, > + prot_desc->total_clocks_for_one_frame); This looks like your clock calculation stuff may be replicating framework features. > + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > + default: > + case SND_SOC_DAIFMT_NB_NF: > + break; > + > + case SND_SOC_DAIFMT_NB_IF: > + msp_config->tx_frame_sync_pol ^= 1 << TFSPOL_SHIFT; > + msp_config->rx_frame_sync_pol ^= 1 << RFSPOL_SHIFT; > + break; > + } Shold be returning an error for unsupported modes. > + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) { > + dev_dbg(dai->dev, "%s: Codec is MASTER.\n", This should be a switch statement and again should error out on unsupporetd modes. Use of if statements instead of switch statements happens elsewhere too. > + /* To avoid division by zero in I2S-driver (i2s_setup) */ > + prot_desc->total_clocks_for_one_frame = 1; Also many_of_these_variable_names_are_getting_so_long_it_is_hard_to_use_them_without_word_wrapping > + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: > + dev_dbg(dai->dev, "%s: SND_SOC_DAIFMT_I2S.\n", > + __func__); In general if you find this logging useful I'd add it to the framework; perhaps other people will too and it'll save everyone open coding it. > + > + msp_config->default_protocol_desc = 1; > + msp_config->protocol = MSP_I2S_PROTOCOL; > + break; > + > + default: > + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: Are you *positive* that all modes you don't explicitly support are identical to _I2S | _CBM_CFM? > + /* If already configured -> not errors reported */ > + if (mode_playback) { > + if ((drvdata->configured & PLAYBACK_CONFIGURED) && > + (drvdata->playback_active)) > + goto cleanup; > + } else { > + if ((drvdata->configured & CAPTURE_CONFIGURED) && > + (drvdata->capture_active)) > + goto cleanup; > + } This looks suspicious... why are you doing this? > + dev_err(dai->dev, "%s: Error: PCM TDM format requires channels " > + "to match active slots " > + "(channels = %d, active slots = %d)!\n", > + __func__, params_channels(params), slots_active); Don't split error messages over multiple lines, it makes them hard to search for. > + if (!(slots == 1 || slots == 2 || slots == 8 || slots == 16)) { > + dev_err(dai->dev, "%s: Error: Unsupported slots (%d)! " > + "Supported values are 1/2/8/16.\n", > + __func__, slots); > + return -EINVAL; > + } This looks like a switch statement.... > + switch (slots) { > + default: > + case 1: > + cap = 0x01; > + break; > + case 2: > + cap = 0x03; > + break; > + case 8: > + cap = 0xFF; > + break; > + case 16: > + cap = 0xFFFF; > + break; > + } ...in fact it could be combined with this one. > +static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, > + int cmd, struct snd_soc_dai *dai) > +{ > + int ret = 0; > + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; > + > + dev_dbg(dai->dev, "%s: MSP %d (%s): Enter (msp->id = %d, cmd = %d).\n", > + __func__, dai->id, stream_str(substream), > + (int)drvdata->msp_i2s_drvdata->id, cmd); > + > + ret = ux500_msp_i2s_trigger(drvdata->msp_i2s_drvdata, cmd, substream->stream); > + > + return ret; > +} This function does nothing, just inline the function you're calling. > + .suspend = NULL, > + .resume = NULL, No need to assign nonexistant ops like this. > + .ops = (struct snd_soc_dai_ops[]) { > + { > + .set_sysclk = ux500_msp_dai_set_dai_sysclk, > + .set_fmt = ux500_msp_dai_set_dai_fmt, > + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, > + .startup = ux500_msp_dai_startup, > + .shutdown = ux500_msp_dai_shutdown, > + .prepare = ux500_msp_dai_prepare, > + .trigger = ux500_msp_dai_trigger, > + .hw_params = ux500_msp_dai_hw_params, > + } Just declare the struct as a variable, if you need casts like this something is going wrong and doing this will prevent sharing the ops struct between the DAIs which is the whole point. > +EXPORT_SYMBOL(ux500_msp_dai_drv); Why are you doing this? > + dev_err(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__, pdev->name); Remove stuff like this, it's not an error. You've got lots of logging that should at most be at dev_dbg() level. > + platform_data = (struct msp_i2s_platform_data *)pdev->dev.platform_data; Why not just inline this function? > + msp_i2s_drvdata = ux500_msp_i2s_init(pdev, platform_data); > + if (!msp_i2s_drvdata) { > + dev_err(&pdev->dev, "%s: ERROR: ux500_msp_i2s_init failed!", __func__); > + return -1; > + } Are you sure that -EPERM is the best error code? If you are then you should use the constant not the number. > + dev_info(&pdev->dev, "%s: Registering ux500-msp-dai SoC CPU-DAI.\n", __func__); For things like this the core will already log for you. > +int ux500_msp_drv_suspend(struct platform_device *pdev, pm_message_t state) > +{ > + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata = dev_get_drvdata(&pdev->dev); > + > + dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__, pdev->name); > + > + return ux500_msp_i2s_suspend(msp_i2s_drvdata); Once again, this function does nothing and should just have the contents inline. This should also be done from the ASoC level callbacks so that it's joined up with the suspend of the audio subsystem as a whole. > +static struct platform_driver msp_i2s_driver = { > + .driver = { > + .name = "ux500-msp-i2s", > + .owner = THIS_MODULE, > + }, > + .probe = ux500_msp_drv_probe, > + .remove = ux500_msp_drv_remove, It looks like this just registers one device for all the ports in the system. Are all the ports really part of one IP or should the driver be controlling a single port and letting the SoC instantiate whatever is present on a given chip? > +static int __init ux500_msp_init(void) module_platform_driver. > +#ifndef CONFIG_SND_SOC_UX500_AB5500 > +#define UX500_MSP_INTERNAL_CLOCK_FREQ 40000000 > +#define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ > +#else > +#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000 > +#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2) > +#endif This should probably be coming out of the clock framework. > +int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay); What is this?

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[alsa-devel] Triple implementation of WM8766 and quadruple WM8776!
by Ondrej Zary 13 Mar '12

13 Mar '12

Hello, I'm trying do implement support for Philips PSC724 Ultimate Edge card, which is based on VT1722 + WM8776 + WM8766 chips. Found that the best file to base my work on is sound/pci/ice1712/se.c (as SE-200PCI card contains both WM8776 and WM8766 chips). While doing this, found that there are three implementations of WM8766 codec control, two of them in the same driver(!): sound/pci/ice1712/se.c sound/pci/ice1712/prodigy_hifi.c sound/pci/oxygen/xonar_wm87x6.c (+sound/pci/oxygen/wm8766.h) And WM8776 status is even worse, there are four implementations, again two being in the same driver: sound/pci/ice1712/se.c sound/pci/ice1712/maya44.c sound/pci/oxygen/xonar_wm87x6.c (+sound/pci/oxygen/wm8776.h) sound/soc/codecs/wm8776.c (this one being official and stand-alone module but it does not seem to be usable as a part of sound card driver) My code would enlarge both of these lists and I don't like that. What to do? -- Ondrej Zary

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[alsa-devel] [PATCH 3/3] ASoC: Ux500: Add driver for Ux500/AB8500 platform/codec
by Ola Lilja 13 Mar '12

13 Mar '12

Add the ST-Ericsson Ux500 ASoC platform-driver, AB8500 codec-driver and machine-driver for U8500 hardware. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 195 ++ arch/arm/mach-ux500/board-mop500-regulators.c | 28 + arch/arm/mach-ux500/board-mop500.c | 7 +- arch/arm/mach-ux500/board-mop500.h | 1 + arch/arm/mach-ux500/clock.c | 8 +- arch/arm/mach-ux500/devices-common.h | 2 +- arch/arm/mach-ux500/include/mach/msp.h | 1030 +++++++++ include/sound/ux500_ab8500.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/Makefile.rej | 10 + sound/soc/codecs/ab8500_audio.c | 2928 +++++++++++++++++++++++++ sound/soc/codecs/ab8500_audio.h | 673 ++++++ sound/soc/ux500/Kconfig | 33 + sound/soc/ux500/Makefile | 25 + sound/soc/ux500/u8500.c | 150 ++ sound/soc/ux500/ux500_ab8500.c | 790 +++++++ sound/soc/ux500/ux500_msp_dai.c | 998 +++++++++ sound/soc/ux500/ux500_msp_dai.h | 83 + sound/soc/ux500/ux500_msp_i2s.c | 1004 +++++++++ sound/soc/ux500/ux500_msp_i2s.h | 40 + sound/soc/ux500/ux500_pcm.c | 428 ++++ sound/soc/ux500/ux500_pcm.h | 44 + 26 files changed, 8519 insertions(+), 9 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/include/mach/msp.h create mode 100644 include/sound/ux500_ab8500.h create mode 100644 sound/soc/codecs/Makefile.rej create mode 100644 sound/soc/codecs/ab8500_audio.c create mode 100644 sound/soc/codecs/ab8500_audio.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/u8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.c create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h diff --git a/arch/arm/mach-ux500/Makefile b/arch/arm/mach-ux500/Makefile index 6bd2f45..6ee4e84 100644 --- a/arch/arm/mach-ux500/Makefile +++ b/arch/arm/mach-ux500/Makefile @@ -11,7 +11,8 @@ obj-$(CONFIG_MACH_U8500) += board-mop500.o board-mop500-sdi.o \ board-mop500-regulators.o \ board-mop500-uib.o board-mop500-stuib.o \ board-mop500-u8500uib.o \ - board-mop500-pins.o + board-mop500-pins.o \ + board-mop500-msp.o obj-$(CONFIG_MACH_U5500) += board-u5500.o board-u5500-sdi.o obj-$(CONFIG_SMP) += platsmp.o headsmp.o obj-$(CONFIG_HOTPLUG_CPU) += hotplug.o diff --git a/arch/arm/mach-ux500/board-mop500-msp.c b/arch/arm/mach-ux500/board-mop500-msp.c new file mode 100644 index 0000000..c6a73a0 --- /dev/null +++ b/arch/arm/mach-ux500/board-mop500-msp.c @@ -0,0 +1,195 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * License terms: GNU General Public License (GPL), version 2 + */ + +#include <linux/platform_device.h> +#include <linux/init.h> +#include <linux/gpio.h> +#include <plat/gpio-nomadik.h> + +#include <plat/pincfg.h> +#include <plat/ste_dma40.h> + +#include <mach/devices.h> +#include <ste-dma40-db8500.h> +#include <mach/hardware.h> +#include <mach/irqs.h> +#include <mach/msp.h> + +#include <mach/msp.h> +#include "board-mop500.h" +#include "devices-db8500.h" +#include "pins-db8500.h" + +/* MSP1/3 Tx/Rx usage protection */ +static DEFINE_SPINLOCK(msp_rxtx_lock); + +/* Reference Count */ +static int msp_rxtx_ref; + +static pin_cfg_t mop500_msp1_pins_init[] = { + GPIO33_MSP1_TXD | PIN_OUTPUT_LOW | PIN_SLPM_WAKEUP_DISABLE, + GPIO34_MSP1_TFS | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, + GPIO35_MSP1_TCK | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, + GPIO36_MSP1_RXD | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_DISABLE, +}; + +static pin_cfg_t mop500_msp1_pins_exit[] = { + GPIO33_MSP1_TXD | PIN_OUTPUT_LOW | PIN_SLPM_WAKEUP_ENABLE, + GPIO34_MSP1_TFS | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, + GPIO35_MSP1_TCK | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, + GPIO36_MSP1_RXD | PIN_INPUT_NOPULL | PIN_SLPM_WAKEUP_ENABLE, +}; + +int msp13_i2s_init(void) +{ + int retval = 0; + unsigned long flags; + + spin_lock_irqsave(&msp_rxtx_lock, flags); + if (msp_rxtx_ref == 0) + retval = nmk_config_pins( + ARRAY_AND_SIZE(mop500_msp1_pins_init)); + if (!retval) + msp_rxtx_ref++; + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + + return retval; +} + +int msp13_i2s_exit(void) +{ + int retval = 0; + unsigned long flags; + + spin_lock_irqsave(&msp_rxtx_lock, flags); + WARN_ON(!msp_rxtx_ref); + msp_rxtx_ref--; + if (msp_rxtx_ref == 0) + retval = nmk_config_pins_sleep( + ARRAY_AND_SIZE(mop500_msp1_pins_exit)); + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + + return retval; +} + +static struct stedma40_chan_cfg msp0_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV31_MSP0_RX_SLIM0_CH0_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp0_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV31_MSP0_TX_SLIM0_CH0_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp0_platform_data = { + .id = MSP_0_I2S_CONTROLLER, + .msp_i2s_dma_rx = &msp0_dma_rx, + .msp_i2s_dma_tx = &msp0_dma_tx, +}; + +static struct stedma40_chan_cfg msp1_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV30_MSP3_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp1_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV30_MSP1_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp1_platform_data = { + .id = MSP_1_I2S_CONTROLLER, + .msp_i2s_dma_rx = NULL, + .msp_i2s_dma_tx = &msp1_dma_tx, + .msp_i2s_init = msp13_i2s_init, + .msp_i2s_exit = msp13_i2s_exit, +}; + +static struct stedma40_chan_cfg msp2_dma_rx = { + .high_priority = true, + .dir = STEDMA40_PERIPH_TO_MEM, + + .src_dev_type = DB8500_DMA_DEV14_MSP2_RX, + .dst_dev_type = STEDMA40_DEV_DST_MEMORY, + + /* MSP2 DMA doesn't work with PSIZE == 4 on DB8500v2 */ + .src_info.psize = STEDMA40_PSIZE_LOG_1, + .dst_info.psize = STEDMA40_PSIZE_LOG_1, + + /* data_width is set during configuration */ +}; + +static struct stedma40_chan_cfg msp2_dma_tx = { + .high_priority = true, + .dir = STEDMA40_MEM_TO_PERIPH, + + .src_dev_type = STEDMA40_DEV_DST_MEMORY, + .dst_dev_type = DB8500_DMA_DEV14_MSP2_TX, + + .src_info.psize = STEDMA40_PSIZE_LOG_4, + .dst_info.psize = STEDMA40_PSIZE_LOG_4, + + .use_fixed_channel = true, + .phy_channel = 1, + + /* data_width is set during configuration */ +}; + +static struct msp_i2s_platform_data msp2_platform_data = { + .id = MSP_2_I2S_CONTROLLER, + .msp_i2s_dma_rx = &msp2_dma_rx, + .msp_i2s_dma_tx = &msp2_dma_tx, +}; + +static struct msp_i2s_platform_data msp3_platform_data = { + .id = MSP_3_I2S_CONTROLLER, + .msp_i2s_dma_rx = &msp1_dma_rx, + .msp_i2s_dma_tx = NULL, + .msp_i2s_init = msp13_i2s_init, + .msp_i2s_exit = msp13_i2s_exit, +}; + +void __init mop500_msp_init(void) +{ + pr_err("DORIAN WAS HERE!\n"); + db8500_add_msp0_i2s(&msp0_platform_data); + db8500_add_msp1_i2s(&msp1_platform_data); + db8500_add_msp2_i2s(&msp2_platform_data); + db8500_add_msp3_i2s(&msp3_platform_data); +} diff --git a/arch/arm/mach-ux500/board-mop500-regulators.c b/arch/arm/mach-ux500/board-mop500-regulators.c index 2735d03..52426a4 100644 --- a/arch/arm/mach-ux500/board-mop500-regulators.c +++ b/arch/arm/mach-ux500/board-mop500-regulators.c @@ -74,6 +74,26 @@ static struct regulator_consumer_supply ab8500_vtvout_consumers[] = { REGULATOR_SUPPLY("vddadc", "ab8500-gpadc.0"), }; +static struct regulator_consumer_supply ab8500_vaud_consumers[] = { + /* AB8500 audio-codec main supply */ + REGULATOR_SUPPLY("vaud", "ab8500-codec.0"), +}; + +static struct regulator_consumer_supply ab8500_vamic1_consumers[] = { + /* AB8500 audio-codec Mic1 supply */ + REGULATOR_SUPPLY("vamic1", "ab8500-codec.0"), +}; + +static struct regulator_consumer_supply ab8500_vamic2_consumers[] = { + /* AB8500 audio-codec Mic2 supply */ + REGULATOR_SUPPLY("vamic2", "ab8500-codec.0"), +}; + +static struct regulator_consumer_supply ab8500_vdmic_consumers[] = { + /* AB8500 audio-codec DMic supply */ + REGULATOR_SUPPLY("vdmic", "ab8500-codec.0"), +}; + static struct regulator_consumer_supply ab8500_vintcore_consumers[] = { /* SoC core supply, no device */ REGULATOR_SUPPLY("v-intcore", NULL), @@ -323,6 +343,8 @@ struct regulator_init_data ab8500_regulators[AB8500_NUM_REGULATORS] = { .name = "V-AUD", .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, + .num_consumer_supplies = ARRAY_SIZE(ab8500_vaud_consumers), + .consumer_supplies = ab8500_vaud_consumers, }, /* supply for v-anamic1 VAMic1-LDO */ [AB8500_LDO_ANAMIC1] = { @@ -330,6 +352,8 @@ struct regulator_init_data ab8500_regulators[AB8500_NUM_REGULATORS] = { .name = "V-AMIC1", .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, + .num_consumer_supplies = ARRAY_SIZE(ab8500_vamic1_consumers), + .consumer_supplies = ab8500_vamic1_consumers, }, /* supply for v-amic2, VAMIC2 LDO, reuse constants for AMIC1 */ [AB8500_LDO_ANAMIC2] = { @@ -337,6 +361,8 @@ struct regulator_init_data ab8500_regulators[AB8500_NUM_REGULATORS] = { .name = "V-AMIC2", .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, + .num_consumer_supplies = ARRAY_SIZE(ab8500_vamic2_consumers), + .consumer_supplies = ab8500_vamic2_consumers, }, /* supply for v-dmic, VDMIC LDO */ [AB8500_LDO_DMIC] = { @@ -344,6 +370,8 @@ struct regulator_init_data ab8500_regulators[AB8500_NUM_REGULATORS] = { .name = "V-DMIC", .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, + .num_consumer_supplies = ARRAY_SIZE(ab8500_vdmic_consumers), + .consumer_supplies = ab8500_vdmic_consumers, }, /* supply for v-intcore12, VINTCORE12 LDO */ [AB8500_LDO_INTCORE] = { diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index 2df23ed..4b017a1 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -604,7 +604,6 @@ static struct platform_device *snowball_platform_devs[] __initdata = { static void __init mop500_init_machine(void) { int i2c0_devs; - mop500_gpio_keys[0].gpio = GPIO_PROX_SENSOR; u8500_init_devices(); @@ -613,12 +612,11 @@ static void __init mop500_init_machine(void) platform_add_devices(mop500_platform_devs, ARRAY_SIZE(mop500_platform_devs)); - mop500_i2c_init(); mop500_sdi_init(); + mop500_msp_init(); mop500_spi_init(); mop500_uart_init(); - i2c0_devs = ARRAY_SIZE(mop500_i2c0_devices); i2c_register_board_info(0, mop500_i2c0_devices, i2c0_devs); @@ -642,6 +640,7 @@ static void __init snowball_init_machine(void) mop500_i2c_init(); snowball_sdi_init(); + mop500_msp_init(); mop500_spi_init(); mop500_uart_init(); @@ -657,7 +656,6 @@ static void __init snowball_init_machine(void) static void __init hrefv60_init_machine(void) { int i2c0_devs; - /* * The HREFv60 board removed a GPIO expander and routed * all these GPIO pins to the internal GPIO controller @@ -674,6 +672,7 @@ static void __init hrefv60_init_machine(void) mop500_i2c_init(); hrefv60_sdi_init(); + mop500_msp_init(); mop500_spi_init(); mop500_uart_init(); diff --git a/arch/arm/mach-ux500/board-mop500.h b/arch/arm/mach-ux500/board-mop500.h index f926d3d..b377bbe 100644 --- a/arch/arm/mach-ux500/board-mop500.h +++ b/arch/arm/mach-ux500/board-mop500.h @@ -81,6 +81,7 @@ extern void hrefv60_sdi_init(void); extern void mop500_sdi_tc35892_init(void); void __init mop500_u8500uib_init(void); void __init mop500_stuib_init(void); +void __init mop500_msp_init(void); void __init mop500_pins_init(void); void __init hrefv60_pins_init(void); void __init snowball_pins_init(void); diff --git a/arch/arm/mach-ux500/clock.c b/arch/arm/mach-ux500/clock.c index ec35f0a..700042c 100644 --- a/arch/arm/mach-ux500/clock.c +++ b/arch/arm/mach-ux500/clock.c @@ -336,6 +336,7 @@ static DEFINE_PRCMU_CLK(uiccclk, 0x4, 1, UICCCLK); /* v1 */ */ /* Peripheral Cluster #1 */ +static DEFINE_PRCC_CLK(1, msp3, 11, 10, &clk_msp1clk); static DEFINE_PRCC_CLK(1, i2c4, 10, 9, &clk_i2cclk); static DEFINE_PRCC_CLK(1, gpio0, 9, -1, NULL); static DEFINE_PRCC_CLK(1, slimbus0, 8, 8, &clk_slimclk); @@ -405,7 +406,7 @@ static struct clk_lookup u8500_clks[] = { CLK(slimbus0, "slimbus0", NULL), CLK(i2c2, "nmk-i2c.2", NULL), CLK(sdi0, "sdi0", NULL), - CLK(msp0, "msp0", NULL), + CLK(msp0, "ux500-msp-i2s.0", NULL), CLK(i2c1, "nmk-i2c.1", NULL), CLK(uart1, "uart1", NULL), CLK(uart0, "uart0", NULL), @@ -455,7 +456,8 @@ static struct clk_lookup u8500_clks[] = { /* Peripheral Cluster #1 */ CLK(i2c4, "nmk-i2c.4", NULL), CLK(spi3, "spi3", NULL), - CLK(msp1, "msp1", NULL), + CLK(msp1, "ux500-msp-i2s.1", NULL), + CLK(msp3, "ux500-msp-i2s.3", NULL), /* Peripheral Cluster #2 */ CLK(gpio1, "gpio.6", NULL), @@ -465,7 +467,7 @@ static struct clk_lookup u8500_clks[] = { CLK(spi0, "spi0", NULL), CLK(sdi3, "sdi3", NULL), CLK(sdi1, "sdi1", NULL), - CLK(msp2, "msp2", NULL), + CLK(msp2, "ux500-msp-i2s.2", NULL), CLK(sdi4, "sdi4", NULL), CLK(pwl, "pwl", NULL), CLK(spi1, "spi1", NULL), diff --git a/arch/arm/mach-ux500/devices-common.h b/arch/arm/mach-ux500/devices-common.h index 7825705..3e5bd4b 100644 --- a/arch/arm/mach-ux500/devices-common.h +++ b/arch/arm/mach-ux500/devices-common.h @@ -69,7 +69,7 @@ static inline struct platform_device * dbx500_add_msp_i2s(int id, resource_size_t base, int irq, struct msp_i2s_platform_data *pdata) { - return dbx500_add_platform_device_4k1irq("MSP_I2S", id, base, irq, + return dbx500_add_platform_device_4k1irq("ux500-msp-i2s", id, base, irq, pdata); } diff --git a/arch/arm/mach-ux500/include/mach/msp.h b/arch/arm/mach-ux500/include/mach/msp.h new file mode 100644 index 0000000..9055364 --- /dev/null +++ b/arch/arm/mach-ux500/include/mach/msp.h @@ -0,0 +1,1030 @@ +/* + * Copyright (c) 2009 STMicroelectronics + * + * 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 _STM_MSP_HEADER +#define _STM_MSP_HEADER +#include <linux/device.h> +#include <linux/spinlock.h> +#include <linux/semaphore.h> +#include <linux/dmaengine.h> +#include <linux/irqreturn.h> +#include <linux/bitops.h> +#include <plat/ste_dma40.h> +#include <linux/gpio.h> + +enum msp_data_size { + MSP_DATA_BITS_DEFAULT = -1, + MSP_DATA_BITS_8 = 0x00, + MSP_DATA_BITS_10, + MSP_DATA_BITS_12, + MSP_DATA_BITS_14, + MSP_DATA_BITS_16, + MSP_DATA_BITS_20, + MSP_DATA_BITS_24, + MSP_DATA_BITS_32, +}; + +enum msp_state { + MSP_STATE_IDLE = 0, + MSP_STATE_CONFIGURED = 1, + MSP_STATE_RUN = 2, +}; + +enum msp_rx_comparison_enable_mode { + MSP_COMPARISON_DISABLED = 0, + MSP_COMPARISON_NONEQUAL_ENABLED = 2, + MSP_COMPARISON_EQUAL_ENABLED = 3 +}; + +#define RMCEN_BIT 0 +#define RMCSF_BIT 1 +#define RCMPM_BIT 3 +#define TMCEN_BIT 5 +#define TNCSF_BIT 6 + +struct msp_multichannel_config { + bool rx_multichannel_enable; + bool tx_multichannel_enable; + enum msp_rx_comparison_enable_mode rx_comparison_enable_mode; + u8 padding; + u32 comparison_value; + u32 comparison_mask; + u32 rx_channel_0_enable; + u32 rx_channel_1_enable; + u32 rx_channel_2_enable; + u32 rx_channel_3_enable; + u32 tx_channel_0_enable; + u32 tx_channel_1_enable; + u32 tx_channel_2_enable; + u32 tx_channel_3_enable; +}; + +/** + * struct msp_protocol_desc- MSP Protocol desc structure per MSP. + * @rx_phase_mode: rx_phase_mode whether single or dual. + * @tx_phase_mode: tx_phase_mode whether single or dual. + * @rx_phase2_start_mode: rx_phase2_start_mode whether imediate or after + * some delay. + * @tx_phase2_start_mode: tx_phase2_start_mode whether imediate or after + * some delay. + * @rx_bit_transfer_format: MSP or LSB. + * @tx_bit_transfer_format: MSP or LSB. + * @rx_frame_length_1: Frame1 length 1,2,3.. + * @rx_frame_length_2: Frame2 length 1,2,3.. + * @tx_frame_length_1: Frame1 length 1,2,3.. + * @tx_frame_length_2: Frame2 length 1,2,3.. + * @rx_element_length_1: Element1 length 1,2,... + * @rx_element_length_2: Element2 length 1,2,... + * @tx_element_length_1: Element1 length 1,2,... + * @tx_element_length_2: Element2 length 1,2,... + * @rx_data_delay: Delay in clk cycle after frame sync + * @tx_data_delay: Delay in clk cycle after frame sync + * @rx_clock_pol: Rxpol whether rising or falling.It indicates pol of bit clock. + * @tx_clock_pol: Txpol whether rising or falling.It indicates pol of bit clock. + * @rx_frame_sync_pol: Frame sync pol whether rising or Falling. + * @tx_frame_sync_pol: Frame sync pol whether rising or Falling. + * @rx_half_word_swap: Word swap half word, full word. + * @tx_half_word_swap: Word swap half word, full word. + * @compression_mode: Compression mode whether Alaw or Ulaw or disabled. + * @expansion_mode: Compression mode whether Alaw or Ulaw or disabled. + * @spi_clk_mode: Spi clock mode to be enabled or not. + * @spi_burst_mode: Spi burst mode to be enabled or not. + * @frame_sync_ignore: Frame sync to be ignored or not. Ignore in case of Audio + * codec acting as Master. + * @frame_period: Frame period (clk cycles) after which new frame sync occurs. + * @frame_width: Frame width (clk cycles) after which frame sycn changes state. + * @total_clocks_for_one_frame: No. of clk cycles per frame. + * + * Main Msp protocol descriptor data structure to be used to store various info + * in transmit or recevie configuration registers of an MSP. + */ + +struct msp_protocol_desc { + u32 rx_phase_mode; + u32 tx_phase_mode; + u32 rx_phase2_start_mode; + u32 tx_phase2_start_mode; + u32 rx_bit_transfer_format; + u32 tx_bit_transfer_format; + u32 rx_frame_length_1; + u32 rx_frame_length_2; + u32 tx_frame_length_1; + u32 tx_frame_length_2; + u32 rx_element_length_1; + u32 rx_element_length_2; + u32 tx_element_length_1; + u32 tx_element_length_2; + u32 rx_data_delay; + u32 tx_data_delay; + u32 rx_clock_pol; + u32 tx_clock_pol; + u32 rx_frame_sync_pol; + u32 tx_frame_sync_pol; + u32 rx_half_word_swap; + u32 tx_half_word_swap; + u32 compression_mode; + u32 expansion_mode; + u32 spi_clk_mode; + u32 spi_burst_mode; + u32 frame_sync_ignore; + u32 frame_period; + u32 frame_width; + u32 total_clocks_for_one_frame; +}; + +enum i2s_direction_t { + I2S_DIRECTION_TX = 0, + I2S_DIRECTION_RX = 1, + I2S_DIRECTION_BOTH = 2 +}; + +enum i2s_transfer_mode_t { + I2S_TRANSFER_MODE_SINGLE_DMA = 0, + I2S_TRANSFER_MODE_CYCLIC_DMA = 1, + I2S_TRANSFER_MODE_INF_LOOPBACK = 2, + I2S_TRANSFER_MODE_NON_DMA = 4, +}; + +struct i2s_message { + enum i2s_direction_t i2s_direction; + void *txdata; + void *rxdata; + size_t txbytes; + size_t rxbytes; + int dma_flag; + int tx_offset; + int rx_offset; + /* cyclic dma */ + bool cyclic_dma; + dma_addr_t buf_addr; + size_t buf_len; + size_t period_len; +}; + +enum i2s_flag { + DISABLE_ALL = 0, + DISABLE_TRANSMIT = 1, + DISABLE_RECEIVE = 2, +}; + +struct i2s_controller { + struct module *owner; + unsigned int id; + unsigned int class; + const struct i2s_algorithm *algo; /* the algorithm to access the bus */ + void *data; + struct mutex bus_lock; + struct device dev; /* the controller device */ + char name[48]; +}; +#define to_i2s_controller(d) container_of(d, struct i2s_controller, dev) + +/** + * struct trans_data - MSP transfer data structure used during xfer. + * @message: i2s message. + * @msp: msp structure. + * @tx_handler: callback handler for transmit path. + * @rx_handler: callback handler for receive path. + * @tx_callback_data: callback data for transmit. + * @rx_callback_data: callback data for receive. + * + */ +struct trans_data { + struct i2s_message message; + struct msp *msp; + void (*tx_handler) (void *data); + void (*rx_handler) (void *data); + void *tx_callback_data; + void *rx_callback_data; +}; + +/** + * struct msp_config- MSP configuration structure used by i2s client. + * @input_clock_freq: Input clock frequency default is 48MHz. + * @rx_clock_sel: Receive clock selection (Provided by Sample Gen or external + * source). + * @tx_clock_sel: Transmit clock selection (Provided by Sample Gen or external. + * source). + * @srg_clock_sel: APB clock or clock dervied from Slave (Audio codec). + * @rx_frame_sync_pol: Receive frame sync polarity. + * @tx_frame_sync_pol: Transmit frame sync polarity. + * @rx_frame_sync_sel: Rx frame sync signal is provided by which source. + * External source or by frame generator logic. + * @tx_frame_sync_sel: Tx frame sync signal is provided by which source. + * External source or by frame generator logic. + * @rx_fifo_config: Receive fifo enable or not. + * @tx_fifo_config: Transmit fifo enable or not. + * @spi_clk_mode: In case of SPI protocol spi modes: Normal, Zero delay or + * half cycle delay. + * @spi_burst_mode: Spi burst mode is enabled or not. + * @loopback_enable: Loopback mode. + * @tx_data_enable: Transmit extra delay enable. + * @default_protocol_desc: Flag to indicate client defined protocol desc or + * statically defined in msp.h. + * @protocol_desc: Protocol desc structure filled by i2s client driver. + * In case client defined default_prtocol_desc as 0. + * @multichannel_configured: multichannel configuration structure. + * @multichannel_config: multichannel is enabled or not. + * @direction: Transmit, Receive or Both. + * @work_mode: Dma, Polling or Interrupt. + * @protocol: I2S, PCM, etc. + * @frame_freq: Sampling freq at which data is sampled. + * @frame_size: size of element. + * @data_size: data size which defines the format in which data is written on + * transmit or receive fifo. Only three modes 8,16,32 are supported. + * @def_elem_len: Flag to indicate whether default element length is to be used + * or should be changed acc to data size defined by user at run time. + * @iodelay: value for the MSP_IODLY register + * @handler: callback handler in case of interrupt or dma. + * @tx_callback_data: Callback data for transmit. + * @rx_callback_data: Callback data for receive. + * + * Main Msp configuration data structure used by i2s client driver to fill + * various info like data size, frequency etc. + */ +struct msp_config { + unsigned int input_clock_freq; + unsigned int rx_clock_sel; + unsigned int tx_clock_sel; + unsigned int srg_clock_sel; + unsigned int rx_frame_sync_pol; + unsigned int tx_frame_sync_pol; + unsigned int rx_frame_sync_sel; + unsigned int tx_frame_sync_sel; + unsigned int rx_fifo_config; + unsigned int tx_fifo_config; + unsigned int spi_clk_mode; + unsigned int spi_burst_mode; + unsigned int loopback_enable; + unsigned int tx_data_enable; + unsigned int default_protocol_desc; + struct msp_protocol_desc protocol_desc; + int multichannel_configured; + struct msp_multichannel_config multichannel_config; + unsigned int direction; + unsigned int work_mode; + unsigned int protocol; + unsigned int frame_freq; + unsigned int frame_size; + enum msp_data_size data_size; + unsigned int def_elem_len; + unsigned int iodelay; + void (*handler) (void *data); + void *tx_callback_data; + void *rx_callback_data; + +}; + +/*** Protocols ***/ +enum msp_protocol { + MSP_I2S_PROTOCOL, + MSP_PCM_PROTOCOL, + MSP_PCM_COMPAND_PROTOCOL, + MSP_AC97_PROTOCOL, + MSP_MASTER_SPI_PROTOCOL, + MSP_SLAVE_SPI_PROTOCOL, + MSP_INVALID_PROTOCOL +}; + +/*** Sample Frequencies ***/ +/* These are no longer required, frequencies in Hz can be used directly */ +enum msp_sample_freq { + MSP_SAMPLE_FREQ_NOT_SUPPORTED = -1, + MSP_SAMPLE_FREQ_8KHZ = 8000, + MSP_SAMPLE_FREQ_12KHZ = 12000, + MSP_SAMPLE_FREQ_16KHZ = 16000, + MSP_SAMPLE_FREQ_24KHZ = 24000, + MSP_SAMPLE_FREQ_32KHZ = 32000, + MSP_SAMPLE_FREQ_44KHZ = 44000, + MSP_SAMPLE_FREQ_48KHZ = 48000, + MSP_SAMPLE_FREQ_64KHZ = 64000, + MSP_SAMPLE_FREQ_88KHZ = 88000, + MSP_SAMPLE_FREQ_96KHZ = 96000, + MSP_SAMPLE_FREQ_22KHZ = 22000, + MSP_SAMPLE_FREQ_11KHZ = 11000 +}; + +/*** Input Frequencies ***/ +/* These are no longer required, frequencies in Hz can be used directly */ +enum msp_in_clock_freq { + MSP_INPUT_FREQ_1MHZ = 1000, + MSP_INPUT_FREQ_2MHZ = 2000, + MSP_INPUT_FREQ_3MHZ = 3000, + MSP_INPUT_FREQ_4MHZ = 4000, + MSP_INPUT_FREQ_5MHZ = 5000, + MSP_INPUT_FREQ_6MHZ = 6000, + MSP_INPUT_FREQ_8MHZ = 8000, + MSP_INPUT_FREQ_11MHZ = 11000, + MSP_INPUT_FREQ_12MHZ = 12000, + MSP_INPUT_FREQ_16MHZ = 16000, + MSP_INPUT_FREQ_22MHZ = 22000, + MSP_INPUT_FREQ_24MHZ = 24000, + MSP_INPUT_FREQ_48MHZ = 48000 +}; + +#define MSP_INPUT_FREQ_APB 48000000 + +/*** Stereo mode. Used for APB data accesses as 16 bits accesses (mono), + * 32 bits accesses (stereo). + ***/ +enum msp_stereo_mode { + MSP_MONO, + MSP_STEREO +}; + +/* Direction (Transmit/Receive mode) */ +enum msp_direction { + MSP_TRANSMIT_MODE, + MSP_RECEIVE_MODE, + MSP_BOTH_T_R_MODE +}; + +/* Dma mode should be used for large transfers, + * polling mode should be used for transfers of a few bytes + */ +enum msp_xfer_mode { + MSP_DMA_MODE, + MSP_POLLING_MODE, + MSP_INTERRUPT_MODE +}; + +/* User client for the MSP */ +enum msp_user { + MSP_NO_USER = 0, + MSP_USER_SPI, + MSP_USER_ALSA, + MSP_USER_SAA, +}; + +/*Flag structure for MSPx*/ +struct msp_flag { + struct semaphore lock; + enum msp_user user; +}; + +/* User client for the MSP */ +enum msp_mode { + MSP_NO_MODE = 0, + MSP_MODE_SPI, + MSP_MODE_NON_SPI, +}; + +/* Transmit and receive configuration register */ +#define MSP_BIG_ENDIAN 0x00000000 +#define MSP_LITTLE_ENDIAN 0x00001000 +#define MSP_UNEXPECTED_FS_ABORT 0x00000000 +#define MSP_UNEXPECTED_FS_IGNORE 0x00008000 +#define MSP_NON_MODE_BIT_MASK 0x00009000 + +/* Global configuration register */ +#define RX_ENABLE 0x00000001 +#define RX_FIFO_ENABLE 0x00000002 +#define RX_SYNC_SRG 0x00000010 +#define RX_CLK_POL_RISING 0x00000020 +#define RX_CLK_SEL_SRG 0x00000040 +#define TX_ENABLE 0x00000100 +#define TX_FIFO_ENABLE 0x00000200 +#define TX_SYNC_SRG_PROG 0x00001800 +#define TX_SYNC_SRG_AUTO 0x00001000 +#define TX_CLK_POL_RISING 0x00002000 +#define TX_CLK_SEL_SRG 0x00004000 +#define TX_EXTRA_DELAY_ENABLE 0x00008000 +#define SRG_ENABLE 0x00010000 +#define FRAME_GEN_ENABLE 0x00100000 +#define SRG_CLK_SEL_APB 0x00000000 +#define RX_FIFO_SYNC_HI 0x00000000 +#define TX_FIFO_SYNC_HI 0x00000000 +#define SPI_CLK_MODE_NORMAL 0x00000000 + +/* SPI Clock Modes enumertion + * SPI clock modes of MSP provides compatibility with + * the SPI protocol.MSP supports 2 SPI transfer formats. + * MSP_ZERO_DELAY_SPI_MODE:MSP transmits data over Tx/Rx + * Lines immediately after MSPTCK/MSPRCK rising/falling edge. + * MSP_HALF_CYCLE_DELY_SPI_MODE:MSP transmits data one-half cycle + * ahead of the rising/falling edge of the MSPTCK + */ + +#define MSP_FRAME_SIZE_AUTO -1 + + +#define MSP_DR 0x00 +#define MSP_GCR 0x04 +#define MSP_TCF 0x08 +#define MSP_RCF 0x0c +#define MSP_SRG 0x10 +#define MSP_FLR 0x14 +#define MSP_DMACR 0x18 + +#define MSP_IMSC 0x20 +#define MSP_RIS 0x24 +#define MSP_MIS 0x28 +#define MSP_ICR 0x2c +#define MSP_MCR 0x30 +#define MSP_RCV 0x34 +#define MSP_RCM 0x38 + +#define MSP_TCE0 0x40 +#define MSP_TCE1 0x44 +#define MSP_TCE2 0x48 +#define MSP_TCE3 0x4c + +#define MSP_RCE0 0x60 +#define MSP_RCE1 0x64 +#define MSP_RCE2 0x68 +#define MSP_RCE3 0x6c +#define MSP_IODLY 0x70 + +#define MSP_ITCR 0x80 +#define MSP_ITIP 0x84 +#define MSP_ITOP 0x88 +#define MSP_TSTDR 0x8c + +#define MSP_PID0 0xfe0 +#define MSP_PID1 0xfe4 +#define MSP_PID2 0xfe8 +#define MSP_PID3 0xfec + +#define MSP_CID0 0xff0 +#define MSP_CID1 0xff4 +#define MSP_CID2 0xff8 +#define MSP_CID3 0xffc + +/* Single or dual phase mode */ +enum msp_phase_mode { + MSP_SINGLE_PHASE, + MSP_DUAL_PHASE +}; + +/* Frame length */ +enum msp_frame_length { + MSP_FRAME_LENGTH_1 = 0, + MSP_FRAME_LENGTH_2 = 1, + MSP_FRAME_LENGTH_4 = 3, + MSP_FRAME_LENGTH_8 = 7, + MSP_FRAME_LENGTH_12 = 11, + MSP_FRAME_LENGTH_16 = 15, + MSP_FRAME_LENGTH_20 = 19, + MSP_FRAME_LENGTH_32 = 31, + MSP_FRAME_LENGTH_48 = 47, + MSP_FRAME_LENGTH_64 = 63 +}; + +/* Element length */ +enum msp_elem_length { + MSP_ELEM_LENGTH_8 = 0, + MSP_ELEM_LENGTH_10 = 1, + MSP_ELEM_LENGTH_12 = 2, + MSP_ELEM_LENGTH_14 = 3, + MSP_ELEM_LENGTH_16 = 4, + MSP_ELEM_LENGTH_20 = 5, + MSP_ELEM_LENGTH_24 = 6, + MSP_ELEM_LENGTH_32 = 7 +}; + +enum msp_data_xfer_width { + MSP_DATA_TRANSFER_WIDTH_BYTE, + MSP_DATA_TRANSFER_WIDTH_HALFWORD, + MSP_DATA_TRANSFER_WIDTH_WORD +}; + +enum msp_frame_sync { + MSP_FRAME_SYNC_UNIGNORE = 0, + MSP_FRAME_SYNC_IGNORE = 1, + +}; + +enum msp_phase2_start_mode { + MSP_PHASE2_START_MODE_IMEDIATE, + MSP_PHASE2_START_MODE_FRAME_SYNC +}; + +enum msp_btf { + MSP_BTF_MS_BIT_FIRST = 0, + MSP_BTF_LS_BIT_FIRST = 1 +}; + +enum msp_frame_sync_pol { + MSP_FRAME_SYNC_POL_ACTIVE_HIGH = 0, + MSP_FRAME_SYNC_POL_ACTIVE_LOW = 1 +}; + +/* Data delay (in bit clock cycles) */ +enum msp_delay { + MSP_DELAY_0 = 0, + MSP_DELAY_1 = 1, + MSP_DELAY_2 = 2, + MSP_DELAY_3 = 3 +}; + +/* Configurations of clocks (transmit, receive or sample rate generator) */ +enum msp_edge { + MSP_FALLING_EDGE = 0, + MSP_RISING_EDGE = 1, +}; + +enum msp_hws { + MSP_HWS_NO_SWAP = 0, + MSP_HWS_BYTE_SWAP_IN_WORD = 1, + MSP_HWS_BYTE_SWAP_IN_EACH_HALF_WORD = 2, + MSP_HWS_HALF_WORD_SWAP_IN_WORD = 3 +}; + +enum msp_compress_mode { + MSP_COMPRESS_MODE_LINEAR = 0, + MSP_COMPRESS_MODE_MU_LAW = 2, + MSP_COMPRESS_MODE_A_LAW = 3 +}; + +enum msp_spi_clock_mode { + MSP_SPI_CLOCK_MODE_NON_SPI = 0, + MSP_SPI_CLOCK_MODE_ZERO_DELAY = 2, + MSP_SPI_CLOCK_MODE_HALF_CYCLE_DELAY = 3 +}; + +enum msp_spi_burst_mode { + MSP_SPI_BURST_MODE_DISABLE = 0, + MSP_SPI_BURST_MODE_ENABLE = 1 +}; + +enum msp_expand_mode { + MSP_EXPAND_MODE_LINEAR = 0, + MSP_EXPAND_MODE_LINEAR_SIGNED = 1, + MSP_EXPAND_MODE_MU_LAW = 2, + MSP_EXPAND_MODE_A_LAW = 3 +}; + +/* Protocol dependant parameters list */ +#define RX_ENABLE_MASK BIT(0) +#define RX_FIFO_ENABLE_MASK BIT(1) +#define RX_FRAME_SYNC_MASK BIT(2) +#define DIRECT_COMPANDING_MASK BIT(3) +#define RX_SYNC_SEL_MASK BIT(4) +#define RX_CLK_POL_MASK BIT(5) +#define RX_CLK_SEL_MASK BIT(6) +#define LOOPBACK_MASK BIT(7) +#define TX_ENABLE_MASK BIT(8) +#define TX_FIFO_ENABLE_MASK BIT(9) +#define TX_FRAME_SYNC_MASK BIT(10) +#define TX_MSP_TDR_TSR BIT(11) +#define TX_SYNC_SEL_MASK (BIT(12) | BIT(11)) +#define TX_CLK_POL_MASK BIT(13) +#define TX_CLK_SEL_MASK BIT(14) +#define TX_EXTRA_DELAY_MASK BIT(15) +#define SRG_ENABLE_MASK BIT(16) +#define SRG_CLK_POL_MASK BIT(17) +#define SRG_CLK_SEL_MASK (BIT(19) | BIT(18)) +#define FRAME_GEN_EN_MASK BIT(20) +#define SPI_CLK_MODE_MASK (BIT(22) | BIT(21)) +#define SPI_BURST_MODE_MASK BIT(23) + +#define RXEN_SHIFT 0 +#define RFFEN_SHIFT 1 +#define RFSPOL_SHIFT 2 +#define DCM_SHIFT 3 +#define RFSSEL_SHIFT 4 +#define RCKPOL_SHIFT 5 +#define RCKSEL_SHIFT 6 +#define LBM_SHIFT 7 +#define TXEN_SHIFT 8 +#define TFFEN_SHIFT 9 +#define TFSPOL_SHIFT 10 +#define TFSSEL_SHIFT 11 +#define TCKPOL_SHIFT 13 +#define TCKSEL_SHIFT 14 +#define TXDDL_SHIFT 15 +#define SGEN_SHIFT 16 +#define SCKPOL_SHIFT 17 +#define SCKSEL_SHIFT 18 +#define FGEN_SHIFT 20 +#define SPICKM_SHIFT 21 +#define TBSWAP_SHIFT 28 + +#define RCKPOL_MASK BIT(0) +#define TCKPOL_MASK BIT(0) +#define SPICKM_MASK (BIT(1) | BIT(0)) +#define MSP_RX_CLKPOL_BIT(n) ((n & RCKPOL_MASK) << RCKPOL_SHIFT) +#define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT) +#define MSP_SPI_CLK_MODE_BITS(n) ((n & SPICKM_MASK) << SPICKM_SHIFT) + + + +/* Use this to clear the clock mode bits to non-spi */ +#define MSP_NON_SPI_CLK_MASK (BIT(22) | BIT(21)) + +#define P1ELEN_SHIFT 0 +#define P1FLEN_SHIFT 3 +#define DTYP_SHIFT 10 +#define ENDN_SHIFT 12 +#define DDLY_SHIFT 13 +#define FSIG_SHIFT 15 +#define P2ELEN_SHIFT 16 +#define P2FLEN_SHIFT 19 +#define P2SM_SHIFT 26 +#define P2EN_SHIFT 27 +#define FRAME_SYNC_SHIFT 15 + + +#define P1ELEN_MASK 0x00000007 +#define P2ELEN_MASK 0x00070000 +#define P1FLEN_MASK 0x00000378 +#define P2FLEN_MASK 0x03780000 +#define DDLY_MASK 0x00003000 +#define DTYP_MASK 0x00000600 +#define P2SM_MASK 0x04000000 +#define P2EN_MASK 0x08000000 +#define ENDN_MASK 0x00001000 +#define TFSPOL_MASK 0x00000400 +#define TBSWAP_MASK 0x30000000 +#define COMPANDING_MODE_MASK 0x00000c00 +#define FRAME_SYNC_MASK 0x00008000 + +#define MSP_P1_ELEM_LEN_BITS(n) (n & P1ELEN_MASK) +#define MSP_P2_ELEM_LEN_BITS(n) (((n) << P2ELEN_SHIFT) & P2ELEN_MASK) +#define MSP_P1_FRAME_LEN_BITS(n) (((n) << P1FLEN_SHIFT) & P1FLEN_MASK) +#define MSP_P2_FRAME_LEN_BITS(n) (((n) << P2FLEN_SHIFT) & P2FLEN_MASK) +#define MSP_DATA_DELAY_BITS(n) (((n) << DDLY_SHIFT) & DDLY_MASK) +#define MSP_DATA_TYPE_BITS(n) (((n) << DTYP_SHIFT) & DTYP_MASK) +#define MSP_P2_START_MODE_BIT(n) ((n << P2SM_SHIFT) & P2SM_MASK) +#define MSP_P2_ENABLE_BIT(n) ((n << P2EN_SHIFT) & P2EN_MASK) +#define MSP_SET_ENDIANNES_BIT(n) ((n << ENDN_SHIFT) & ENDN_MASK) +#define MSP_FRAME_SYNC_POL(n) ((n << TFSPOL_SHIFT) & TFSPOL_MASK) +#define MSP_DATA_WORD_SWAP(n) ((n << TBSWAP_SHIFT) & TBSWAP_MASK) +#define MSP_SET_COMPANDING_MODE(n) ((n << DTYP_SHIFT) & COMPANDING_MODE_MASK) +#define MSP_SET_FRAME_SYNC_IGNORE(n) ((n << FRAME_SYNC_SHIFT) & \ + FRAME_SYNC_MASK) + +/* Flag register */ +#define RX_BUSY BIT(0) +#define RX_FIFO_EMPTY BIT(1) +#define RX_FIFO_FULL BIT(2) +#define TX_BUSY BIT(3) +#define TX_FIFO_EMPTY BIT(4) +#define TX_FIFO_FULL BIT(5) + +#define RBUSY_SHIFT 0 +#define RFE_SHIFT 1 +#define RFU_SHIFT 2 +#define TBUSY_SHIFT 3 +#define TFE_SHIFT 4 +#define TFU_SHIFT 5 + +/* Multichannel control register */ +#define RMCEN_SHIFT 0 +#define RMCSF_SHIFT 1 +#define RCMPM_SHIFT 3 +#define TMCEN_SHIFT 5 +#define TNCSF_SHIFT 6 + +/* Sample rate generator register */ +#define SCKDIV_SHIFT 0 +#define FRWID_SHIFT 10 +#define FRPER_SHIFT 16 + +#define SCK_DIV_MASK 0x0000003FF +#define FRAME_WIDTH_BITS(n) (((n) << FRWID_SHIFT) & 0x0000FC00) +#define FRAME_PERIOD_BITS(n) (((n) << FRPER_SHIFT) & 0x1FFF0000) + +/* DMA controller register */ +#define RX_DMA_ENABLE BIT(0) +#define TX_DMA_ENABLE BIT(1) + +#define RDMAE_SHIFT 0 +#define TDMAE_SHIFT 1 + +/* Interrupt Register */ +#define RECEIVE_SERVICE_INT BIT(0) +#define RECEIVE_OVERRUN_ERROR_INT BIT(1) +#define RECEIVE_FRAME_SYNC_ERR_INT BIT(2) +#define RECEIVE_FRAME_SYNC_INT BIT(3) +#define TRANSMIT_SERVICE_INT BIT(4) +#define TRANSMIT_UNDERRUN_ERR_INT BIT(5) +#define TRANSMIT_FRAME_SYNC_ERR_INT BIT(6) +#define TRANSMIT_FRAME_SYNC_INT BIT(7) +#define ALL_INT 0x000000ff + +/* MSP test control register */ +#define MSP_ITCR_ITEN BIT(0) +#define MSP_ITCR_TESTFIFO BIT(1) + +/* + * Protocol configuration values I2S: + * Single phase, 16 bits, 2 words per frame + */ +#define I2S_PROTOCOL_DESC \ +{ \ + MSP_SINGLE_PHASE, \ + MSP_SINGLE_PHASE, \ + MSP_PHASE2_START_MODE_IMEDIATE, \ + MSP_PHASE2_START_MODE_IMEDIATE, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_ELEM_LENGTH_32, \ + MSP_ELEM_LENGTH_32, \ + MSP_ELEM_LENGTH_32, \ + MSP_ELEM_LENGTH_32, \ + MSP_DELAY_1, \ + MSP_DELAY_1, \ + MSP_RISING_EDGE, \ + MSP_FALLING_EDGE, \ + MSP_FRAME_SYNC_POL_ACTIVE_LOW, \ + MSP_FRAME_SYNC_POL_ACTIVE_LOW, \ + MSP_HWS_NO_SWAP, \ + MSP_HWS_NO_SWAP, \ + MSP_COMPRESS_MODE_LINEAR, \ + MSP_EXPAND_MODE_LINEAR, \ + MSP_SPI_CLOCK_MODE_NON_SPI, \ + MSP_SPI_BURST_MODE_DISABLE, \ + MSP_FRAME_SYNC_IGNORE, \ + 31, \ + 15, \ + 32, \ +} + +#define PCM_PROTOCOL_DESC \ +{ \ + MSP_DUAL_PHASE, \ + MSP_DUAL_PHASE, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_ELEM_LENGTH_16, \ + MSP_ELEM_LENGTH_16, \ + MSP_ELEM_LENGTH_16, \ + MSP_ELEM_LENGTH_16, \ + MSP_DELAY_0, \ + MSP_DELAY_0, \ + MSP_RISING_EDGE, \ + MSP_FALLING_EDGE, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_HWS_NO_SWAP, \ + MSP_HWS_NO_SWAP, \ + MSP_COMPRESS_MODE_LINEAR, \ + MSP_EXPAND_MODE_LINEAR, \ + MSP_SPI_CLOCK_MODE_NON_SPI, \ + MSP_SPI_BURST_MODE_DISABLE, \ + MSP_FRAME_SYNC_IGNORE, \ + 255, \ + 0, \ + 256, \ +} + +/* Companded PCM: Single phase, 8 bits, 1 word per frame */ +#define PCM_COMPAND_PROTOCOL_DESC \ +{ \ + MSP_SINGLE_PHASE, \ + MSP_SINGLE_PHASE, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_DELAY_0, \ + MSP_DELAY_0, \ + MSP_RISING_EDGE, \ + MSP_RISING_EDGE, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_HWS_NO_SWAP, \ + MSP_HWS_NO_SWAP, \ + MSP_COMPRESS_MODE_LINEAR, \ + MSP_EXPAND_MODE_LINEAR, \ + MSP_SPI_CLOCK_MODE_NON_SPI, \ + MSP_SPI_BURST_MODE_DISABLE, \ + MSP_FRAME_SYNC_IGNORE, \ + 255, \ + 0, \ + 256, \ +} + +/* + * AC97: Double phase, 1 element of 16 bits during first phase, + * 12 elements of 20 bits in second phase. + */ +#define AC97_PROTOCOL_DESC \ +{ \ + MSP_DUAL_PHASE, \ + MSP_DUAL_PHASE, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_12, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_12, \ + MSP_ELEM_LENGTH_16, \ + MSP_ELEM_LENGTH_20, \ + MSP_ELEM_LENGTH_16, \ + MSP_ELEM_LENGTH_20, \ + MSP_DELAY_1, \ + MSP_DELAY_1, \ + MSP_RISING_EDGE, \ + MSP_RISING_EDGE, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_HWS_NO_SWAP, \ + MSP_HWS_NO_SWAP, \ + MSP_COMPRESS_MODE_LINEAR, \ + MSP_EXPAND_MODE_LINEAR, \ + MSP_SPI_CLOCK_MODE_NON_SPI, \ + MSP_SPI_BURST_MODE_DISABLE, \ + MSP_FRAME_SYNC_IGNORE, \ + 255, \ + 0, \ + 256, \ +} + +#define SPI_MASTER_PROTOCOL_DESC \ +{ \ + MSP_SINGLE_PHASE, \ + MSP_SINGLE_PHASE, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_DELAY_1, \ + MSP_DELAY_1, \ + MSP_FALLING_EDGE, \ + MSP_FALLING_EDGE, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_HWS_NO_SWAP, \ + MSP_HWS_NO_SWAP, \ + MSP_COMPRESS_MODE_LINEAR, \ + MSP_EXPAND_MODE_LINEAR, \ + MSP_SPI_CLOCK_MODE_NON_SPI, \ + MSP_SPI_BURST_MODE_DISABLE, \ + MSP_FRAME_SYNC_IGNORE, \ + 255, \ + 0, \ + 256, \ +} + +#define SPI_SLAVE_PROTOCOL_DESC \ +{ \ + MSP_SINGLE_PHASE, \ + MSP_SINGLE_PHASE, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_PHASE2_START_MODE_FRAME_SYNC, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_BTF_MS_BIT_FIRST, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_FRAME_LENGTH_1, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_ELEM_LENGTH_8, \ + MSP_DELAY_1, \ + MSP_DELAY_1, \ + MSP_FALLING_EDGE, \ + MSP_FALLING_EDGE, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_FRAME_SYNC_POL_ACTIVE_HIGH, \ + MSP_HWS_NO_SWAP, \ + MSP_HWS_NO_SWAP, \ + MSP_COMPRESS_MODE_LINEAR, \ + MSP_EXPAND_MODE_LINEAR, \ + MSP_SPI_CLOCK_MODE_NON_SPI, \ + MSP_SPI_BURST_MODE_DISABLE, \ + MSP_FRAME_SYNC_IGNORE, \ + 255, \ + 0, \ + 256, \ +} + +#define MSP_FRAME_PERIOD_IN_MONO_MODE 256 +#define MSP_FRAME_PERIOD_IN_STEREO_MODE 32 +#define MSP_FRAME_WIDTH_IN_STEREO_MODE 16 + +/* + * No of registers to backup during + * suspend resume + */ +#define MAX_MSP_BACKUP_REGS 36 + +enum enum_i2s_controller { + MSP_0_I2S_CONTROLLER = 0, + MSP_1_I2S_CONTROLLER, + MSP_2_I2S_CONTROLLER, + MSP_3_I2S_CONTROLLER, +}; + +/** + * struct msp - Main msp controller data structure per MSP. + * @work_mode: Mode i.e dma, polling or interrupt. + * @id: Controller id like MSP1 or MSP2 etc. + * @msp_io_error: To indicate error while transferring. + * @registers: MSP's register base address. + * @actual_data_size: Data size in which data needs to send or receive. + * @irq: MSP's irq number. + * @i2s_cont: MSP's Controller's structure pointer created per MSP. + * @lock: semaphore lock acquired while configuring msp. + * @dma_cfg_tx: TX DMA configuration + * @dma_cfg_rx: RX DMA configuration + * @tx_pipeid: TX DMA channel + * @rx_pipeid: RX DMA channel + * @msp_state: Current state of msp. + * @read: Function pointer for read, u8_msp_read,u16_msp_read,u32_msp_read. + * @write: Function pointer for write, u8_msp_write,u16_msp_write,u32_msp_write. + * @transfer: Function pointer for type of transfer i.e dma,polling or interrupt + * @xfer_data: MSP's transfer data structure. Contains info about current xfer. + * @plat_init: MSP's initialization function. + * @plat_exit: MSP's Exit function. + * @notify_timer: Timer used in Polling mode to prevent hang. + * @polling_flag: Flag used in error handling while polling. + * @def_elem_len: Flag indicates whether default elem len to be used in + * protocol_desc or not. + * @reg_enabled: Flag indicates whether regulator has been enabled or not. + * @vape_opp_constraint: 1 if constraint is applied to have vape at 100OPP; 0 otherwise + * @infinite: true if an infinite transfer has been configured + * + * Main Msp private data structure to be used to store various info of a + * particular MSP.Longer description + */ +struct msp { + int work_mode; + enum enum_i2s_controller id; + int msp_io_error; + void __iomem *registers; + enum msp_data_size actual_data_size; + struct device *dev; + int irq; + struct i2s_controller *i2s_cont; + struct semaphore lock; + struct stedma40_chan_cfg *dma_cfg_rx; + struct stedma40_chan_cfg *dma_cfg_tx; + struct dma_chan *tx_pipeid; + struct dma_chan *rx_pipeid; + enum msp_state msp_state; + void (*read) (struct trans_data *xfer_data); + void (*write) (struct trans_data *xfer_data); + int (*transfer) (struct msp *msp, struct i2s_message *message); + struct trans_data xfer_data; + int (*plat_init) (void); + int (*plat_exit) (void); + struct timer_list notify_timer; + int polling_flag; + int def_elem_len; + struct clk *clk; + unsigned int direction; + int users; + int reg_enabled; + int loopback_enable; + u32 backup_regs[MAX_MSP_BACKUP_REGS]; + int vape_opp_constraint; + bool infinite; +}; + +/** + * struct msp_i2s_platform_data - Main msp controller platform data structure. + * @id: Controller id like MSP1 or MSP2 etc. + * @msp_i2s_dma_rx: RX DMA channel config + * @msp_i2s_dma_tx: RX DMA channel config + * @msp_i2s_init: MSP's initialization function. + * @msp_i2s_exit: MSP's Exit function. + * @backup_regs: used for backup registers during suspend resume. + * + * Platform data structure passed by devices.c file. + */ +struct msp_i2s_platform_data { + enum enum_i2s_controller id; + struct stedma40_chan_cfg *msp_i2s_dma_rx; + struct stedma40_chan_cfg *msp_i2s_dma_tx; + int (*msp_i2s_init) (void); + int (*msp_i2s_exit) (void); +}; + +#endif diff --git a/include/sound/ux500_ab8500.h b/include/sound/ux500_ab8500.h new file mode 100644 index 0000000..7858bfd --- /dev/null +++ b/include/sound/ux500_ab8500.h @@ -0,0 +1,36 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Jarmo K. Kuronen <jarmo.kuronen(a)symbio.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 UX500_AB8500_H +#define UX500_AB8500_H + +extern struct snd_soc_ops ux500_ab8500_ops[]; + +struct snd_soc_pcm_runtime; + +int ux500_ab8500_startup(struct snd_pcm_substream *substream); + +void ux500_ab8500_shutdown(struct snd_pcm_substream *substream); + +int ux500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); + +int ux500_ab8500_soc_machine_drv_init(void); + +void ux500_ab8500_soc_machine_drv_cleanup(void); + +int ux500_ab8500_machine_codec_init(struct snd_soc_pcm_runtime *runtime); + +extern void ux500_ab8500_jack_report(int); + +#endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 35e662d..10c66dc 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -45,6 +45,7 @@ source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" +source "sound/soc/ux500/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9ea8ac8..c5d3966 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -22,3 +22,4 @@ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ +obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c205e7..a02b2c3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 + select SND_SOC_AB8500 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI @@ -124,6 +125,9 @@ config SND_SOC_WM_HUBS default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m +config SND_SOC_AB8500 + tristate + config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de80781..7b3e406 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-88pm860x-objs := 88pm860x-codec.o +snd-soc-ab8500_audio-objs := ab8500_audio.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o @@ -101,6 +102,7 @@ snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o +obj-$(CONFIG_SND_SOC_AB8500) += snd-soc-ab8500_audio.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o @@ -201,3 +203,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o + +ifdef CONFIG_SND_SOC_UX500_DEBUG +CFLAGS_ab8500_audio.o := -DDEBUG +endif diff --git a/sound/soc/codecs/Makefile.rej b/sound/soc/codecs/Makefile.rej new file mode 100644 index 0000000..893304b --- /dev/null +++ b/sound/soc/codecs/Makefile.rej @@ -0,0 +1,10 @@ +--- sound/soc/codecs/Makefile ++++ sound/soc/codecs/Makefile +@@ -197,3 +199,7 @@ + obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o + obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o + obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o ++ ++ifdef CONFIG_SND_SOC_UX500_DEBUG ++CFLAGS_ab8500_audio.o := -DDEBUG ++endif diff --git a/sound/soc/codecs/ab8500_audio.c b/sound/soc/codecs/ab8500_audio.c new file mode 100644 index 0000000..6b7b1c0 --- /dev/null +++ b/sound/soc/codecs/ab8500_audio.c @@ -0,0 +1,2928 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Mikko J. Lehto <mikko.lehto(a)symbio.com>, + * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, + * Jarmo K. Kuronen <jarmo.kuronen(a)symbio.com>, + * Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <linux/mfd/abx500/ab8500.h> +#include <linux/mfd/abx500.h> +#include <linux/mfd/abx500/ab5500.h> +#include <linux/mfd/abx500/ab8500-sysctrl.h> +#include "ab8500_audio.h" + +/* To convert register definition shifts to masks */ +#define BMASK(bsft) (1 << (bsft)) + +/* Macrocell value definitions */ +#define CLK_32K_OUT2_DISABLE 0x01 +#define INACTIVE_RESET_AUDIO 0x02 +#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK 0x10 +#define ENABLE_VINTCORE12_SUPPLY 0x04 +#define GPIO27_DIR_OUTPUT 0x04 +#define GPIO29_DIR_OUTPUT 0x10 +#define GPIO31_DIR_OUTPUT 0x40 + +/* Macrocell register definitions */ +#define AB8500_CTRL3_REG 0x0200 +#define AB8500_GPIO_DIR4_REG 0x1013 + +/* Nr of FIR/IIR-coeff banks in ANC-block */ +#define AB8500_NR_OF_ANC_COEFF_BANKS 2 + +/* Macros to simplify implementation of register write sequences and error handling */ +#define AB8500_SET_BIT_LOCKED(xreg, xbit, xerr, xerr_hdl) { \ + xerr = snd_soc_update_bits_locked(ab8500_codec, xreg, REG_MASK_NONE, BMASK(xbit)); \ + if (xerr < 0) \ + goto xerr_hdl; } +#define AB8500_CLEAR_BIT_LOCKED(xreg, xbit, xerr, xerr_hdl) { \ + xerr = snd_soc_update_bits_locked(ab8500_codec, xreg, BMASK(xbit), REG_MASK_NONE); \ + if (xerr < 0) \ + goto xerr_hdl; } +#define AB8500_WRITE(xreg, xvalue, xerr, xerr_hdl) { \ + xerr = snd_soc_write(ab8500_codec, xreg, xvalue); \ + if (xerr < 0) \ + goto xerr_hdl; } + +/* + * AB8500 register cache & default register settings + */ +static const u8 ab8500_reg_cache[AB8500_CACHEREGNUM] = { + 0x00, /* REG_POWERUP (0x00) */ + 0x00, /* REG_AUDSWRESET (0x01) */ + 0x00, /* REG_ADPATHENA (0x02) */ + 0x00, /* REG_DAPATHENA (0x03) */ + 0x00, /* REG_ANACONF1 (0x04) */ + 0x0F, /* REG_ANACONF2 (0x05) */ + 0x00, /* REG_DIGMICCONF (0x06) */ + 0x00, /* REG_ANACONF3 (0x07) */ + 0x00, /* REG_ANACONF4 (0x08) */ + 0x00, /* REG_DAPATHCONF (0x09) */ + 0x40, /* REG_MUTECONF (0x0A) */ + 0x00, /* REG_SHORTCIRCONF (0x0B) */ + 0x01, /* REG_ANACONF5 (0x0C) */ + 0x00, /* REG_ENVCPCONF (0x0D) */ + 0x00, /* REG_SIGENVCONF (0x0E) */ + 0x3F, /* REG_PWMGENCONF1 (0x0F) */ + 0x32, /* REG_PWMGENCONF2 (0x10) */ + 0x32, /* REG_PWMGENCONF3 (0x11) */ + 0x32, /* REG_PWMGENCONF4 (0x12) */ + 0x32, /* REG_PWMGENCONF5 (0x13) */ + 0x0F, /* REG_ANAGAIN1 (0x14) */ + 0x0F, /* REG_ANAGAIN2 (0x15) */ + 0x22, /* REG_ANAGAIN3 (0x16) */ + 0x55, /* REG_ANAGAIN4 (0x17) */ + 0x13, /* REG_DIGLINHSLGAIN (0x18) */ + 0x13, /* REG_DIGLINHSRGAIN (0x19) */ + 0x00, /* REG_ADFILTCONF (0x1A) */ + 0x00, /* REG_DIGIFCONF1 (0x1B) */ + 0x02, /* REG_DIGIFCONF2 (0x1C) */ + 0x00, /* REG_DIGIFCONF3 (0x1D) */ + 0x02, /* REG_DIGIFCONF4 (0x1E) */ + 0xCC, /* REG_ADSLOTSEL1 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL2 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL3 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL4 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL5 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL6 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL7 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL8 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL9 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL10 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL11 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL12 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL13 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL14 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL15 (0xCC) */ + 0xCC, /* REG_ADSLOTSEL16 (0xCC) */ + 0x00, /* REG_ADSLOTHIZCTRL1 (0x2F) */ + 0x00, /* REG_ADSLOTHIZCTRL2 (0x30) */ + 0x00, /* REG_ADSLOTHIZCTRL3 (0x31) */ + 0x00, /* REG_ADSLOTHIZCTRL4 (0x32) */ + 0x08, /* REG_DASLOTCONF1 (0x33) */ + 0x08, /* REG_DASLOTCONF2 (0x34) */ + 0x08, /* REG_DASLOTCONF3 (0x35) */ + 0x08, /* REG_DASLOTCONF4 (0x36) */ + 0x08, /* REG_DASLOTCONF5 (0x37) */ + 0x08, /* REG_DASLOTCONF6 (0x38) */ + 0x08, /* REG_DASLOTCONF7 (0x39) */ + 0x08, /* REG_DASLOTCONF8 (0x3A) */ + 0x00, /* REG_CLASSDCONF1 (0x3B) */ + 0x00, /* REG_CLASSDCONF2 (0x3C) */ + 0x84, /* REG_CLASSDCONF3 (0x3D) */ + 0x00, /* REG_DMICFILTCONF (0x3E) */ + 0xFE, /* REG_DIGMULTCONF1 (0x3F) */ + 0xC0, /* REG_DIGMULTCONF2 (0x40) */ + 0x3F, /* REG_ADDIGGAIN1 (0x41) */ + 0x3F, /* REG_ADDIGGAIN2 (0x42) */ + 0x1F, /* REG_ADDIGGAIN3 (0x43) */ + 0x1F, /* REG_ADDIGGAIN4 (0x44) */ + 0x3F, /* REG_ADDIGGAIN5 (0x45) */ + 0x3F, /* REG_ADDIGGAIN6 (0x46) */ + 0x1F, /* REG_DADIGGAIN1 (0x47) */ + 0x1F, /* REG_DADIGGAIN2 (0x48) */ + 0x3F, /* REG_DADIGGAIN3 (0x49) */ + 0x3F, /* REG_DADIGGAIN4 (0x4A) */ + 0x3F, /* REG_DADIGGAIN5 (0x4B) */ + 0x3F, /* REG_DADIGGAIN6 (0x4C) */ + 0x3F, /* REG_ADDIGLOOPGAIN1 (0x4D) */ + 0x3F, /* REG_ADDIGLOOPGAIN2 (0x4E) */ + 0x00, /* REG_HSLEARDIGGAIN (0x4F) */ + 0x00, /* REG_HSRDIGGAIN (0x50) */ + 0x1F, /* REG_SIDFIRGAIN1 (0x51) */ + 0x1F, /* REG_SIDFIRGAIN2 (0x52) */ + 0x00, /* REG_ANCCONF1 (0x53) */ + 0x00, /* REG_ANCCONF2 (0x54) */ + 0x00, /* REG_ANCCONF3 (0x55) */ + 0x00, /* REG_ANCCONF4 (0x56) */ + 0x00, /* REG_ANCCONF5 (0x57) */ + 0x00, /* REG_ANCCONF6 (0x58) */ + 0x00, /* REG_ANCCONF7 (0x59) */ + 0x00, /* REG_ANCCONF8 (0x5A) */ + 0x00, /* REG_ANCCONF9 (0x5B) */ + 0x00, /* REG_ANCCONF10 (0x5C) */ + 0x00, /* REG_ANCCONF11 (0x5D) - read only */ + 0x00, /* REG_ANCCONF12 (0x5E) - read only */ + 0x00, /* REG_ANCCONF13 (0x5F) - read only */ + 0x00, /* REG_ANCCONF14 (0x60) - read only */ + 0x00, /* REG_SIDFIRADR (0x61) */ + 0x00, /* REG_SIDFIRCOEF1 (0x62) */ + 0x00, /* REG_SIDFIRCOEF2 (0x63) */ + 0x00, /* REG_SIDFIRCONF (0x64) */ + 0x00, /* REG_AUDINTMASK1 (0x65) */ + 0x00, /* REG_AUDINTSOURCE1 (0x66) - read only */ + 0x00, /* REG_AUDINTMASK2 (0x67) */ + 0x00, /* REG_AUDINTSOURCE2 (0x68) - read only */ + 0x00, /* REG_FIFOCONF1 (0x69) */ + 0x00, /* REG_FIFOCONF2 (0x6A) */ + 0x00, /* REG_FIFOCONF3 (0x6B) */ + 0x00, /* REG_FIFOCONF4 (0x6C) */ + 0x00, /* REG_FIFOCONF5 (0x6D) */ + 0x00, /* REG_FIFOCONF6 (0x6E) */ + 0x02, /* REG_AUDREV (0x6F) - read only */ +}; + +static struct snd_soc_codec *ab8500_codec; + +/* Signed multi register array controls. */ +struct soc_smra_control { + unsigned int *reg; + const unsigned int rcount, count, invert; + long min, max; + const char **texts; + long *values; +}; + +/* ANC FIR- & IIR-coeff caches */ +static int ab8500_anc_fir_coeff_cache[REG_ANC_FIR_COEFFS]; +static int ab8500_anc_iir_coeff_cache[REG_ANC_IIR_COEFFS]; + +/* ANC states */ +enum anc_states { + ANC_UNCONFIGURED = 0, + ANC_CONFIGURE_FIR_IIR = 1, + ANC_FIR_IIR_CONFIGURED = 2, + ANC_CONFIGURE_FIR = 3, + ANC_FIR_CONFIGURED = 4, + ANC_CONFIGURE_IIR = 5, + ANC_IIR_CONFIGURED = 6, + ANC_ERROR = 7 +}; +static int ab8500_anc_status = ANC_UNCONFIGURED; + +/* ANC configuration lock */ +static DEFINE_MUTEX(ab8500_anc_conf_lock); + +/* Reads an arbitrary register from the ab8500 chip. +*/ +static int ab8500_codec_read_reg(struct snd_soc_codec *codec, unsigned int bank, + unsigned int reg) +{ + u8 value; + int status = abx500_get_register_interruptible( + codec->dev, bank, reg, &value); + + if (status < 0) { + pr_err("%s: Register (%02x:%02x) read failed (%d).\n", + __func__, (u8)bank, (u8)reg, status); + } else { + pr_debug("Read 0x%02x from register %02x:%02x\n", + (u8)value, (u8)bank, (u8)reg); + status = value; + } + + return status; +} + +/* Writes an arbitrary register to the ab8500 chip. + */ +static int ab8500_codec_write_reg(struct snd_soc_codec *codec, unsigned int bank, + unsigned int reg, unsigned int value) +{ + int status = abx500_set_register_interruptible( + codec->dev, bank, reg, value); + + if (status < 0) { + pr_err("%s: Register (%02x:%02x) write failed (%d).\n", + __func__, (u8)bank, (u8)reg, status); + } else { + pr_debug("Wrote 0x%02x into register %02x:%02x\n", + (u8)value, (u8)bank, (u8)reg); + } + + return status; +} + +/* Reads an audio register from the cache. + */ +static unsigned int ab8500_codec_read_reg_audio(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + return cache[reg]; +} + +/* Reads an audio register from the hardware. + */ +static int ab8500_codec_read_reg_audio_nocache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + int value = ab8500_codec_read_reg(codec, AB8500_AUDIO, reg); + + if (value >= 0) + cache[reg] = value; + + return value; +} + +/* Writes an audio register to the hardware and cache. + */ +static int ab8500_codec_write_reg_audio(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + int status = ab8500_codec_write_reg(codec, AB8500_AUDIO, reg, value); + + if (status >= 0) + cache[reg] = value; + + return status; +} + +/* Dumps all audio registers. + */ +static inline void ab8500_codec_dump_all_reg(struct snd_soc_codec *codec) +{ + int i; + + pr_debug("%s Enter.\n", __func__); + + for (i = AB8500_FIRST_REG; i <= AB8500_LAST_REG; i++) + ab8500_codec_read_reg_audio_nocache(codec, i); +} + +/* Updates an audio register. + */ +static inline int ab8500_codec_update_reg_audio(struct snd_soc_codec *codec, + unsigned int reg, unsigned int clr, unsigned int ins) +{ + unsigned int new, old; + + old = ab8500_codec_read_reg_audio(codec, reg); + new = (old & ~clr) | ins; + if (old == new) + return 0; + + return ab8500_codec_write_reg_audio(codec, reg, new); +} + +/* Generic soc info for signed register controls. */ +int snd_soc_info_s(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_smra_control *smra = + (struct soc_smra_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = smra->count; + uinfo->value.integer.min = smra->min; + uinfo->value.integer.max = smra->max; + + return 0; +} + +/* Generic soc get for signed multi register controls. */ +int snd_soc_get_smr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_smra_control *smra = + (struct soc_smra_control *)kcontrol->private_value; + unsigned int *reg = smra->reg; + unsigned int rcount = smra->rcount; + long min = smra->min; + long max = smra->max; + unsigned int invert = smra->invert; + unsigned long mask = abs(min) | abs(max); + long value = 0; + int i, rvalue; + + for (i = 0; i < rcount; i++) { + rvalue = snd_soc_read(codec, reg[i]) & REG_MASK_ALL; + value |= rvalue << (8 * (rcount - i - 1)); + } + value &= mask; + if (min < 0 && value > max) + value |= ~mask; + if (invert) + value = ~value; + ucontrol->value.integer.value[0] = value; + + return 0; +} + +/* Generic soc put for signed multi register controls. */ +int snd_soc_put_smr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_smra_control *smra = + (struct soc_smra_control *)kcontrol->private_value; + unsigned int *reg = smra->reg; + unsigned int rcount = smra->rcount; + long min = smra->min; + long max = smra->max; + unsigned int invert = smra->invert; + unsigned long mask = abs(min) | abs(max); + long value = ucontrol->value.integer.value[0]; + int i, rvalue, err; + + if (invert) + value = ~value; + if (value > max) + value = max; + else if (value < min) + value = min; + value &= mask; + for (i = 0; i < rcount; i++) { + rvalue = (value >> (8 * (rcount - i - 1))) & REG_MASK_ALL; + err = snd_soc_write(codec, reg[i], rvalue); + if (err < 0) + return 0; + } + + return 1; +} + +/* Generic soc get for signed array controls. */ +static int snd_soc_get_sa(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_smra_control *smra = + (struct soc_smra_control *)kcontrol->private_value; + long *values = smra->values; + unsigned int count = smra->count; + unsigned int idx; + + for (idx = 0; idx < count; idx++) + ucontrol->value.integer.value[idx] = values[idx]; + + return 0; +} + +/* Generic soc put for signed array controls. */ +static int snd_soc_put_sa(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_smra_control *smra = + (struct soc_smra_control *) kcontrol->private_value; + long *values = smra->values; + unsigned int count = smra->count; + long min = smra->min; + long max = smra->max; + unsigned int idx; + long value; + + for (idx = 0; idx < count; idx++) { + value = ucontrol->value.integer.value[idx]; + if (value > max) + value = max; + else if (value < min) + value = min; + values[idx] = value; + } + + return 0; +} + +/* Generic soc get for enum strobe controls. */ +int snd_soc_get_enum_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = + (struct soc_enum *)kcontrol->private_value; + unsigned int reg = e->reg; + unsigned int bit = e->shift_l; + unsigned int invert = e->shift_r != 0; + unsigned int value = snd_soc_read(codec, reg) & BMASK(bit); + + if (bit != 0 && value != 0) + value = value >> bit; + ucontrol->value.enumerated.item[0] = value ^ invert; + + return 0; +} + +/* Generic soc put for enum strobe controls. */ +int snd_soc_put_enum_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = + (struct soc_enum *)kcontrol->private_value; + unsigned int reg = e->reg; + unsigned int bit = e->shift_l; + unsigned int invert = e->shift_r != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int set_mask = REG_MASK_NONE; + unsigned int clr_mask = REG_MASK_NONE; + unsigned int err; + + if (strobe ^ invert) + set_mask = BMASK(bit); + else + clr_mask = BMASK(bit); + err = snd_soc_update_bits_locked(codec, reg, clr_mask, set_mask); + if (err < 0) + return err; + return snd_soc_update_bits_locked(codec, reg, set_mask, clr_mask); +} + +static const char * const enum_ena_dis[] = {"Enabled", "Disabled"}; +static const char * const enum_dis_ena[] = {"Disabled", "Enabled"}; + +/* Controls - Non-DAPM Non-ASoC */ + +/* Sidetone */ + +static int st_fir_value_control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (REG_MASK_ALL+1) * (REG_MASK_ALL+1) - 1; + + return 0; +} + +static int st_fir_value_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int st_fir_value_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret; + unsigned int val_msb = (int)ucontrol->value.integer.value[0] / (REG_MASK_ALL+1); + unsigned int val_lsb = (int)ucontrol->value.integer.value[0] - + val_msb * (REG_MASK_ALL+1); + ret = ab8500_codec_write_reg_audio(ab8500_codec, REG_SIDFIRCOEF1, val_msb); + ret |= ab8500_codec_write_reg_audio(ab8500_codec, REG_SIDFIRCOEF2, val_lsb); + if (ret < 0) { + pr_err("%s: ERROR: Failed to write FIR-coeffecient!\n", __func__); + return 0; + } + return 1; +} + +static const struct snd_kcontrol_new st_fir_value_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sidetone FIR Coeffecient Value", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = st_fir_value_control_info, + .get = st_fir_value_control_get, + .put = st_fir_value_control_put, + .private_value = 1 /* ULPCLK */ +}; + +static int st_fir_apply_control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item) { + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, "Apply"); + } else { + strcpy(uinfo->value.enumerated.name, "Ready"); + } + return 0; +} + +static int st_fir_apply_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int reg = ab8500_codec_read_reg_audio(ab8500_codec, REG_SIDFIRADR); + ucontrol->value.enumerated.item[0] = reg & BMASK(REG_SIDFIRADR_FIRSIDSET); + + return 0; +} + +static int st_fir_apply_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret; + + if (ucontrol->value.enumerated.item[0] != 0) { + ret = ab8500_codec_write_reg_audio(ab8500_codec, + REG_SIDFIRADR, + BMASK(REG_SIDFIRADR_FIRSIDSET)); + if (ret < 0) { + pr_err("%s: ERROR: Failed to apply FIR-coeffecients!\n", __func__); + return 0; + } + pr_debug("%s: FIR-coeffecients applied.\n", __func__); + } + + ret = ab8500_codec_write_reg_audio(ab8500_codec, REG_SIDFIRADR, 0); + if (ret < 0) + pr_err("%s: ERROR: Going to ready failed!\n", __func__); + + return 1; +} + +static const struct snd_kcontrol_new st_fir_apply_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sidetone FIR Apply Coeffecients", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = st_fir_apply_control_info, + .get = st_fir_apply_control_get, + .put = st_fir_apply_control_put, + .private_value = 0 /* Ready */ +}; + +/* Controls - DAPM */ + +/* Inverted order - Ascending/Descending */ +enum control_inversion { + NORMAL = 0, + INVERT = 1 +}; + +/* Headset */ + +/* Headset left - Mute */ +static const struct snd_kcontrol_new dapm_hsl_mute[] = { + SOC_DAPM_SINGLE("Playback Switch", REG_MUTECONF, REG_MUTECONF_MUTHSL, 1, INVERT), +}; + +/* Headset right - Mute */ +static const struct snd_kcontrol_new dapm_hsr_mute[] = { + SOC_DAPM_SINGLE("Playback Switch", REG_MUTECONF, REG_MUTECONF_MUTHSR, 1, INVERT), +}; + +/* Earpiece */ + +/* Earpiece - Mute */ +static const struct snd_kcontrol_new dapm_ear_mute[] = { + SOC_DAPM_SINGLE("Playback Switch", REG_MUTECONF, REG_MUTECONF_MUTEAR, 1, INVERT), +}; + +/* Earpiece source selector */ +static const char * const enum_ear_lineout_source[] = {"Headset Left", "IHF Left"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, REG_DMICFILTCONF, + REG_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source); +static const struct snd_kcontrol_new dapm_ear_lineout_source = + SOC_DAPM_ENUM("Earpiece or LineOut Mono Source", dapm_enum_ear_lineout_source); + +/* LineOut */ + +/* LineOut source selector */ +static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"}; +static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, REG_ANACONF5, + REG_ANACONF5_HSLDACTOLOL, REG_ANACONF5_HSRDACTOLOR, enum_lineout_source); +static const struct snd_kcontrol_new dapm_lineout_source[] = { + SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source), +}; + +/* LineOut */ + +/* LineOut Left - Enable/Disable */ +static const struct soc_enum enum_lineout_left = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); +static const struct snd_kcontrol_new dapm_lineout_left_mux = + SOC_DAPM_ENUM_VIRT("LineOut Left", enum_lineout_left); + +/* LineOut Right - Enable/Disable */ +static const struct soc_enum enum_lineout_right = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); +static const struct snd_kcontrol_new dapm_lineout_right_mux = + SOC_DAPM_ENUM_VIRT("LineOut Right", enum_lineout_right); + +/* LineOut/IHF - Select */ +static const char * const enum_ihf_or_lineout_select_sel[] = {"IHF", "LineOut"}; +static const struct soc_enum enum_ihf_or_lineout_select = SOC_ENUM_SINGLE(0, 0, 2, enum_ihf_or_lineout_select_sel); +static const struct snd_kcontrol_new dapm_ihf_or_lineout_select_mux = + SOC_DAPM_ENUM_VIRT("IHF or LineOut Select", enum_ihf_or_lineout_select); + + +/* IHF */ + +/* IHF - Enable/Disable */ +static const struct soc_enum enum_ihf_left = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); +static const struct snd_kcontrol_new dapm_ihf_left_mux = + SOC_DAPM_ENUM_VIRT("IHF Left", enum_ihf_left); + +static const struct soc_enum enum_ihf_right = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); +static const struct snd_kcontrol_new dapm_ihf_right_mux = + SOC_DAPM_ENUM_VIRT("IHF Right", enum_ihf_right); + +/* IHF left - ANC selector */ +static const char * const enum_ihfx_sel[] = {"Audio Path", "ANC"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ihfl_sel, REG_DIGMULTCONF2, + REG_DIGMULTCONF2_HFLSEL, enum_ihfx_sel); +static const struct snd_kcontrol_new dapm_ihfl_select[] = { + SOC_DAPM_ENUM("IHF Left Source", dapm_enum_ihfl_sel), +}; + +/* IHF right - ANC selector */ +static SOC_ENUM_SINGLE_DECL(dapm_enum_ihfr_sel, REG_DIGMULTCONF2, + REG_DIGMULTCONF2_HFRSEL, enum_ihfx_sel); +static const struct snd_kcontrol_new dapm_ihfr_select[] = { + SOC_DAPM_ENUM("IHF Right Source", dapm_enum_ihfr_sel), +}; + +/* Mic 1 */ + +/* Mic 1 - Mute */ +static const struct snd_kcontrol_new dapm_mic1_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_ANACONF2, REG_ANACONF2_MUTMIC1, 1, INVERT), +}; + +/* Mic 1 - Mic 1A or 1B selector */ +static const char * const enum_mic1ab_sel[] = {"Mic 1A", "Mic 1B"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, REG_ANACONF3, + REG_ANACONF3_MIC1SEL, enum_mic1ab_sel); +static const struct snd_kcontrol_new dapm_mic1ab_select[] = { + SOC_DAPM_ENUM("Mic 1A or 1B Select", dapm_enum_mic1ab_sel), +}; + +/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */ +static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, REG_DIGMULTCONF1, + REG_DIGMULTCONF1_AD3SEL, enum_ad3_sel); +static const struct snd_kcontrol_new dapm_ad3_select[] = { + SOC_DAPM_ENUM("AD 3 Select", dapm_enum_ad3_sel), +}; + +/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */ +static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, REG_DIGMULTCONF1, + REG_DIGMULTCONF1_AD6SEL, enum_ad6_sel); +static const struct snd_kcontrol_new dapm_ad6_select[] = { + SOC_DAPM_ENUM("AD 6 Select", dapm_enum_ad6_sel), +}; + +/* Mic 2 */ + +/* Mic 2 - Mute */ +static const struct snd_kcontrol_new dapm_mic2_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_ANACONF2, REG_ANACONF2_MUTMIC2, 1, INVERT), +}; + +/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */ +static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, REG_DIGMULTCONF1, + REG_DIGMULTCONF1_AD5SEL, enum_ad5_sel); +static const struct snd_kcontrol_new dapm_ad5_select[] = { + SOC_DAPM_ENUM("AD 5 Select", dapm_enum_ad5_sel), +}; + +/* LineIn */ + +/* LineIn left - Mute */ +static const struct snd_kcontrol_new dapm_linl_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_ANACONF2, REG_ANACONF2_MUTLINL, 1, INVERT), +}; + +/* LineIn left - AD1 - LineIn Left or DMic 1 selector */ +static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, REG_DIGMULTCONF1, + REG_DIGMULTCONF1_AD1SEL, enum_ad1_sel); +static const struct snd_kcontrol_new dapm_ad1_select[] = { + SOC_DAPM_ENUM("AD 1 Select", dapm_enum_ad1_sel), +}; + +/* LineIn right - Mute */ +static const struct snd_kcontrol_new dapm_linr_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_ANACONF2, REG_ANACONF2_MUTLINR, 1, INVERT), +}; + +/* LineIn right - Mic 2 or LineIn Right selector */ +static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, REG_ANACONF3, + REG_ANACONF3_LINRSEL, enum_mic2lr_sel); +static const struct snd_kcontrol_new dapm_mic2lr_select[] = { + SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel), +}; + +/* LineIn right - AD2 - LineIn Right or DMic2 selector */ +static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, REG_DIGMULTCONF1, + REG_DIGMULTCONF1_AD2SEL, enum_ad2_sel); +static const struct snd_kcontrol_new dapm_ad2_select[] = { + SOC_DAPM_ENUM("AD 2 Select", dapm_enum_ad2_sel), +}; + +/* DMic */ + +/* DMic 1 - Mute */ +static const struct snd_kcontrol_new dapm_dmic1_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_DIGMICCONF, + REG_DIGMICCONF_ENDMIC1, 1, NORMAL), +}; + +/* DMic 2 - Mute */ +static const struct snd_kcontrol_new dapm_dmic2_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_DIGMICCONF, + REG_DIGMICCONF_ENDMIC2, 1, NORMAL), +}; + +/* DMic 3 - Mute */ +static const struct snd_kcontrol_new dapm_dmic3_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_DIGMICCONF, + REG_DIGMICCONF_ENDMIC3, 1, NORMAL), +}; + +/* DMic 4 - Mute */ +static const struct snd_kcontrol_new dapm_dmic4_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_DIGMICCONF, + REG_DIGMICCONF_ENDMIC4, 1, NORMAL), +}; + +/* DMic 5 - Mute */ +static const struct snd_kcontrol_new dapm_dmic5_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_DIGMICCONF, + REG_DIGMICCONF_ENDMIC5, 1, NORMAL), +}; + +/* DMic 6 - Mute */ +static const struct snd_kcontrol_new dapm_dmic6_mute[] = { + SOC_DAPM_SINGLE("Capture Switch", REG_DIGMICCONF, + REG_DIGMICCONF_ENDMIC6, 1, NORMAL), +}; + +/* ANC */ + +static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6", "Mic 2 / DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, REG_DMICFILTCONF, + REG_DMICFILTCONF_ANCINSEL, enum_anc_in_sel); +static const struct snd_kcontrol_new dapm_anc_in_select[] = { + SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel), +}; + +/* ANC - Enable/Disable */ +static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_enable, REG_ANCCONF1, + REG_ANCCONF1_ENANC, enum_dis_ena); +static const struct snd_kcontrol_new dapm_anc_enable[] = { + SOC_DAPM_ENUM("ANC", dapm_enum_anc_enable), +}; + +/* ANC to Earpiece - Mute */ +static const struct snd_kcontrol_new dapm_anc_ear_mute[] = { + SOC_DAPM_SINGLE("Playback Switch", REG_DIGMULTCONF1, + REG_DIGMULTCONF1_ANCSEL, 1, NORMAL), +}; + +/* Sidetone left */ + +/* Sidetone left - Input selector */ +static const char * const enum_stfir1_in_sel[] = { + "LineIn Left", "LineIn Right", "Mic 1", "Headset Left"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, REG_DIGMULTCONF2, + REG_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel); +static const struct snd_kcontrol_new dapm_stfir1_in_select[] = { + SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel), +}; + +/* Sidetone right path */ + +/* Sidetone right - Input selector */ +static const char * const enum_stfir2_in_sel[] = { + "LineIn Right", "Mic 1", "DMic 4", "Headset Right"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, REG_DIGMULTCONF2, + REG_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel); +static const struct snd_kcontrol_new dapm_stfir2_in_select[] = { + SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel), +}; + +/* Vibra */ + +/* Vibra 1 - Enable/Disable */ +static const struct soc_enum enum_vibra1 = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); +static const struct snd_kcontrol_new dapm_vibra1_mux = + SOC_DAPM_ENUM_VIRT("Vibra 1", enum_vibra1); + +/* Vibra 2 - Enable/Disable */ +static const struct soc_enum enum_vibra2 = SOC_ENUM_SINGLE(0, 0, 2, enum_dis_ena); +static const struct snd_kcontrol_new dapm_vibra2_mux = + SOC_DAPM_ENUM_VIRT("Vibra 2", enum_vibra2); + +static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, REG_PWMGENCONF1, + REG_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib1[] = { + SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1), +}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, REG_PWMGENCONF1, + REG_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib2[] = { + SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2), +}; + +static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { + + /* DA/AD */ + + SND_SOC_DAPM_INPUT("ADC Input"), + SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC", "ab8500_0p", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("DAC Output"), + + SND_SOC_DAPM_AIF_IN("DA_IN1", "ab8500_0p", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN2", "ab8500_0p", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN3", "ab8500_0p", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN4", "ab8500_0p", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN5", "ab8500_0p", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN6", "ab8500_0p", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT1", "ab8500_0c", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT2", "ab8500_0c", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT3", "ab8500_0c", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT4", "ab8500_0c", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT57", "ab8500_0c", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT68", "ab8500_0c", 0, SND_SOC_NOPM, 0, 0), + + /* Headset path */ + + SND_SOC_DAPM_SUPPLY("Charge Pump", REG_ANACONF5, REG_ANACONF5_ENCPHS, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p", + REG_DAPATHENA, REG_DAPATHENA_ENDA1, 0), + SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p", + REG_DAPATHENA, REG_DAPATHENA_ENDA2, 0), + + SND_SOC_DAPM_SWITCH("Headset Left", SND_SOC_NOPM, 0, 0, dapm_hsl_mute), + SND_SOC_DAPM_SWITCH("Headset Right", SND_SOC_NOPM, 0, 0, dapm_hsr_mute), + + SND_SOC_DAPM_PGA("HSL Digital Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HSR Digital Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p", + REG_DAPATHCONF, REG_DAPATHCONF_ENDACHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p", + REG_DAPATHCONF, REG_DAPATHCONF_ENDACHSR, 0), + SND_SOC_DAPM_MIXER("HSL DAC Mute", REG_MUTECONF, REG_MUTECONF_MUTDACHSL, + INVERT, NULL, 0), + SND_SOC_DAPM_MIXER("HSR DAC Mute", REG_MUTECONF, REG_MUTECONF_MUTDACHSR, + INVERT, NULL, 0), + SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p", + REG_ANACONF3, REG_ANACONF3_ENDRVHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p", + REG_ANACONF3, REG_ANACONF3_ENDRVHSR, 0), + + SND_SOC_DAPM_MIXER("HSL Mute", REG_MUTECONF, REG_MUTECONF_MUTHSL, + INVERT, NULL, 0), + SND_SOC_DAPM_MIXER("HSR Mute", REG_MUTECONF, REG_MUTECONF_MUTHSR, + INVERT, NULL, 0), + SND_SOC_DAPM_MIXER("HSL Enable", REG_ANACONF4, REG_ANACONF4_ENHSL, + NORMAL, NULL, 0), + SND_SOC_DAPM_MIXER("HSR Enable", REG_ANACONF4, REG_ANACONF4_ENHSR, + NORMAL, NULL, 0), + SND_SOC_DAPM_PGA("HSL Gain", SND_SOC_NOPM, 0, + 0, NULL, 0), + SND_SOC_DAPM_PGA("HSR Gain", SND_SOC_NOPM, 0, + 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HSL"), + SND_SOC_DAPM_OUTPUT("HSR"), + + /* LineOut path */ + + SND_SOC_DAPM_MUX("LineOut Source Playback Route", + SND_SOC_NOPM, 0, 0, dapm_lineout_source), + + SND_SOC_DAPM_MIXER("LOL Enable", REG_ANACONF5, + REG_ANACONF5_ENLOL, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOR Enable", REG_ANACONF5, + REG_ANACONF5_ENLOR, 0, NULL, 0), + + SND_SOC_DAPM_MUX("LineOut Left", + SND_SOC_NOPM, 0, 0, &dapm_lineout_left_mux), + + SND_SOC_DAPM_MUX("LineOut Right", + SND_SOC_NOPM, 0, 0, &dapm_lineout_right_mux), + + /* Earpiece path */ + + SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source", + SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source), + + SND_SOC_DAPM_MIXER("EAR DAC", REG_DAPATHCONF, + REG_DAPATHCONF_ENDACEAR, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Earpiece", SND_SOC_NOPM, 0, 0, dapm_ear_mute), + + SND_SOC_DAPM_MIXER("EAR Enable", REG_ANACONF4, + REG_ANACONF4_ENEAR, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("EAR"), + + /* Handsfree path */ + + SND_SOC_DAPM_MIXER("DA3 Channel Gain", REG_DAPATHENA, + REG_DAPATHENA_ENDA3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("DA4 Channel Gain", REG_DAPATHENA, + REG_DAPATHENA_ENDA4, 0, NULL, 0), + + SND_SOC_DAPM_MUX("IHF Left Source Playback Route", + SND_SOC_NOPM, 0, 0, dapm_ihfl_select), + SND_SOC_DAPM_MUX("IHF Right Source Playback Route", + SND_SOC_NOPM, 0, 0, dapm_ihfr_select), + + SND_SOC_DAPM_MUX("IHF Left", SND_SOC_NOPM, 0, 0, &dapm_ihf_left_mux), + SND_SOC_DAPM_MUX("IHF Right", SND_SOC_NOPM, 0, 0, &dapm_ihf_right_mux), + + SND_SOC_DAPM_MUX("IHF or LineOut Select", SND_SOC_NOPM, + 0, 0, &dapm_ihf_or_lineout_select_mux), + + SND_SOC_DAPM_MIXER("IHFL DAC", REG_DAPATHCONF, + REG_DAPATHCONF_ENDACHFL, 0, NULL, 0), + SND_SOC_DAPM_MIXER("IHFR DAC", REG_DAPATHCONF, + REG_DAPATHCONF_ENDACHFR, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR", REG_DIGMULTCONF2, + REG_DIGMULTCONF2_DATOHFREN, 0, NULL, 0), + SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL", REG_DIGMULTCONF2, + REG_DIGMULTCONF2_DATOHFLEN, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("IHFL Enable", REG_ANACONF4, + REG_ANACONF4_ENHFL, 0, NULL, 0), + SND_SOC_DAPM_MIXER("IHFR Enable", REG_ANACONF4, + REG_ANACONF4_ENHFR, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("IHFL"), + SND_SOC_DAPM_OUTPUT("IHFR"), + + /* Vibrator path */ + + SND_SOC_DAPM_MUX("Vibra 1", SND_SOC_NOPM, 0, 0, &dapm_vibra1_mux), + SND_SOC_DAPM_MUX("Vibra 2", SND_SOC_NOPM, 0, 0, &dapm_vibra2_mux), + SND_SOC_DAPM_MIXER("DA5 Channel Gain", REG_DAPATHENA, + REG_DAPATHENA_ENDA5, 0, NULL, 0), + SND_SOC_DAPM_MIXER("DA6 Channel Gain", REG_DAPATHENA, + REG_DAPATHENA_ENDA6, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("VIB1 DAC", REG_DAPATHCONF, + REG_DAPATHCONF_ENDACVIB1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 DAC", REG_DAPATHCONF, + REG_DAPATHCONF_ENDACVIB2, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("PWMGEN1"), + SND_SOC_DAPM_INPUT("PWMGEN2"), + + SND_SOC_DAPM_MUX("Vibra 1 Controller Playback Route", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib1), + SND_SOC_DAPM_MUX("Vibra 2 Controller Playback Route", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib2), + + SND_SOC_DAPM_MIXER("VIB1 Enable", REG_ANACONF4, + REG_ANACONF4_ENVIB1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 Enable", REG_ANACONF4, + REG_ANACONF4_ENVIB2, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("VIB1"), + SND_SOC_DAPM_OUTPUT("VIB2"), + + /* LineIn & Microphone 2 path */ + + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + SND_SOC_DAPM_INPUT("MIC2 Input"), + + SND_SOC_DAPM_SWITCH("LineIn Left", SND_SOC_NOPM, 0, 0, dapm_linl_mute), + SND_SOC_DAPM_SWITCH("LineIn Right", SND_SOC_NOPM, 0, 0, dapm_linr_mute), + SND_SOC_DAPM_SWITCH("Mic 2", SND_SOC_NOPM, 0, 0, dapm_mic2_mute), + + SND_SOC_DAPM_MIXER("LINL Enable", REG_ANACONF2, + REG_ANACONF2_ENLINL, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LINR Enable", REG_ANACONF2, + REG_ANACONF2_ENLINR, 0, NULL, 0), + SND_SOC_DAPM_MIXER("MIC2 Enable", REG_ANACONF2, + REG_ANACONF2_ENMIC2, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Mic 2 or LINR Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_mic2lr_select), + + SND_SOC_DAPM_MIXER("LINL ADC", REG_ANACONF3, + REG_ANACONF3_ENADCLINL, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LINR ADC", REG_ANACONF3, + REG_ANACONF3_ENADCLINR, 0, NULL, 0), + + SND_SOC_DAPM_MUX("AD 1 Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_ad1_select), + SND_SOC_DAPM_MUX("AD 2 Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_ad2_select), + + SND_SOC_DAPM_MIXER("AD1 Channel Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("AD2 Channel Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("AD12 Enable", REG_ADPATHENA, + REG_ADPATHENA_ENAD12, 0, NULL, 0), + + /* Microphone 1 path */ + + SND_SOC_DAPM_INPUT("MIC1 Input"), + + SND_SOC_DAPM_MUX("Mic 1A or 1B Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_mic1ab_select), + + SND_SOC_DAPM_SWITCH("Mic 1", SND_SOC_NOPM, 0, 0, dapm_mic1_mute), + + SND_SOC_DAPM_MIXER("MIC1 Enable", REG_ANACONF2, + REG_ANACONF2_ENMIC1, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("MIC1 ADC", REG_ANACONF3, + REG_ANACONF3_ENADCMIC, 0, NULL, 0), + + SND_SOC_DAPM_MUX("AD 3 Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_ad3_select), + + SND_SOC_DAPM_MIXER("AD3 Channel Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("AD3 Enable", REG_ADPATHENA, + REG_ADPATHENA_ENAD34, 0, NULL, 0), + + /* HD Capture path */ + + SND_SOC_DAPM_MUX("AD 5 Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_ad5_select), + SND_SOC_DAPM_MUX("AD 6 Select Capture Route", + SND_SOC_NOPM, 0, 0, dapm_ad6_select), + + SND_SOC_DAPM_MIXER("AD5 Channel Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("AD6 Channel Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("AD57 Enable", REG_ADPATHENA, + REG_ADPATHENA_ENAD5768, 0, NULL, 0), + SND_SOC_DAPM_MIXER("AD68 Enable", REG_ADPATHENA, + REG_ADPATHENA_ENAD5768, 0, NULL, 0), + + /* Digital Microphone path */ + + SND_SOC_DAPM_INPUT("DMIC Input"), + + SND_SOC_DAPM_SWITCH("DMic 1", SND_SOC_NOPM, 0, 0, dapm_dmic1_mute), + SND_SOC_DAPM_SWITCH("DMic 2", SND_SOC_NOPM, 0, 0, dapm_dmic2_mute), + SND_SOC_DAPM_SWITCH("DMic 3", SND_SOC_NOPM, 0, 0, dapm_dmic3_mute), + SND_SOC_DAPM_SWITCH("DMic 4", SND_SOC_NOPM, 0, 0, dapm_dmic4_mute), + SND_SOC_DAPM_SWITCH("DMic 5", SND_SOC_NOPM, 0, 0, dapm_dmic5_mute), + SND_SOC_DAPM_SWITCH("DMic 6", SND_SOC_NOPM, 0, 0, dapm_dmic6_mute), + + SND_SOC_DAPM_MIXER("AD4 Channel Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("AD4 Enable", REG_ADPATHENA, + REG_ADPATHENA_ENAD34, 0, NULL, 0), + + /* LineIn Bypass path */ + + SND_SOC_DAPM_MIXER("LINL to HSL Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LINR to HSR Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Acoustical Noise Cancellation path */ + + SND_SOC_DAPM_MUX("ANC Source Playback Route", + SND_SOC_NOPM, 0, 0, dapm_anc_in_select), + + SND_SOC_DAPM_MUX("ANC Playback Switch", + SND_SOC_NOPM, 0, 0, dapm_anc_enable), + + SND_SOC_DAPM_SWITCH("ANC to Earpiece", + SND_SOC_NOPM, 0, 0, dapm_anc_ear_mute), + + /* Sidetone Filter path */ + + SND_SOC_DAPM_MUX("Sidetone Left Source Playback Route", + SND_SOC_NOPM, 0, 0, dapm_stfir1_in_select), + SND_SOC_DAPM_MUX("Sidetone Right Source Playback Route", + SND_SOC_NOPM, 0, 0, dapm_stfir2_in_select), + + SND_SOC_DAPM_MIXER("STFIR1 Control", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Control", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("STFIR1 Gain", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Gain", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + +/* DAPM-routes */ + +static const struct snd_soc_dapm_route dapm_routes[] = { + /* AD/DA */ + {"ADC", NULL, "ADC Input"}, + {"DAC Output", NULL, "DAC"}, + + /* Powerup charge pump if DA1/2 is in use */ + {"DA_IN1", NULL, "Charge Pump"}, + {"DA_IN2", NULL, "Charge Pump"}, + + /* Headset path */ + + {"DA1 Enable", NULL, "DA_IN1"}, + {"DA2 Enable", NULL, "DA_IN2"}, + + {"HSL Digital Gain", NULL, "DA1 Enable"}, + {"HSR Digital Gain", NULL, "DA2 Enable"}, + + {"HSL DAC", NULL, "HSL Digital Gain"}, + {"HSR DAC", NULL, "HSR Digital Gain"}, + + {"HSL DAC Mute", NULL, "HSL DAC"}, + {"HSR DAC Mute", NULL, "HSR DAC"}, + + {"HSL DAC Driver", NULL, "HSL DAC Mute"}, + {"HSR DAC Driver", NULL, "HSR DAC Mute"}, + + {"HSL Mute", NULL, "HSL DAC Driver"}, + {"HSR Mute", NULL, "HSR DAC Driver"}, + + {"Headset Left", "Playback Switch", "HSL Mute"}, + {"Headset Right", "Playback Switch", "HSR Mute"}, + + {"HSL Enable", NULL, "Headset Left"}, + {"HSR Enable", NULL, "Headset Right"}, + + {"HSL Gain", NULL, "HSL Enable"}, + {"HSR Gain", NULL, "HSR Enable"}, + + {"HSL", NULL, "HSL Gain"}, + {"HSR", NULL, "HSR Gain"}, + + /* IHF or LineOut path */ + + {"DA3 Channel Gain", NULL, "DA_IN3"}, + {"DA4 Channel Gain", NULL, "DA_IN4"}, + + {"IHF Left Source Playback Route", "Audio Path", "DA3 Channel Gain"}, + {"IHF Right Source Playback Route", "Audio Path", "DA4 Channel Gain"}, + + {"DA3 or ANC path to HfL", NULL, "IHF Left Source Playback Route"}, + {"DA4 or ANC path to HfR", NULL, "IHF Right Source Playback Route"}, + + /* IHF path */ + + {"IHF Left", "Enabled", "DA3 or ANC path to HfL"}, + {"IHF Right", "Enabled", "DA4 or ANC path to HfR"}, + + {"IHFL DAC", NULL, "IHF Left"}, + {"IHFR DAC", NULL, "IHF Right"}, + + {"IHFL Enable", NULL, "IHFL DAC"}, + {"IHFR Enable", NULL, "IHFR DAC"}, + + {"IHF or LineOut Select", "IHF", "IHFL Enable"}, + {"IHF or LineOut Select", "IHF", "IHFR Enable"}, + + /* Earpiece path */ + + {"Earpiece or LineOut Mono Source", "Headset Left", "HSL Digital Gain"}, + {"Earpiece or LineOut Mono Source", "IHF Left", "DA3 or ANC path to HfL"}, + + {"EAR DAC", NULL, "Earpiece or LineOut Mono Source"}, + + {"Earpiece", "Playback Switch", "EAR DAC"}, + + {"EAR Enable", NULL, "Earpiece"}, + + {"EAR", NULL, "EAR Enable"}, + + /* LineOut path stereo */ + + {"LineOut Source Playback Route", "Stereo Path", "HSL DAC Driver"}, + {"LineOut Source Playback Route", "Stereo Path", "HSR DAC Driver"}, + + /* LineOut path mono */ + + {"LineOut Source Playback Route", "Mono Path", "EAR DAC"}, + + /* LineOut path */ + + {"LineOut Left", "Enabled", "LineOut Source Playback Route"}, + {"LineOut Right", "Enabled", "LineOut Source Playback Route"}, + + {"LOL Enable", NULL, "LineOut Left"}, + {"LOR Enable", NULL, "LineOut Right"}, + + {"IHF or LineOut Select", "LineOut", "LOL Enable"}, + {"IHF or LineOut Select", "LineOut", "LOR Enable"}, + + /* IHF path */ + + {"IHFL", NULL, "IHF or LineOut Select"}, + {"IHFR", NULL, "IHF or LineOut Select"}, + + /* Vibrator path */ + + {"DA5 Channel Gain", NULL, "DA_IN5"}, + {"DA6 Channel Gain", NULL, "DA_IN6"}, + + {"VIB1 DAC", NULL, "DA5 Channel Gain"}, + {"VIB2 DAC", NULL, "DA6 Channel Gain"}, + + {"Vibra 1 Controller Playback Route", "Audio Path", "VIB1 DAC"}, + {"Vibra 2 Controller Playback Route", "Audio Path", "VIB2 DAC"}, + {"Vibra 1 Controller Playback Route", "PWM Generator", "PWMGEN1"}, + {"Vibra 2 Controller Playback Route", "PWM Generator", "PWMGEN2"}, + + {"Vibra 1", "Enabled", "Vibra 1 Controller Playback Route"}, + {"Vibra 2", "Enabled", "Vibra 2 Controller Playback Route"}, + + {"VIB1 Enable", NULL, "Vibra 1"}, + {"VIB2 Enable", NULL, "Vibra 2"}, + + {"VIB1", NULL, "VIB1 Enable"}, + {"VIB2", NULL, "VIB2 Enable"}, + + /* LineIn & Microphone 2 path */ + + {"LineIn Left", "Capture Switch", "LINL"}, + {"LineIn Right", "Capture Switch", "LINR"}, + {"Mic 2", "Capture Switch", "MIC2 Input"}, + + {"LINL Enable", NULL, "LineIn Left"}, + {"LINR Enable", NULL, "LineIn Right"}, + {"MIC2 Enable", NULL, "Mic 2"}, + + {"Mic 2 or LINR Select Capture Route", "LineIn Right", "LINR Enable"}, + {"Mic 2 or LINR Select Capture Route", "Mic 2", "MIC2 Enable"}, + + {"LINL ADC", NULL, "LINL Enable"}, + {"LINR ADC", NULL, "Mic 2 or LINR Select Capture Route"}, + + {"AD 1 Select Capture Route", "LineIn Left", "LINL ADC"}, + {"AD 2 Select Capture Route", "LineIn Right", "LINR ADC"}, + + {"AD1 Channel Gain", NULL, "AD 1 Select Capture Route"}, + {"AD2 Channel Gain", NULL, "AD 2 Select Capture Route"}, + + {"AD12 Enable", NULL, "AD1 Channel Gain"}, + {"AD12 Enable", NULL, "AD2 Channel Gain"}, + + {"AD_OUT1", NULL, "AD12 Enable"}, + {"AD_OUT2", NULL, "AD12 Enable"}, + + /* Microphone 1 path */ + + {"Mic 1A or 1B Select Capture Route", "Mic 1A", "MIC1 Input"}, + {"Mic 1A or 1B Select Capture Route", "Mic 1B", "MIC1 Input"}, + + {"Mic 1", "Capture Switch", "Mic 1A or 1B Select Capture Route"}, + + {"MIC1 Enable", NULL, "Mic 1"}, + + {"MIC1 ADC", NULL, "MIC1 Enable"}, + + {"AD 3 Select Capture Route", "Mic 1", "MIC1 ADC"}, + + {"AD3 Channel Gain", NULL, "AD 3 Select Capture Route"}, + + {"AD3 Enable", NULL, "AD3 Channel Gain"}, + + {"AD_OUT3", NULL, "AD3 Enable"}, + + /* HD Capture path */ + + {"AD 5 Select Capture Route", "Mic 2", "LINR ADC"}, + {"AD 6 Select Capture Route", "Mic 1", "MIC1 ADC"}, + + {"AD5 Channel Gain", NULL, "AD 5 Select Capture Route"}, + {"AD6 Channel Gain", NULL, "AD 6 Select Capture Route"}, + + {"AD57 Enable", NULL, "AD5 Channel Gain"}, + {"AD68 Enable", NULL, "AD6 Channel Gain"}, + + {"AD_OUT57", NULL, "AD57 Enable"}, + {"AD_OUT68", NULL, "AD68 Enable"}, + + /* Digital Microphone path */ + + {"DMic 1", "Capture Switch", "DMIC Input"}, + {"DMic 2", "Capture Switch", "DMIC Input"}, + {"DMic 3", "Capture Switch", "DMIC Input"}, + {"DMic 4", "Capture Switch", "DMIC Input"}, + {"DMic 5", "Capture Switch", "DMIC Input"}, + {"DMic 6", "Capture Switch", "DMIC Input"}, + + {"AD 1 Select Capture Route", "DMic 1", "DMic 1"}, + {"AD 2 Select Capture Route", "DMic 2", "DMic 2"}, + {"AD 3 Select Capture Route", "DMic 3", "DMic 3"}, + {"AD 5 Select Capture Route", "DMic 5", "DMic 5"}, + {"AD 6 Select Capture Route", "DMic 6", "DMic 6"}, + + {"AD4 Channel Gain", NULL, "DMic 4"}, + + {"AD4 Enable", NULL, "AD4 Channel Gain"}, + + {"AD_OUT4", NULL, "AD4 Enable"}, + + /* LineIn Bypass path */ + + {"LINL to HSL Gain", NULL, "LINL Enable"}, + {"LINR to HSR Gain", NULL, "LINR Enable"}, + + {"HSL DAC Driver", NULL, "LINL to HSL Gain"}, + {"HSR DAC Driver", NULL, "LINR to HSR Gain"}, + + /* Acoustical Noise Cancellation path */ + + {"ANC Source Playback Route", "Mic 2 / DMic 5", "AD5 Channel Gain"}, + {"ANC Source Playback Route", "Mic 1 / DMic 6", "AD6 Channel Gain"}, + + {"ANC Playback Switch", "Enabled", "ANC Source Playback Route"}, + + {"IHF Left Source Playback Route", "ANC", "ANC Playback Switch"}, + {"IHF Right Source Playback Route", "ANC", "ANC Playback Switch"}, + {"ANC to Earpiece", "Playback Switch", "ANC Playback Switch"}, + + {"HSL Digital Gain", NULL, "ANC to Earpiece"}, + + /* Sidetone Filter path */ + + {"Sidetone Left Source Playback Route", "LineIn Left", "AD12 Enable"}, + {"Sidetone Left Source Playback Route", "LineIn Right", "AD12 Enable"}, + {"Sidetone Left Source Playback Route", "Mic 1", "AD3 Enable"}, + {"Sidetone Left Source Playback Route", "Headset Left", "DA_IN1"}, + {"Sidetone Right Source Playback Route", "LineIn Right", "AD12 Enable"}, + {"Sidetone Right Source Playback Route", "Mic 1", "AD3 Enable"}, + {"Sidetone Right Source Playback Route", "DMic 4", "AD4 Enable"}, + {"Sidetone Right Source Playback Route", "Headset Right", "DA_IN2"}, + + {"STFIR1 Control", NULL, "Sidetone Left Source Playback Route"}, + {"STFIR2 Control", NULL, "Sidetone Right Source Playback Route"}, + + {"STFIR1 Gain", NULL, "STFIR1 Control"}, + {"STFIR2 Gain", NULL, "STFIR2 Control"}, + + {"DA1 Enable", NULL, "STFIR1 Gain"}, + {"DA2 Enable", NULL, "STFIR2 Gain"}, +}; + +/* Controls - Non-DAPM ASoC */ + +/* from -31 to 31 dB in 1 dB steps (mute instead of -32 dB) */ +static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1); + +/* from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) */ +static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1); + +/* from 0 to 8 dB in 1 dB steps (mute instead of -1 dB) */ +static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1); + +/* from -30 to 0 dB in 1 dB steps (mute instead of -31 dB) */ +static DECLARE_TLV_DB_SCALE(stfir_dig_gain_tlv, -3100, 100, 1); + +/* from -32 to -20 dB in 4 dB steps / from -18 to 2 dB in 2 dB steps */ +static const unsigned int hs_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0), + 4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0), +}; + +/* from 0 to 31 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0); + +/* from -10 to 20 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0); + +/* from -36 to 0 dB in 2 dB steps (mute instead of -38 dB) */ +static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1); + +static SOC_ENUM_SINGLE_DECL(soc_enum_hshpen, + REG_ANACONF1, REG_ANACONF1_HSHPEN, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_hslowpow, + REG_ANACONF1, REG_ANACONF1_HSLOWPOW, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_daclowpow1, + REG_ANACONF1, REG_ANACONF1_DACLOWPOW1, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_daclowpow0, + REG_ANACONF1, REG_ANACONF1_DACLOWPOW0, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, + REG_ANACONF1, REG_ANACONF1_EARDACLOWPOW, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, + REG_ANACONF1, REG_ANACONF1_EARDRVLOWPOW, enum_dis_ena); + +static const char * const enum_earselcm[] = {"0.95V", "1.10V", "1.27V", "1.58V"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_earselcm, + REG_ANACONF1, REG_ANACONF1_EARSELCM, enum_earselcm); + +static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms", "5ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed, + REG_DIGMICCONF, REG_DIGMICCONF_HSFADSPEED, enum_hsfadspeed); + +static const char * const enum_envdetthre[] = { + "250mV", "300mV", "350mV", "400mV", + "450mV", "500mV", "550mV", "600mV", + "650mV", "700mV", "750mV", "800mV", + "850mV", "900mV", "950mV", "1.00V" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdetcpen, + REG_SIGENVCONF, REG_SIGENVCONF_ENVDETCPEN, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre, + REG_ENVCPCONF, REG_ENVCPCONF_ENVDETHTHRE, enum_envdetthre); +static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre, + REG_ENVCPCONF, REG_ENVCPCONF_ENVDETLTHRE, enum_envdetthre); + +static const char * const enum_envdettime[] = { + "26.6us", "53.2us", "106us", "213us", + "426us", "851us", "1.70ms", "3.40ms", + "6.81ms", "13.6ms", "27.2ms", "54.5ms", + "109ms", "218ms", "436ms", "872ms" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime, + REG_SIGENVCONF, REG_SIGENVCONF_ENVDETTIME, enum_envdettime); + +static const char * const enum_ensemicx[] = {"Differential", "Single Ended"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_ensemic1, + REG_ANAGAIN1, REG_ANAGAINX_ENSEMICX, enum_ensemicx); +static SOC_ENUM_SINGLE_DECL(soc_enum_ensemic2, + REG_ANAGAIN2, REG_ANAGAINX_ENSEMICX, enum_ensemicx); +static SOC_ENUM_SINGLE_DECL(soc_enum_lowpowmic1, + REG_ANAGAIN1, REG_ANAGAINX_LOWPOWMICX, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_lowpowmic2, + REG_ANAGAIN2, REG_ANAGAINX_LOWPOWMICX, enum_dis_ena); + +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12nh, REG_ADFILTCONF, + REG_ADFILTCONF_AD1NH, REG_ADFILTCONF_AD2NH, enum_ena_dis); +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34nh, REG_ADFILTCONF, + REG_ADFILTCONF_AD3NH, REG_ADFILTCONF_AD4NH, enum_ena_dis); + +static const char * const enum_av_mode[] = {"Audio", "Voice"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, REG_ADFILTCONF, + REG_ADFILTCONF_AD1VOICE, REG_ADFILTCONF_AD2VOICE, enum_av_mode); +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, REG_ADFILTCONF, + REG_ADFILTCONF_AD3VOICE, REG_ADFILTCONF_AD4VOICE, enum_av_mode); + +static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice, + REG_DASLOTCONF1, REG_DASLOTCONF1_DA12VOICE, enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice, + REG_DASLOTCONF3, REG_DASLOTCONF3_DA34VOICE, enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice, + REG_DASLOTCONF5, REG_DASLOTCONF5_DA56VOICE, enum_av_mode); + +static SOC_ENUM_SINGLE_DECL(soc_enum_swapda12_34, + REG_DASLOTCONF1, REG_DASLOTCONF1_SWAPDA12_34, enum_dis_ena); + +static SOC_ENUM_DOUBLE_DECL(soc_enum_vib12swap, REG_CLASSDCONF1, + REG_CLASSDCONF1_VIB1SWAPEN, REG_CLASSDCONF1_VIB2SWAPEN, enum_dis_ena); +static SOC_ENUM_DOUBLE_DECL(soc_enum_hflrswap, REG_CLASSDCONF1, + REG_CLASSDCONF1_HFLSWAPEN, REG_CLASSDCONF1_HFRSWAPEN, enum_dis_ena); + +static SOC_ENUM_DOUBLE_DECL(soc_enum_fir01byp, REG_CLASSDCONF2, + REG_CLASSDCONF2_FIRBYP0, REG_CLASSDCONF2_FIRBYP1, enum_dis_ena); +static SOC_ENUM_DOUBLE_DECL(soc_enum_fir23byp, REG_CLASSDCONF2, + REG_CLASSDCONF2_FIRBYP2, REG_CLASSDCONF2_FIRBYP3, enum_dis_ena); +static SOC_ENUM_DOUBLE_DECL(soc_enum_highvol01, REG_CLASSDCONF2, + REG_CLASSDCONF2_HIGHVOLEN0, REG_CLASSDCONF2_HIGHVOLEN1, enum_dis_ena); +static SOC_ENUM_DOUBLE_DECL(soc_enum_highvol23, REG_CLASSDCONF2, + REG_CLASSDCONF2_HIGHVOLEN2, REG_CLASSDCONF2_HIGHVOLEN3, enum_dis_ena); + +static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, REG_DMICFILTCONF, + REG_DMICFILTCONF_DMIC1SINC3, REG_DMICFILTCONF_DMIC2SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, REG_DMICFILTCONF, + REG_DMICFILTCONF_DMIC3SINC3, REG_DMICFILTCONF_DMIC4SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, REG_DMICFILTCONF, + REG_DMICFILTCONF_DMIC5SINC3, REG_DMICFILTCONF_DMIC6SINC3, enum_sinc53); + +static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, REG_DIGMULTCONF1, + REG_DIGMULTCONF1_DATOHSLEN, REG_DIGMULTCONF1_DATOHSREN, enum_da2hslr); + +static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, + REG_HSLEARDIGGAIN, REG_HSLEARDIGGAIN_HSSINC1, enum_sinc31); + +static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, + REG_HSRDIGGAIN, REG_HSRDIGGAIN_FADESPEED, enum_fadespeed); + +/* Digital interface - Clocks */ +static SOC_ENUM_SINGLE_DECL(soc_enum_mastgen, + REG_DIGIFCONF1, REG_DIGIFCONF1_ENMASTGEN, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_fsbitclk0, + REG_DIGIFCONF1, REG_DIGIFCONF1_ENFSBITCLK0, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_fsbitclk1, + REG_DIGIFCONF1, REG_DIGIFCONF1_ENFSBITCLK1, enum_dis_ena); + +/* Digital interface - DA from slot mapping */ +static const char * const enum_da_from_slot_map[] = {"SLOT0", + "SLOT1", + "SLOT2", + "SLOT3", + "SLOT4", + "SLOT5", + "SLOT6", + "SLOT7", + "SLOT8", + "SLOT9", + "SLOT10", + "SLOT11", + "SLOT12", + "SLOT13", + "SLOT14", + "SLOT15", + "SLOT16", + "SLOT17", + "SLOT18", + "SLOT19", + "SLOT20", + "SLOT21", + "SLOT22", + "SLOT23", + "SLOT24", + "SLOT25", + "SLOT26", + "SLOT27", + "SLOT28", + "SLOT29", + "SLOT30", + "SLOT31"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap, + REG_DASLOTCONF1, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap, + REG_DASLOTCONF2, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap, + REG_DASLOTCONF3, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap, + REG_DASLOTCONF4, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap, + REG_DASLOTCONF5, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap, + REG_DASLOTCONF6, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap, + REG_DASLOTCONF7, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap, + REG_DASLOTCONF8, REG_DASLOTCONFX_SLTODAX_SHIFT, enum_da_from_slot_map); + +/* Digital interface - AD to slot mapping */ +static const char * const enum_ad_to_slot_map[] = {"AD_OUT1", + "AD_OUT2", + "AD_OUT3", + "AD_OUT4", + "AD_OUT5", + "AD_OUT6", + "AD_OUT7", + "AD_OUT8", + "zeroes", + "tristate"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map, + REG_ADSLOTSEL1, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map, + REG_ADSLOTSEL1, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map, + REG_ADSLOTSEL2, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map, + REG_ADSLOTSEL2, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map, + REG_ADSLOTSEL3, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map, + REG_ADSLOTSEL3, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map, + REG_ADSLOTSEL4, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map, + REG_ADSLOTSEL4, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map, + REG_ADSLOTSEL5, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map, + REG_ADSLOTSEL5, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map, + REG_ADSLOTSEL6, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map, + REG_ADSLOTSEL6, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map, + REG_ADSLOTSEL7, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map, + REG_ADSLOTSEL7, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map, + REG_ADSLOTSEL8, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map, + REG_ADSLOTSEL8, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map, + REG_ADSLOTSEL9, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map, + REG_ADSLOTSEL9, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map, + REG_ADSLOTSEL10, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map, + REG_ADSLOTSEL10, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map, + REG_ADSLOTSEL11, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map, + REG_ADSLOTSEL11, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map, + REG_ADSLOTSEL12, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map, + REG_ADSLOTSEL12, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map, + REG_ADSLOTSEL13, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map, + REG_ADSLOTSEL13, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map, + REG_ADSLOTSEL14, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map, + REG_ADSLOTSEL14, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map, + REG_ADSLOTSEL15, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map, + REG_ADSLOTSEL15, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map, + REG_ADSLOTSEL16, REG_ADSLOTSELX_EVEN_SHIFT, enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map, + REG_ADSLOTSEL16, REG_ADSLOTSELX_ODD_SHIFT, enum_ad_to_slot_map); + +/* Digital interface - Digital loopback */ +static SOC_ENUM_SINGLE_DECL(soc_enum_ad1loop, + REG_DASLOTCONF1, REG_DASLOTCONF1_DAI7TOADO1, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad2loop, + REG_DASLOTCONF2, REG_DASLOTCONF2_DAI8TOADO2, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad3loop, + REG_DASLOTCONF3, REG_DASLOTCONF3_DAI7TOADO3, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad4loop, + REG_DASLOTCONF4, REG_DASLOTCONF4_DAI8TOADO4, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad5loop, + REG_DASLOTCONF5, REG_DASLOTCONF5_DAI7TOADO5, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad6loop, + REG_DASLOTCONF6, REG_DASLOTCONF6_DAI8TOADO6, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad7loop, + REG_DASLOTCONF7, REG_DASLOTCONF7_DAI8TOADO7, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_ad8loop, + REG_DASLOTCONF8, REG_DASLOTCONF8_DAI7TOADO8, enum_dis_ena); + +/* Digital interface - Burst mode */ +static SOC_ENUM_SINGLE_DECL(soc_enum_if0fifoen, + REG_DIGIFCONF3, REG_DIGIFCONF3_IF0BFIFOEN, enum_dis_ena); +static const char * const enum_mask[] = {"Unmasked", "Masked"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask, + REG_FIFOCONF1, REG_FIFOCONF1_BFIFOMASK, enum_mask); +static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2, + REG_FIFOCONF1, REG_FIFOCONF1_BFIFO19M2, enum_bitclk0); +static const char * const enum_slavemaster[] = {"Slave", "Master"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast, + REG_FIFOCONF3, REG_FIFOCONF3_BFIFOMAST_SHIFT, enum_slavemaster); +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifoint, + REG_FIFOCONF3, REG_FIFOCONF3_BFIFORUN_SHIFT, enum_dis_ena); + +/* TODO: move to DAPM */ +static SOC_ENUM_SINGLE_DECL(soc_enum_enfirsids, + REG_SIDFIRCONF, REG_SIDFIRCONF_ENFIRSIDS, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_parlhf, + REG_CLASSDCONF1, REG_CLASSDCONF1_PARLHF, enum_dis_ena); +static SOC_ENUM_SINGLE_DECL(soc_enum_parlvib, + REG_CLASSDCONF1, REG_CLASSDCONF1_PARLVIB, enum_dis_ena); + +static struct snd_kcontrol_new ab8500_snd_controls[] = { + SOC_ENUM("Headset High Pass Playback Switch", soc_enum_hshpen), + SOC_ENUM("Headset Low Power Playback Switch", soc_enum_hslowpow), + SOC_ENUM("Headset DAC Low Power Playback Switch", soc_enum_daclowpow1), + SOC_ENUM("Headset DAC Drv Low Power Playback Switch", + soc_enum_daclowpow0), + SOC_ENUM("Earpiece DAC Low Power Playback Switch", + soc_enum_eardaclowpow), + SOC_ENUM("Earpiece DAC Drv Low Power Playback Switch", + soc_enum_eardrvlowpow), + SOC_ENUM("Earpiece Common Mode Playback Switch", soc_enum_earselcm), + + SOC_ENUM("Headset Fade Speed Playback Switch", soc_enum_hsfadspeed), + + SOC_ENUM("Charge Pump High Threshold For Low Voltage", + soc_enum_envdeththre), + SOC_ENUM("Charge Pump Low Threshold For Low Voltage", + soc_enum_envdetlthre), + SOC_ENUM("Charge Pump Envelope Detection", soc_enum_envdetcpen), + SOC_ENUM("Charge Pump Envelope Detection Decay Time", + soc_enum_envdettime), + + SOC_ENUM("Mic 1 Type Capture Switch", soc_enum_ensemic1), + SOC_ENUM("Mic 2 Type Capture Switch", soc_enum_ensemic2), + SOC_ENUM("Mic 1 Low Power Capture Switch", soc_enum_lowpowmic1), + SOC_ENUM("Mic 2 Low Power Capture Switch", soc_enum_lowpowmic2), + + SOC_ENUM("LineIn High Pass Capture Switch", soc_enum_ad12nh), + SOC_ENUM("Mic High Pass Capture Switch", soc_enum_ad34nh), + SOC_ENUM("LineIn Mode Capture Switch", soc_enum_ad12voice), + SOC_ENUM("Mic Mode Capture Switch", soc_enum_ad34voice), + + SOC_ENUM("Headset Mode Playback Switch", soc_enum_da12voice), + SOC_ENUM("IHF Mode Playback Switch", soc_enum_da34voice), + SOC_ENUM("Vibra Mode Playback Switch", soc_enum_da56voice), + + SOC_ENUM("IHF and Headset Swap Playback Switch", soc_enum_swapda12_34), + + SOC_ENUM("IHF Low EMI Mode Playback Switch", soc_enum_hflrswap), + SOC_ENUM("Vibra Low EMI Mode Playback Switch", soc_enum_vib12swap), + + SOC_ENUM("IHF FIR Bypass Playback Switch", soc_enum_fir01byp), + SOC_ENUM("Vibra FIR Bypass Playback Switch", soc_enum_fir23byp), + + /* TODO: Cannot be changed on the fly with digital channel enabled. */ + SOC_ENUM("IHF High Volume Playback Switch", soc_enum_highvol01), + SOC_ENUM("Vibra High Volume Playback Switch", soc_enum_highvol23), + + SOC_SINGLE("ClassD High Pass Gain Playback Volume", + REG_CLASSDCONF3, REG_CLASSDCONF3_DITHHPGAIN, + REG_CLASSDCONF3_DITHHPGAIN_MAX, NORMAL), + SOC_SINGLE("ClassD White Gain Playback Volume", + REG_CLASSDCONF3, REG_CLASSDCONF3_DITHWGAIN, + REG_CLASSDCONF3_DITHWGAIN_MAX, NORMAL), + + SOC_ENUM("LineIn Filter Capture Switch", soc_enum_dmic12sinc), + SOC_ENUM("Mic Filter Capture Switch", soc_enum_dmic34sinc), + SOC_ENUM("HD Mic Filter Capture Switch", soc_enum_dmic56sinc), + + SOC_ENUM("Headset Source Playback Route", soc_enum_da2hslr), + + /* TODO: Cannot be changed on the fly with digital channel enabled. */ + SOC_ENUM("Headset Filter Playback Switch", soc_enum_hsesinc), + + SOC_ENUM("Digital Gain Fade Speed Switch", soc_enum_fadespeed), + + SOC_DOUBLE_R("Vibra PWM Duty Cycle N Playback Volume", + REG_PWMGENCONF3, REG_PWMGENCONF5, + REG_PWMGENCONFX_PWMVIBXDUTCYC, + REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX, NORMAL), + SOC_DOUBLE_R("Vibra PWM Duty Cycle P Playback Volume", + REG_PWMGENCONF2, REG_PWMGENCONF4, + REG_PWMGENCONFX_PWMVIBXDUTCYC, + REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX, NORMAL), + + /* TODO: move to DAPM */ + SOC_ENUM("Sidetone Playback Switch", soc_enum_enfirsids), + SOC_ENUM("IHF L and R Bridge Playback Route", soc_enum_parlhf), + SOC_ENUM("Vibra 1 and 2 Bridge Playback Route", soc_enum_parlvib), + + /* Digital gains for AD side */ + + SOC_DOUBLE_R_TLV("LineIn Master Gain Capture Volume", + REG_ADDIGGAIN1, REG_ADDIGGAIN2, + 0, REG_ADDIGGAINX_ADXGAIN_MAX, INVERT, adx_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Mic Master Gain Capture Volume", + REG_ADDIGGAIN3, REG_ADDIGGAIN4, + 0, REG_ADDIGGAINX_ADXGAIN_MAX, INVERT, adx_dig_gain_tlv), + SOC_DOUBLE_R_TLV("HD Mic Master Gain Capture Volume", + REG_ADDIGGAIN5, REG_ADDIGGAIN6, + 0, REG_ADDIGGAINX_ADXGAIN_MAX, INVERT, adx_dig_gain_tlv), + + /* Digital gains for DA side */ + + SOC_DOUBLE_R_TLV("Headset Master Gain Playback Volume", + REG_DADIGGAIN1, REG_DADIGGAIN2, + 0, REG_DADIGGAINX_DAXGAIN_MAX, INVERT, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("IHF Master Gain Playback Volume", + REG_DADIGGAIN3, REG_DADIGGAIN4, + 0, REG_DADIGGAINX_DAXGAIN_MAX, INVERT, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Vibra Master Gain Playback Volume", + REG_DADIGGAIN5, REG_DADIGGAIN6, + 0, REG_DADIGGAINX_DAXGAIN_MAX, INVERT, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Analog Loopback Gain Playback Volume", + REG_ADDIGLOOPGAIN1, REG_ADDIGLOOPGAIN2, + 0, REG_ADDIGLOOPGAINX_ADXLBGAIN_MAX, INVERT, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Headset Digital Gain Playback Volume", + REG_HSLEARDIGGAIN, REG_HSRDIGGAIN, + 0, REG_HSLEARDIGGAIN_HSLDGAIN_MAX, INVERT, hs_ear_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Sidetone Digital Gain Playback Volume", + REG_SIDFIRGAIN1, REG_SIDFIRGAIN2, + 0, REG_SIDFIRGAINX_FIRSIDXGAIN_MAX, INVERT, stfir_dig_gain_tlv), + + /* Analog gains */ + + SOC_DOUBLE_TLV("Headset Gain Playback Volume", + REG_ANAGAIN3, + REG_ANAGAIN3_HSLGAIN, REG_ANAGAIN3_HSRGAIN, + REG_ANAGAIN3_HSXGAIN_MAX, INVERT, hs_gain_tlv), + SOC_SINGLE_TLV("Mic 1 Capture Volume", + REG_ANAGAIN1, + REG_ANAGAINX_MICXGAIN, + REG_ANAGAINX_MICXGAIN_MAX, NORMAL, mic_gain_tlv), + SOC_SINGLE_TLV("Mic 2 Capture Volume", + REG_ANAGAIN2, + REG_ANAGAINX_MICXGAIN, + REG_ANAGAINX_MICXGAIN_MAX, NORMAL, mic_gain_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + REG_ANAGAIN4, + REG_ANAGAIN4_LINLGAIN, REG_ANAGAIN4_LINRGAIN, + REG_ANAGAIN4_LINXGAIN_MAX, NORMAL, lin_gain_tlv), + SOC_DOUBLE_R_TLV("LineIn to Headset Bypass Playback Volume", + REG_DIGLINHSLGAIN, REG_DIGLINHSRGAIN, + REG_DIGLINHSXGAIN_LINTOHSXGAIN, + REG_DIGLINHSXGAIN_LINTOHSXGAIN_MAX, INVERT, lin2hs_gain_tlv), + + /* Digital interface - Clocks */ + SOC_ENUM("Digital Interface Master Generator Switch", soc_enum_mastgen), + SOC_ENUM("Digital Interface 0 Bit-clock Switch", soc_enum_fsbitclk0), + SOC_ENUM("Digital Interface 1 Bit-clock Switch", soc_enum_fsbitclk1), + + /* Digital interface - DA from slot mapping */ + SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap), + SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap), + SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap), + SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap), + SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap), + SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap), + SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap), + SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap), + + /* Digital interface - AD to slot mapping */ + SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map), + SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map), + SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map), + SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map), + SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map), + SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map), + SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map), + SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map), + SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map), + SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map), + SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map), + SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map), + SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map), + SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map), + SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map), + SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map), + SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map), + SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map), + SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map), + SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map), + SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map), + SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map), + SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map), + SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map), + SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map), + SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map), + SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map), + SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map), + SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map), + SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map), + SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map), + SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map), + + /* Digital interface - Loopback */ + SOC_ENUM("Digital Interface AD 1 Loopback Switch", soc_enum_ad1loop), + SOC_ENUM("Digital Interface AD 2 Loopback Switch", soc_enum_ad2loop), + SOC_ENUM("Digital Interface AD 3 Loopback Switch", soc_enum_ad3loop), + SOC_ENUM("Digital Interface AD 4 Loopback Switch", soc_enum_ad4loop), + SOC_ENUM("Digital Interface AD 5 Loopback Switch", soc_enum_ad5loop), + SOC_ENUM("Digital Interface AD 6 Loopback Switch", soc_enum_ad6loop), + SOC_ENUM("Digital Interface AD 7 Loopback Switch", soc_enum_ad7loop), + SOC_ENUM("Digital Interface AD 8 Loopback Switch", soc_enum_ad8loop), + + /* Digital interface - Burst FIFO */ + SOC_ENUM("Digital Interface 0 FIFO Enable Switch", soc_enum_if0fifoen), + SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask), + SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2), + SOC_SINGLE("Burst FIFO Threshold", + REG_FIFOCONF1, + REG_FIFOCONF1_BFIFOINT_SHIFT, + REG_FIFOCONF1_BFIFOINT_MAX, + NORMAL), + SOC_SINGLE("Burst FIFO Length", + REG_FIFOCONF2, + REG_FIFOCONF2_BFIFOTX_SHIFT, + REG_FIFOCONF2_BFIFOTX_MAX, + NORMAL), + SOC_SINGLE("Burst FIFO EOS Extra Slots", + REG_FIFOCONF3, + REG_FIFOCONF3_BFIFOEXSL_SHIFT, + REG_FIFOCONF3_BFIFOEXSL_MAX, + NORMAL), + SOC_SINGLE("Burst FIFO FS Extra Bit-clocks", + REG_FIFOCONF3, + REG_FIFOCONF3_PREBITCLK0_SHIFT, + REG_FIFOCONF3_PREBITCLK0_MAX, + NORMAL), + SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast), + SOC_ENUM("Burst FIFO Interface Switch", soc_enum_bfifoint), + SOC_SINGLE("Burst FIFO Switch Frame Number", + REG_FIFOCONF4, + REG_FIFOCONF4_BFIFOFRAMSW_SHIFT, + REG_FIFOCONF4_BFIFOFRAMSW_MAX, + NORMAL), + SOC_SINGLE("Burst FIFO Wake Up Delay", + REG_FIFOCONF5, + REG_FIFOCONF5_BFIFOWAKEUP_SHIFT, + REG_FIFOCONF5_BFIFOWAKEUP_MAX, + NORMAL), + SOC_SINGLE("Burst FIFO Samples In FIFO", + REG_FIFOCONF6, + REG_FIFOCONF6_BFIFOSAMPLE_SHIFT, + REG_FIFOCONF6_BFIFOSAMPLE_MAX, + NORMAL), + + /* Sidetone */ + SOC_SINGLE("Sidetone FIR Coeffecient Index", + REG_SIDFIRADR, + REG_SIDFIRADR_ADDRESS_SHIFT, + REG_SIDFIRADR_ADDRESS_MAX, + NORMAL), + + /* ANC */ + SOC_SINGLE_S1R("ANC Warp Delay Shift", + REG_ANCCONF2, + REG_ANCCONF2_VALUE_MIN, + REG_ANCCONF2_VALUE_MAX, + NORMAL), + SOC_SINGLE_S1R("ANC FIR Output Shift", + REG_ANCCONF3, + REG_ANCCONF3_VALUE_MIN, + REG_ANCCONF3_VALUE_MAX, + NORMAL), + SOC_SINGLE_S1R("ANC IIR Output Shift", + REG_ANCCONF4, + REG_ANCCONF4_VALUE_MIN, + REG_ANCCONF4_VALUE_MAX, + NORMAL), + SOC_SINGLE_S2R("ANC Warp Delay", + REG_ANCCONF9, REG_ANCCONF10, + REG_ANC_WARP_DELAY_MIN, + REG_ANC_WARP_DELAY_MAX, + NORMAL), + SOC_MULTIPLE_SA("ANC FIR Coefficients", + ab8500_anc_fir_coeff_cache, + REG_ANC_FIR_COEFF_MIN, + REG_ANC_FIR_COEFF_MAX, + NORMAL), + SOC_MULTIPLE_SA("ANC IIR Coefficients", + ab8500_anc_iir_coeff_cache, + REG_ANC_IIR_COEFF_MIN, + REG_ANC_IIR_COEFF_MAX, + NORMAL), +}; + +static int ab8500_codec_set_format_if1(struct snd_soc_codec *codec, unsigned int fmt) +{ + unsigned int clear_mask, set_mask; + + /* Master or slave */ + + clear_mask = BMASK(REG_DIGIFCONF3_IF1MASTER); + set_mask = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */ + pr_debug("%s: IF1 Master-mode: AB8500 master.\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF3_IF1MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */ + pr_debug("%s: IF1 Master-mode: AB8500 slave.\n", __func__); + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */ + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + pr_err("%s: ERROR: The device is either a master or a slave.\n", + __func__); + default: + pr_err("%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + ab8500_codec_update_reg_audio(codec, + REG_DIGIFCONF3, + BMASK(REG_DIGIFCONF3_IF1MASTER), + BMASK(REG_DIGIFCONF3_IF1MASTER)); + + /* I2S or TDM */ + + clear_mask = BMASK(REG_DIGIFCONF4_FSYNC1P) | + BMASK(REG_DIGIFCONF4_BITCLK1P) | + BMASK(REG_DIGIFCONF4_IF1FORMAT1) | + BMASK(REG_DIGIFCONF4_IF1FORMAT0); + set_mask = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: /* I2S mode */ + pr_debug("%s: IF1 Protocol: I2S\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF4_IF1FORMAT1); + break; + case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */ + pr_debug("%s: IF1 Protocol: DSP B (TDM)\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF4_IF1FORMAT0); + break; + default: + pr_err("%s: ERROR: Unsupported format (0x%x)!\n", + __func__, + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF4, clear_mask, set_mask); + + return 0; +} + +static int ab8500_codec_set_word_length_if1(struct snd_soc_codec *codec, unsigned int wl) +{ + unsigned int clear_mask, set_mask; + + clear_mask = BMASK(REG_DIGIFCONF4_IF1WL1) | BMASK(REG_DIGIFCONF4_IF1WL0); + set_mask = 0; + + switch (wl) { + case 16: + break; + case 20: + set_mask |= BMASK(REG_DIGIFCONF4_IF1WL0); + break; + case 24: + set_mask |= BMASK(REG_DIGIFCONF4_IF1WL1); + break; + case 32: + set_mask |= BMASK(REG_DIGIFCONF2_IF0WL1) | + BMASK(REG_DIGIFCONF2_IF0WL0); + break; + default: + pr_err("%s: Unsupporter word-length 0x%x\n", __func__, wl); + return -EINVAL; + } + + pr_debug("%s: Word-length: %d bits.\n", __func__, wl); + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF4, clear_mask, set_mask); + + return 0; +} + +static int ab8500_codec_set_bit_delay_if1(struct snd_soc_codec *codec, unsigned int delay) +{ + unsigned int clear_mask, set_mask; + + clear_mask = BMASK(REG_DIGIFCONF4_IF1DEL); + set_mask = 0; + + switch (delay) { + case 0: + break; + case 1: + set_mask |= BMASK(REG_DIGIFCONF4_IF1DEL); + break; + default: + pr_err("%s: ERROR: Unsupported bit-delay (0x%x)!\n", __func__, delay); + return -EINVAL; + } + + pr_debug("%s: IF1 Bit-delay: %d bits.\n", __func__, delay); + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF4, clear_mask, set_mask); + + return 0; +} + +/* Configures audio macrocell into the AB8500 Chip */ +static void ab8500_codec_configure_audio_macrocell(struct snd_soc_codec *codec) +{ + int data, ret; + + ret = ab8500_sysctrl_write(AB8500_STW4500CTRL3, + AB8500_STW4500CTRL3_CLK32KOUT2DIS | AB8500_STW4500CTRL3_RESETAUDN, + AB8500_STW4500CTRL3_RESETAUDN); + if (ret < 0) + pr_err("%s: WARN: Unable to set reg STW4500CTRL3!\n", __func__); + + data = ab8500_codec_read_reg(codec, AB8500_MISC, AB8500_GPIO_DIR4_REG); + data |= GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | GPIO31_DIR_OUTPUT; + ab8500_codec_write_reg(codec, AB8500_MISC, AB8500_GPIO_DIR4_REG, data); +} + +/* Extended interface for codec-driver */ + +void ab8500_audio_power_control(bool power_on) +{ + if (ab8500_codec == NULL) { + pr_err("%s: ERROR: AB8500 ASoC-driver not yet probed!\n", __func__); + return; + } + + if (power_on) { + unsigned int set_mask; + pr_debug("Enabling AB8500."); + set_mask = BMASK(REG_POWERUP_POWERUP) | BMASK(REG_POWERUP_ENANA); + ab8500_codec_update_reg_audio(ab8500_codec, REG_POWERUP, 0x00, set_mask); + } else { + unsigned int clear_mask; + pr_debug("Disabling AB8500."); + clear_mask = BMASK(REG_POWERUP_POWERUP) | BMASK(REG_POWERUP_ENANA); + ab8500_codec_update_reg_audio(ab8500_codec, REG_POWERUP, clear_mask, 0x00); + } +} + +void ab8500_audio_pwm_vibra(unsigned char speed_left_pos, + unsigned char speed_left_neg, + unsigned char speed_right_pos, + unsigned char speed_right_neg) +{ + unsigned int clear_mask, set_mask; + bool vibra_on; + + if (ab8500_codec == NULL) { + pr_err("%s: ERROR: AB8500 ASoC-driver not yet probed!\n", __func__); + return; + } + + vibra_on = speed_left_pos | speed_left_neg | speed_right_pos | speed_right_neg; + if (!vibra_on) { + speed_left_pos = 0; + speed_left_neg = 0; + speed_right_pos = 0; + speed_right_neg = 0; + } + + pr_debug("%s: PWM-vibra (%d, %d, %d, %d).\n", + __func__, + speed_left_pos, + speed_left_neg, + speed_right_pos, + speed_right_neg); + + set_mask = BMASK(REG_PWMGENCONF1_PWMTOVIB1) | + BMASK(REG_PWMGENCONF1_PWMTOVIB2) | + BMASK(REG_PWMGENCONF1_PWM1CTRL) | + BMASK(REG_PWMGENCONF1_PWM2CTRL) | + BMASK(REG_PWMGENCONF1_PWM1NCTRL) | + BMASK(REG_PWMGENCONF1_PWM1PCTRL) | + BMASK(REG_PWMGENCONF1_PWM2NCTRL) | + BMASK(REG_PWMGENCONF1_PWM2PCTRL); + ab8500_codec_update_reg_audio(ab8500_codec, REG_PWMGENCONF1, 0x00, set_mask); + + if (speed_left_pos > REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX) + speed_left_pos = REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX; + ab8500_codec_update_reg_audio(ab8500_codec, REG_PWMGENCONF3, REG_MASK_ALL, speed_left_pos); + + if (speed_left_neg > REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX) + speed_left_neg = REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX; + ab8500_codec_update_reg_audio(ab8500_codec, REG_PWMGENCONF2, REG_MASK_ALL, speed_left_neg); + + if (speed_right_pos > REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX) + speed_right_pos = REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX; + ab8500_codec_update_reg_audio(ab8500_codec, REG_PWMGENCONF5, REG_MASK_ALL, speed_right_pos); + + if (speed_right_neg > REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX) + speed_right_neg = REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX; + ab8500_codec_update_reg_audio(ab8500_codec, REG_PWMGENCONF4, REG_MASK_ALL, speed_right_neg); + + if (vibra_on) { + clear_mask = 0; + set_mask = BMASK(REG_ANACONF4_ENVIB1) | BMASK(REG_ANACONF4_ENVIB2); + } else { + clear_mask = BMASK(REG_ANACONF4_ENVIB1) | BMASK(REG_ANACONF4_ENVIB2); + set_mask = 0; + }; + ab8500_codec_update_reg_audio(ab8500_codec, REG_ANACONF4, clear_mask, set_mask); +} + +int ab8500_audio_set_word_length(struct snd_soc_dai *dai, unsigned int wl) +{ + unsigned int clear_mask, set_mask; + struct snd_soc_codec *codec = dai->codec; + + clear_mask = BMASK(REG_DIGIFCONF2_IF0WL0) | BMASK(REG_DIGIFCONF2_IF0WL1); + set_mask = 0; + + switch (wl) { + case 16: + break; + case 20: + set_mask |= BMASK(REG_DIGIFCONF2_IF0WL0); + break; + case 24: + set_mask |= BMASK(REG_DIGIFCONF2_IF0WL1); + break; + case 32: + set_mask |= BMASK(REG_DIGIFCONF2_IF0WL1) | + BMASK(REG_DIGIFCONF2_IF0WL0); + break; + default: + pr_err("%s: Unsupporter word-length 0x%x\n", __func__, wl); + return -EINVAL; + } + + pr_debug("%s: IF0 Word-length: %d bits.\n", __func__, wl); + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF2, clear_mask, set_mask); + + return 0; +} + +int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, unsigned int delay) +{ + unsigned int clear_mask, set_mask; + struct snd_soc_codec *codec = dai->codec; + + clear_mask = BMASK(REG_DIGIFCONF2_IF0DEL); + set_mask = 0; + + switch (delay) { + case 0: + break; + case 1: + set_mask |= BMASK(REG_DIGIFCONF2_IF0DEL); + break; + default: + pr_err("%s: ERROR: Unsupported bit-delay (0x%x)!\n", __func__, delay); + return -EINVAL; + } + + pr_debug("%s: IF0 Bit-delay: %d bits.\n", __func__, delay); + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF2, clear_mask, set_mask); + + return 0; +} + +int ab8500_audio_setup_if1(struct snd_soc_codec *codec, + unsigned int fmt, + unsigned int wl, + unsigned int delay) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + ret = ab8500_codec_set_format_if1(codec, fmt); + if (ret) + return -1; + + ret = ab8500_codec_set_bit_delay_if1(codec, delay); + if (ret) + return -1; + + + ret = ab8500_codec_set_word_length_if1(codec, wl); + if (ret) + return -1; + + return 0; +} + +/* ANC block current configuration status */ +unsigned int ab8500_audio_anc_status(void) +{ + return ab8500_anc_status; +} + +/* ANC IIR-/FIR-coefficients configuration sequence */ +int ab8500_audio_anc_configure(unsigned int req_state) +{ + bool configure_fir = req_state == ANC_CONFIGURE_FIR || req_state == ANC_CONFIGURE_FIR_IIR; + bool configure_iir = req_state == ANC_CONFIGURE_IIR || req_state == ANC_CONFIGURE_FIR_IIR; + unsigned int bank, param; + int ret; + + if (req_state == ANC_UNCONFIGURED + || req_state == ANC_FIR_IIR_CONFIGURED + || req_state == ANC_FIR_CONFIGURED + || req_state == ANC_IIR_CONFIGURED + || req_state == ANC_ERROR) + return -1; + + mutex_lock(&ab8500_anc_conf_lock); + + if (configure_fir) + AB8500_CLEAR_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ENANC, ret, cleanup) + + AB8500_SET_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ENANC, ret, cleanup) + + if (configure_fir) { + for (bank = 0; bank < AB8500_NR_OF_ANC_COEFF_BANKS; bank++) { + for (param = 0; param < REG_ANC_FIR_COEFFS; param++) { + if (param == 0 && bank == 0) + AB8500_SET_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ANCFIRUPDATE, ret, cleanup) + + AB8500_WRITE(REG_ANCCONF5, ab8500_anc_fir_coeff_cache[param] >> 8 & REG_MASK_ALL, ret, cleanup) + AB8500_WRITE(REG_ANCCONF6, ab8500_anc_fir_coeff_cache[param] & REG_MASK_ALL, ret, cleanup) + + if (param == REG_ANC_FIR_COEFFS - 1 && bank == 1) + AB8500_CLEAR_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ANCFIRUPDATE, ret, cleanup) + } + } + if (ab8500_anc_status == ANC_IIR_CONFIGURED) + ab8500_anc_status = ANC_FIR_IIR_CONFIGURED; + else if (ab8500_anc_status != ANC_FIR_IIR_CONFIGURED) + ab8500_anc_status = ANC_FIR_CONFIGURED; + } + + if (configure_iir) { + for (bank = 0; bank < AB8500_NR_OF_ANC_COEFF_BANKS; bank++) { + for (param = 0; param < REG_ANC_IIR_COEFFS; param++) { + if (param == 0) { + if (bank == 0) { + AB8500_SET_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ANCIIRINIT, ret, cleanup) + udelay(2000); + AB8500_CLEAR_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ANCIIRINIT, ret, cleanup) + udelay(2000); + } else { + AB8500_SET_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ANCIIRUPDATE, ret, cleanup) + } + } else if (param > 3) { + AB8500_WRITE(REG_ANCCONF7, REG_MASK_NONE, ret, cleanup) + AB8500_WRITE(REG_ANCCONF8, ab8500_anc_iir_coeff_cache[param] >> 16 & REG_MASK_ALL, ret, cleanup) + } + + AB8500_WRITE(REG_ANCCONF7, ab8500_anc_iir_coeff_cache[param] >> 8 & REG_MASK_ALL, ret, cleanup) + AB8500_WRITE(REG_ANCCONF8, ab8500_anc_iir_coeff_cache[param] & REG_MASK_ALL, ret, cleanup) + + if (param == REG_ANC_IIR_COEFFS - 1 && bank == 1) + AB8500_CLEAR_BIT_LOCKED(REG_ANCCONF1, REG_ANCCONF1_ANCIIRUPDATE, ret, cleanup) + } + } + if (ab8500_anc_status == ANC_FIR_CONFIGURED) + ab8500_anc_status = ANC_FIR_IIR_CONFIGURED; + else if (ab8500_anc_status != ANC_FIR_IIR_CONFIGURED) + ab8500_anc_status = ANC_IIR_CONFIGURED; + } + + mutex_unlock(&ab8500_anc_conf_lock); + + return 0; + +cleanup: + ret |= snd_soc_update_bits_locked(ab8500_codec + , REG_ANCCONF1 + , BMASK(REG_ANCCONF1_ENANC) + | BMASK(REG_ANCCONF1_ANCIIRINIT) + | BMASK(REG_ANCCONF1_ANCIIRUPDATE) + | BMASK(REG_ANCCONF1_ANCFIRUPDATE) + , REG_MASK_NONE); + + ab8500_anc_status = ANC_ERROR; + + mutex_unlock(&ab8500_anc_conf_lock); + + return ret; +} + +bool ab8500_audio_dapm_path_active(enum ab8500_audio_dapm_path dapm_path) +{ + int reg, reg_mask; + + switch (dapm_path) { + case AB8500_AUDIO_DAPM_PATH_DMIC: + reg = ab8500_codec_read_reg_audio(ab8500_codec, REG_DIGMICCONF); + reg_mask = BMASK(REG_DIGMICCONF_ENDMIC1) | + BMASK(REG_DIGMICCONF_ENDMIC2) | + BMASK(REG_DIGMICCONF_ENDMIC3) | + BMASK(REG_DIGMICCONF_ENDMIC4) | + BMASK(REG_DIGMICCONF_ENDMIC5) | + BMASK(REG_DIGMICCONF_ENDMIC6); + return reg & reg_mask; + + case AB8500_AUDIO_DAPM_PATH_AMIC1: + reg = ab8500_codec_read_reg_audio(ab8500_codec, REG_ANACONF2); + reg_mask = BMASK(REG_ANACONF2_MUTMIC1); + return !(reg & reg_mask); + + case AB8500_AUDIO_DAPM_PATH_AMIC2: + reg = ab8500_codec_read_reg_audio(ab8500_codec, REG_ANACONF2); + reg_mask = BMASK(REG_ANACONF2_MUTMIC2); + return !(reg & reg_mask); + + default: + return false; + } +} + +static int ab8500_codec_add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(&codec->dapm, ab8500_dapm_widgets, + ARRAY_SIZE(ab8500_dapm_widgets)); + if (ret < 0) { + pr_err("%s: Failed to create DAPM controls (%d).\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&codec->dapm, dapm_routes, ARRAY_SIZE(dapm_routes)); + if (ret < 0) { + pr_err("%s: Failed to add DAPM routes (%d).\n", + __func__, ret); + return ret; + } + + return 0; +} + +static int ab8500_codec_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + pr_debug("%s Enter.\n", __func__); + return 0; +} + +static int ab8500_codec_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pr_debug("%s Enter.\n", __func__); + + return 0; +} + +static int ab8500_codec_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pr_debug("%s Enter.\n", __func__); + + /* Clear interrupt status registers by reading them. */ + ab8500_codec_read_reg_audio(dai->codec, REG_AUDINTSOURCE1); + ab8500_codec_read_reg_audio(dai->codec, REG_AUDINTSOURCE2); + + return 0; +} + +static void ab8500_codec_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pr_debug("%s Enter.\n", __func__); + + ab8500_codec_dump_all_reg(dai->codec); +} + +static int ab8500_codec_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + pr_err("%s Enter.\n", __func__); + + return 0; +} + +/* Gates clocking according format mask */ +static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec, unsigned int fmt) +{ + unsigned int clear_mask; + unsigned int set_mask; + + clear_mask = BMASK(REG_DIGIFCONF1_ENMASTGEN) | + BMASK(REG_DIGIFCONF1_ENFSBITCLK0); + + set_mask = BMASK(REG_DIGIFCONF1_ENMASTGEN); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: /* continuous clock */ + pr_debug("%s: IF0 Clock is continous.\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF1_ENFSBITCLK0); + break; + case SND_SOC_DAIFMT_GATED: /* clock is gated */ + pr_debug("%s: IF0 Clock is gated.\n", __func__); + break; + default: + pr_err("%s: ERROR: Unsupporter clock mask (0x%x)!\n", + __func__, + fmt & SND_SOC_DAIFMT_CLOCK_MASK); + return -EINVAL; + } + + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF1, clear_mask, set_mask); + + return 0; +} + +static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + unsigned int clear_mask; + unsigned int set_mask; + struct snd_soc_codec *codec = dai->codec; + int err; + + pr_debug("%s: Enter (fmt = 0x%x)\n", __func__, fmt); + + clear_mask = BMASK(REG_DIGIFCONF3_IF1DATOIF0AD) | + BMASK(REG_DIGIFCONF3_IF1CLKTOIF0CLK) | + BMASK(REG_DIGIFCONF3_IF0BFIFOEN) | + BMASK(REG_DIGIFCONF3_IF0MASTER); + set_mask = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */ + pr_debug("%s: IF0 Master-mode: AB8500 master.\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF3_IF0MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */ + pr_debug("%s: IF0 Master-mode: AB8500 slave.\n", __func__); + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */ + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + pr_err("%s: ERROR: The device is either a master or a slave.\n", __func__); + default: + pr_err("%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, + (fmt & SND_SOC_DAIFMT_MASTER_MASK)); + return -EINVAL; + break; + } + + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF3, clear_mask, set_mask); + + /* Set clock gating */ + err = ab8500_codec_set_dai_clock_gate(codec, fmt); + if (err) { + pr_err("%s: ERRROR: Failed to set clock gate (%d).\n", __func__, err); + return err; + } + + /* Setting data transfer format */ + + clear_mask = BMASK(REG_DIGIFCONF2_IF0FORMAT0) | + BMASK(REG_DIGIFCONF2_IF0FORMAT1) | + BMASK(REG_DIGIFCONF2_FSYNC0P) | + BMASK(REG_DIGIFCONF2_BITCLK0P); + set_mask = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: /* I2S mode */ + pr_debug("%s: IF0 Protocol: I2S\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF2_IF0FORMAT1); + + /* 32 bit, 0 delay */ + ab8500_audio_set_word_length(dai, 32); + ab8500_audio_set_bit_delay(dai, 0); + + break; + case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */ + pr_debug("%s: IF0 Protocol: DSP A (TDM)\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF2_IF0FORMAT0); + break; + case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */ + pr_debug("%s: IF0 Protocol: DSP B (TDM)\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF2_IF0FORMAT0); + break; + default: + pr_err("%s: ERROR: Unsupporter format (0x%x)!\n", + __func__, + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ + pr_debug("%s: IF0: Normal bit clock, normal frame\n", __func__); + break; + case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */ + pr_debug("%s: IF0: Normal bit clock, inverted frame\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF2_FSYNC0P); + break; + case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */ + pr_debug("%s: IF0: Inverted bit clock, normal frame\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF2_BITCLK0P); + break; + case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */ + pr_debug("%s: IF0: Inverted bit clock, inverted frame\n", __func__); + set_mask |= BMASK(REG_DIGIFCONF2_FSYNC0P); + set_mask |= BMASK(REG_DIGIFCONF2_BITCLK0P); + break; + default: + pr_err("%s: ERROR: Unsupported INV mask 0x%x\n", + __func__, + (fmt & SND_SOC_DAIFMT_INV_MASK)); + return -EINVAL; + break; + } + + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF2, clear_mask, set_mask); + + return 0; +} + +static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int set_mask, clear_mask, slots_active; + + /* Only 16 bit slot width is supported at the moment in TDM mode */ + if (slot_width != 16) { + pr_err("%s: ERROR: Unsupported slot_width %d.\n", + __func__, slot_width); + return -EINVAL; + } + + /* Setup TDM clocking according to slot count */ + pr_debug("%s: Slots, total: %d\n", __func__, slots); + clear_mask = BMASK(REG_DIGIFCONF1_IF0BITCLKOS0) | + BMASK(REG_DIGIFCONF1_IF0BITCLKOS1); + switch (slots) { + case 2: + set_mask = REG_MASK_NONE; + break; + case 4: + set_mask = BMASK(REG_DIGIFCONF1_IF0BITCLKOS0); + break; + case 8: + set_mask = BMASK(REG_DIGIFCONF1_IF0BITCLKOS1); + break; + case 16: + set_mask = BMASK(REG_DIGIFCONF1_IF0BITCLKOS0) | + BMASK(REG_DIGIFCONF1_IF0BITCLKOS1); + break; + default: + pr_err("%s: ERROR: Unsupported number of slots (%d)!\n", __func__, slots); + return -EINVAL; + } + ab8500_codec_update_reg_audio(codec, REG_DIGIFCONF1, clear_mask, set_mask); + + /* Setup TDM DA according to active tx slots */ + clear_mask = REG_DASLOTCONFX_SLTODAX_MASK; + slots_active = hweight32(tx_mask); + pr_debug("%s: Slots, active, TX: %d\n", __func__, slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* Slot 9 -> DA_IN1 & DA_IN3 */ + ab8500_codec_update_reg_audio(codec, REG_DASLOTCONF1, clear_mask, 9); + ab8500_codec_update_reg_audio(codec, REG_DASLOTCONF3, clear_mask, 9); + break; + case 2: + /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */ + ab8500_codec_update_reg_audio(codec, REG_DASLOTCONF1, clear_mask, 9); + ab8500_codec_update_reg_audio(codec, REG_DASLOTCONF3, clear_mask, 9); + ab8500_codec_update_reg_audio(codec, REG_DASLOTCONF2, clear_mask, 11); + ab8500_codec_update_reg_audio(codec, REG_DASLOTCONF4, clear_mask, 11); + + break; + case 8: + pr_debug("%s: In 8-channel mode DA-from-slot mapping is set manually.", __func__); + break; + default: + pr_err("%s: Unsupported number of active TX-slots (%d)!\n", __func__, slots_active); + return -EINVAL; + } + + /* Setup TDM AD according to active RX-slots */ + slots_active = hweight32(rx_mask); + pr_debug("%s: Slots, active, RX: %d\n", __func__, slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* AD_OUT3 -> slot 0 & 1 */ + ab8500_codec_update_reg_audio(codec, REG_ADSLOTSEL1, + REG_MASK_ALL, + REG_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + REG_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD); + break; + case 2: + /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */ + ab8500_codec_update_reg_audio(codec, REG_ADSLOTSEL1, + REG_MASK_ALL, + REG_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + REG_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD); + break; + case 8: + pr_debug("%s: In 8-channel mode AD-to-slot mapping is set manually.", __func__); + break; + default: + pr_err("%s: Unsupported number of active RX-slots (%d)!\n", __func__, slots_active); + return -EINVAL; + } + + return 0; +} + +struct snd_soc_dai_driver ab8500_codec_dai[] = { + { + .name = "ab8500-codec-dai.0", + .id = 0, + .playback = { + .stream_name = "ab8500_0p", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .startup = ab8500_codec_pcm_startup, + .prepare = ab8500_codec_pcm_prepare, + .hw_params = ab8500_codec_pcm_hw_params, + .shutdown = ab8500_codec_pcm_shutdown, + .set_sysclk = ab8500_codec_set_dai_sysclk, + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + }, + { + .name = "ab8500-codec-dai.1", + .id = 1, + .capture = { + .stream_name = "ab8500_0c", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .startup = ab8500_codec_pcm_startup, + .prepare = ab8500_codec_pcm_prepare, + .hw_params = ab8500_codec_pcm_hw_params, + .shutdown = ab8500_codec_pcm_shutdown, + .set_sysclk = ab8500_codec_set_dai_sysclk, + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + } +}; + +static int ab8500_codec_probe(struct snd_soc_codec *codec) +{ + int i, ret; + u8 *cache = codec->reg_cache; + + pr_debug("%s: Enter.\n", __func__); + + ab8500_codec_configure_audio_macrocell(codec); + + for (i = REG_AUDREV; i >= REG_POWERUP; i--) + ab8500_codec_write_reg_audio(codec, i, cache[i]); + + /* Add controls */ + ret = snd_soc_add_controls(codec, ab8500_snd_controls, + ARRAY_SIZE(ab8500_snd_controls)); + if (ret < 0) { + pr_err("%s: failed to add soc controls (%d).\n", + __func__, ret); + return ret; + } + + /* Add controls with events */ + snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&st_fir_value_control, codec)); + snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&st_fir_apply_control, codec)); + + /* Add DAPM-widgets */ + ret = ab8500_codec_add_widgets(codec); + if (ret < 0) { + pr_err("%s: Failed add widgets (%d).\n", __func__, ret); + return ret; + } + + ab8500_codec = codec; + + return ret; +} + +static int ab8500_codec_remove(struct snd_soc_codec *codec) +{ + snd_soc_dapm_free(&codec->dapm); + ab8500_codec = NULL; + + return 0; +} + +static int ab8500_codec_suspend(struct snd_soc_codec *codec, + pm_message_t state) +{ + pr_debug("%s Enter.\n", __func__); + + return 0; +} + +static int ab8500_codec_resume(struct snd_soc_codec *codec) +{ + pr_debug("%s Enter.\n", __func__); + + return 0; +} + +struct snd_soc_codec_driver ab8500_codec_driver = { + .probe = ab8500_codec_probe, + .remove = ab8500_codec_remove, + .suspend = ab8500_codec_suspend, + .resume = ab8500_codec_resume, + .read = ab8500_codec_read_reg_audio, + .write = ab8500_codec_write_reg_audio, + .reg_cache_size = ARRAY_SIZE(ab8500_reg_cache), + .reg_word_size = sizeof(u8), + .reg_cache_default = ab8500_reg_cache, +}; + +static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev) +{ + int err; + + pr_debug("%s: Enter.\n", __func__); + + pr_info("%s: Register codec.\n", __func__); + err = snd_soc_register_codec(&pdev->dev, + &ab8500_codec_driver, + ab8500_codec_dai, + ARRAY_SIZE(ab8500_codec_dai)); + + if (err < 0) { + pr_err("%s: Error: Failed to register codec (%d).\n", + __func__, err); + } + + return err; +} + +static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev) +{ + pr_info("%s Enter.\n", __func__); + + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static int ab8500_codec_driver_suspend(struct platform_device *pdev, + pm_message_t state) +{ + pr_debug("%s Enter.\n", __func__); + + return 0; +} + +static int ab8500_codec_driver_resume(struct platform_device *pdev) +{ + pr_debug("%s Enter.\n", __func__); + + return 0; +} + +static struct platform_driver ab8500_codec_platform_driver = { + .driver = { + .name = "ab8500-codec", + .owner = THIS_MODULE, + }, + .probe = ab8500_codec_driver_probe, + .remove = __devexit_p(ab8500_codec_driver_remove), + .suspend = ab8500_codec_driver_suspend, + .resume = ab8500_codec_driver_resume, +}; + +static int __devinit ab8500_codec_platform_driver_init(void) +{ + int ret; + + pr_info("%s: Enter.\n", __func__); + + ret = platform_driver_register(&ab8500_codec_platform_driver); + if (ret != 0) { + pr_err("%s: Failed to register AB8500 platform driver (%d)!\n", + __func__, ret); + } + + return ret; +} + +static void __exit ab8500_codec_platform_driver_exit(void) +{ + pr_info("%s: Enter.\n", __func__); + + platform_driver_unregister(&ab8500_codec_platform_driver); +} + +module_init(ab8500_codec_platform_driver_init); +module_exit(ab8500_codec_platform_driver_exit); + diff --git a/sound/soc/codecs/ab8500_audio.h b/sound/soc/codecs/ab8500_audio.h new file mode 100644 index 0000000..ad6cd16 --- /dev/null +++ b/sound/soc/codecs/ab8500_audio.h @@ -0,0 +1,673 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Mikko J. Lehto <mikko.lehto(a)symbio.com>, + * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, + * Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 AB8500_CODEC_REGISTERS_H +#define AB8500_CODEC_REGISTERS_H + +#define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000) +#define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE) + +extern struct snd_soc_dai_driver ab8500_codec_dai[]; +extern struct snd_soc_codec_driver soc_codec_dev_ab8500; + +/* Extended interface for codec-driver */ + +void ab8500_audio_power_control(bool power_on); +void ab8500_audio_pwm_vibra(unsigned char speed_left_pos, + unsigned char speed_left_neg, + unsigned char speed_right_pos, + unsigned char speed_right_neg); +int ab8500_audio_set_word_length(struct snd_soc_dai *dai, unsigned int wl); +int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, unsigned int delay); +int ab8500_audio_setup_if1(struct snd_soc_codec *codec, + unsigned int fmt, + unsigned int wl, + unsigned int delay); +unsigned int ab8500_audio_anc_status(void); +int ab8500_audio_anc_configure(unsigned int req_state); + +enum ab8500_audio_dapm_path { + AB8500_AUDIO_DAPM_PATH_DMIC, + AB8500_AUDIO_DAPM_PATH_AMIC1, + AB8500_AUDIO_DAPM_PATH_AMIC2 +}; +bool ab8500_audio_dapm_path_active(enum ab8500_audio_dapm_path dapm_path); + +#define SOC_SINGLE_VALUE_S1R(xreg0, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_smra_control) \ + { .reg = ((unsigned int[]){ xreg0 }), \ + .rcount = 1, .count = xcount, \ + .invert = xinvert, .min = xmin, .max = xmax}) + +#define SOC_SINGLE_VALUE_S2R(xreg0, xreg1, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_smra_control) \ + {.reg = ((unsigned int[]){ xreg0, xreg1 }), \ + .rcount = 2, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) + +#define SOC_SINGLE_VALUE_S4R(xreg0, xreg1, xreg2, xreg3, \ + xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_smra_control) \ + {.reg = ((unsigned int[]){ xreg0, xreg1, xreg2, xreg3 }), \ + .rcount = 4, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) + +#define SOC_SINGLE_VALUE_S8R(xreg0, xreg1, xreg2, xreg3, \ + xreg4, xreg5, xreg6, xreg7, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_smra_control) \ + {.reg = ((unsigned int[]){ xreg0, xreg1, xreg2, xreg3, \ + xreg4, xreg5, xreg6, xreg7 }), \ + .rcount = 8, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) + +#define SOC_MULTIPLE_VALUE_SA(xvalues, xcount, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_smra_control) \ + {.values = xvalues, .rcount = 1, .count = xcount, \ + .min = xmin, .max = xmax, .invert = xinvert}) + +#define SOC_ENUM_STROBE_DECL(name, xreg, xbit, xinvert, xtexts) \ + struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xbit, \ + xinvert, 2, xtexts) + +/* Extended SOC macros */ + +#define SOC_SINGLE_S1R(xname, reg0, min, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_soc_info_s, .get = snd_soc_get_smr, .put = snd_soc_put_smr, \ + .private_value = SOC_SINGLE_VALUE_S1R(reg0, 1, min, max, invert) } + +#define SOC_SINGLE_S2R(xname, reg0, reg1, min, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_soc_info_s, .get = snd_soc_get_smr, .put = snd_soc_put_smr, \ + .private_value = SOC_SINGLE_VALUE_S2R(reg0, reg1, 1, min, max, invert) } + +#define SOC_SINGLE_S4R(xname, reg0, reg1, reg2, reg3, min, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_soc_info_s, .get = snd_soc_get_smr, .put = snd_soc_put_smr, \ + .private_value = SOC_SINGLE_VALUE_S4R(reg0, reg1, reg2, reg3, \ + 1, min, max, invert) } + +#define SOC_SINGLE_S8R(xname, reg0, reg1, reg2, reg3, \ + reg4, reg5, reg6, reg7, min, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_soc_info_s, .get = snd_soc_get_smr, .put = snd_soc_put_smr, \ + .private_value = SOC_SINGLE_VALUE_S4R(reg0, reg1, reg2, reg3, \ + reg4, reg5, reg6, reg7\ + 1, min, max, invert) } + +#define SOC_MULTIPLE_SA(xname, values, min, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_soc_info_s, .get = snd_soc_get_sa, .put = snd_soc_put_sa, \ + .private_value = SOC_MULTIPLE_VALUE_SA((long *)values, ARRAY_SIZE(values), \ + min, max, invert) } + +#define SOC_ENUM_STROBE(xname, enum) \ + SOC_ENUM_EXT(xname, xenum, \ + snd_soc_get_enum_strobe, \ + snd_soc_put_enum_strobe) + +/* AB8500 audio bank (0x0d) register definitions */ + +#define REG_POWERUP 0x00 +#define REG_AUDSWRESET 0x01 +#define REG_ADPATHENA 0x02 +#define REG_DAPATHENA 0x03 +#define REG_ANACONF1 0x04 +#define REG_ANACONF2 0x05 +#define REG_DIGMICCONF 0x06 +#define REG_ANACONF3 0x07 +#define REG_ANACONF4 0x08 +#define REG_DAPATHCONF 0x09 +#define REG_MUTECONF 0x0A +#define REG_SHORTCIRCONF 0x0B +#define REG_ANACONF5 0x0C +#define REG_ENVCPCONF 0x0D +#define REG_SIGENVCONF 0x0E +#define REG_PWMGENCONF1 0x0F +#define REG_PWMGENCONF2 0x10 +#define REG_PWMGENCONF3 0x11 +#define REG_PWMGENCONF4 0x12 +#define REG_PWMGENCONF5 0x13 +#define REG_ANAGAIN1 0x14 +#define REG_ANAGAIN2 0x15 +#define REG_ANAGAIN3 0x16 +#define REG_ANAGAIN4 0x17 +#define REG_DIGLINHSLGAIN 0x18 +#define REG_DIGLINHSRGAIN 0x19 +#define REG_ADFILTCONF 0x1A +#define REG_DIGIFCONF1 0x1B +#define REG_DIGIFCONF2 0x1C +#define REG_DIGIFCONF3 0x1D +#define REG_DIGIFCONF4 0x1E +#define REG_ADSLOTSEL1 0x1F +#define REG_ADSLOTSEL2 0x20 +#define REG_ADSLOTSEL3 0x21 +#define REG_ADSLOTSEL4 0x22 +#define REG_ADSLOTSEL5 0x23 +#define REG_ADSLOTSEL6 0x24 +#define REG_ADSLOTSEL7 0x25 +#define REG_ADSLOTSEL8 0x26 +#define REG_ADSLOTSEL9 0x27 +#define REG_ADSLOTSEL10 0x28 +#define REG_ADSLOTSEL11 0x29 +#define REG_ADSLOTSEL12 0x2A +#define REG_ADSLOTSEL13 0x2B +#define REG_ADSLOTSEL14 0x2C +#define REG_ADSLOTSEL15 0x2D +#define REG_ADSLOTSEL16 0x2E +#define REG_ADSLOTHIZCTRL1 0x2F +#define REG_ADSLOTHIZCTRL2 0x30 +#define REG_ADSLOTHIZCTRL3 0x31 +#define REG_ADSLOTHIZCTRL4 0x32 +#define REG_DASLOTCONF1 0x33 +#define REG_DASLOTCONF2 0x34 +#define REG_DASLOTCONF3 0x35 +#define REG_DASLOTCONF4 0x36 +#define REG_DASLOTCONF5 0x37 +#define REG_DASLOTCONF6 0x38 +#define REG_DASLOTCONF7 0x39 +#define REG_DASLOTCONF8 0x3A +#define REG_CLASSDCONF1 0x3B +#define REG_CLASSDCONF2 0x3C +#define REG_CLASSDCONF3 0x3D +#define REG_DMICFILTCONF 0x3E +#define REG_DIGMULTCONF1 0x3F +#define REG_DIGMULTCONF2 0x40 +#define REG_ADDIGGAIN1 0x41 +#define REG_ADDIGGAIN2 0x42 +#define REG_ADDIGGAIN3 0x43 +#define REG_ADDIGGAIN4 0x44 +#define REG_ADDIGGAIN5 0x45 +#define REG_ADDIGGAIN6 0x46 +#define REG_DADIGGAIN1 0x47 +#define REG_DADIGGAIN2 0x48 +#define REG_DADIGGAIN3 0x49 +#define REG_DADIGGAIN4 0x4A +#define REG_DADIGGAIN5 0x4B +#define REG_DADIGGAIN6 0x4C +#define REG_ADDIGLOOPGAIN1 0x4D +#define REG_ADDIGLOOPGAIN2 0x4E +#define REG_HSLEARDIGGAIN 0x4F +#define REG_HSRDIGGAIN 0x50 +#define REG_SIDFIRGAIN1 0x51 +#define REG_SIDFIRGAIN2 0x52 +#define REG_ANCCONF1 0x53 +#define REG_ANCCONF2 0x54 +#define REG_ANCCONF3 0x55 +#define REG_ANCCONF4 0x56 +#define REG_ANCCONF5 0x57 +#define REG_ANCCONF6 0x58 +#define REG_ANCCONF7 0x59 +#define REG_ANCCONF8 0x5A +#define REG_ANCCONF9 0x5B +#define REG_ANCCONF10 0x5C +#define REG_ANCCONF11 0x5D +#define REG_ANCCONF12 0x5E +#define REG_ANCCONF13 0x5F +#define REG_ANCCONF14 0x60 +#define REG_SIDFIRADR 0x61 +#define REG_SIDFIRCOEF1 0x62 +#define REG_SIDFIRCOEF2 0x63 +#define REG_SIDFIRCONF 0x64 +#define REG_AUDINTMASK1 0x65 +#define REG_AUDINTSOURCE1 0x66 +#define REG_AUDINTMASK2 0x67 +#define REG_AUDINTSOURCE2 0x68 +#define REG_FIFOCONF1 0x69 +#define REG_FIFOCONF2 0x6A +#define REG_FIFOCONF3 0x6B +#define REG_FIFOCONF4 0x6C +#define REG_FIFOCONF5 0x6D +#define REG_FIFOCONF6 0x6E +#define REG_AUDREV 0x6F + +#define AB8500_FIRST_REG REG_POWERUP +#define AB8500_LAST_REG REG_AUDREV +#define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1) + + +#define REG_MASK_ALL 0xFF +#define REG_MASK_NONE 0x00 + +/* REG_POWERUP */ +#define REG_POWERUP_POWERUP 7 +#define REG_POWERUP_ENANA 3 + +/* REG_AUDSWRESET */ +#define REG_AUDSWRESET_SWRESET 7 + +/* REG_ADPATHENA */ +#define REG_ADPATHENA_ENAD12 7 +#define REG_ADPATHENA_ENAD34 5 +#define REG_ADPATHENA_ENAD5768 3 + +/* REG_DAPATHENA */ +#define REG_DAPATHENA_ENDA1 7 +#define REG_DAPATHENA_ENDA2 6 +#define REG_DAPATHENA_ENDA3 5 +#define REG_DAPATHENA_ENDA4 4 +#define REG_DAPATHENA_ENDA5 3 +#define REG_DAPATHENA_ENDA6 2 + +/* REG_ANACONF1 */ +#define REG_ANACONF1_HSLOWPOW 7 +#define REG_ANACONF1_DACLOWPOW1 6 +#define REG_ANACONF1_DACLOWPOW0 5 +#define REG_ANACONF1_EARDACLOWPOW 4 +#define REG_ANACONF1_EARSELCM 2 +#define REG_ANACONF1_HSHPEN 1 +#define REG_ANACONF1_EARDRVLOWPOW 0 + +/* REG_ANACONF2 */ +#define REG_ANACONF2_ENMIC1 7 +#define REG_ANACONF2_ENMIC2 6 +#define REG_ANACONF2_ENLINL 5 +#define REG_ANACONF2_ENLINR 4 +#define REG_ANACONF2_MUTMIC1 3 +#define REG_ANACONF2_MUTMIC2 2 +#define REG_ANACONF2_MUTLINL 1 +#define REG_ANACONF2_MUTLINR 0 + +/* REG_DIGMICCONF */ +#define REG_DIGMICCONF_ENDMIC1 7 +#define REG_DIGMICCONF_ENDMIC2 6 +#define REG_DIGMICCONF_ENDMIC3 5 +#define REG_DIGMICCONF_ENDMIC4 4 +#define REG_DIGMICCONF_ENDMIC5 3 +#define REG_DIGMICCONF_ENDMIC6 2 +#define REG_DIGMICCONF_HSFADSPEED 0 + +/* REG_ANACONF3 */ +#define REG_ANACONF3_MIC1SEL 7 +#define REG_ANACONF3_LINRSEL 6 +#define REG_ANACONF3_ENDRVHSL 5 +#define REG_ANACONF3_ENDRVHSR 4 +#define REG_ANACONF3_ENADCMIC 2 +#define REG_ANACONF3_ENADCLINL 1 +#define REG_ANACONF3_ENADCLINR 0 + +/* REG_ANACONF4 */ +#define REG_ANACONF4_DISPDVSS 7 +#define REG_ANACONF4_ENEAR 6 +#define REG_ANACONF4_ENHSL 5 +#define REG_ANACONF4_ENHSR 4 +#define REG_ANACONF4_ENHFL 3 +#define REG_ANACONF4_ENHFR 2 +#define REG_ANACONF4_ENVIB1 1 +#define REG_ANACONF4_ENVIB2 0 + +/* REG_DAPATHCONF */ +#define REG_DAPATHCONF_ENDACEAR 6 +#define REG_DAPATHCONF_ENDACHSL 5 +#define REG_DAPATHCONF_ENDACHSR 4 +#define REG_DAPATHCONF_ENDACHFL 3 +#define REG_DAPATHCONF_ENDACHFR 2 +#define REG_DAPATHCONF_ENDACVIB1 1 +#define REG_DAPATHCONF_ENDACVIB2 0 + +/* REG_MUTECONF */ +#define REG_MUTECONF_MUTEAR 6 +#define REG_MUTECONF_MUTHSL 5 +#define REG_MUTECONF_MUTHSR 4 +#define REG_MUTECONF_MUTDACEAR 2 +#define REG_MUTECONF_MUTDACHSL 1 +#define REG_MUTECONF_MUTDACHSR 0 + + +/* REG_SHORTCIRCONF */ + +/* REG_ANACONF5 */ +#define REG_ANACONF5_ENCPHS 7 +#define REG_ANACONF5_HSLDACTOLOL 5 +#define REG_ANACONF5_HSRDACTOLOR 4 +#define REG_ANACONF5_ENLOL 3 +#define REG_ANACONF5_ENLOR 2 +#define REG_ANACONF5_HSAUTOEN 0 + +/* REG_ENVCPCONF */ +#define REG_ENVCPCONF_ENVDETHTHRE 4 +#define REG_ENVCPCONF_ENVDETLTHRE 0 +#define REG_ENVCPCONF_ENVDETHTHRE_MAX 0x0F +#define REG_ENVCPCONF_ENVDETLTHRE_MAX 0x0F + +/* REG_SIGENVCONF */ +#define REG_SIGENVCONF_CPLVEN 5 +#define REG_SIGENVCONF_ENVDETCPEN 4 +#define REG_SIGENVCONF_ENVDETTIME 0 +#define REG_SIGENVCONF_ENVDETTIME_MAX 0x0F + +/* REG_PWMGENCONF1 */ +#define REG_PWMGENCONF1_PWMTOVIB1 7 +#define REG_PWMGENCONF1_PWMTOVIB2 6 +#define REG_PWMGENCONF1_PWM1CTRL 5 +#define REG_PWMGENCONF1_PWM2CTRL 4 +#define REG_PWMGENCONF1_PWM1NCTRL 3 +#define REG_PWMGENCONF1_PWM1PCTRL 2 +#define REG_PWMGENCONF1_PWM2NCTRL 1 +#define REG_PWMGENCONF1_PWM2PCTRL 0 + +/* REG_PWMGENCONF2 */ +/* REG_PWMGENCONF3 */ +/* REG_PWMGENCONF4 */ +/* REG_PWMGENCONF5 */ +#define REG_PWMGENCONFX_PWMVIBXPOL 7 +#define REG_PWMGENCONFX_PWMVIBXDUTCYC 0 +#define REG_PWMGENCONFX_PWMVIBXDUTCYC_MAX 0x64 + +/* REG_ANAGAIN1 */ +/* REG_ANAGAIN2 */ +#define REG_ANAGAINX_ENSEMICX 7 +#define REG_ANAGAINX_LOWPOWMICX 6 +#define REG_ANAGAINX_MICXGAIN 0 +#define REG_ANAGAINX_MICXGAIN_MAX 0x1F + +/* REG_ANAGAIN3 */ +#define REG_ANAGAIN3_HSLGAIN 4 +#define REG_ANAGAIN3_HSRGAIN 0 +#define REG_ANAGAIN3_HSXGAIN_MAX 0x0F + +/* REG_ANAGAIN4 */ +#define REG_ANAGAIN4_LINLGAIN 4 +#define REG_ANAGAIN4_LINRGAIN 0 +#define REG_ANAGAIN4_LINXGAIN_MAX 0x0F + +/* REG_DIGLINHSLGAIN */ +/* REG_DIGLINHSRGAIN */ +#define REG_DIGLINHSXGAIN_LINTOHSXGAIN 0 +#define REG_DIGLINHSXGAIN_LINTOHSXGAIN_MAX 0x13 + +/* REG_ADFILTCONF */ +#define REG_ADFILTCONF_AD1NH 7 +#define REG_ADFILTCONF_AD2NH 6 +#define REG_ADFILTCONF_AD3NH 5 +#define REG_ADFILTCONF_AD4NH 4 +#define REG_ADFILTCONF_AD1VOICE 3 +#define REG_ADFILTCONF_AD2VOICE 2 +#define REG_ADFILTCONF_AD3VOICE 1 +#define REG_ADFILTCONF_AD4VOICE 0 + +/* REG_DIGIFCONF1 */ +#define REG_DIGIFCONF1_ENMASTGEN 7 +#define REG_DIGIFCONF1_IF1BITCLKOS1 6 +#define REG_DIGIFCONF1_IF1BITCLKOS0 5 +#define REG_DIGIFCONF1_ENFSBITCLK1 4 +#define REG_DIGIFCONF1_IF0BITCLKOS1 2 +#define REG_DIGIFCONF1_IF0BITCLKOS0 1 +#define REG_DIGIFCONF1_ENFSBITCLK0 0 + +/* REG_DIGIFCONF2 */ +#define REG_DIGIFCONF2_FSYNC0P 6 +#define REG_DIGIFCONF2_BITCLK0P 5 +#define REG_DIGIFCONF2_IF0DEL 4 +#define REG_DIGIFCONF2_IF0FORMAT1 3 +#define REG_DIGIFCONF2_IF0FORMAT0 2 +#define REG_DIGIFCONF2_IF0WL1 1 +#define REG_DIGIFCONF2_IF0WL0 0 + +/* REG_DIGIFCONF3 */ +#define REG_DIGIFCONF3_IF0DATOIF1AD 7 +#define REG_DIGIFCONF3_IF0CLKTOIF1CLK 6 +#define REG_DIGIFCONF3_IF1MASTER 5 +#define REG_DIGIFCONF3_IF1DATOIF0AD 3 +#define REG_DIGIFCONF3_IF1CLKTOIF0CLK 2 +#define REG_DIGIFCONF3_IF0MASTER 1 +#define REG_DIGIFCONF3_IF0BFIFOEN 0 + +/* REG_DIGIFCONF4 */ +#define REG_DIGIFCONF4_FSYNC1P 6 +#define REG_DIGIFCONF4_BITCLK1P 5 +#define REG_DIGIFCONF4_IF1DEL 4 +#define REG_DIGIFCONF4_IF1FORMAT1 3 +#define REG_DIGIFCONF4_IF1FORMAT0 2 +#define REG_DIGIFCONF4_IF1WL1 1 +#define REG_DIGIFCONF4_IF1WL0 0 + +/* REG_ADSLOTSELX */ +#define REG_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 +#define REG_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01 +#define REG_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02 +#define REG_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03 +#define REG_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04 +#define REG_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05 +#define REG_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06 +#define REG_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07 +#define REG_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08 +#define REG_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F +#define REG_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 +#define REG_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10 +#define REG_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20 +#define REG_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30 +#define REG_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40 +#define REG_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50 +#define REG_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60 +#define REG_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70 +#define REG_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80 +#define REG_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0 +#define REG_ADSLOTSELX_EVEN_SHIFT 0 +#define REG_ADSLOTSELX_ODD_SHIFT 4 + +/* REG_ADSLOTHIZCTRL1 */ +/* REG_ADSLOTHIZCTRL2 */ +/* REG_ADSLOTHIZCTRL3 */ +/* REG_ADSLOTHIZCTRL4 */ +/* REG_DASLOTCONF1 */ +#define REG_DASLOTCONF1_DA12VOICE 7 +#define REG_DASLOTCONF1_SWAPDA12_34 6 +#define REG_DASLOTCONF1_DAI7TOADO1 5 + +/* REG_DASLOTCONF2 */ +#define REG_DASLOTCONF2_DAI8TOADO2 5 + +/* REG_DASLOTCONF3 */ +#define REG_DASLOTCONF3_DA34VOICE 7 +#define REG_DASLOTCONF3_DAI7TOADO3 5 + +/* REG_DASLOTCONF4 */ +#define REG_DASLOTCONF4_DAI8TOADO4 5 + +/* REG_DASLOTCONF5 */ +#define REG_DASLOTCONF5_DA56VOICE 7 +#define REG_DASLOTCONF5_DAI7TOADO5 5 + +/* REG_DASLOTCONF6 */ +#define REG_DASLOTCONF6_DAI8TOADO6 5 + +/* REG_DASLOTCONF7 */ +#define REG_DASLOTCONF7_DAI8TOADO7 5 + +/* REG_DASLOTCONF8 */ +#define REG_DASLOTCONF8_DAI7TOADO8 5 + +#define REG_DASLOTCONFX_SLTODAX_SHIFT 0 +#define REG_DASLOTCONFX_SLTODAX_MASK 0x1F + +/* REG_CLASSDCONF1 */ +#define REG_CLASSDCONF1_PARLHF 7 +#define REG_CLASSDCONF1_PARLVIB 6 +#define REG_CLASSDCONF1_VIB1SWAPEN 3 +#define REG_CLASSDCONF1_VIB2SWAPEN 2 +#define REG_CLASSDCONF1_HFLSWAPEN 1 +#define REG_CLASSDCONF1_HFRSWAPEN 0 + +/* REG_CLASSDCONF2 */ +#define REG_CLASSDCONF2_FIRBYP3 7 +#define REG_CLASSDCONF2_FIRBYP2 6 +#define REG_CLASSDCONF2_FIRBYP1 5 +#define REG_CLASSDCONF2_FIRBYP0 4 +#define REG_CLASSDCONF2_HIGHVOLEN3 3 +#define REG_CLASSDCONF2_HIGHVOLEN2 2 +#define REG_CLASSDCONF2_HIGHVOLEN1 1 +#define REG_CLASSDCONF2_HIGHVOLEN0 0 + +/* REG_CLASSDCONF3 */ +#define REG_CLASSDCONF3_DITHHPGAIN 4 +#define REG_CLASSDCONF3_DITHHPGAIN_MAX 0x0A +#define REG_CLASSDCONF3_DITHWGAIN 0 +#define REG_CLASSDCONF3_DITHWGAIN_MAX 0x0A + +/* REG_DMICFILTCONF */ +#define REG_DMICFILTCONF_ANCINSEL 7 +#define REG_DMICFILTCONF_DA3TOEAR 6 +#define REG_DMICFILTCONF_DMIC1SINC3 5 +#define REG_DMICFILTCONF_DMIC2SINC3 4 +#define REG_DMICFILTCONF_DMIC3SINC3 3 +#define REG_DMICFILTCONF_DMIC4SINC3 2 +#define REG_DMICFILTCONF_DMIC5SINC3 1 +#define REG_DMICFILTCONF_DMIC6SINC3 0 + +/* REG_DIGMULTCONF1 */ +#define REG_DIGMULTCONF1_DATOHSLEN 7 +#define REG_DIGMULTCONF1_DATOHSREN 6 +#define REG_DIGMULTCONF1_AD1SEL 5 +#define REG_DIGMULTCONF1_AD2SEL 4 +#define REG_DIGMULTCONF1_AD3SEL 3 +#define REG_DIGMULTCONF1_AD5SEL 2 +#define REG_DIGMULTCONF1_AD6SEL 1 +#define REG_DIGMULTCONF1_ANCSEL 0 + +/* REG_DIGMULTCONF2 */ +#define REG_DIGMULTCONF2_DATOHFREN 7 +#define REG_DIGMULTCONF2_DATOHFLEN 6 +#define REG_DIGMULTCONF2_HFRSEL 5 +#define REG_DIGMULTCONF2_HFLSEL 4 +#define REG_DIGMULTCONF2_FIRSID1SEL 2 +#define REG_DIGMULTCONF2_FIRSID2SEL 0 + +/* REG_ADDIGGAIN1 */ +/* REG_ADDIGGAIN2 */ +/* REG_ADDIGGAIN3 */ +/* REG_ADDIGGAIN4 */ +/* REG_ADDIGGAIN5 */ +/* REG_ADDIGGAIN6 */ +#define REG_ADDIGGAINX_FADEDISADX 6 +#define REG_ADDIGGAINX_ADXGAIN_MAX 0x3F + +/* REG_DADIGGAIN1 */ +/* REG_DADIGGAIN2 */ +/* REG_DADIGGAIN3 */ +/* REG_DADIGGAIN4 */ +/* REG_DADIGGAIN5 */ +/* REG_DADIGGAIN6 */ +#define REG_DADIGGAINX_FADEDISDAX 6 +#define REG_DADIGGAINX_DAXGAIN_MAX 0x3F + +/* REG_ADDIGLOOPGAIN1 */ +/* REG_ADDIGLOOPGAIN2 */ +#define REG_ADDIGLOOPGAINX_FADEDISADXL 6 +#define REG_ADDIGLOOPGAINX_ADXLBGAIN_MAX 0x3F + +/* REG_HSLEARDIGGAIN */ +#define REG_HSLEARDIGGAIN_HSSINC1 7 +#define REG_HSLEARDIGGAIN_FADEDISHSL 4 +#define REG_HSLEARDIGGAIN_HSLDGAIN_MAX 0x09 + +/* REG_HSRDIGGAIN */ +#define REG_HSRDIGGAIN_FADESPEED 6 +#define REG_HSRDIGGAIN_FADEDISHSR 4 +#define REG_HSRDIGGAIN_HSRDGAIN_MAX 0x09 + +/* REG_SIDFIRGAIN1 */ +/* REG_SIDFIRGAIN2 */ +#define REG_SIDFIRGAINX_FIRSIDXGAIN_MAX 0x1F + +/* REG_ANCCONF1 */ +#define REG_ANCCONF1_ANCIIRUPDATE 3 +#define REG_ANCCONF1_ENANC 2 +#define REG_ANCCONF1_ANCIIRINIT 1 +#define REG_ANCCONF1_ANCFIRUPDATE 0 + +/* REG_ANCCONF2 */ +#define REG_ANCCONF2_VALUE_MIN -0x10 +#define REG_ANCCONF2_VALUE_MAX 0x0F +/* REG_ANCCONF3 */ +#define REG_ANCCONF3_VALUE_MIN -0x10 +#define REG_ANCCONF3_VALUE_MAX 0x0F +/* REG_ANCCONF4 */ +#define REG_ANCCONF4_VALUE_MIN -0x10 +#define REG_ANCCONF4_VALUE_MAX 0x0F +/* REG_ANC_FIR_COEFFS */ +#define REG_ANC_FIR_COEFF_MIN -0x8000 +#define REG_ANC_FIR_COEFF_MAX 0x7FFF +#define REG_ANC_FIR_COEFFS 0xF +/* REG_ANC_IIR_COEFFS */ +#define REG_ANC_IIR_COEFF_MIN -0x800000 +#define REG_ANC_IIR_COEFF_MAX 0x7FFFFF +#define REG_ANC_IIR_COEFFS 0x18 +/* REG_ANC_WARP_DELAY */ +#define REG_ANC_WARP_DELAY_MIN 0x0000 +#define REG_ANC_WARP_DELAY_MAX 0xFFFF +/* REG_ANCCONF11 */ +/* REG_ANCCONF12 */ +/* REG_ANCCONF13 */ +/* REG_ANCCONF14 */ + +/* REG_SIDFIRADR */ +#define REG_SIDFIRADR_FIRSIDSET 7 +#define REG_SIDFIRADR_ADDRESS_SHIFT 0 +#define REG_SIDFIRADR_ADDRESS_MAX 0x7F + +/* REG_SIDFIRCOEF1 */ +/* REG_SIDFIRCOEF2 */ +#define REG_SIDFIRCOEFX_VALUE_SHIFT 0 +#define REG_SIDFIRCOEFX_VALUE_MAX 0xFF + +/* REG_SIDFIRCONF */ +#define REG_SIDFIRCONF_ENFIRSIDS 2 +#define REG_SIDFIRCONF_FIRSIDSTOIF1 1 +#define REG_SIDFIRCONF_FIRSIDBUSY 0 + +/* REG_AUDINTMASK1 */ +/* REG_AUDINTSOURCE1 */ +/* REG_AUDINTMASK2 */ +/* REG_AUDINTSOURCE2 */ + +/* REG_FIFOCONF1 */ +#define REG_FIFOCONF1_BFIFOMASK 0x80 +#define REG_FIFOCONF1_BFIFO19M2 0x40 +#define REG_FIFOCONF1_BFIFOINT_SHIFT 0 +#define REG_FIFOCONF1_BFIFOINT_MAX 0x3F + +/* REG_FIFOCONF2 */ +#define REG_FIFOCONF2_BFIFOTX_SHIFT 0 +#define REG_FIFOCONF2_BFIFOTX_MAX 0xFF + +/* REG_FIFOCONF3 */ +#define REG_FIFOCONF3_BFIFOEXSL_SHIFT 5 +#define REG_FIFOCONF3_BFIFOEXSL_MAX 0x5 +#define REG_FIFOCONF3_PREBITCLK0_SHIFT 2 +#define REG_FIFOCONF3_PREBITCLK0_MAX 0x7 +#define REG_FIFOCONF3_BFIFOMAST_SHIFT 1 +#define REG_FIFOCONF3_BFIFORUN_SHIFT 0 + +/* REG_FIFOCONF4 */ +#define REG_FIFOCONF4_BFIFOFRAMSW_SHIFT 0 +#define REG_FIFOCONF4_BFIFOFRAMSW_MAX 0xFF + +/* REG_FIFOCONF5 */ +#define REG_FIFOCONF5_BFIFOWAKEUP_SHIFT 0 +#define REG_FIFOCONF5_BFIFOWAKEUP_MAX 0xFF + +/* REG_FIFOCONF6 */ +#define REG_FIFOCONF6_BFIFOSAMPLE_SHIFT 0 +#define REG_FIFOCONF6_BFIFOSAMPLE_MAX 0xFF + +/* REG_AUDREV */ + +#endif diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig new file mode 100644 index 0000000..bcf1ff8 --- /dev/null +++ b/sound/soc/ux500/Kconfig @@ -0,0 +1,33 @@ +# +# Ux500 SoC audio configuration +# + +menuconfig SND_SOC_UX500 + bool "SoC Audio support for Ux500 platform" + depends on SND_SOC + default n + help + Say Y if you want to add support for the Ux500 platform. + +choice + prompt "Ux500 Platform" + depends on SND_SOC_UX500 + default SND_SOC_U8500 + config SND_SOC_U8500 + bool "Platform - U8500" +endchoice + +config SND_SOC_UX500_AB8500 + bool "Codec - AB8500" + depends on SND_SOC_UX500 && UX500_SOC_DB8500 && AB8500_CORE && AB8500_GPADC + select SND_SOC_AB8500 + default n + help + Select this to enable support for AB8500 in the Ux500 machine-driver. + +config SND_SOC_UX500_DEBUG + bool "Activate Ux500 platform debug-mode (pr_debug)" + depends on SND_SOC_UX500 + default n + help + Say Y if you want to add debug level prints for Ux500 code-files. diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile new file mode 100644 index 0000000..eeb86f6 --- /dev/null +++ b/sound/soc/ux500/Makefile @@ -0,0 +1,25 @@ +# Ux500 Platform Support + +ifdef CONFIG_SND_SOC_UX500_DEBUG +CFLAGS_u8500.o := -DDEBUG +CFLAGS_ux500_pcm.o := -DDEBUG +CFLAGS_ux500_msp_dai.o := -DDEBUG +CFLAGS_ux500_ab8500.o := -DDEBUG +CFLAGS_ux500_msp_i2s.o := -DDEBUG +endif + +ifdef CONFIG_UX500_SOC_DB8500 +snd-soc-ux500-platform-objs := ux500_pcm.o ux500_msp_dai.o ux500_msp_i2s.o +obj-y += snd-soc-ux500-platform.o +endif + +ifdef CONFIG_SND_SOC_UX500_AB8500 +snd-soc-ux500-machine-objs += ux500_ab8500.o +endif + +obj-y += snd-soc-ux500-machine.o + +ifdef CONFIG_UX500_SOC_DB8500 +snd-soc-u8500-objs := u8500.o +obj-y += snd-soc-u8500.o +endif diff --git a/sound/soc/ux500/u8500.c b/sound/soc/ux500/u8500.c new file mode 100644 index 0000000..7a5c61d --- /dev/null +++ b/sound/soc/ux500/u8500.c @@ -0,0 +1,150 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/mach-types.h> + +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#ifdef CONFIG_SND_SOC_UX500_AB8500 +#include <sound/ux500_ab8500.h> +#endif + +static struct platform_device *u8500_platform_dev; + +/* Create dummy devices for platform drivers */ + +static struct platform_device ux500_pcm = { + .name = "ux500-pcm", + .id = 0, + .dev = { + .platform_data = NULL, + }, +}; + +/* Define the whole U8500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link u8500_dai_links[] = { + #ifdef CONFIG_SND_SOC_UX500_AB8500 + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = ux500_ab8500_machine_codec_init, + .ops = ux500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = ux500_ab8500_ops, + }, + #endif +}; + +static struct snd_soc_card u8500_drvdata = { + .name = "U8500-card", + .probe = NULL, + .dai_link = u8500_dai_links, + .num_links = ARRAY_SIZE(u8500_dai_links), +}; + +static int __init u8500_soc_init(void) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + #ifdef CONFIG_SND_SOC_UX500_AB8500 + pr_debug("%s: Calling init-function for AB8500 machine driver.\n", + __func__); + ret = ux500_ab8500_soc_machine_drv_init(); + if (ret) + pr_err("%s: ux500_ab8500_soc_machine_drv_init failed (%d).\n", + __func__, ret); + #endif + + pr_debug("%s: Register device to generate a probe for Ux500-pcm platform.\n", + __func__); + platform_device_register(&ux500_pcm); + + pr_debug("%s: Allocate platform device 'soc-audio'.\n", + __func__); + u8500_platform_dev = platform_device_alloc("soc-audio", -1); + if (!u8500_platform_dev) + return -ENOMEM; + + pr_debug("%s: Card %s: num_links = %d\n", + __func__, + u8500_drvdata.name, + u8500_drvdata.num_links); + pr_debug("%s: Card %s: DAI-link 0: name = %s\n", + __func__, + u8500_drvdata.name, + u8500_drvdata.dai_link[0].name); + pr_debug("%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, + u8500_drvdata.name, + u8500_drvdata.dai_link[0].stream_name); + + pr_debug("%s: Card %s: Set platform drvdata.\n", + __func__, + u8500_drvdata.name); + platform_set_drvdata(u8500_platform_dev, &u8500_drvdata); + u8500_drvdata.dev = &u8500_platform_dev->dev; + + pr_debug("%s: Card %s: Add platform device.\n", + __func__, + u8500_drvdata.name); + ret = platform_device_add(u8500_platform_dev); + if (ret) { + pr_err("%s: Error: Failed to add platform device (%s).\n", + __func__, + u8500_drvdata.name); + platform_device_put(u8500_platform_dev); + } + + return ret; +} + +static void __exit u8500_soc_exit(void) +{ + pr_debug("%s: Enter.\n", __func__); + + #ifdef CONFIG_SND_SOC_UX500_AB8500 + pr_debug("%s: Calling exit-function for AB8500 machine driver.\n", + __func__); + ux500_ab8500_soc_machine_drv_cleanup(); + #endif + + pr_debug("%s: Unregister platform device (%s).\n", + __func__, + u8500_drvdata.name); + platform_device_unregister(u8500_platform_dev); +} + +module_init(u8500_soc_init); +module_exit(u8500_soc_exit); + diff --git a/sound/soc/ux500/ux500_ab8500.c b/sound/soc/ux500/ux500_ab8500.c new file mode 100644 index 0000000..e507f93 --- /dev/null +++ b/sound/soc/ux500/ux500_ab8500.c @@ -0,0 +1,790 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Mikko J. Lehto <mikko.lehto(a)symbio.com>, + * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, + * Jarmo K. Kuronen <jarmo.kuronen(a)symbio.com>. + * Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <mach/hardware.h> +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500_audio.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +static struct snd_soc_jack jack; +static bool vibra_on; + +/* Power-control */ +static DEFINE_MUTEX(power_lock); +static int ab8500_power_count; + +/* Clocks */ +/* audioclk -> intclk -> sysclk/ulpclk */ +static int master_clock_sel; +static struct clk *clk_ptr_audioclk; +static struct clk *clk_ptr_intclk; +static struct clk *clk_ptr_sysclk; +static struct clk *clk_ptr_ulpclk; +static struct clk *clk_ptr_gpio1; + +/* Temporary solution until the Ux500 clock-framework is in place.*/ +#ifndef clk_set_parent + +int clock_nosupport(void) +{ + pr_warn("%s: WARNING: Clock-framework does not support " \ + "needed clocks!\n", __func__); + return 0; +} +#define CLK_SET_PARENT(x, y) clock_nosupport() +#define CLK_ENABLE(x) clock_nosupport() +#define CLK_DISABLE(x) clock_nosupport() +#define CLK_GET(x, y) clock_nosupport() + +#else + +#define CLK_SET_PARENT(x, y) clk_set_parent(c, y) +#define CLK_ENABLE(x) clk_enable(x) +#define CLK_DISABLE(x) clk_disable(x) +#define CLK_GET(x, y) clk_get(x, y) + +#endif + +/* ANC States */ +static const char *enum_anc_state[] = { + "Unconfigured", + "Configure FIR+IIR", + "FIR+IIR Configured", + "Configure FIR", + "FIR Configured", + "Configure IIR", + "IIR Configured", + "Error" +}; +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state); + +/* Regulators */ +enum regulator_idx { + REGULATOR_AUDIO, + REGULATOR_DMIC, + REGULATOR_AMIC1, + REGULATOR_AMIC2 +}; +static struct regulator_bulk_data reg_info[4] = { + { .supply = "vaud" }, + { .supply = "vdmic" }, + { .supply = "vamic1" }, + { .supply = "vamic2" }, +}; +static bool reg_enabled[4] = { + false, + false, + false, + false +}; + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Regulators */ + +static int enable_regulator(enum regulator_idx idx) +{ + int ret; + + if (reg_enabled[idx]) + return 0; + + ret = regulator_enable(reg_info[idx].consumer); + if (ret != 0) { + pr_err("%s: Failure to enable regulator '%s' (ret = %d)\n", + __func__, reg_info[idx].supply, ret); + return ret; + }; + + reg_enabled[idx] = true; + pr_debug("%s: Enabled regulator '%s', status: %d, %d, %d, %d\n", + __func__, + reg_info[idx].supply, + (int)reg_enabled[0], + (int)reg_enabled[1], + (int)reg_enabled[2], + (int)reg_enabled[3]); + return 0; +} + +static void disable_regulator(enum regulator_idx idx) +{ + if (!reg_enabled[idx]) + return; + + regulator_disable(reg_info[idx].consumer); + + reg_enabled[idx] = false; + pr_debug("%s: Disabled regulator '%s', status: %d, %d, %d, %d\n", + __func__, + reg_info[idx].supply, + (int)reg_enabled[0], + (int)reg_enabled[1], + (int)reg_enabled[2], + (int)reg_enabled[3]); +} + +static int create_regulators(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int i, status = 0; + + pr_debug("%s: Enter.\n", __func__); + + for (i = 0; i < ARRAY_SIZE(reg_info); ++i) { + reg_info[i].consumer = regulator_get(codec->dev, reg_info[i].supply); + if (IS_ERR(reg_info[i].consumer)) { + status = PTR_ERR(reg_info[i].consumer); + pr_err("%s: ERROR: Failed to get regulator '%s' (ret = %d)!\n", + __func__, reg_info[i].supply, status); + reg_info[i].consumer = NULL; + goto err_get; + } + } + + return 0; + +err_get: + + for (i = 0; i < ARRAY_SIZE(reg_info); ++i) { + if (reg_info[i].consumer) { + regulator_put(reg_info[i].consumer); + reg_info[i].consumer = NULL; + } + } + + return status; +} + +/* Power/clock control */ + +static int ux500_ab8500_power_control_inc(void) +{ + int ret = 0; + + mutex_lock(&power_lock); + + ab8500_power_count++; + pr_debug("%s: ab8500_power_count changed from %d to %d", + __func__, + ab8500_power_count-1, + ab8500_power_count); + + if (ab8500_power_count == 1) { + /* Turn on audio-regulator */ + ret = enable_regulator(REGULATOR_AUDIO); + + if (ret) + goto out; + + /* Enable audio-clock */ + ret = CLK_SET_PARENT(clk_ptr_intclk, + (master_clock_sel == 0) ? clk_ptr_sysclk : clk_ptr_ulpclk); + if (ret) { + pr_err("%s: ERROR: Setting master-clock to %s failed (ret = %d)!", + __func__, + (master_clock_sel == 0) ? "SYSCLK" : "ULPCLK", + ret); + goto clk_err; + } + pr_debug("%s: Enabling master-clock (%s).", + __func__, + (master_clock_sel == 0) ? "SYSCLK" : "ULPCLK"); + ret = CLK_ENABLE(clk_ptr_audioclk); + if (ret) { + pr_err("%s: ERROR: clk_enable failed (ret = %d)!", __func__, ret); + ab8500_power_count = 0; + goto clk_err; + } + + /* Power on audio-parts of AB8500 */ + ab8500_audio_power_control(true); + } + + goto out; + +clk_err: + disable_regulator(REGULATOR_AUDIO); + +out: + mutex_unlock(&power_lock); + + return ret; +} + +static void ux500_ab8500_power_control_dec(void) +{ + mutex_lock(&power_lock); + + ab8500_power_count--; + + pr_debug("%s: ab8500_power_count changed from %d to %d", + __func__, + ab8500_power_count+1, + ab8500_power_count); + + if (ab8500_power_count == 0) { + /* Power off audio-parts of AB8500 */ + ab8500_audio_power_control(false); + + /* Disable audio-clock */ + pr_debug("%s: Disabling master-clock (%s).", + __func__, + (master_clock_sel == 0) ? "SYSCLK" : "ULPCLK"); + CLK_DISABLE(clk_ptr_audioclk); + + /* Turn off audio-regulator */ + disable_regulator(REGULATOR_AUDIO); + } + + mutex_unlock(&power_lock); +} + +/* Controls - Non-DAPM Non-ASoC */ + +static int mclk_input_control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item) { + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, "ULPCLK"); + } else { + strcpy(uinfo->value.enumerated.name, "SYSCLK"); + } + return 0; +} + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = master_clock_sel; + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int val; + + val = (ucontrol->value.enumerated.item[0] != 0); + if (master_clock_sel == val) + return 0; + + master_clock_sel = val; + + return 1; +} + +static const struct snd_kcontrol_new mclk_input_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Clock Select", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = mclk_input_control_info, + .get = mclk_input_control_get, + .put = mclk_input_control_put, + .private_value = 1 /* ULPCLK */ +}; + +static int anc_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = ab8500_audio_anc_status(); + + return 0; +} + +static int anc_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret; + int req_state = ucontrol->value.integer.value[0]; + + ret = ux500_ab8500_power_control_inc(); + if (ret) + goto cleanup; + + ret = ab8500_audio_anc_configure(req_state); + + ux500_ab8500_power_control_dec(); + +cleanup: + if (ret) { + pr_err("%s: Unable to configure ANC! (ret = %d)\n", + __func__, ret); + return 0; + } + + return 1; +} + +static const struct snd_kcontrol_new anc_status_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ANC Status", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_enum_ext, + .get = anc_status_control_get, .put = anc_status_control_put, + .private_value = (unsigned long) &soc_enum_ancstate +}; + +/* DAPM-events */ + +static int dapm_audioreg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + ux500_ab8500_power_control_inc(); + else + ux500_ab8500_power_control_dec(); + + return 0; +} + + +static int dapm_mic1reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = enable_regulator(REGULATOR_AMIC1); + else + disable_regulator(REGULATOR_AMIC1); + + return ret; +} + +static int dapm_mic2reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = enable_regulator(REGULATOR_AMIC2); + else + disable_regulator(REGULATOR_AMIC2); + + return ret; +} + +static int dapm_dmicreg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = enable_regulator(REGULATOR_DMIC); + else + disable_regulator(REGULATOR_DMIC); + + return ret; +} + +/* DAPM-widgets */ + +static const struct snd_soc_dapm_widget ux500_ab8500_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("AUDIO Regulator", SND_SOC_NOPM, 0, 0, dapm_audioreg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AMIC1 Regulator", SND_SOC_NOPM, 0, 0, dapm_mic1reg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AMIC2 Regulator", SND_SOC_NOPM, 0, 0, dapm_mic2reg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC Regulator", SND_SOC_NOPM, 0, 0, dapm_dmicreg_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +/* DAPM-routes */ + +static const struct snd_soc_dapm_route ux500_ab8500_dapm_intercon[] = { + + /* Power AB8500 audio-block when AD/DA is active */ + {"DAC", NULL, "AUDIO Regulator"}, + {"ADC", NULL, "AUDIO Regulator"}, + + /* Power AMIC1-regulator when MIC1 is enabled */ + {"MIC1 Enable", NULL, "AMIC1 Regulator"}, + + /* Power AMIC2-regulator when MIC1 is enabled */ + {"MIC2 Enable", NULL, "AMIC2 Regulator"}, + + /* Power DMIC-regulator when any digital mic is enabled */ + {"DMic 1", NULL, "DMIC Regulator"}, + {"DMic 2", NULL, "DMIC Regulator"}, + {"DMic 3", NULL, "DMIC Regulator"}, + {"DMic 4", NULL, "DMIC Regulator"}, + {"DMic 5", NULL, "DMIC Regulator"}, + {"DMic 6", NULL, "DMIC Regulator"}, +}; + + +static int add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(&codec->dapm, + ux500_ab8500_dapm_widgets, + ARRAY_SIZE(ux500_ab8500_dapm_widgets)); + if (ret < 0) { + pr_err("%s: Failed to create DAPM controls (%d).\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&codec->dapm, + ux500_ab8500_dapm_intercon, + ARRAY_SIZE(ux500_ab8500_dapm_intercon)); + if (ret < 0) { + pr_err("%s: Failed to add DAPM routes (%d).\n", + __func__, ret); + return ret; + } + + return 0; +} + +/* ASoC */ + +int ux500_ab8500_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; + + pr_debug("%s: Enter\n", __func__); + + /* Enable gpio.1-clock (needed by DSP in burst mode) */ + ret = CLK_ENABLE(clk_ptr_gpio1); + if (ret) { + pr_err("%s: ERROR: CLK_ENABLE(gpio.1) failed (ret = %d)!", __func__, ret); + return ret; + } + + return 0; +} + +void ux500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + pr_debug("%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; + + CLK_DISABLE(clk_ptr_gpio1); +} + +int ux500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, fmt_if1; + int channels, ret = 0, slots, slot_width, driver_mode; + bool streamIsPlayback; + + pr_debug("%s: Enter\n", __func__); + + pr_debug("%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + /* Setup codec depending on driver-mode */ + driver_mode = (channels == 8) ? + DRIVERMODE_CODEC_ONLY : DRIVERMODE_NORMAL; + pr_debug("%s: Driver-mode: %s.\n", + __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + ab8500_audio_set_bit_delay(codec_dai, 1); + + if (driver_mode == DRIVERMODE_NORMAL) { + ab8500_audio_set_word_length(codec_dai, 16); + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + ab8500_audio_set_word_length(codec_dai, 20); + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err("%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, + ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err("%s: ERROR: snd_soc_dai_set_fmt for cpu_dai (ret = %d)!\n", + __func__, + ret); + return ret; + } + + ux500_msp_dai_set_data_delay(cpu_dai, MSP_DELAY_1); + + /* Setup TDM-slots */ + + streamIsPlayback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + slot_width = 16; + tx_slots = (streamIsPlayback) ? TX_SLOT_MONO : 0; + rx_slots = (streamIsPlayback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + slot_width = 16; + tx_slots = (streamIsPlayback) ? TX_SLOT_STEREO : 0; + rx_slots = (streamIsPlayback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + slot_width = 16; + tx_slots = (streamIsPlayback) ? TX_SLOT_8CH : 0; + rx_slots = (streamIsPlayback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + pr_debug("%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", + __func__, tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, slot_width); + if (ret) + return ret; + + pr_debug("%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", + __func__, tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, slot_width); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_debug("%s: Setup IF1 for FM-radio.\n", __func__); + fmt_if1 = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_I2S; + ret = ab8500_audio_setup_if1(codec_dai->codec, fmt_if1, 16, 1); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_ops ux500_ab8500_ops[] = { + { + .hw_params = ux500_ab8500_hw_params, + .startup = ux500_ab8500_startup, + .shutdown = ux500_ab8500_shutdown, + } +}; + +int ux500_ab8500_machine_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + pr_debug("%s Enter.\n", __func__); + + ret = snd_soc_jack_new(codec, + "AB8500 Hs Status", + SND_JACK_HEADPHONE | + SND_JACK_MICROPHONE | + SND_JACK_HEADSET | + SND_JACK_LINEOUT | + SND_JACK_MECHANICAL | + SND_JACK_VIDEOOUT, + &jack); + if (ret < 0) { + pr_err("%s: ERROR: Failed to create Jack (ret = %d)!\n", __func__, ret); + return ret; + } + + /* Add controls */ + snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&mclk_input_control, codec)); + snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&anc_status_control, codec)); + + ret = create_regulators(rtd); + if (ret < 0) { + pr_err("%s: ERROR: Failed to instantiate regulators (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Get references to clock-nodes */ + clk_ptr_sysclk = NULL; + clk_ptr_ulpclk = NULL; + clk_ptr_intclk = NULL; + clk_ptr_audioclk = NULL; + clk_ptr_gpio1 = NULL; + clk_ptr_sysclk = CLK_GET(codec->dev, "sysclk"); + if (IS_ERR(clk_ptr_sysclk)) { + pr_err("ERROR: clk_get failed (ret = %d)!", -EFAULT); + return -EFAULT; + } + clk_ptr_ulpclk = CLK_GET(codec->dev, "ulpclk"); + if (IS_ERR(clk_ptr_sysclk)) { + pr_err("ERROR: clk_get failed (ret = %d)!", -EFAULT); + return -EFAULT; + } + clk_ptr_intclk = CLK_GET(codec->dev, "intclk"); + if (IS_ERR(clk_ptr_audioclk)) { + pr_err("ERROR: clk_get failed (ret = %d)!", -EFAULT); + return -EFAULT; + } + clk_ptr_audioclk = CLK_GET(codec->dev, "audioclk"); + if (IS_ERR(clk_ptr_audioclk)) { + pr_err("ERROR: clk_get failed (ret = %d)!", -EFAULT); + return -EFAULT; + } + clk_ptr_gpio1 = clk_get_sys("gpio.1", NULL); + if (IS_ERR(clk_ptr_gpio1)) { + pr_err("ERROR: clk_get_sys(gpio.1) failed (ret = %d)!", -EFAULT); + return -EFAULT; + } + + /* Set intclk default parent to ulpclk */ + ret = CLK_SET_PARENT(clk_ptr_intclk, clk_ptr_ulpclk); + if (ret) { + pr_err("%s: ERROR: Setting intclk parent to ulpclk failed (ret = %d)!", + __func__, + ret); + return -EFAULT; + } + + master_clock_sel = 1; + + ab8500_power_count = 0; + + /* Add DAPM-widgets */ + ret = add_widgets(codec); + if (ret < 0) { + pr_err("%s: Failed add widgets (%d).\n", __func__, ret); + return ret; + } + + return 0; +} + +int ux500_ab8500_soc_machine_drv_init(void) +{ + pr_debug("%s: Enter.\n", __func__); + + vibra_on = false; + + return 0; +} + +void ux500_ab8500_soc_machine_drv_cleanup(void) +{ + pr_debug("%s: Enter.\n", __func__); + + regulator_bulk_free(ARRAY_SIZE(reg_info), reg_info); + + if (clk_ptr_sysclk != NULL) + clk_put(clk_ptr_sysclk); + if (clk_ptr_ulpclk != NULL) + clk_put(clk_ptr_ulpclk); + if (clk_ptr_intclk != NULL) + clk_put(clk_ptr_intclk); + if (clk_ptr_audioclk != NULL) + clk_put(clk_ptr_audioclk); + if (clk_ptr_gpio1 != NULL) + clk_put(clk_ptr_gpio1); +} + +/* Extended interface */ + +void ux500_ab8500_audio_pwm_vibra(unsigned char speed_left_pos, + unsigned char speed_left_neg, + unsigned char speed_right_pos, + unsigned char speed_right_neg) +{ + bool vibra_on_new; + + vibra_on_new = speed_left_pos | speed_left_neg | speed_right_pos | speed_right_neg; + if ((!vibra_on_new) && (vibra_on)) { + pr_debug("%s: PWM-vibra off.\n", __func__); + vibra_on = false; + + ux500_ab8500_power_control_dec(); + } + + if ((vibra_on_new) && (!vibra_on)) { + pr_debug("%s: PWM-vibra on.\n", __func__); + vibra_on = true; + + ux500_ab8500_power_control_inc(); + } + + ab8500_audio_pwm_vibra(speed_left_pos, + speed_left_neg, + speed_right_pos, + speed_right_neg); +} + +void ux500_ab8500_jack_report(int value) +{ + if (jack.jack) + snd_soc_jack_report(&jack, value, 0xFF); +} +EXPORT_SYMBOL(ux500_ab8500_jack_report); + diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c new file mode 100644 index 0000000..bc1a7ea --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -0,0 +1,998 @@ +/* + * Copyright (C) ST-Ericsson SA 2011 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> + +#include <mach/hardware.h> +#include <mach/msp.h> + +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "ux500_msp_i2s.h" +#include "ux500_msp_dai.h" +#include "ux500_pcm.h" + +static struct ux500_platform_drvdata platform_drvdata[UX500_NBR_OF_DAI] = { + { + .msp_i2s_drvdata = NULL, + .fmt = 0, + .slots = 1, + .tx_mask = 0x01, + .rx_mask = 0x01, + .slot_width = 16, + .playback_active = false, + .capture_active = false, + .configured = 0, + .data_delay = MSP_DELAY_0, + .master_clk = UX500_MSP_INTERNAL_CLOCK_FREQ, + }, + { + .msp_i2s_drvdata = NULL, + .fmt = 0, + .slots = 1, + .tx_mask = 0x01, + .rx_mask = 0x01, + .slot_width = 16, + .playback_active = false, + .capture_active = false, + .configured = 0, + .data_delay = MSP_DELAY_0, + .master_clk = UX500_MSP1_INTERNAL_CLOCK_FREQ, + }, + { + .msp_i2s_drvdata = NULL, + .fmt = 0, + .slots = 1, + .tx_mask = 0x01, + .rx_mask = 0x01, + .slot_width = 16, + .playback_active = false, + .capture_active = false, + .configured = 0, + .data_delay = MSP_DELAY_0, + .master_clk = UX500_MSP_INTERNAL_CLOCK_FREQ, + }, + { + .msp_i2s_drvdata = NULL, + .fmt = 0, + .slots = 1, + .tx_mask = 0x01, + .rx_mask = 0x01, + .slot_width = 16, + .playback_active = false, + .capture_active = false, + .configured = 0, + .data_delay = MSP_DELAY_0, + .master_clk = UX500_MSP1_INTERNAL_CLOCK_FREQ, + }, +}; + +bool ux500_msp_dai_i2s_get_underrun_status(int dai_idx) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai_idx]; + int status = ux500_msp_i2s_hw_status(drvdata->msp_i2s_drvdata); + return (bool)(status & TRANSMIT_UNDERRUN_ERR_INT); +} + +int ux500_msp_dai_i2s_configure_sg(dma_addr_t dma_addr, + int period_cnt, + size_t period_len, + int dai_idx, + int stream_id) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai_idx]; + struct i2s_message message; + int ret = 0; + bool playback_req_valid = + (drvdata->playback_active && + stream_id == SNDRV_PCM_STREAM_PLAYBACK); + bool capture_req_valid = + (drvdata->capture_active && + stream_id == SNDRV_PCM_STREAM_CAPTURE); + + pr_debug("%s: Enter (MSP Index: %u, period-cnt: %u, period-len: %u).\n", + __func__, + dai_idx, + period_cnt, + period_len); + + if (!playback_req_valid && !capture_req_valid) { + pr_err("%s: The I2S controller is not available." + "MSP index:%d\n", + __func__, + dai_idx); + return ret; + } + + message.i2s_direction = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + I2S_DIRECTION_TX : + I2S_DIRECTION_RX; + message.buf_addr = dma_addr; + message.buf_len = period_cnt * period_len; + message.period_len = period_len; + + ret = ux500_msp_i2s_transfer(drvdata->msp_i2s_drvdata, &message); + if (ret < 0) { + pr_err("%s: Error: i2s_transfer failed. MSP index: %d\n", + __func__, + dai_idx); + } + + return ret; +} + +static const char *stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + +static int ux500_msp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + bool mode_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + pr_debug("%s: MSP %d (%s): Enter.\n", __func__, dai->id, stream_str(substream)); + + if ((mode_playback && drvdata->playback_active) || + (!mode_playback && drvdata->capture_active)) { + pr_err("%s: Error: MSP %d (%s): Stream already active.\n", + __func__, + dai->id, + stream_str(substream)); + return -EBUSY; + } + + if (mode_playback) + drvdata->playback_active = true; + else + drvdata->capture_active = true; + + return 0; +} + +static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + bool mode_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + pr_debug("%s: MSP %d (%s): Enter.\n", __func__, dai->id, stream_str(substream)); + + if (drvdata == NULL) + return; + + if (mode_playback) + drvdata->playback_active = false; + else + drvdata->capture_active = false; + + if (ux500_msp_i2s_close(drvdata->msp_i2s_drvdata, + mode_playback ? DISABLE_TRANSMIT : DISABLE_RECEIVE)) { + pr_err("%s: Error: MSP %d (%s): Unable to close i2s.\n", + __func__, + dai->id, + stream_str(substream)); + } + + if (mode_playback) + drvdata->configured &= ~PLAYBACK_CONFIGURED; + else + drvdata->configured &= ~CAPTURE_CONFIGURED; +} + +static void ux500_msp_dai_setup_multichannel(struct ux500_platform_drvdata *private, + struct msp_config *msp_config) +{ + struct msp_multichannel_config *multi = &msp_config->multichannel_config; + + if (private->slots > 1) { + msp_config->multichannel_configured = 1; + + multi->tx_multichannel_enable = true; + multi->rx_multichannel_enable = true; + multi->rx_comparison_enable_mode = MSP_COMPARISON_DISABLED; + + multi->tx_channel_0_enable = private->tx_mask; + multi->tx_channel_1_enable = 0; + multi->tx_channel_2_enable = 0; + multi->tx_channel_3_enable = 0; + + multi->rx_channel_0_enable = private->rx_mask; + multi->rx_channel_1_enable = 0; + multi->rx_channel_2_enable = 0; + multi->rx_channel_3_enable = 0; + + pr_debug("%s: Multichannel enabled." + "Slots: %d TX: %u RX: %u\n", + __func__, + private->slots, + multi->tx_channel_0_enable, + multi->rx_channel_0_enable); + } +} + +static void ux500_msp_dai_setup_frameper(struct ux500_platform_drvdata *private, + unsigned int rate, + struct msp_protocol_desc *prot_desc) +{ + switch (private->slots) { + default: + case 1: + switch (rate) { + case 8000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_8_KHZ; + break; + case 16000: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_16_KHZ; + break; + case 44100: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_44_1_KHZ; + break; + case 48000: + default: + prot_desc->frame_period = + FRAME_PER_SINGLE_SLOT_48_KHZ; + break; + } + break; + + case 2: + prot_desc->frame_period = FRAME_PER_2_SLOTS; + break; + + case 8: + prot_desc->frame_period = + FRAME_PER_8_SLOTS; + break; + + case 16: + prot_desc->frame_period = + FRAME_PER_16_SLOTS; + break; + } + + prot_desc->total_clocks_for_one_frame = + prot_desc->frame_period+1; + + pr_debug("%s: Total clocks per frame: %u\n", + __func__, + prot_desc->total_clocks_for_one_frame); +} + +static void ux500_msp_dai_setup_framing_pcm(struct ux500_platform_drvdata *private, + unsigned int rate, + struct msp_protocol_desc *prot_desc) +{ + u32 frame_length = MSP_FRAME_LENGTH_1; + prot_desc->frame_width = 0; + + switch (private->slots) { + default: + case 1: + frame_length = MSP_FRAME_LENGTH_1; + break; + + case 2: + frame_length = MSP_FRAME_LENGTH_2; + break; + + case 8: + frame_length = MSP_FRAME_LENGTH_8; + break; + + case 16: + frame_length = MSP_FRAME_LENGTH_16; + break; + } + + prot_desc->tx_frame_length_1 = frame_length; + prot_desc->rx_frame_length_1 = frame_length; + prot_desc->tx_frame_length_2 = frame_length; + prot_desc->rx_frame_length_2 = frame_length; + + prot_desc->tx_element_length_1 = MSP_ELEM_LENGTH_16; + prot_desc->rx_element_length_1 = MSP_ELEM_LENGTH_16; + prot_desc->tx_element_length_2 = MSP_ELEM_LENGTH_16; + prot_desc->rx_element_length_2 = MSP_ELEM_LENGTH_16; + + ux500_msp_dai_setup_frameper(private, rate, prot_desc); +} + +static void ux500_msp_dai_setup_clocking(unsigned int fmt, + struct msp_config *msp_config) +{ + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + default: + case SND_SOC_DAIFMT_NB_NF: + break; + + case SND_SOC_DAIFMT_NB_IF: + msp_config->tx_frame_sync_pol ^= 1 << TFSPOL_SHIFT; + msp_config->rx_frame_sync_pol ^= 1 << RFSPOL_SHIFT; + break; + } + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) { + pr_debug("%s: Codec is MASTER.\n", + __func__); + msp_config->iodelay = 0x20; + msp_config->rx_frame_sync_sel = 0; + msp_config->tx_frame_sync_sel = 1 << TFSSEL_SHIFT; + msp_config->tx_clock_sel = 0; + msp_config->rx_clock_sel = 0; + msp_config->srg_clock_sel = 0x2 << SCKSEL_SHIFT; + + } else { + pr_debug("%s: Codec is SLAVE.\n", + __func__); + + msp_config->tx_clock_sel = TX_CLK_SEL_SRG; + msp_config->tx_frame_sync_sel = TX_SYNC_SRG_PROG; + msp_config->rx_clock_sel = RX_CLK_SEL_SRG; + msp_config->rx_frame_sync_sel = RX_SYNC_SRG; + msp_config->srg_clock_sel = 1 << SCKSEL_SHIFT; + } +} + +static void ux500_msp_dai_compile_prot_desc_pcm(unsigned int fmt, + struct msp_protocol_desc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->tx_phase_mode = MSP_SINGLE_PHASE; + prot_desc->rx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->tx_phase2_start_mode = MSP_PHASE2_START_MODE_IMEDIATE; + prot_desc->rx_bit_transfer_format = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_bit_transfer_format = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_frame_sync_pol = MSP_FRAME_SYNC_POL(MSP_FRAME_SYNC_POL_ACTIVE_HIGH); + prot_desc->rx_frame_sync_pol = MSP_FRAME_SYNC_POL_ACTIVE_HIGH << RFSPOL_SHIFT; + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_A) { + pr_debug("%s: DSP_A.\n", + __func__); + prot_desc->rx_clock_pol = MSP_RISING_EDGE; + prot_desc->tx_clock_pol = MSP_FALLING_EDGE; + } else { + pr_debug("%s: DSP_B.\n", + __func__); + prot_desc->rx_clock_pol = MSP_FALLING_EDGE; + prot_desc->tx_clock_pol = MSP_RISING_EDGE; + } + + prot_desc->rx_half_word_swap = MSP_HWS_NO_SWAP; + prot_desc->tx_half_word_swap = MSP_HWS_NO_SWAP; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->spi_clk_mode = MSP_SPI_CLOCK_MODE_NON_SPI; + prot_desc->spi_burst_mode = MSP_SPI_BURST_MODE_DISABLE; + prot_desc->frame_sync_ignore = MSP_FRAME_SYNC_IGNORE; +} + +static void ux500_msp_dai_compile_prot_desc_i2s(struct msp_protocol_desc *prot_desc) +{ + prot_desc->rx_phase_mode = MSP_DUAL_PHASE; + prot_desc->tx_phase_mode = MSP_DUAL_PHASE; + prot_desc->rx_phase2_start_mode = + MSP_PHASE2_START_MODE_FRAME_SYNC; + prot_desc->tx_phase2_start_mode = + MSP_PHASE2_START_MODE_FRAME_SYNC; + prot_desc->rx_bit_transfer_format = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_bit_transfer_format = MSP_BTF_MS_BIT_FIRST; + prot_desc->tx_frame_sync_pol = MSP_FRAME_SYNC_POL(MSP_FRAME_SYNC_POL_ACTIVE_LOW); + prot_desc->rx_frame_sync_pol = MSP_FRAME_SYNC_POL_ACTIVE_LOW << RFSPOL_SHIFT; + + prot_desc->rx_frame_length_1 = MSP_FRAME_LENGTH_1; + prot_desc->rx_frame_length_2 = MSP_FRAME_LENGTH_1; + prot_desc->tx_frame_length_1 = MSP_FRAME_LENGTH_1; + prot_desc->tx_frame_length_2 = MSP_FRAME_LENGTH_1; + prot_desc->rx_element_length_1 = MSP_ELEM_LENGTH_16; + prot_desc->rx_element_length_2 = MSP_ELEM_LENGTH_16; + prot_desc->tx_element_length_1 = MSP_ELEM_LENGTH_16; + prot_desc->tx_element_length_2 = MSP_ELEM_LENGTH_16; + + prot_desc->rx_clock_pol = MSP_RISING_EDGE; + prot_desc->tx_clock_pol = MSP_FALLING_EDGE; + + prot_desc->tx_half_word_swap = MSP_HWS_NO_SWAP; + prot_desc->rx_half_word_swap = MSP_HWS_NO_SWAP; + prot_desc->compression_mode = MSP_COMPRESS_MODE_LINEAR; + prot_desc->expansion_mode = MSP_EXPAND_MODE_LINEAR; + prot_desc->spi_clk_mode = MSP_SPI_CLOCK_MODE_NON_SPI; + prot_desc->spi_burst_mode = MSP_SPI_BURST_MODE_DISABLE; + prot_desc->frame_sync_ignore = MSP_FRAME_SYNC_IGNORE; +} + +static void ux500_msp_dai_compile_msp_config(struct snd_pcm_substream *substream, + struct ux500_platform_drvdata *private, + unsigned int rate, + struct msp_config *msp_config) +{ + struct msp_protocol_desc *prot_desc = &msp_config->protocol_desc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int fmt = private->fmt; + + memset(msp_config, 0, sizeof(*msp_config)); + + msp_config->input_clock_freq = private->master_clk; + + msp_config->tx_fifo_config = TX_FIFO_ENABLE; + msp_config->rx_fifo_config = RX_FIFO_ENABLE; + msp_config->spi_clk_mode = SPI_CLK_MODE_NORMAL; + msp_config->spi_burst_mode = 0; + msp_config->handler = ux500_pcm_dma_eot_handler; + msp_config->tx_callback_data = + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + substream : NULL; + msp_config->rx_callback_data = + substream->stream == SNDRV_PCM_STREAM_CAPTURE ? + substream : NULL; + msp_config->def_elem_len = 1; + msp_config->direction = + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + MSP_TRANSMIT_MODE : MSP_RECEIVE_MODE; + msp_config->data_size = MSP_DATA_BITS_32; + msp_config->work_mode = MSP_DMA_MODE; + msp_config->frame_freq = rate; + + pr_debug("%s: input_clock_freq = %u, frame_freq = %u.\n", + __func__, msp_config->input_clock_freq, msp_config->frame_freq); + /* To avoid division by zero in I2S-driver (i2s_setup) */ + prot_desc->total_clocks_for_one_frame = 1; + + prot_desc->rx_data_delay = private->data_delay; + prot_desc->tx_data_delay = private->data_delay; + + pr_debug("%s: rate: %u channels: %d.\n", + __func__, + rate, + runtime->channels); + switch (fmt & + (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + pr_debug("%s: SND_SOC_DAIFMT_I2S.\n", + __func__); + + msp_config->default_protocol_desc = 1; + msp_config->protocol = MSP_I2S_PROTOCOL; + break; + + default: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + pr_debug("%s: SND_SOC_DAIFMT_I2S.\n", + __func__); + + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_I2S_PROTOCOL; + + ux500_msp_dai_compile_prot_desc_i2s(prot_desc); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + pr_debug("%s: PCM format.\n", + __func__); + msp_config->data_size = MSP_DATA_BITS_16; + msp_config->protocol = MSP_PCM_PROTOCOL; + + ux500_msp_dai_compile_prot_desc_pcm(fmt, prot_desc); + ux500_msp_dai_setup_multichannel(private, msp_config); + ux500_msp_dai_setup_framing_pcm(private, rate, prot_desc); + break; + } + + ux500_msp_dai_setup_clocking(fmt, msp_config); +} + +static int ux500_msp_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + struct snd_pcm_runtime *runtime = substream->runtime; + struct msp_config msp_config; + bool mode_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + pr_debug("%s: MSP %d (%s): Enter.\n", __func__, dai->id, stream_str(substream)); + + /* If already configured -> not errors reported */ + if (mode_playback) { + if ((drvdata->configured & PLAYBACK_CONFIGURED) && + (drvdata->playback_active)) + goto cleanup; + } else { + if ((drvdata->configured & CAPTURE_CONFIGURED) && + (drvdata->capture_active)) + goto cleanup; + } + + pr_debug("%s: Setup dai (Rate: %u).\n", __func__, runtime->rate); + ux500_msp_dai_compile_msp_config(substream, + drvdata, + runtime->rate, + &msp_config); + + ret = ux500_msp_i2s_open(drvdata->msp_i2s_drvdata, &msp_config); + if (ret < 0) { + pr_err("%s: Error: msp_setup failed (ret = %d)!\n", __func__, ret); + goto cleanup; + } + + drvdata->configured |= mode_playback ? + PLAYBACK_CONFIGURED : CAPTURE_CONFIGURED; + +cleanup: + return ret; +} + +static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + unsigned int mask, slots_active; + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + + pr_debug("%s: MSP %d (%s): Enter.\n", + __func__, + dai->id, + stream_str(substream)); + + switch (drvdata->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + if (params_channels(params) != 2) { + pr_err("%s: Error: I2S requires channels = 2 " + "(channels = %d)!\n", + __func__, + params_channels(params)); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_DSP_A: + + mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + drvdata->tx_mask : + drvdata->rx_mask; + + slots_active = hweight32(mask); + + pr_debug("TDM slots active: %d", slots_active); + + if (params_channels(params) != slots_active) { + pr_err("%s: Error: PCM TDM format requires channels " + "to match active slots " + "(channels = %d, active slots = %d)!\n", + __func__, + params_channels(params), + slots_active); + return -EINVAL; + } + break; + + default: + break; + } + + return 0; +} + +int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + + pr_debug("%s: MSP %d: Enter.\n", __func__, dai->id); + + switch (delay) { + case MSP_DELAY_0: + case MSP_DELAY_1: + case MSP_DELAY_2: + case MSP_DELAY_3: + break; + default: + goto unsupported_delay; + } + + drvdata->data_delay = delay; + return 0; + +unsupported_delay: + pr_err("%s: MSP %d: Error: Unsupported DAI delay (%d)!\n", + __func__, + dai->id, + delay); + return -EINVAL; +} + +static int ux500_msp_dai_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + + pr_debug("%s: MSP %d: Enter.\n", __func__, dai->id); + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + break; + + default: + goto unsupported_format; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + break; + + default: + goto unsupported_format; + } + + drvdata->fmt = fmt; + return 0; + +unsupported_format: + pr_err("%s: MSP %d: Error: Unsupported DAI format (0x%x)!\n", + __func__, + dai->id, + fmt); + return -EINVAL; +} + +static int ux500_msp_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, + int slot_width) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + unsigned int cap; + + if (!(slots == 1 || slots == 2 || slots == 8 || slots == 16)) { + pr_err("%s: Error: Unsupported slots (%d)! " + "Supported values are 1/2/8/16.\n", + __func__, + slots); + return -EINVAL; + } + drvdata->slots = slots; + + if (!(slot_width == 16)) { + pr_err("%s: Error: Unsupported slots_width (%d)!. " + "Supported value is 16.\n", + __func__, + slot_width); + return -EINVAL; + } + drvdata->slot_width = slot_width; + + switch (slots) { + default: + case 1: + cap = 0x01; + break; + case 2: + cap = 0x03; + break; + case 8: + cap = 0xFF; + break; + case 16: + cap = 0xFFFF; + break; + } + + drvdata->tx_mask = tx_mask & cap; + drvdata->rx_mask = rx_mask & cap; + + return 0; +} + +static int ux500_msp_dai_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, + unsigned int freq, + int dir) +{ + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + + pr_debug("%s: MSP %d: Enter. Clk id: %d, freq: %u.\n", + __func__, + dai->id, + clk_id, + freq); + + switch (clk_id) { + case UX500_MSP_MASTER_CLOCK: + drvdata->master_clk = freq; + break; + + default: + pr_err("%s: MSP %d: Invalid clkid: %d.\n", + __func__, + dai->id, + clk_id); + } + + return 0; +} + +static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, + int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct ux500_platform_drvdata *drvdata = &platform_drvdata[dai->id]; + + pr_debug("%s: MSP %d (%s): Enter (msp->id = %d, cmd = %d).\n", + __func__, + dai->id, + stream_str(substream), + (int)drvdata->msp_i2s_drvdata->id, + cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = 0; + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = 0; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = 0; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = 0; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = { + { + .name = "ux500-msp-i2s.0", + .id = 0, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } + }, + }, + { + .name = "ux500-msp-i2s.1", + .id = 1, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } + }, + }, + { + .name = "ux500-msp-i2s.2", + .id = 2, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } + }, + }, + { + .name = "ux500-msp-i2s.3", + .id = 3, + .suspend = NULL, + .resume = NULL, + .playback = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .capture = { + .channels_min = UX500_MSP_MIN_CHANNELS, + .channels_max = UX500_MSP_MAX_CHANNELS, + .rates = UX500_I2S_RATES, + .formats = UX500_I2S_FORMATS, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_sysclk = ux500_msp_dai_set_dai_sysclk, + .set_fmt = ux500_msp_dai_set_dai_fmt, + .set_tdm_slot = ux500_msp_dai_set_tdm_slot, + .startup = ux500_msp_dai_startup, + .shutdown = ux500_msp_dai_shutdown, + .prepare = ux500_msp_dai_prepare, + .trigger = ux500_msp_dai_trigger, + .hw_params = ux500_msp_dai_hw_params, + } + }, + }, +}; +EXPORT_SYMBOL(ux500_msp_dai_drv); + +static int ux500_msp_drv_probe(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata; + struct ux500_platform_drvdata *drvdata; + struct msp_i2s_platform_data *platform_data; + int id; + int ret = 0; + + pr_err("%s: Enter (pdev->name = %s).\n", __func__, pdev->name); + + platform_data = (struct msp_i2s_platform_data *)pdev->dev.platform_data; + msp_i2s_drvdata = ux500_msp_i2s_init(pdev, platform_data); + if (!msp_i2s_drvdata) { + pr_err("%s: ERROR: ux500_msp_i2s_init failed!", __func__); + return -1; + } + + id = msp_i2s_drvdata->id; + drvdata = &platform_drvdata[id]; + drvdata->msp_i2s_drvdata = msp_i2s_drvdata; + + pr_info("%s: Registering ux500-msp-dai SoC CPU-DAI.\n", __func__); + ret = snd_soc_register_dai(&pdev->dev, &ux500_msp_dai_drv[id]); + if (ret < 0) { + pr_err("Error: %s: Failed to register MSP %d.\n", __func__, id); + return ret; + } + + return ret; +} + +static int ux500_msp_drv_remove(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata = dev_get_drvdata(&pdev->dev); + struct ux500_platform_drvdata *drvdata = &platform_drvdata[msp_i2s_drvdata->id]; + + pr_info("%s: Unregister ux500-msp-dai ASoC CPU-DAI.\n", __func__); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); + + ux500_msp_i2s_exit(msp_i2s_drvdata); + drvdata->msp_i2s_drvdata = NULL; + + return 0; +} + +int ux500_msp_drv_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata = dev_get_drvdata(&pdev->dev); + + pr_debug("%s: Enter (pdev->name = %s).\n", __func__, pdev->name); + + return ux500_msp_i2s_suspend(msp_i2s_drvdata); +} + +int ux500_msp_drv_resume(struct platform_device *pdev) +{ + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata = dev_get_drvdata(&pdev->dev); + + pr_debug("%s: Enter (pdev->name = %s).\n", __func__, pdev->name); + + return ux500_msp_i2s_resume(msp_i2s_drvdata); +} + +static struct platform_driver msp_i2s_driver = { + .driver = { + .name = "ux500-msp-i2s", + .owner = THIS_MODULE, + }, + .probe = ux500_msp_drv_probe, + .remove = ux500_msp_drv_remove, + .suspend = ux500_msp_drv_suspend, + .resume = ux500_msp_drv_resume, +}; + +static int __init ux500_msp_init(void) +{ + pr_info("%s: Register ux500-msp-dai platform driver.\n", __func__); + return platform_driver_register(&msp_i2s_driver); +} + +static void __exit ux500_msp_exit(void) +{ + pr_info("%s: Unregister ux500-msp-dai platform driver.\n", __func__); + platform_driver_unregister(&msp_i2s_driver); +} + +module_init(ux500_msp_init); +module_exit(ux500_msp_exit); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h new file mode 100644 index 0000000..b9625e6 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -0,0 +1,83 @@ +/* + * Copyright (C) ST-Ericsson SA 2011 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 UX500_msp_dai_H +#define UX500_msp_dai_H + +#include <linux/types.h> +#include <linux/spinlock.h> + +#include <mach/msp.h> + +#define UX500_NBR_OF_DAI 4 + +#define UX500_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define UX500_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +#define FRAME_PER_SINGLE_SLOT_8_KHZ 31 +#define FRAME_PER_SINGLE_SLOT_16_KHZ 124 +#define FRAME_PER_SINGLE_SLOT_44_1_KHZ 63 +#define FRAME_PER_SINGLE_SLOT_48_KHZ 49 +#define FRAME_PER_2_SLOTS 31 +#define FRAME_PER_8_SLOTS 138 +#define FRAME_PER_16_SLOTS 277 + +#ifndef CONFIG_SND_SOC_UX500_AB5500 +#define UX500_MSP_INTERNAL_CLOCK_FREQ 40000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ UX500_MSP_INTERNAL_CLOCK_FREQ +#else +#define UX500_MSP_INTERNAL_CLOCK_FREQ 13000000 +#define UX500_MSP1_INTERNAL_CLOCK_FREQ (UX500_MSP_INTERNAL_CLOCK_FREQ * 2) +#endif + +#define UX500_MSP_MIN_CHANNELS 1 +#define UX500_MSP_MAX_CHANNELS 8 + +#define PLAYBACK_CONFIGURED 1 +#define CAPTURE_CONFIGURED 2 + +enum ux500_msp_clock_id { + UX500_MSP_MASTER_CLOCK, +}; + +struct ux500_platform_drvdata { + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata; + unsigned int fmt; + unsigned int tx_mask; + unsigned int rx_mask; + int slots; + int slot_width; + bool playback_active; + bool capture_active; + u8 configured; + int data_delay; + unsigned int master_clk; +}; + +extern struct snd_soc_dai ux500_msp_dai[UX500_NBR_OF_DAI]; + +bool ux500_msp_dai_i2s_get_underrun_status(int dai_idx); +int ux500_msp_dai_i2s_configure_sg(dma_addr_t dma_addr, + int perod_cnt, + size_t period_len, + int dai_idx, + int stream_id); +int ux500_msp_dai_i2s_send_data(void *data, size_t bytes, int dai_idx); +int ux500_msp_dai_i2s_receive_data(void *data, size_t bytes, int dai_idx); + +int ux500_msp_dai_set_data_delay(struct snd_soc_dai *dai, int delay); + +#endif diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c new file mode 100644 index 0000000..c52012c --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -0,0 +1,1004 @@ +/* + * Copyright (C) ST-Ericsson SA 2011 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Sandeep Kaushik <sandeep.kaushik(a)st.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/regulator/consumer.h> +#include <linux/mfd/dbx500-prcmu.h> + +#include <plat/ste_dma40.h> + +#include <mach/hardware.h> +#include <mach/msp.h> + +#include "ux500_msp_i2s.h" + + /* Protocol desciptors */ +static const struct msp_protocol_desc prot_descs[] = { + I2S_PROTOCOL_DESC, + PCM_PROTOCOL_DESC, + PCM_COMPAND_PROTOCOL_DESC, + AC97_PROTOCOL_DESC, + SPI_MASTER_PROTOCOL_DESC, + SPI_SLAVE_PROTOCOL_DESC, +}; + +static void ux500_msp_i2s_set_prot_desc_tx(struct msp *msp, + struct msp_protocol_desc *protocol_desc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protocol_desc->tx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protocol_desc->tx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protocol_desc->tx_frame_length_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protocol_desc->tx_frame_length_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protocol_desc->tx_element_length_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protocol_desc->tx_element_length_2); + if (protocol_desc->tx_element_length_1 == + protocol_desc->tx_element_length_2) { + msp->actual_data_size = protocol_desc->tx_element_length_1; + } else { + msp->actual_data_size = data_size; + } + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + msp->actual_data_size = data_size; + } + temp_reg |= MSP_DATA_DELAY_BITS(protocol_desc->tx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protocol_desc->tx_bit_transfer_format); + temp_reg |= MSP_FRAME_SYNC_POL(protocol_desc->tx_frame_sync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protocol_desc->tx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protocol_desc->compression_mode); + temp_reg |= MSP_SET_FRAME_SYNC_IGNORE(protocol_desc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_TCF); +} + +static void ux500_msp_i2s_set_prot_desc_rx(struct msp *msp, + struct msp_protocol_desc *protocol_desc, + enum msp_data_size data_size) +{ + u32 temp_reg = 0; + + temp_reg |= MSP_P2_ENABLE_BIT(protocol_desc->rx_phase_mode); + temp_reg |= MSP_P2_START_MODE_BIT(protocol_desc->rx_phase2_start_mode); + temp_reg |= MSP_P1_FRAME_LEN_BITS(protocol_desc->rx_frame_length_1); + temp_reg |= MSP_P2_FRAME_LEN_BITS(protocol_desc->rx_frame_length_2); + if (msp->def_elem_len) { + temp_reg |= MSP_P1_ELEM_LEN_BITS(protocol_desc->rx_element_length_1); + temp_reg |= MSP_P2_ELEM_LEN_BITS(protocol_desc->rx_element_length_2); + if (protocol_desc->rx_element_length_1 == + protocol_desc->rx_element_length_2) { + msp->actual_data_size = protocol_desc->rx_element_length_1; + } else { + msp->actual_data_size = data_size; + } + } else { + temp_reg |= MSP_P1_ELEM_LEN_BITS(data_size); + temp_reg |= MSP_P2_ELEM_LEN_BITS(data_size); + msp->actual_data_size = data_size; + } + + temp_reg |= MSP_DATA_DELAY_BITS(protocol_desc->rx_data_delay); + temp_reg |= MSP_SET_ENDIANNES_BIT(protocol_desc->rx_bit_transfer_format); + temp_reg |= MSP_FRAME_SYNC_POL(protocol_desc->rx_frame_sync_pol); + temp_reg |= MSP_DATA_WORD_SWAP(protocol_desc->rx_half_word_swap); + temp_reg |= MSP_SET_COMPANDING_MODE(protocol_desc->expansion_mode); + temp_reg |= MSP_SET_FRAME_SYNC_IGNORE(protocol_desc->frame_sync_ignore); + + writel(temp_reg, msp->registers + MSP_RCF); + +} + +static int ux500_msp_i2s_configure_protocol(struct msp *msp, + struct msp_config *config) +{ + int direction; + struct msp_protocol_desc *protocol_desc; + enum msp_data_size data_size; + u32 temp_reg = 0; + + data_size = config->data_size; + msp->def_elem_len = config->def_elem_len; + direction = config->direction; + if (config->default_protocol_desc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + pr_err("%s: ERROR: Invalid protocol!\n", __func__); + return -EINVAL; + } + protocol_desc = + (struct msp_protocol_desc *)&prot_descs[config->protocol]; + } else { + protocol_desc = (struct msp_protocol_desc *)&config->protocol_desc; + } + + if (data_size < MSP_DATA_BITS_DEFAULT || data_size > MSP_DATA_BITS_32) { + pr_err("%s: ERROR: Invalid data-size requested (data_size = %d)!\n", + __func__, data_size); + return -EINVAL; + } + + switch (direction) { + case MSP_TRANSMIT_MODE: + ux500_msp_i2s_set_prot_desc_tx(msp, protocol_desc, data_size); + break; + case MSP_RECEIVE_MODE: + ux500_msp_i2s_set_prot_desc_rx(msp, protocol_desc, data_size); + break; + case MSP_BOTH_T_R_MODE: + ux500_msp_i2s_set_prot_desc_tx(msp, protocol_desc, data_size); + ux500_msp_i2s_set_prot_desc_rx(msp, protocol_desc, data_size); + break; + default: + pr_err("%s: ERROR: Invalid direction requested (direction = %d)!\n", + __func__, direction); + return -EINVAL; + } + + /* The below code is needed for both Rx and Tx path. Can't separate them. */ + temp_reg = readl(msp->registers + MSP_GCR) & ~TX_CLK_POL_RISING; + temp_reg |= MSP_TX_CLKPOL_BIT(~protocol_desc->tx_clock_pol); + writel(temp_reg, msp->registers + MSP_GCR); + temp_reg = readl(msp->registers + MSP_GCR) & ~RX_CLK_POL_RISING; + temp_reg |= MSP_RX_CLKPOL_BIT(protocol_desc->rx_clock_pol); + writel(temp_reg, msp->registers + MSP_GCR); + + return 0; +} + +static int ux500_msp_i2s_configure_clock(struct msp *msp, struct msp_config *config) +{ + u32 reg_val_GCR; + u32 frame_per = 0; + u32 sck_div = 0; + u32 frame_width = 0; + u32 temp_reg = 0; + u32 bit_clock = 0; + struct msp_protocol_desc *protocol_desc = NULL; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~SRG_ENABLE, msp->registers + MSP_GCR); + + if (config->default_protocol_desc) + protocol_desc = + (struct msp_protocol_desc *)&prot_descs[config->protocol]; + else + protocol_desc = (struct msp_protocol_desc *)&config->protocol_desc; + + switch (config->protocol) { + case MSP_PCM_PROTOCOL: + case MSP_PCM_COMPAND_PROTOCOL: + frame_width = protocol_desc->frame_width; + sck_div = config->input_clock_freq / (config->frame_freq * + (protocol_desc->total_clocks_for_one_frame)); + frame_per = protocol_desc->frame_period; + break; + case MSP_I2S_PROTOCOL: + frame_width = protocol_desc->frame_width; + sck_div = config->input_clock_freq / (config->frame_freq * + (protocol_desc->total_clocks_for_one_frame)); + frame_per = protocol_desc->frame_period; + break; + case MSP_AC97_PROTOCOL: + /* Not supported */ + pr_err("%s: ERROR: AC97 protocol not supported!\n", __func__); + return -ENOSYS; + default: + pr_err("%s: ERROR: Unknown protocol (%d)!\n", + __func__, + config->protocol); + return -EINVAL; + } + + temp_reg = (sck_div - 1) & SCK_DIV_MASK; + temp_reg |= FRAME_WIDTH_BITS(frame_width); + temp_reg |= FRAME_PERIOD_BITS(frame_per); + writel(temp_reg, msp->registers + MSP_SRG); + + bit_clock = (config->input_clock_freq)/(sck_div + 1); + /* If the bit clock is higher than 19.2MHz, Vape should be run in 100% OPP + * Only consider OPP 100% when bit-clock is used, i.e. MSP master mode + */ + if ((bit_clock > 19200000) && ((config->tx_clock_sel != 0) || (config->rx_clock_sel != 0))) { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, "ux500-msp-i2s", 100); + msp->vape_opp_constraint = 1; + } else { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, "ux500-msp-i2s", 50); + msp->vape_opp_constraint = 0; + } + + /* Enable clock */ + udelay(100); + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | SRG_ENABLE, msp->registers + MSP_GCR); + udelay(100); + + return 0; +} + +static int ux500_msp_i2s_configure_multichannel(struct msp *msp, struct msp_config *config) +{ + struct msp_protocol_desc *protocol_desc; + struct msp_multichannel_config *mcfg; + u32 reg_val_MCR; + + if (config->default_protocol_desc == 1) { + if (config->protocol >= MSP_INVALID_PROTOCOL) { + pr_err("%s: ERROR: Invalid protocol (%d)!\n", + __func__, + config->protocol); + return -EINVAL; + } + protocol_desc = (struct msp_protocol_desc *) + &prot_descs[config->protocol]; + } else { + protocol_desc = (struct msp_protocol_desc *)&config->protocol_desc; + } + + mcfg = &config->multichannel_config; + if (mcfg->tx_multichannel_enable) { + if (protocol_desc->tx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | + (mcfg->tx_multichannel_enable ? 1 << TMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->tx_channel_0_enable, + msp->registers + MSP_TCE0); + writel(mcfg->tx_channel_1_enable, + msp->registers + MSP_TCE1); + writel(mcfg->tx_channel_2_enable, + msp->registers + MSP_TCE2); + writel(mcfg->tx_channel_3_enable, + msp->registers + MSP_TCE3); + } else { + pr_err("%s: ERROR: Only single-phase supported (TX-mode: %d)!\n", + __func__, protocol_desc->tx_phase_mode); + return -EINVAL; + } + } + if (mcfg->rx_multichannel_enable) { + if (protocol_desc->rx_phase_mode == MSP_SINGLE_PHASE) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | + (mcfg->rx_multichannel_enable ? 1 << RMCEN_BIT : 0), + msp->registers + MSP_MCR); + writel(mcfg->rx_channel_0_enable, + msp->registers + MSP_RCE0); + writel(mcfg->rx_channel_1_enable, + msp->registers + MSP_RCE1); + writel(mcfg->rx_channel_2_enable, + msp->registers + MSP_RCE2); + writel(mcfg->rx_channel_3_enable, + msp->registers + MSP_RCE3); + } else { + pr_err("%s: ERROR: Only single-phase supported (RX-mode: %d)!\n", + __func__, protocol_desc->rx_phase_mode); + return -EINVAL; + } + if (mcfg->rx_comparison_enable_mode) { + reg_val_MCR = readl(msp->registers + MSP_MCR); + writel(reg_val_MCR | + (mcfg->rx_comparison_enable_mode << RCMPM_BIT), + msp->registers + MSP_MCR); + + writel(mcfg->comparison_mask, + msp->registers + MSP_RCM); + writel(mcfg->comparison_value, + msp->registers + MSP_RCV); + + } + } + + return 0; +} + +void ux500_msp_i2s_configure_dma(struct msp *msp, struct msp_config *config) +{ + struct stedma40_chan_cfg *rx_dma_info = msp->dma_cfg_rx; + struct stedma40_chan_cfg *tx_dma_info = msp->dma_cfg_tx; + dma_cap_mask_t mask; + u16 word_width; + bool rx_active, tx_active; + + if (msp->tx_pipeid != NULL) { + dma_release_channel(msp->tx_pipeid); + msp->tx_pipeid = NULL; + } + + switch (config->data_size) { + case MSP_DATA_BITS_32: + word_width = STEDMA40_WORD_WIDTH; + break; + case MSP_DATA_BITS_16: + word_width = STEDMA40_HALFWORD_WIDTH; + break; + case MSP_DATA_BITS_8: + word_width = STEDMA40_BYTE_WIDTH; + break; + default: + word_width = STEDMA40_WORD_WIDTH; + pr_warn("%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, config->data_size); + } + + rx_active = (config->direction == MSP_RECEIVE_MODE || + config->direction == MSP_BOTH_T_R_MODE); + tx_active = (config->direction == MSP_TRANSMIT_MODE || + config->direction == MSP_BOTH_T_R_MODE); + + if (rx_active) { + rx_dma_info->src_info.data_width = word_width; + rx_dma_info->dst_info.data_width = word_width; + } + if (tx_active) { + tx_dma_info->src_info.data_width = word_width; + tx_dma_info->dst_info.data_width = word_width; + } + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + if (rx_active) + msp->rx_pipeid = dma_request_channel(mask, stedma40_filter, rx_dma_info); + + if (tx_active) + msp->tx_pipeid = dma_request_channel(mask, stedma40_filter, tx_dma_info); +} + +static int ux500_msp_i2s_dma_xfer(struct msp *msp, struct i2s_message *msg) +{ + dma_cookie_t status_submit; + int direction, enable_bit; + u32 reg_val_GCR; + struct dma_chan *pipeid; + struct dma_async_tx_descriptor *cdesc; + + if (msg->i2s_direction == I2S_DIRECTION_TX) { + direction = DMA_TO_DEVICE; + pipeid = msp->tx_pipeid; + enable_bit = TX_ENABLE; + pr_debug("%s: Direction: TX\n", __func__); + } else { + direction = DMA_FROM_DEVICE; + pipeid = msp->rx_pipeid; + enable_bit = RX_ENABLE; + pr_debug("%s: Direction: RX\n", __func__); + } + + pr_debug("%s: msg->buf_addr = %p\n", __func__, (void *)msg->buf_addr); + pr_debug("%s: buf_len = %d\n", __func__, msg->buf_len); + pr_debug("%s: perios_len = %d\n", __func__, msg->period_len); + + /* setup the cyclic description */ + cdesc = pipeid->device->device_prep_dma_cyclic(pipeid, + msg->buf_addr, + msg->buf_len, + msg->period_len, + direction); + if (IS_ERR(cdesc)) { + pr_err("%s: ERROR: device_prep_dma_cyclic failed (%ld)!\n", + __func__, + PTR_ERR(cdesc)); + return -EINVAL; + } + + /* Submit to the dma */ + if (msg->i2s_direction == I2S_DIRECTION_TX) { + cdesc->callback = msp->xfer_data.tx_handler; + cdesc->callback_param = msp->xfer_data.tx_callback_data; + } else { + cdesc->callback = msp->xfer_data.rx_handler; + cdesc->callback_param = msp->xfer_data.rx_callback_data; + } + status_submit = dmaengine_submit(cdesc); + if (dma_submit_error(status_submit)) { + pr_err("%s: ERROR: dmaengine_submit failed!\n", __func__); + return -EINVAL; + } + + /* Start the dma */ + dma_async_issue_pending(pipeid); + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | enable_bit, msp->registers + MSP_GCR); + + return 0; +} + +static int ux500_msp_i2s_enable(struct msp *msp, struct msp_config *config) +{ + int status = 0; + u32 reg_val_DMACR, reg_val_GCR; + + if (config->work_mode != MSP_DMA_MODE) { + pr_err("%s: ERROR: Only DMA-mode is supported (msp->work_mode = %d)\n", + __func__, + msp->work_mode); + return -EINVAL; + } + msp->work_mode = config->work_mode; + + /* Check msp state whether in RUN or CONFIGURED Mode */ + if (msp->msp_state == MSP_STATE_IDLE) { + if (msp->plat_init) { + status = msp->plat_init(); + if (status) { + pr_err("%s: ERROR: Failed to init MSP (%d)!\n", + __func__, + status); + return status; + } + } + } + + /* Configure msp with protocol dependent settings */ + ux500_msp_i2s_configure_protocol(msp, config); + ux500_msp_i2s_configure_clock(msp, config); + if (config->multichannel_configured == 1) { + status = ux500_msp_i2s_configure_multichannel(msp, config); + if (status) + pr_warn("%s: WARN: ux500_msp_i2s_configure_multichannel failed (%d)!\n", + __func__, status); + } + + /* Make sure the correct DMA-directions are configured */ + if ((config->direction == MSP_RECEIVE_MODE) || + (config->direction == MSP_BOTH_T_R_MODE)) + if (!msp->dma_cfg_rx) { + pr_err("%s: ERROR: MSP RX-mode is not configured!", __func__); + return -EINVAL; + } + if ((config->direction == MSP_TRANSMIT_MODE) || + (config->direction == MSP_BOTH_T_R_MODE)) + if (!msp->dma_cfg_tx) { + pr_err("%s: ERROR: MSP TX-mode is not configured!", __func__); + return -EINVAL; + } + + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + switch (config->direction) { + case MSP_TRANSMIT_MODE: + writel(reg_val_DMACR | TX_DMA_ENABLE, + msp->registers + MSP_DMACR); + + msp->xfer_data.tx_handler = config->handler; + msp->xfer_data.tx_callback_data = config->tx_callback_data; + + break; + case MSP_RECEIVE_MODE: + writel(reg_val_DMACR | RX_DMA_ENABLE, + msp->registers + MSP_DMACR); + + msp->xfer_data.rx_handler = config->handler; + msp->xfer_data.rx_callback_data = config->rx_callback_data; + + break; + case MSP_BOTH_T_R_MODE: + writel(reg_val_DMACR | RX_DMA_ENABLE | TX_DMA_ENABLE, + msp->registers + MSP_DMACR); + + msp->xfer_data.tx_handler = config->handler; + msp->xfer_data.rx_handler = config->handler; + msp->xfer_data.tx_callback_data = config->tx_callback_data; + msp->xfer_data.rx_callback_data = config->rx_callback_data; + + break; + default: + pr_err("%s: ERROR: Illegal MSP direction (config->direction = %d)!", + __func__, + config->direction); + if (msp->plat_exit) + msp->plat_exit(); + return -EINVAL; + } + ux500_msp_i2s_configure_dma(msp, config); + + msp->transfer = ux500_msp_i2s_dma_xfer; + + writel(config->iodelay, msp->registers + MSP_IODLY); + + /* Enable frame generation logic */ + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | FRAME_GEN_ENABLE, msp->registers + MSP_GCR); + + return status; +} + +static void flush_fifo_rx(struct msp *msp) +{ + u32 reg_val_DR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | RX_ENABLE, msp->registers + MSP_GCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & RX_FIFO_EMPTY) && limit--) { + reg_val_DR = readl(msp->registers + MSP_DR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +static void flush_fifo_tx(struct msp *msp) +{ + u32 reg_val_TSTDR, reg_val_GCR, reg_val_FLR; + u32 limit = 32; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | TX_ENABLE, msp->registers + MSP_GCR); + writel(MSP_ITCR_ITEN | MSP_ITCR_TESTFIFO, msp->registers + MSP_ITCR); + + reg_val_FLR = readl(msp->registers + MSP_FLR); + while (!(reg_val_FLR & TX_FIFO_EMPTY) && limit--) { + reg_val_TSTDR = readl(msp->registers + MSP_TSTDR); + reg_val_FLR = readl(msp->registers + MSP_FLR); + } + writel(0x0, msp->registers + MSP_ITCR); + writel(reg_val_GCR, msp->registers + MSP_GCR); +} + +int ux500_msp_i2s_open(struct ux500_msp_i2s_drvdata *drvdata, struct msp_config *msp_config) +{ + struct msp *msp = drvdata->msp; + u32 old_reg, new_reg, mask; + int res; + + if (in_interrupt()) { + pr_err("%s: ERROR: Open called in interrupt context!\n", __func__); + return -1; + } + + /* Two simultanous configuring msp is avoidable */ + down(&msp->lock); + + /* Don't enable regulator if its msp1 or msp3 */ + if (!(msp->reg_enabled) && msp->id != MSP_1_I2S_CONTROLLER + && msp->id != MSP_3_I2S_CONTROLLER) { + res = regulator_enable(drvdata->reg_vape); + if (res != 0) { + pr_err("%s: Failed to enable regulator!\n", __func__); + up(&msp->lock); + return res; + } + msp->reg_enabled = 1; + } + + switch (msp->users) { + case 0: + clk_enable(msp->clk); + msp->direction = msp_config->direction; + break; + case 1: + if (msp->direction == MSP_BOTH_T_R_MODE || + msp_config->direction == msp->direction || + msp_config->direction == MSP_BOTH_T_R_MODE) { + pr_warn("%s: WARN: MSP is in use (direction = %d)!\n", + __func__, msp_config->direction); + up(&msp->lock); + return -EBUSY; + } + msp->direction = MSP_BOTH_T_R_MODE; + break; + default: + pr_warn("%s: MSP in use in (both directions)!\n", __func__); + up(&msp->lock); + return -EBUSY; + } + msp->users++; + + /* First do the global config register */ + mask = + RX_CLK_SEL_MASK | TX_CLK_SEL_MASK | RX_FRAME_SYNC_MASK | + TX_FRAME_SYNC_MASK | RX_SYNC_SEL_MASK | TX_SYNC_SEL_MASK | + RX_FIFO_ENABLE_MASK | TX_FIFO_ENABLE_MASK | SRG_CLK_SEL_MASK | + LOOPBACK_MASK | TX_EXTRA_DELAY_MASK; + + new_reg = (msp_config->tx_clock_sel | msp_config->rx_clock_sel | + msp_config->rx_frame_sync_pol | msp_config->tx_frame_sync_pol | + msp_config->rx_frame_sync_sel | msp_config->tx_frame_sync_sel | + msp_config->rx_fifo_config | msp_config->tx_fifo_config | + msp_config->srg_clock_sel | msp_config->loopback_enable | + msp_config->tx_data_enable); + + old_reg = readl(msp->registers + MSP_GCR); + old_reg &= ~mask; + new_reg |= old_reg; + writel(new_reg, msp->registers + MSP_GCR); + + if (ux500_msp_i2s_enable(msp, msp_config) != 0) { + pr_err("%s: ERROR: ux500_msp_i2s_enable failed!\n", __func__); + return -EBUSY; + } + if (msp_config->loopback_enable & 0x80) + msp->loopback_enable = 1; + + /* Flush FIFOs */ + flush_fifo_tx(msp); + flush_fifo_rx(msp); + + msp->msp_state = MSP_STATE_CONFIGURED; + up(&msp->lock); + return 0; +} + +static void func_notify_timer(unsigned long data) +{ + struct msp *msp = (struct msp *)data; + if (msp->polling_flag) { + msp->msp_io_error = 1; + pr_err("%s: ERROR: Polling timeout!\n", __func__); + del_timer(&msp->notify_timer); + } +} + +int ux500_msp_i2s_transfer(struct ux500_msp_i2s_drvdata *drvdata, struct i2s_message *message) +{ + struct msp *msp = drvdata->msp; + int status = 0; + + if (!message || (msp->msp_state == MSP_STATE_IDLE)) { + pr_err("%s: ERROR: i2s_message == NULL!\n", __func__); + return -EINVAL; + } + if (msp->msp_state == MSP_STATE_IDLE) { + pr_err("%s: ERROR: MSP in idle-state!\n", __func__); + return -EPERM; + } + + msp->msp_state = MSP_STATE_RUN; + if (msp->transfer) + status = msp->transfer(msp, message); + + if (msp->msp_state == MSP_STATE_RUN) + msp->msp_state = MSP_STATE_CONFIGURED; + + return status; +} + +static void ux500_msp_i2s_disable_rx(struct msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~RX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~RX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(RECEIVE_SERVICE_INT | RECEIVE_OVERRUN_ERROR_INT), + msp->registers + MSP_IMSC); + msp->xfer_data.message.rxbytes = 0; + msp->xfer_data.message.rx_offset = 0; + msp->xfer_data.message.rxdata = NULL; + msp->read = NULL; +} + +static void ux500_msp_i2s_disable_tx(struct msp *msp) +{ + u32 reg_val_GCR, reg_val_DMACR, reg_val_IMSC; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR & ~TX_ENABLE, msp->registers + MSP_GCR); + reg_val_DMACR = readl(msp->registers + MSP_DMACR); + writel(reg_val_DMACR & ~TX_DMA_ENABLE, msp->registers + MSP_DMACR); + reg_val_IMSC = readl(msp->registers + MSP_IMSC); + writel(reg_val_IMSC & + ~(TRANSMIT_SERVICE_INT | TRANSMIT_UNDERRUN_ERR_INT), + msp->registers + MSP_IMSC); + msp->xfer_data.message.txbytes = 0; + msp->xfer_data.message.tx_offset = 0; + msp->xfer_data.message.txdata = NULL; + msp->write = NULL; +} + +static int ux500_msp_i2s_disable(struct msp *msp, int direction, enum i2s_flag flag) +{ + u32 reg_val_GCR; + int status = 0; + + reg_val_GCR = readl(msp->registers + MSP_GCR); + if (!(reg_val_GCR & (TX_ENABLE | RX_ENABLE))) + return 0; + + if (flag == DISABLE_ALL || flag == DISABLE_TRANSMIT) { + if (msp->tx_pipeid != NULL) { + dmaengine_terminate_all(msp->tx_pipeid); + dma_release_channel(msp->tx_pipeid); + msp->tx_pipeid = NULL; + } + } + if ((flag == DISABLE_ALL || flag == DISABLE_RECEIVE)) { + if (msp->rx_pipeid != NULL) { + dmaengine_terminate_all(msp->rx_pipeid); + dma_release_channel(msp->rx_pipeid); + msp->rx_pipeid = NULL; + } + } + + if (flag == DISABLE_TRANSMIT) + ux500_msp_i2s_disable_tx(msp); + else if (flag == DISABLE_RECEIVE) + ux500_msp_i2s_disable_rx(msp); + else { + reg_val_GCR = readl(msp->registers + MSP_GCR); + writel(reg_val_GCR | LOOPBACK_MASK, + msp->registers + MSP_GCR); + + /* Flush TX-FIFO */ + flush_fifo_tx(msp); + + /* Disable TX-channel */ + writel((readl(msp->registers + MSP_GCR) & + (~TX_ENABLE)), msp->registers + MSP_GCR); + + /* Flush RX-FIFO */ + flush_fifo_rx(msp); + + /* Disable Loopback and Receive channel */ + writel((readl(msp->registers + MSP_GCR) & + (~(RX_ENABLE | LOOPBACK_MASK))), + msp->registers + MSP_GCR); + + ux500_msp_i2s_disable_tx(msp); + ux500_msp_i2s_disable_rx(msp); + + } + + /* disable sample rate and frame generators */ + if (flag == DISABLE_ALL) { + msp->msp_state = MSP_STATE_IDLE; + writel((readl(msp->registers + MSP_GCR) & + (~(FRAME_GEN_ENABLE | SRG_ENABLE))), + msp->registers + MSP_GCR); + memset(&msp->xfer_data, 0, sizeof(struct trans_data)); + if (msp->plat_exit) + status = msp->plat_exit(); + if (status) + pr_warn("%s: WARN: ux500_msp_i2s_exit failed (%d)!\n", + __func__, status); + msp->transfer = NULL; + writel(0, msp->registers + MSP_GCR); + writel(0, msp->registers + MSP_TCF); + writel(0, msp->registers + MSP_RCF); + writel(0, msp->registers + MSP_DMACR); + writel(0, msp->registers + MSP_SRG); + writel(0, msp->registers + MSP_MCR); + writel(0, msp->registers + MSP_RCM); + writel(0, msp->registers + MSP_RCV); + writel(0, msp->registers + MSP_TCE0); + writel(0, msp->registers + MSP_TCE1); + writel(0, msp->registers + MSP_TCE2); + writel(0, msp->registers + MSP_TCE3); + writel(0, msp->registers + MSP_RCE0); + writel(0, msp->registers + MSP_RCE1); + writel(0, msp->registers + MSP_RCE2); + writel(0, msp->registers + MSP_RCE3); + } + + return status; +} + +int ux500_msp_i2s_close(struct ux500_msp_i2s_drvdata *drvdata, enum i2s_flag flag) +{ + struct msp *msp = drvdata->msp; + int status = 0; + + pr_debug("%s: Enter.\n", __func__); + + down(&msp->lock); + + if (msp->users == 0) { + pr_err("%s: ERROR: MSP already closed!\n", __func__); + status = -EINVAL; + goto end; + } + pr_debug("%s: msp->users = %d, flag = %d\n", __func__, msp->users, flag); + + /* We need to call it twice for DISABLE_ALL*/ + msp->users = flag == DISABLE_ALL ? 0 : msp->users - 1; + if (msp->users) + status = ux500_msp_i2s_disable(msp, MSP_BOTH_T_R_MODE, flag); + else { + status = ux500_msp_i2s_disable(msp, MSP_BOTH_T_R_MODE, DISABLE_ALL); + clk_disable(msp->clk); + if (msp->reg_enabled) { + status = regulator_disable(drvdata->reg_vape); + msp->reg_enabled = 0; + } + if (status != 0) { + pr_err("%s: ERROR: Failed to disable regulator (%d)!\n", + __func__, status); + clk_enable(msp->clk); + goto end; + } + } + if (status) + goto end; + if (msp->users) + msp->direction = flag == DISABLE_TRANSMIT ? + MSP_RECEIVE_MODE : MSP_TRANSMIT_MODE; + + if (msp->vape_opp_constraint == 1) { + prcmu_qos_update_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s", 50); + msp->vape_opp_constraint = 0; + } +end: + up(&msp->lock); + return status; + +} + +int ux500_msp_i2s_hw_status(struct ux500_msp_i2s_drvdata *drvdata) +{ + struct msp *msp = drvdata->msp; + int status; + + pr_debug("%s: Enter.\n", __func__); + + status = readl(msp->registers + MSP_RIS) & 0xee; + if (status) + writel(status, msp->registers + MSP_ICR); + + return status; +} + +struct ux500_msp_i2s_drvdata *ux500_msp_i2s_init(struct platform_device *pdev, + struct msp_i2s_platform_data *platform_data) +{ + struct ux500_msp_i2s_drvdata *msp_i2s_drvdata; + int irq; + struct resource *res = NULL; + struct i2s_controller *i2s_cont; + struct msp *msp; + + pr_debug("%s: Enter (pdev->name = %s).\n", __func__, pdev->name); + + msp_i2s_drvdata = kzalloc(sizeof(struct ux500_msp_i2s_drvdata), GFP_KERNEL); + msp_i2s_drvdata->msp = kzalloc(sizeof(struct msp), GFP_KERNEL); + msp = msp_i2s_drvdata->msp; + + msp->id = platform_data->id; + msp_i2s_drvdata->id = msp->id; + pr_debug("msp_i2s_drvdata->id = %d\n", msp_i2s_drvdata->id); + + msp->plat_init = platform_data->msp_i2s_init; + msp->plat_exit = platform_data->msp_i2s_exit; + msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; + msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; + + sema_init(&msp->lock, 1); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + pr_err("%s: ERROR: Unable to get resource!\n", __func__); + goto free_msp; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) + goto free_msp; + msp->irq = irq; + + msp->registers = ioremap(res->start, (res->end - res->start + 1)); + if (msp->registers == NULL) + goto free_msp; + + msp_i2s_drvdata->reg_vape = regulator_get(NULL, "v-ape"); + if (IS_ERR(msp_i2s_drvdata->reg_vape)) { + pr_err("%s: ERROR: Failed to get Vape supply (%d)!\n", + __func__, (int)PTR_ERR(msp_i2s_drvdata->reg_vape)); + goto free_irq; + } + dev_set_drvdata(&pdev->dev, msp_i2s_drvdata); + + prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50); + msp->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(msp->clk)) { + pr_err("%s: ERROR: clk_get failed (%d)!\n", + __func__, (int)PTR_ERR(msp->clk)); + goto free_irq; + } + + init_timer(&msp->notify_timer); + msp->notify_timer.expires = jiffies + msecs_to_jiffies(1000); + msp->notify_timer.function = func_notify_timer; + msp->notify_timer.data = (unsigned long)msp; + + msp->rx_pipeid = NULL; + msp->tx_pipeid = NULL; + msp->read = NULL; + msp->write = NULL; + msp->transfer = NULL; + msp->msp_state = MSP_STATE_IDLE; + msp->loopback_enable = 0; + + /* I2S Controller is allocated and added in I2S controller class. */ + i2s_cont = kzalloc(sizeof(*i2s_cont), GFP_KERNEL); + if (!i2s_cont) { + pr_err("%s: ERROR: Failed to allocate struct i2s_cont (kzalloc)!\n", + __func__); + goto del_timer; + } + i2s_cont->dev.parent = &pdev->dev; + i2s_cont->data = (void *)msp; + i2s_cont->id = (s16)msp->id; + snprintf(i2s_cont->name, + sizeof(i2s_cont->name), + "ux500-msp-i2s.%04x", + msp->id); + pr_debug("I2S device-name :%s\n", i2s_cont->name); + msp->i2s_cont = i2s_cont; + + return msp_i2s_drvdata; + +del_timer: + del_timer_sync(&msp->notify_timer); + clk_put(msp->clk); +free_irq: + iounmap(msp->registers); +free_msp: + kfree(msp); + return NULL; +} + +int ux500_msp_i2s_exit(struct ux500_msp_i2s_drvdata *drvdata) +{ + struct msp *msp = drvdata->msp; + int status = 0; + + pr_debug("%s: Enter (drvdata->id = %d).\n", __func__, drvdata->id); + + device_unregister(&msp->i2s_cont->dev); + del_timer_sync(&msp->notify_timer); + clk_put(msp->clk); + iounmap(msp->registers); + prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); + regulator_put(drvdata->reg_vape); + kfree(msp); + + return status; +} + +int ux500_msp_i2s_suspend(struct ux500_msp_i2s_drvdata *drvdata) +{ + struct msp *msp = drvdata->msp; + + pr_debug("%s: Enter (drvdata->id = %d).\n", __func__, drvdata->id); + + down(&msp->lock); + if (msp->users > 0) { + up(&msp->lock); + return -EBUSY; + } + up(&msp->lock); + + return 0; +} + +int ux500_msp_i2s_resume(struct ux500_msp_i2s_drvdata *drvdata) +{ + pr_debug("%s: Enter (drvdata->id = %d).\n", __func__, drvdata->id); + + return 0; +} + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h new file mode 100644 index 0000000..5c12f31 --- /dev/null +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -0,0 +1,40 @@ +/* + * Copyright (C) ST-Ericsson SA 2011 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * for ST-Ericsson. + * + * License terms: + * + * 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 UX500_MSP_I2S_H +#define UX500_MSP_I2S_H + +#include <linux/platform_device.h> + +#include <mach/msp.h> + +struct ux500_msp_i2s_drvdata { + int id; + struct msp *msp; + struct regulator *reg_vape; +}; + +struct ux500_msp_i2s_drvdata *ux500_msp_i2s_init(struct platform_device *pdev, + struct msp_i2s_platform_data *platform_data); +int ux500_msp_i2s_exit(struct ux500_msp_i2s_drvdata *drvdata); +int ux500_msp_i2s_open(struct ux500_msp_i2s_drvdata *drvdata, struct msp_config *msp_config); +int ux500_msp_i2s_close(struct ux500_msp_i2s_drvdata *drvdata, enum i2s_flag flag); +int ux500_msp_i2s_transfer(struct ux500_msp_i2s_drvdata *drvdata, struct i2s_message *message); +int ux500_msp_i2s_hw_status(struct ux500_msp_i2s_drvdata *drvdata); + +int ux500_msp_i2s_suspend(struct ux500_msp_i2s_drvdata *drvdata); +int ux500_msp_i2s_resume(struct ux500_msp_i2s_drvdata *drvdata); + +#endif + diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..85b9a7b --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,428 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static const char *stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent( + buf->dev.dev, + buf->bytes, + buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +void ux500_pcm_dma_eot_handler(void *data) +{ + struct snd_pcm_substream *substream = data; + struct snd_pcm_runtime *runtime; + struct ux500_pcm_private *private; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + pr_debug("%s: MSP %d (%s): Enter.\n", __func__, dai->id, stream_str(substream)); + + if (substream) { + runtime = substream->runtime; + private = substream->runtime->private_data; + + if (ux500_msp_dai_i2s_get_underrun_status(private->msp_id)) { + private->no_of_underruns++; + pr_debug("%s: Nr of underruns (%d)\n", __func__, + private->no_of_underruns); + } + + /* calc the offset in the circular buffer */ + private->offset += frames_to_bytes(runtime, + runtime->period_size); + private->offset %= frames_to_bytes(runtime, + runtime->period_size) * runtime->periods; + + snd_pcm_period_elapsed(substream); + } +} +EXPORT_SYMBOL(ux500_pcm_dma_eot_handler); + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_pcm_private *private; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + int ret; + + pr_debug("%s: MSP %d (%s): Enter.\n", __func__, dai->id, stream_str(substream)); + + pr_debug("%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + pr_err("%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, + ret); + return ret; + } + + pr_debug("%s: Init runtime private data.\n", __func__); + private = kzalloc(sizeof(struct ux500_pcm_private), GFP_KERNEL); + if (private == NULL) + return -ENOMEM; + private->msp_id = dai->id; + runtime->private_data = private; + + pr_debug("%s: Set hw-struct for %s.\n", __func__, stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct ux500_pcm_private *private = substream->runtime->private_data; + + pr_debug("%s: Enter\n", __func__); + + kfree(private); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + int ret = 0; + int size; + + pr_debug("%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent( + NULL, + size, + &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + pr_debug("%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_prepare(struct snd_pcm_substream *substream) +{ + pr_debug("%s: Enter\n", __func__); + return 0; +} + +static int ux500_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_pcm_private *private = runtime->private_data; + int stream_id = substream->pstr->stream; + + pr_debug("%s: Enter\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("%s: START/PAUSE-RELEASE\n", __func__); + if (runtime->status->state == SNDRV_PCM_STATE_XRUN) { + pr_debug("XRUN occurred\n"); + return 0; + } + + private->no_of_underruns = 0; + private->offset = 0; + ret = ux500_msp_dai_i2s_configure_sg(runtime->dma_addr, + runtime->periods, + frames_to_bytes(runtime, runtime->period_size), + private->msp_id, + stream_id); + if (ret) { + pr_err("%s: Failed to configure I2S!\n", __func__); + return -EINVAL; + } + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("%s: SNDRV_PCM_TRIGGER_STOP\n", __func__); + pr_debug("%s: no_of_underruns = %u\n", + __func__, + private->no_of_underruns); + break; + + default: + pr_err("%s: Invalid command in pcm trigger\n", + __func__); + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t ux500_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct ux500_pcm_private *private = runtime->private_data; + + pr_debug("%s: dma_offset %d frame %ld\n", __func__, private->offset, + bytes_to_frames(substream->runtime, private->offset)); + + return bytes_to_frames(substream->runtime, private->offset); +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + pr_debug("%s: Enter.\n", __func__); + + return dma_mmap_coherent( + NULL, + vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static const struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .prepare = ux500_pcm_prepare, + .trigger = ux500_pcm_trigger, + .pointer = ux500_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + pr_debug("%s: pcm = %d\n", __func__, (int)pcm); + + pcm->info_flags = 0; + strcpy(pcm->name, "UX500_PCM"); + + pr_debug("%s: pcm->name = %s.\n", __func__, pcm->name); + + return 0; +} + +static void ux500_pcm_free(struct snd_pcm *pcm) +{ + pr_debug("%s: Enter\n", __func__); +} + +static int ux500_pcm_suspend(struct snd_soc_dai *dai) +{ + pr_debug("%s: Enter\n", __func__); + + return 0; +} + +static int ux500_pcm_resume(struct snd_soc_dai *dai) +{ + pr_debug("%s: Enter\n", __func__); + + return 0; +} + +struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, + .pcm_free = ux500_pcm_free, + .suspend = ux500_pcm_suspend, + .resume = ux500_pcm_resume, +}; +EXPORT_SYMBOL(ux500_pcm_soc_drv); + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + pr_info("%s: Register ux500-pcm SoC platform driver.\n", __func__); + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + pr_err("%s: Error: Failed to register " + "ux500-pcm SoC platform driver (%d)!\n", + __func__, + ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + pr_info("%s: Unregister ux500-pcm SoC platform driver.\n", __func__); + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; + +static int __init ux500_pcm_drv_init(void) +{ + pr_debug("%s: Register ux500-pcm platform driver.\n", __func__); + + return platform_driver_register(&ux500_pcm_driver); +} + +static void __exit ux500_pcm_drv_exit(void) +{ + pr_debug("%s: Unregister ux500-pcm platform driver.\n", __func__); + + platform_driver_unregister(&ux500_pcm_driver); +} + +module_init(ux500_pcm_drv_init); +module_exit(ux500_pcm_drv_exit); + diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..50f4661 --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,44 @@ +/* + * Copyright (C) ST-Ericsson SA 2010 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 UX500_PCM_H +#define UX500_PCM_H + +#include <mach/msp.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +extern struct snd_soc_platform ux500_soc_platform; + +struct ux500_pcm_private { + int msp_id; + int stream_id; + unsigned int no_of_underruns; + unsigned int offset; +}; + +void ux500_pcm_dma_eot_handler(void *data); + +#endif -- 1.7.8.3

3 3

[alsa-devel] [PATCH] ASoC: jack: Push locking for jacks down to the jack
by Mark Brown 13 Mar '12

13 Mar '12

Currently operations on jack reporting take the CODEC mutex both to protect the current jack status and also to protect the DAPM run which is triggered on status updates. Since the addition of a DAPM-specific lock we no longer need to worry about locking DAPM as it has its own finer grained lock so create a per jack lock to take care of the jack status. This is both cleaner where the jack isn't specifically associated with a CODEC and clearer as it's much more obvious what the lock is protecting. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 5 +++-- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0989987..b8163dd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -518,6 +518,7 @@ struct snd_soc_jack_gpio { #endif struct snd_soc_jack { + struct mutex mutex; struct snd_jack *jack; struct snd_soc_codec *codec; struct list_head pins; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 17717fc..9db15ba 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -36,6 +36,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { + mutex_init(&jack->mutex); jack->codec = codec; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); @@ -75,7 +76,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) codec = jack->codec; dapm = &codec->dapm; - mutex_lock(&codec->mutex); + mutex_lock(&jack->mutex); oldstatus = jack->status; @@ -109,7 +110,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_jack_report(jack->jack, jack->status); out: - mutex_unlock(&codec->mutex); + mutex_unlock(&jack->mutex); } EXPORT_SYMBOL_GPL(snd_soc_jack_report); -- 1.7.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - fix printing of high HDMI sample rates
by Anssi Hannula 13 Mar '12

13 Mar '12

A previous commit af65cbf296 (ALSA: hdmi: fix printout of SAD sampling rates) fixed the sample rates shown in /proc/asound/cardX/eldY and kernel log to not be entirely wrong. However, a missing rate from the array added in the patch causes HDMI rates 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz to be shown as 96 kHz, 176.4 kHz, 192 kHz, and 384 kHz, respectively. Fix the reporting by adding the ALSA rate 64 kHz into the conversion array between 48 kHz and 88.2 kHz. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: stable(a)kernel.org --- sound/pci/hda/hda_eld.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c1da422..b58b4b1 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -385,8 +385,8 @@ error: static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) { static unsigned int alsa_rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, - 96000, 176400, 192000, 384000 + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, 384000 }; int i, j; -- 1.7.7.2

2 1

[alsa-devel] [RFC] [PATCH 0/5] Add vmaster hook for HD-audio mute-LED controls
by Takashi Iwai 13 Mar '12

13 Mar '12

This is yet another hack for HD-audio. Currently the mute-LED is controlled via abusing the powerstatus check callback for power-saving feature. Thus the mute-LED won't be enabled when user didn't build the driver with CONFIG_SND_HDA_POWER_SAVE=y. With this patchset, the driver will refer to only Master switch to follow the mute-LED status. The first patch is to add a hook to vmaster control. It's simple and small. The rest are implementations of hooks and replacements of the existing codes with the vmaster hook. The last patch is the addition of the mute-EAPD support on Conexant codec. This was one of the major reason I wanted to implement before 3.4, since it's the biggest missing piece in Conexant auto-parser. I've checked quickly on a few machines. The patches are found in my sound-unstable tree topic/cxt-fix branch, too. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git topic/cxt-fix Let me know if you see any problems with this patchset. thanks, Takashi

2 11

[alsa-devel] [PATCH 00/16] Support U8500/AB8500 ASoC - patch-set 2
by Ola Lilja 13 Mar '12

13 Mar '12

OK, we have done some serious rewriting of the patch-set. And it now consists of several patches, one for each platform-driver and a few more. Generic code is put in specific generic-patches. All code outside the ASoC-driver has been moved to seperate patches and are upstreamed by Linus Walleij. I will reply to the last mail with answers to your comments. PS. These are the full patch-set to be able to build. The following patches has already been upstreamed by Linus Walleij: ARM: ux500: Add DMA-channels for MSP arm: ux500: Add audio-regulators BR, Ola Kristoffer KARLSSON (7): ASoC: core: Add multi reg control struct & macros ASoC: core: Add 8bit multi reg control accessors ASoC: core: Add range of reg control accessors ASoC: core: Add info accessor for mreg control ASoC: core: Add strobe control ASoC: core: Add macros for 8bit hwdep multi reg cntrl ASoC: core: Add macro for hwdep range of regs control Ola Lilja (9): ARM: ux500: Add DMA-channels for MSP arm: ux500: Add audio-regulators arm: ux500: Add support for MSP I2S-devices ARM: ux500: Add placeholder for clk_set_parent ASoC: Ux500: Add MSP I2S-driver ASoC: codecs: Add AB8500 codec-driver ASoC: Ux500: Add platform-driver ASoC: Ux500: Activate the Ux500 ASoC-driver ASoC: Ux500: Add machine-driver arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 194 ++ arch/arm/mach-ux500/board-mop500-regulators.c | 28 + arch/arm/mach-ux500/board-mop500.c | 7 +- arch/arm/mach-ux500/board-mop500.h | 1 + arch/arm/mach-ux500/clock.c | 15 +- arch/arm/mach-ux500/clock.h | 1 + arch/arm/mach-ux500/devices-common.h | 4 +- arch/arm/mach-ux500/devices-db8500.c | 6 + arch/arm/mach-ux500/include/mach/hardware.h | 2 + arch/arm/mach-ux500/include/mach/msp.h | 29 + include/sound/soc.h | 86 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/ab8500_audio.c | 2508 +++++++++++++++++++++++++ sound/soc/codecs/ab8500_audio.h | 622 ++++++ sound/soc/soc-core.c | 239 +++ sound/soc/ux500/Kconfig | 41 + sound/soc/ux500/Makefile | 30 + sound/soc/ux500/u8500.c | 150 ++ sound/soc/ux500/ux500_ab8500.c | 828 ++++++++ sound/soc/ux500/ux500_ab8500.h | 25 + sound/soc/ux500/ux500_msp.h | 936 +++++++++ sound/soc/ux500/ux500_msp_dai.c | 899 +++++++++ sound/soc/ux500/ux500_msp_dai.h | 74 + sound/soc/ux500/ux500_msp_i2s.c | 815 ++++++++ sound/soc/ux500/ux500_msp_i2s.h | 41 + sound/soc/ux500/ux500_pcm.c | 537 ++++++ sound/soc/ux500/ux500_pcm.h | 52 + 31 files changed, 8179 insertions(+), 9 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/include/mach/msp.h create mode 100644 sound/soc/codecs/ab8500_audio.c create mode 100644 sound/soc/codecs/ab8500_audio.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/u8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.h create mode 100644 sound/soc/ux500/ux500_msp.h create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h -- 1.7.8.3

1 0

[alsa-devel] [PATCH] alsa-tools : hwmixvolume - update README for au88x0
by Raymond Yau 13 Mar '12

13 Mar '12

2012/1/24, Takashi Iwai <tiwai(a)suse.de>: > At Tue, 17 Jan 2012 12:05:11 +0800, > Raymond Yau wrote: >> >> [PATCH] ALSA Au88x0 - Implement subdevice volume controls >> >> - add "PCM Playback Volume" controls for 16 playback subdevices >> This allow application to change the volume of each subdevice >> by using hardware mixer of au88x0 and default is zero gain/attenunation. > > Applied this, but to the branch for the next (3.4) kernel, > as this is no urgent "fix". > > [PATCH] alsa-tools: hwmixvolume - update README for au88x0 Need to use alsa-driver kernel 3.4 for Aureal sound cards * Aureal Vortex/Vortex2/Advantage (driver: snd-au8820, snd-au8830, snd-au8810)

2 1

[alsa-devel] [PATCH v3] ASoC: mx27vis-aic32x4: Convert it to platform driver
by Fabio Estevam 12 Mar '12

12 Mar '12

Convert mx27vis-aic32x4 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Tested-by: Javier Martin <javier.martin(a)vista-silicon.com> --- Changes since v2: - Add missing initialization of mx27vis_aic32x4.dev - Add Tested-by line Changes since v1: - Fix MODULE_ALIAS - Rename it to mx27vis arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 41 ++++++++++++++------------ 2 files changed, 23 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c index 0a1643c..ed6023f 100644 --- a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c +++ b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c @@ -397,6 +397,7 @@ static void __init visstrim_m10_board_init(void) &iclink_tvp5150, sizeof(iclink_tvp5150)); gpio_led_register_device(0, &visstrim_m10_led_data); visstrim_camera_init(); + imx_add_platform_device("mx27vis", 0, NULL, 0, NULL, 0); } static void __init visstrim_m10_timer_init(void) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..f6d04ad 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -188,22 +188,16 @@ static struct snd_soc_card mx27vis_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; -static struct platform_device *mx27vis_aic32x4_snd_device; - -static int __init mx27vis_aic32x4_init(void) +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { int ret; - mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_aic32x4_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); - ret = platform_device_add(mx27vis_aic32x4_snd_device); - + mx27vis_aic32x4.dev = &pdev->dev; + ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_aic32x4_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; } /* Connect SSI0 as clock slave to SSI1 external pins */ @@ -221,22 +215,31 @@ static int __init mx27vis_aic32x4_init(void) ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) { + if (ret) printk(KERN_ERR "ASoC: unable to setup gpios\n"); - platform_device_put(mx27vis_aic32x4_snd_device); - } return ret; } -static void __exit mx27vis_aic32x4_exit(void) +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) { - platform_device_unregister(mx27vis_aic32x4_snd_device); + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; } -module_init(mx27vis_aic32x4_init); -module_exit(mx27vis_aic32x4_exit); +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mx27vis"); -- 1.7.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: dmaengine_pcm: Reset pointer position when starting a stream
by Lars-Peter Clausen 12 Mar '12

12 Mar '12

From: Mika Westerberg <mika.westerberg(a)iki.fi> Otherwise a wrong position will be reported after restarting a stream and the first few samples might be skipped. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dmaengine-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 0526cf8..4420b70 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -142,6 +142,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) direction = snd_pcm_substream_to_dma_direction(substream); + prtd->pos = 0; desc = chan->device->device_prep_dma_cyclic(chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), -- 1.7.9.1

3 5

[alsa-devel] [PATCH 3.4 1/2] ASoC: wm8994: Prevent ABBA deadlock with CODEC and accdet mutexes
by Mark Brown 12 Mar '12

12 Mar '12

Currently we can the accdet mutex from within DAPM when updating the device state which means we take accdet then the CODEC mutex but we also do the locking the other way around when responding to the jackdet IRQ. Move all the jackdet use of the CODEC mutex out of the accdet lock to avoid this. Since all the DAPM interactions depend only on a single threaded IRQ this is still serialised. The locking improvements in 3.5 allow a better solution there. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 50 +++++++++++++++++++++++--------------------- 1 files changed, 26 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1a967d9..ac0cd4b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3193,6 +3193,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->codec; int reg; + bool present; mutex_lock(&wm8994->accdet_lock); @@ -3205,11 +3206,10 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) dev_dbg(codec->dev, "JACKDET %x\n", reg); - if (reg & WM1811_JACKDET_LVL) { - dev_dbg(codec->dev, "Jack detected\n"); + present = reg & WM1811_JACKDET_LVL; - snd_soc_jack_report(wm8994->micdet[0].jack, - SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + if (present) { + dev_dbg(codec->dev, "Jack detected\n"); snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); @@ -3227,32 +3227,12 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); - - /* If required for an external cap force MICBIAS on */ - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } } else { dev_dbg(codec->dev, "Jack not detected\n"); snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } - - snd_soc_jack_report(wm8994->micdet[0].jack, 0, - SND_JACK_MECHANICAL | SND_JACK_HEADSET | - wm8994->btn_mask); - /* Enable debounce while removed */ snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, WM1811_JACKDET_DB, WM1811_JACKDET_DB); @@ -3266,6 +3246,28 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) mutex_unlock(&wm8994->accdet_lock); + /* If required for an external cap force MICBIAS on */ + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + + if (present) + snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS2"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); + + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } + + if (present) + snd_soc_jack_report(wm8994->micdet[0].jack, + SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + else + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + return IRQ_HANDLED; } -- 1.7.9.1

1 1

[alsa-devel] [PATCH v2] ASoC: mx27vis-aic32x4: Convert it to platform driver
by Fabio Estevam 12 Mar '12

12 Mar '12

Convert mx27vis-aic32x4 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix MODULE_ALIAS - Rename it to mx27vis arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 40 ++++++++++++++------------- 2 files changed, 22 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c index 0a1643c..ed6023f 100644 --- a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c +++ b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c @@ -397,6 +397,7 @@ static void __init visstrim_m10_board_init(void) &iclink_tvp5150, sizeof(iclink_tvp5150)); gpio_led_register_device(0, &visstrim_m10_led_data); visstrim_camera_init(); + imx_add_platform_device("mx27vis", 0, NULL, 0, NULL, 0); } static void __init visstrim_m10_timer_init(void) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..2554397 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -188,22 +188,15 @@ static struct snd_soc_card mx27vis_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; -static struct platform_device *mx27vis_aic32x4_snd_device; - -static int __init mx27vis_aic32x4_init(void) +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { int ret; - mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_aic32x4_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); - ret = platform_device_add(mx27vis_aic32x4_snd_device); - + ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_aic32x4_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; } /* Connect SSI0 as clock slave to SSI1 external pins */ @@ -221,22 +214,31 @@ static int __init mx27vis_aic32x4_init(void) ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) { + if (ret) printk(KERN_ERR "ASoC: unable to setup gpios\n"); - platform_device_put(mx27vis_aic32x4_snd_device); - } return ret; } -static void __exit mx27vis_aic32x4_exit(void) +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) { - platform_device_unregister(mx27vis_aic32x4_snd_device); + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; } -module_init(mx27vis_aic32x4_init); -module_exit(mx27vis_aic32x4_exit); +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mx27vis"); -- 1.7.1

2 8

[alsa-devel] [GIT PULL] ASoC: Updates for 3.4
by Liam Girdwood 12 Mar '12

12 Mar '12

Mainly Peter's McBSP move from arch/ to sound/ and updates this time. Thanks Liam --- The following changes since commit 303076342b8a30aa98f8a53b539155d0b81ff0c3: Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc (2012-03-06 14:04:16 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc.git for-3.4 Grazvydas Ignotas (1): ASoC: omap-mcbsp: fix snd_pcm_hw_rule_add arguments Peter Ujfalusi (22): OMAP4: panda: Correct cpu version check for 4430 ARM: OMAP: mcbsp: Convert core driver to proper platform driver OMAP: mcbsp: Move core driver under sound/soc/omap ASoC: OMAP: McBSP: Consolidate plat/mcbsp.h content ASoC: omap-mcbsp: Create a single driver for McBSP ARM: OMAP: Do not register omap-mcbsp-dai device ASoC: OMAP: mcbsp.h: Clean up bit definitions ASoC: OMAP McBSP: Remove redundant accessors ASoC: omap-mcbsp: Cleanup of module probe/remove code ASoC: omap-mcbsp: Merge the omap_mcbsp_data into omap_mcbsp structure ASoC: omap-mcbsp: Simplify DMA configuration ASoC: omap: mcbsp: Use uniform st_data pointer initialization ASoC: omap: mcbsp: Remove redundant checks for the st_data pointer ASoC: omap-mcbsp: Configure wakeup in later phase ASoC: omap-mcbsp: When closing the port select PRCM source for CLKS signal ASoC: omap McBSP: Clear rx_irq at probe time for OMAP4 ARM: OMAP2+: McBSP: Correct CLKR/FSR clock source mux configuration ARM/ASoC: OMAP McBSP: Move remainig defines from arch to ASoC header ASoC: omap-mcbsp: Single function CLKR/FSR source mux configuration ASoC: omap-mcbsp: Correct clock muxing for CLKR/FSR signals ARM: OMAP: Remove CONFIG_OMAP_MCBSP references ASoC: OMAP: Build config cleanup for McBSP arch/arm/mach-omap1/Kconfig | 3 - arch/arm/mach-omap1/Makefile | 4 +- arch/arm/mach-omap1/devices.c | 9 - arch/arm/mach-omap1/mcbsp.c | 14 +- arch/arm/mach-omap2/Makefile | 4 +- arch/arm/mach-omap2/board-omap4panda.c | 2 +- arch/arm/mach-omap2/devices.c | 22 - arch/arm/mach-omap2/mcbsp.c | 54 ++- arch/arm/plat-omap/Kconfig | 8 - arch/arm/plat-omap/Makefile | 2 - arch/arm/plat-omap/include/plat/mcbsp.h | 333 -------------- sound/soc/omap/Kconfig | 2 - sound/soc/omap/Makefile | 2 +- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/igep0020.c | 2 +- {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c | 549 +++++------------------- sound/soc/omap/mcbsp.h | 346 +++++++++++++++ sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 321 ++++++++------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/rx51.c | 4 +- sound/soc/omap/sdp3430.c | 4 +- sound/soc/omap/zoom2.c | 4 +- 30 files changed, 706 insertions(+), 1005 deletions(-) rename {arch/arm/plat-omap => sound/soc/omap}/mcbsp.c (66%) create mode 100644 sound/soc/omap/mcbsp.h -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

[alsa-devel] [PATCH 0/3] ASoC: omap-mcbsp: fixes for some underflow problems
by Grazvydas Ignotas 12 Mar '12

12 Mar '12

With these series I'm trying to deal with underflow problems I was complaining about earlier. This mostly tries to get minimum period size large enough for things to work. To illustrate the problems I wrote a testcase with alsalib, hopefully it shows the problem more clearly: http://notaz.gp2x.de/misc/alsa_period_test.c Applies on top of mcbsp move series. Grazvydas Ignotas (3): ASoC: omap-mcbsp: fix snd_pcm_hw_rule_add arguments ASoC: omap-mcbsp: place size constrain on period, not buffer ASoC: omap-mcbsp: make minimum period size larger than FIFO sound/soc/omap/omap-mcbsp.c | 31 +++++++++++++++++++++---------- 1 files changed, 21 insertions(+), 10 deletions(-)

4 19

[alsa-devel] Two patches for Python bindings
by Arvin Schnell 12 Mar '12

12 Mar '12

Hi, please integrate the two attached patches in ALSA. The first one fixes get_volume_array of the Python bindings for mono channels (also reported as bug 5546). The second just updates one of the pyalsa tests (apparently function names changed some time ago). Regards, Arvin -- Arvin Schnell, <aschnell(a)suse.de> Senior Software Engineer, Research & Development SUSE LINUX Products GmbH, GF: Jeff Hawn, Jennifer Guild, Felix Imendörffer, HRB 16746 (AG Nürnberg) Maxfeldstraße 5 90409 Nürnberg Germany

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Export functions for threshold configuration
by Peter Ujfalusi 12 Mar '12

12 Mar '12

ABE has certain requirements towards the FIFO configuration in McBSP. Make it posssible to configure the McBSP FIFO threshold from outside of the McBSP driver stack. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, this patch is needed in preparation to add the ABE support for OMAP4+ platforms. Regards, Peter sound/soc/omap/mcbsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index e5f4444..ca8b028 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -402,6 +402,7 @@ void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) if (threshold && threshold <= mcbsp->max_tx_thres) MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); } +EXPORT_SYMBOL_GPL(omap_mcbsp_set_tx_threshold); /* * omap_mcbsp_set_rx_threshold configures the receive threshold in words. @@ -416,6 +417,7 @@ void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) if (threshold && threshold <= mcbsp->max_rx_thres) MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); } +EXPORT_SYMBOL_GPL(omap_mcbsp_set_rx_threshold); /* * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO -- 1.7.8.5

3 3

[alsa-devel] [PATCH 0/4] mc13783 sound support
by Philippe Rétornaz 12 Mar '12

12 Mar '12

Hello This is patchset enable sound support on boards with mc13783 codec. The first patch adds the codec driver. It's Sascha's code ported to the current kernel with some minors additions. The second patch adds a generic SoC platform driver for imx3 boards with mc13783 codecs. The last two patches enable the sound on the mx31moboard boards and on mx31pdk_3ds boards. This patchset has been tested on mx31moboard and mx31pdk_3ds. It is based on a 3.3-rc6 kernel. I know there are some patches which change how audmux is handled on i.Mx processor but they are not yet merged so I prefer to post this patchset based on a known tree than to add another dependency. If the audmux change is merged I will rebase this patchset. Small note: If you want to test this, make sure you use redboot as bootloader or that you have the patch i.MX 35/5 AIPS Setup applied. Philippe Rétornaz (3): Add imx3sound sound support. mx31moboard: Add ssi configuration imx31pdk: Add sound support Sascha Hauer (1): add a mc13783 codec driver arch/arm/mach-imx/Kconfig | 2 + arch/arm/mach-imx/mach-mx31_3ds.c | 40 ++- arch/arm/mach-imx/mach-mx31moboard.c | 37 ++- drivers/mfd/mc13xxx-core.c | 3 +- include/linux/mfd/mc13xxx.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 1036 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 30 + sound/soc/imx/Kconfig | 10 + sound/soc/imx/Makefile | 2 + sound/soc/imx/imx3sound-mc13783.c | 141 +++++ 12 files changed, 1315 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/imx3sound-mc13783.c

3 12

[alsa-devel] [PATCH 1/2] ASoC: core - Add platform component mutex.
by Liam Girdwood 12 Mar '12

12 Mar '12

Add mutex support for platform IO operations. e.g. can be used for platform DAPM widget IO ops. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + 2 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 82bd773..2ebf787 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,7 @@ struct snd_soc_platform { int id; struct device *dev; struct snd_soc_platform_driver *driver; + struct mutex mutex; unsigned int suspended:1; /* platform is suspended */ unsigned int probed:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7978f6c..c90bb01 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3382,6 +3382,7 @@ int snd_soc_register_platform(struct device *dev, platform->dapm.dev = dev; platform->dapm.platform = platform; platform->dapm.stream_event = platform_drv->stream_event; + mutex_init(&platform->mutex); mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); -- 1.7.5.4

5 15

[alsa-devel] [RFC] Initialize volumes of HD-audio slave ctls
by Takashi Iwai 12 Mar '12

12 Mar '12

Hi, the patch below is an attempt to initialize the volume / mutes of slave controls (such as "Headphone", "Speaker") with vmaster in HD-audio, so that the sound can come out only by changing the master volume/mute. We have thought that such initializations could be done well in alsactl init, but it seems that not everyone installs the latest and greatest alsactl, and there is always a risk that any new controls may be added before alsactl is updated and released. Since the master volume is set muted, the risk by this change should be low. patch_cirrus.c still doesn't support this because it's handling vmaster by itself, but it can be fixed later, too. If anyone has a concern by this, please let me know. thanks, Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0527ae1..d4736b9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -19,6 +19,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/mm.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/slab.h> @@ -2340,6 +2341,52 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) return 1; } +static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) +{ + mm_segment_t fs; + int _tlv[4]; + const int *tlv = NULL; + int val = -1; + + fs = get_fs(); + set_fs(get_ds()); + if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv)) + tlv = _tlv; + } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) + tlv = kctl->tlv.p; + if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) + val = -tlv[2] / tlv[3]; + set_fs(fs); + return val; +} + +static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) +{ + struct snd_ctl_elem_value *ucontrol; + ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL); + if (!ucontrol) + return -ENOMEM; + ucontrol->value.integer.value[0] = val; + ucontrol->value.integer.value[1] = val; + kctl->put(kctl, ucontrol); + kfree(ucontrol); + return 0; +} + +static int init_slave_0dB(void *data, struct snd_kcontrol *slave) +{ + int offset = get_kctl_0dB_offset(slave); + if (offset > 0) + put_kctl_with_value(slave, offset); + return 0; +} + +static int init_slave_unmute(void *data, struct snd_kcontrol *slave) +{ + return put_kctl_with_value(slave, 1); +} + /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec @@ -2347,6 +2394,7 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * @tlv: TLV data (optional) * @slaves: slave control names (optional) * @suffix: suffix string to each slave name (optional) + * @init_slave_vol: initialize slaves to unmute/0dB * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. @@ -2357,9 +2405,9 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * * This function returns zero if successful or a negative error code. */ -int snd_hda_add_vmaster(struct hda_codec *codec, char *name, +int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix) + const char *suffix, bool init_slave_vol) { struct snd_kcontrol *kctl; int err; @@ -2380,9 +2428,16 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, (map_slave_func_t)snd_ctl_add_slave, kctl); if (err < 0) return err; + + /* init with master mute & zero volume */ + put_kctl_with_value(kctl, 0); + if (init_slave_vol) + map_slaves(codec, slaves, suffix, + tlv ? init_slave_0dB : init_slave_unmute, kctl); + return 0; } -EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); +EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 6094dea8..caa6468 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -139,9 +139,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv); struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); -int snd_hda_add_vmaster(struct hda_codec *codec, char *name, +int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix); + const char *suffix, bool init_slave_vol); +#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ + __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true) int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9771b07..f450f2a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -82,6 +82,7 @@ struct ad198x_spec { unsigned int inv_jack_detect: 1;/* inverted jack-detection */ unsigned int inv_eapd: 1; /* inverted EAPD implementation */ unsigned int analog_beep: 1; /* analog beep input present */ + unsigned int avoid_init_slave_vol:1; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -223,11 +224,12 @@ static int ad198x_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", + err = __snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, (spec->slave_vols ? spec->slave_vols : ad_slave_pfxs), - "Playback Volume"); + "Playback Volume", + !spec->avoid_init_slave_vol); if (err < 0) return err; } @@ -3604,6 +3606,7 @@ static int patch_ad1884(struct hda_codec *codec) spec->vmaster_nid = 0x04; /* we need to cover all playback volumes */ spec->slave_vols = ad1884_slave_vols; + spec->avoid_init_slave_vol = 1; codec->patch_ops = ad198x_patch_ops;

3 10

[alsa-devel] [PATCH] ASoC: mx27vis-aic32x4: Convert it to platform driver
by Fabio Estevam 11 Mar '12

11 Mar '12

Convert mx27vis-aic32x4 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 2 + sound/soc/imx/mx27vis-aic32x4.c | 40 ++++++++++++++------------- 2 files changed, 23 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c index 0a1643c..ed5dd41 100644 --- a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c +++ b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c @@ -397,6 +397,8 @@ static void __init visstrim_m10_board_init(void) &iclink_tvp5150, sizeof(iclink_tvp5150)); gpio_led_register_device(0, &visstrim_m10_led_data); visstrim_camera_init(); + imx_add_platform_device("mx27vis_aic32x4", 0, NULL, 0, + NULL, 0); } static void __init visstrim_m10_timer_init(void) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..09448b4 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -188,22 +188,15 @@ static struct snd_soc_card mx27vis_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; -static struct platform_device *mx27vis_aic32x4_snd_device; - -static int __init mx27vis_aic32x4_init(void) +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { int ret; - mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_aic32x4_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); - ret = platform_device_add(mx27vis_aic32x4_snd_device); - + ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_aic32x4_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; } /* Connect SSI0 as clock slave to SSI1 external pins */ @@ -221,22 +214,31 @@ static int __init mx27vis_aic32x4_init(void) ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) { + if (ret) printk(KERN_ERR "ASoC: unable to setup gpios\n"); - platform_device_put(mx27vis_aic32x4_snd_device); - } return ret; } -static void __exit mx27vis_aic32x4_exit(void) +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) { - platform_device_unregister(mx27vis_aic32x4_snd_device); + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; } -module_init(mx27vis_aic32x4_init); -module_exit(mx27vis_aic32x4_exit); +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis_aic32x4", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("mx27vis_aic32x4"); -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: mx27vis-aic32x4: Make sure we are running on the correct machine
by Fabio Estevam 11 Mar '12

11 Mar '12

Only register mx27vis-aic32x4 if we are running on a imx27_visstrim_m10. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/mx27vis-aic32x4.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857..c982bc8 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -194,6 +194,9 @@ static int __init mx27vis_aic32x4_init(void) { int ret; + if (!machine_is_imx27_visstrim_m10()) + return 0; + mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); if (!mx27vis_aic32x4_snd_device) return -ENOMEM; -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Add regulator member to struct dapm_widget.
by Liam Girdwood 11 Mar '12

11 Mar '12

Currently DAPM widgets use the private data for their regulator. Add a regulator * for widgets to use instead of private data. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1beb28d..bde6a52 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -323,6 +323,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct regulator; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -486,6 +487,7 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_context *dapm; void *priv; /* widget specific data */ + struct regulator *regulator; /* attached regulator */ /* dapm control */ short reg; /* negative reg = no direct dapm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e5aa65a..27132db 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -883,9 +883,9 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - return regulator_enable(w->priv); + return regulator_enable(w->regulator); else - return regulator_disable_deferred(w->priv, w->shift); + return regulator_disable_deferred(w->regulator, w->shift); } EXPORT_SYMBOL_GPL(dapm_regulator_event); @@ -2790,9 +2790,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->priv = devm_regulator_get(dapm->dev, w->name); - if (IS_ERR(w->priv)) { - ret = PTR_ERR(w->priv); + w->regulator = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->regulator)) { + ret = PTR_ERR(w->regulator); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); return NULL; -- 1.7.5.4

2 1

Re: [alsa-devel] how to enumerate useful ALSA devices?
by Andrew Eikum 11 Mar '12

11 Mar '12

> The 2008 audio guide recommends > http://0pointer.de/blog/projects/guide-to-sound-apis.html > "Do not use the snd_card_xxx() APIs. For enumerating use > snd_device_name_hint() (and related functions). > snd_card_xxx() is obsolete... Bluetooth audio are not included" > > Is that correct advice? > > Wine still uses snd_card, so it knows only about "default" and > "plughw:x,y". I believe that's wrong but don't know what should be > used instead in order not to come up with a long list of artificial > names mentioned in /usr/share/alsa/alsa.conf such as surround51 when > the poor on-board card knows nothing but 16 bit stereo. > > I'm looking for a list of working devices on this machine, which likely > should contain "default" (usually leading to PA on a recent Ubuntu machine), > "plug:dmix" (PA getting out of the way when trying to use other devices), > "hw:0,0" and presumably not much more. Bump, still hoping for an answer to this. I tried experimenting with snd_device_name_hint(). It gave me what looked like a promising list, consisting of: null front:CARD=SB,DEV=0 surround40:CARD=SB,DEV=0 surround41:CARD=SB,DEV=0 surround50:CARD=SB,DEV=0 surround51:CARD=SB,DEV=0 surround71:CARD=SB,DEV=0 iec958:CARD=SB,DEV=0 This is identical to the list from "aplay -L". But, passing these to aplay fails if I have PulseAudio running ("device in use"). If I kill PA, then they don't fail, but give warnings about rate conversion and suggest the plug plugin. So I stick "plug:" in front of the device name, and they fail again ("unknown parameter front:CARD"). Clearly I'm doing something very wrong. But I don't know what! Then it occurred to me that this list also contains neither "default" nor the "pulse" device in my asound.conf[1]. Both of these work when passed to aplay. Looking to other projects for help, I find that VLC uses the snd_card_ functions, just like Wine currently does. Of course, that fails to find the "pulse" device from asound.conf. And mplayer's libao2 uses a couple of hard-coded device names, which is even worse (right?). So I'm really stuck here. Where am I supposed to get a list of devices that I can present to the user, including their Bluetooth devices and virtual devices from asound.conf? [1] asound.conf from Arch Linux pulseaudio-alsa package: http://projects.archlinux.org/svntogit/packages.git/tree/trunk/asound.conf?… Andrew

4 10

Re: [alsa-devel] Port C Out selector , Port E Out selector of ad1988 and ad1988_3stack_dac_nids
by Raymond Yau 10 Mar '12

10 Mar '12

2011/9/16 Takashi Iwai <tiwai(a)suse.de>: > At Fri, 16 Sep 2011 11:59:54 +0800, > Raymond Yau wrote: >> >> 2011/9/12 Takashi Iwai <tiwai(a)suse.de>: >> > At Wed, 31 Aug 2011 11:21:00 +0800, >> > Raymond Yau wrote: >> >> >> >> HDA - Add Headphone Playback Volume control for ad1988/ad1989 >> >> >> >> - use DAC0 instead of DAC1 for Port-A Headphone >> >> - assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp" >> > >> > Thanks, applied now. >> > >> > >> > Takashi >> > >> >> while testing the headphone volume control , >> >> I discover that the "Port C Out selector" and "Port E Out Selector" >> (i.e. node 0x31 and 0x32 ) by commit in patch_anlog.c >> >> fd66e0d0591dd12eb0bea1e9f3aa194bb93cebbd >> >> >> >> * AD1988 >> * >> * Output pins and routes >> * >> * Pin Mix Sel DAC >> * port-A 0x11 (mute/hp) <- 0x22 <- 0x37 <- 03/04/06 >> * port-B 0x14 (mute/hp) <- 0x2b <- 0x30 <- 03/04/06 >> * port-C 0x15 (mute) <- 0x2c <- 0x31 <- 05/0a >> * port-D 0x12 (mute/hp) <- 0x29 <- 04 >> * port-E 0x17 (mute/hp) <- 0x26 <- 0x32 <- 05/0a >> >> >> >> The connection list of node 0x31 and 0x32 are different from datasheet >> >> >> Table 10. Audio Widget >> >> 31 Audio selector Selects DAC_2 and DAC_4 for PORT-C >> PORT-E Out Selector >> 32 Audio selector Selects DAC_2 and DAC_4 for PORT-E >> PORT-C In Selector >> >> >> >> Codec: Analog Devices AD1988 >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x11d41988 >> Subsystem Id: 0x104381e1 >> Revision Id: 0x100400 >> >> >> Node 0x31 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 2 >> 0x04* 0x0a >> Node 0x32 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 2 >> 0x05* 0x04 >> >> >> >> is the following patch based on real hardware ? >> >> >> commit 8c7c7b8dd2828b42c1230c6b0235e7d1dcf57e5 >> >> hda-codec - Fix surrounds on 3stack mode of AD1988 > > I don't see this commit id in my tree... > > > Takashi > http://git.alsa-project.org/?p=alsa-kernel.git;a=commit;h=fd66e0d0591dd12eb… Thu, 17 Nov 2005 14:31:34 +0000 (15:31 +0100) Add AD1988 codec support to hda-codec driver. http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=f8c7c7b8dd282… Thu, 24 Nov 2005 15:17:20 +0000 (16:17 +0100)] Fixed the swapped surround/CLFE on 3stack mode of AD1988. hda-emu codecs/ad1988a -m "3stack" route -a 0x15 Out[0a] -x Sel[31] -- Mix[2c] -> Pin[15] Out[04] -> Sel[31] -- Mix[2c] -> Pin[15] route -a 0x17 Out[04] -> Sel[32] -- Mix[26] -> Pin[17] Out[05] -x Sel[32] -- Mix[26] -> Pin[17]

2 4

[alsa-devel] [PATCH] USB-Audio: Fixup card name for Logitech web camera
by David Henningsson 10 Mar '12

10 Mar '12

Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index d7c121e..966172f 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -39,7 +39,7 @@ USB-Audio.pcm.iec958_device { # The below don't have digital in/out, so prevent them from being opened. "Logitech USB Headset" 999 - "Logitech Web Camera" 999 + "USB Device 0x46d:0x992" 999 "Blue Snowball" 999 } -- 1.7.9

3 2

[alsa-devel] [PATCH 0/2] ARM/ASoC: OMAP: McBSP config cleanup
by Peter Ujfalusi 09 Mar '12

09 Mar '12

Hi, Since the McBSP driver has been moved out from arch/arm/plat-omap/ and it is combined as a single driver with the sound/soc/omap/omap-mcbsp.c there is no longer need to have CONFIG_OMAP_MCBSP. This series cleans up the Kconfigs, and Makefiles. Also it is fixes the issue that the OMAP audio drivers were not buildable as modules (the ones using McBSP) reported by Grazvydas Ignotas <notasas(a)gmail.com> I have generated this series on top of Liam's for-3.4 branch. The arch/arm/mach-omap* patch applies cleanly on top of todays linux-next, so if this series goes via ASoC we are not going to have conflicts with stuff coming via l-o. Tested on BeagleBoard, and SDP4430, compile tested for OMAP1. This series should go via ASoC along with the rest of the McBSP changes. Regards, Peter --- Peter Ujfalusi (2): ARM: OMAP: Remove CONFIG_OMAP_MCBSP references ASoC: OMAP: Build config cleanup for McBSP arch/arm/mach-omap1/Kconfig | 3 --- arch/arm/mach-omap1/Makefile | 4 +++- arch/arm/mach-omap2/Makefile | 4 +++- sound/soc/omap/Kconfig | 6 ------ sound/soc/omap/Makefile | 3 +-- 5 files changed, 7 insertions(+), 13 deletions(-) -- 1.7.8.5

3 4

[alsa-devel] [PATCH v2] USB-Audio: Blacklist iec958 for some USB devices
by David Henningsson 09 Mar '12

09 Mar '12

Some common USB devices are analog only, so prevent the digital devices from being opened. This simplifies probing the device in PulseAudio. I have a "Logitech USB Headset" myself, the "Logitech Web Camera" was reported by Raymond Yau, and the "Blue Snowball" was reported by Andy Whitcroft. Cc: Andy Whitcroft <andy.whitcroft(a)canonical.com> Cc: Raymond Yau <superquad.vortex2(a)gmail.com> BugLink: https://bugs.launchpad.net/bugs/940145 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 0726c34..d7c121e 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -36,6 +36,11 @@ USB-Audio.pcm.surround40_type { # number for the iec958 device can be changed here. USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 + + # The below don't have digital in/out, so prevent them from being opened. + "Logitech USB Headset" 999 + "Logitech Web Camera" 999 + "Blue Snowball" 999 } -- 1.7.9

3 4

Re: [alsa-devel] amixer: convert percentage into db wrongly
by Takashi Iwai 09 Mar '12

09 Mar '12

At Fri, 9 Mar 2012 18:06:17 +0800, Adam Lee wrote: > > On Fri, Mar 09, 2012 at 10:30:27AM +0100, Takashi Iwai wrote: > > At Fri, 9 Mar 2012 17:07:17 +0800, > > Adam Lee wrote: > > > > > > On Fri, Mar 09, 2012 at 07:40:27AM +0100, Takashi Iwai wrote: > > > > At Fri, 9 Mar 2012 12:22:47 +0800, > > > > Adam Lee wrote: > > > > > > > > > > Add Vincent in cc, because conky read amixer's result. > > > > > > > > > > On Thu, Mar 08, 2012 at 05:45:14PM +0100, Takashi Iwai wrote: > > > > > > > * Adam Lee <adam8157(a)gmail.com> [2012-03-08 20:36 +0800]: > > > > > > > > > > > > > > Package: alsa-utils > > > > > > > Version: 1.0.25-1 > > > > > > > Severity: important > > > > > > > > > > > > > > > > > > > > > db is not linear, but amixer believe it is. > > > > > > > > > > > > > > "amixer get Master" says "Limits: Playback 0 - 74", then everytime I run > > > > > > > "amixer -q sset Master 10%-", there is 8db dec. > > > > > > > > > > > > > > For example, at first Master is 100% and 0db, both alsamixer and amixer > > > > > > > think it is, and after I run "amixer -q sset Master 10%-", both > > > > > > > alsamixer and amixer says Master is -8.00db, but alsamixer says it is > > > > > > > 72%, amixer says it is 89%. > > > > > > > > > > > > > > alsamixer is right, amixer calc and set wrongly. > > > > > > > > > > > > No, both are correct. You are dreaming too much on the world unified > > > > > > percentage representation :) > > > > > > > > > > > > The percentage in amixer has nothing to do with dB level. > > > > > > It's just the percentage of the raw value range of that mixer > > > > > > element. Thus showing 89% is correct. It's 10% down from 100% > > > > > > (1% is because of the resolution of the raw values). > > > > > > > > > > > > Now, alsamixer shows the percentage in a different way. It's > > > > > > explained well in the source code (alsamixer/volume_mapping.c), but > > > > > > not mentioned in the man page, unfortunately. > > > > > > > > > > > > * The mapping is designed so that the position in the interval is proportional > > > > > > * to the volume as a human ear would perceive it (i.e., the position is the > > > > > > * cubic root of the linear sample multiplication factor). For controls with > > > > > > * a small range (24 dB or less), the mapping is linear in the dB values so > > > > > > * that each step has the same size visually. Only for controls without dB > > > > > > * information, a linear mapping of the hardware volume register values is used > > > > > > * (this is the same algorithm as used in the old alsamixer). > > > > > > > > > > > > The percentage representation in alsamixer corresponds to this > > > > > > mapping, thus it's neither dB nor linear percent. > > > > > > > > > > > > > > > > Hi, Takashi > > > > > > > > > > Thank you for replying. But I still insist this is a bug. Three > > > > > questions: > > > > > > > > > > 1, several months ago, it's OK, both amixer and alsamixer use the human > > > > > mapping(0-10% and 90%-100% are the same change by a human ear), why not > > > > > now? > > > > > > > > amixer hasn't been changed until yet. It handles either in raw values > > > > or in dB. No human-ear mapping at all. It's never changed since > > > > years, and won't be changed. If the volume mapping would be > > > > implemented to amixer in future, it must be only optional. > > > > > > > > Only the recent alsamixer introduced the volume mapping to visualize > > > > the volumes reasonably. > > > > > > > > > > OK, thank you. Maybe an optional will make everyone happy. > > > > > > > > 2, conky(Vincent, I mean ${mixer}), some other software, lot of user's > > > > > scripts use amixer to set or get volume, expecting the human mapping, > > > > > why change the behavior? > > > > > > > > You must be smoking something bad. The behavior of amixer hasn't been > > > > changed. > > > > > > > > > > I figured out a reason probably, when the limits range is wide, like > > > 0-65536, amixer's mapping and alsamixer's human-ear mapping are close. > > > > > > If I remember right, my hardware's limits was 0-65536, but it becomes > > > 0-74 after I run "alsactl init", but unfortunately, I don't know how to > > > modify it back. > > > > > > > > 3, alsamixer and amixer use the same dB value, why there is difference > > > > > in percentage? If alsa-utils developer think the human mapping sucks, > > > > > why you guys still use it in alsamixer? There is no "both correct", the > > > > > difference confuses user... > > > > > > > > That's true. alsamixer should have stopped showing the stupid > > > > percentage. > > > > > > > > The biggest understand is that people (including you) think there is > > > > an absolutely perfect percentage definition for the sound level. It's > > > > an illusion. > > > > > > > > > > I don't expect an absolutely perfect percentage definition. I want a > > > human-ear mapping, which alsamixer does well, 100% is about as ten times > > > loud as 10%. > > > > How did you measure _quantitatively_ it's exactly ten times louder? > > And you think 50% is ten times loud as 5% volume, 10% is ten times > > loud as 1%? Things aren't so easy, unfortunately. > > > > > But amixer doesn't work like that, amixer's 100% is about > > > as *one hundred times* as amixer's 10%. > > > > Yes, this is what's amixer expected to behave. It's a value just > > representing the percentage of a "raw value" of the mixer element. > > It never says it's corresponding to any practical volume. This is the > > misunderstanding first of all. > > > > For example, I guess in your 64k case, the raw value is even not in dB > > unit but it's a linear volume. amixer doesn't care. 10% means only > > the 6553 of 65536. That's all. So simple. > > (In addition, amixer can handle dB, e.g. amixer set Master +10dB or > > such. But it's not suitable for percentage unit because dB level > > can't be represented in the absolute percent volume -- what is 10% in > > dB?) > > > > > Maybe it is beautiful, and alsamixer's human-ear mapping is stupid in > > > the sound science universe. But as a common user, I don't think so. And > > > I don't know how you guys put up with it :( > > > > No, you misunderstand what I wrote. alsamixer's mapping is really an > > improvement. That's why it was implemented recently. But if it shows > > a different number, user may wonder, just like you did. If you didn't > > see a number, you didn't notice the difference :) > > > > But amixer wasn't changed. amixer is a tool to manipulate raw > > values. Of course, it'd be also nice to implement the same percentage > > expression, but as already mentioned, it should be activated only via > > an option. The default behavior of amixer must not be changed for > > compatibility reason. > > > > OK. > > Looking forward to the option, if your time and other capacity allow :) The implementation isn't too hard. I wrote a quick patch now as below. It became bigger than I thought since it contains many clean-ups that are needed to adapt the mapped volume. With the patch, amixer shows and reads the percent value with -M option just like in alsamixer. Takashi --- diff --git a/amixer/Makefile.am b/amixer/Makefile.am index fcd0e81..ba67a57 100644 --- a/amixer/Makefile.am +++ b/amixer/Makefile.am @@ -4,7 +4,7 @@ LDADD = -lm # CFLAGS += -g -Wall bin_PROGRAMS = amixer -amixer_SOURCES = amixer.c +amixer_SOURCES = amixer.c ../alsamixer/volume_mapping.c noinst_HEADERS = amixer.h man_MANS = amixer.1 EXTRA_DIST = amixer.1 diff --git a/amixer/amixer.c b/amixer/amixer.c index 9d2855d..a0a7387 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -29,7 +29,9 @@ #include <assert.h> #include <alsa/asoundlib.h> #include <sys/poll.h> +#include <stdint.h> #include "amixer.h" +#include "../alsamixer/volume_mapping.h" #define LEVEL_BASIC (1<<0) #define LEVEL_INACTIVE (1<<1) @@ -68,6 +70,8 @@ static int help(void) printf(" -i,--inactive show also inactive controls\n"); printf(" -a,--abstract L select abstraction level (none or basic)\n"); printf(" -s,--stdin Read and execute commands from stdin sequentially\n"); + printf(" -R,--raw-volume Use the raw value (default)\n"); + printf(" -M,--mapped-volume Use the mapped volume\n"); printf("\nAvailable commands:\n"); printf(" scontrols show all mixer simple controls\n"); printf(" scontents show contents of all mixer simple controls (default command)\n"); @@ -187,9 +191,9 @@ static int convert_db_range(int val, int omin, int omax, int nmin, int nmax) /* Fuction to convert from volume to percentage. val = volume */ -static int convert_prange(int val, int min, int max) +static int convert_prange(long val, long min, long max) { - int range = max - min; + long range = max - min; int tmp; if (range == 0) @@ -204,29 +208,6 @@ static int convert_prange(int val, int min, int max) #define convert_prange1(val, min, max) \ ceil((val) * ((max) - (min)) * 0.01 + (min)) -static const char *get_percent(int val, int min, int max) -{ - static char str[32]; - int p; - - p = convert_prange(val, min, max); - sprintf(str, "%i [%i%%]", val, p); - return str; -} - -#if 0 -static const char *get_percent1(int val, int min, int max, int min_dB, int max_dB) -{ - static char str[32]; - int p, db; - - p = convert_prange(val, min, max); - db = convert_db_range(val, min, max, min_dB, max_dB); - sprintf(str, "%i [%i%%] [%i.%02idB]", val, p, db / 100, abs(db % 100)); - return str; -} -#endif - static long get_integer(char **ptr, long min, long max) { long val = min; @@ -288,26 +269,83 @@ struct volume_ops { int (*get)(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t c, long *value); int (*set)(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t c, - long value); + long value, int dir); }; -enum { VOL_RAW, VOL_DB }; +enum { VOL_RAW, VOL_DB, VOL_MAP }; struct volume_ops_set { int (*has_volume)(snd_mixer_elem_t *elem); - struct volume_ops v[2]; + struct volume_ops v[3]; }; static int set_playback_dB(snd_mixer_elem_t *elem, - snd_mixer_selem_channel_id_t c, long value) + snd_mixer_selem_channel_id_t c, long value, int dir) { - return snd_mixer_selem_set_playback_dB(elem, c, value, 0); + return snd_mixer_selem_set_playback_dB(elem, c, value, dir); } static int set_capture_dB(snd_mixer_elem_t *elem, - snd_mixer_selem_channel_id_t c, long value) + snd_mixer_selem_channel_id_t c, long value, int dir) +{ + return snd_mixer_selem_set_capture_dB(elem, c, value, dir); +} + +static int set_playback_raw_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t c, + long value, int dir) +{ + return snd_mixer_selem_set_playback_volume(elem, c, value); +} + +static int set_capture_raw_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t c, + long value, int dir) +{ + return snd_mixer_selem_set_capture_volume(elem, c, value); +} + +/* FIXME: normalize to int32 space to be compatible with other types */ +#define MAP_VOL_RES (INT32_MAX / 100) + +static int get_mapped_volume_range(snd_mixer_elem_t *elem, + long *pmin, long *pmax) +{ + *pmin = 0; + *pmax = MAP_VOL_RES; + return 0; +} + +static int get_playback_mapped_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t c, + long *value) +{ + *value = (rint)(get_normalized_playback_volume(elem, c) * MAP_VOL_RES); + return 0; +} + +static int set_playback_mapped_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t c, + long value, int dir) { - return snd_mixer_selem_set_capture_dB(elem, c, value, 0); + return set_normalized_playback_volume(elem, c, + (double)value / MAP_VOL_RES, dir); +} + +static int get_capture_mapped_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t c, + long *value) +{ + *value = (rint)(get_normalized_capture_volume(elem, c) * MAP_VOL_RES); + return 0; +} + +static int set_capture_mapped_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t c, + long value, int dir) +{ + return set_normalized_capture_volume(elem, c, + (double)value / MAP_VOL_RES, dir); } static const struct volume_ops_set vol_ops[2] = { @@ -315,29 +353,42 @@ static const struct volume_ops_set vol_ops[2] = { .has_volume = snd_mixer_selem_has_playback_volume, .v = {{ snd_mixer_selem_get_playback_volume_range, snd_mixer_selem_get_playback_volume, - snd_mixer_selem_set_playback_volume }, + set_playback_raw_volume }, { snd_mixer_selem_get_playback_dB_range, snd_mixer_selem_get_playback_dB, - set_playback_dB }}, + set_playback_dB }, + { get_mapped_volume_range, + get_playback_mapped_volume, + set_playback_mapped_volume }, + }, }, { .has_volume = snd_mixer_selem_has_capture_volume, .v = {{ snd_mixer_selem_get_capture_volume_range, snd_mixer_selem_get_capture_volume, - snd_mixer_selem_set_capture_volume }, + set_capture_raw_volume }, { snd_mixer_selem_get_capture_dB_range, snd_mixer_selem_get_capture_dB, - set_capture_dB }}, + set_capture_dB }, + { get_mapped_volume_range, + get_capture_mapped_volume, + set_capture_mapped_volume }, + }, }, }; +static int std_vol_type = VOL_RAW; + static int set_volume_simple(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t chn, char **ptr, int dir) { long val, orig, pmin, pmax; char *p = *ptr, *s; - int invalid = 0, err = 0, vol_type = VOL_RAW; + int invalid = 0, percent = 0, err = 0; + int vol_type = std_vol_type; + double scale = 1.0; + int correct = 0; if (! vol_ops[dir].has_volume(elem)) invalid = 1; @@ -347,10 +398,6 @@ static int set_volume_simple(snd_mixer_elem_t *elem, if (*p == '\0' || (!isdigit(*p) && *p != '-')) goto skip; - if (! invalid && - vol_ops[dir].v[VOL_RAW].get_range(elem, &pmin, &pmax) < 0) - invalid = 1; - s = p; val = strtol(s, &p, 10); if (*p == '.') { @@ -358,32 +405,37 @@ static int set_volume_simple(snd_mixer_elem_t *elem, strtol(p, &p, 10); } if (*p == '%') { - if (! invalid) - val = (long)convert_prange1(strtod(s, NULL), pmin, pmax); + percent = 1; p++; } else if (p[0] == 'd' && p[1] == 'B') { - if (! invalid) { - val = (long)(strtod(s, NULL) * 100.0); - vol_type = VOL_DB; - if (vol_ops[dir].v[vol_type].get_range(elem, &pmin, &pmax) < 0) - invalid = 1; - } + vol_type = VOL_DB; p += 2; - } + scale = 100; + } else + vol_type = VOL_RAW; + + val = (long)(strtod(s, NULL) * scale); + if (vol_ops[dir].v[vol_type].get_range(elem, &pmin, &pmax) < 0) + invalid = 1; + if (percent) + val = (long)convert_prange1(val, pmin, pmax); if (*p == '+' || *p == '-') { if (! invalid) { if (vol_ops[dir].v[vol_type].get(elem, chn, &orig) < 0) invalid = 1; - if (*p == '+') + if (*p == '+') { val = orig + val; - else + correct = -1; + } else { val = orig - val; + correct = 1; + } } p++; } if (! invalid) { val = check_range(val, pmin, pmax); - err = vol_ops[dir].v[vol_type].set(elem, chn, val); + err = vol_ops[dir].v[vol_type].set(elem, chn, val, correct); } skip: if (*p == ',') @@ -711,15 +763,33 @@ static int controls(int level) return 0; } +static void show_selem_volume(snd_mixer_elem_t *elem, + snd_mixer_selem_channel_id_t chn, int dir, + long min, long max) +{ + long raw, val; + vol_ops[dir].v[VOL_RAW].get(elem, chn, &raw); + if (std_vol_type == VOL_RAW) + val = convert_prange(raw, min, max); + else { + vol_ops[dir].v[std_vol_type].get(elem, chn, &val); + val = convert_prange(val, 0, MAP_VOL_RES); + } + printf(" %li [%li%%]", raw, val); + if (!vol_ops[dir].v[VOL_DB].get(elem, chn, &val)) { + printf(" ["); + print_dB(val); + printf("]"); + } +} + static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char *space, int level) { snd_mixer_selem_channel_id_t chn; long pmin = 0, pmax = 0; long cmin = 0, cmax = 0; - long pvol, cvol; int psw, csw; int pmono, cmono, mono_ok = 0; - long db; snd_mixer_elem_t *elem; elem = snd_mixer_find_selem(handle, id); @@ -868,13 +938,7 @@ static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char mono_ok = 1; } if (snd_mixer_selem_has_common_volume(elem)) { - snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_MONO, &pvol); - printf(" %s", get_percent(pvol, pmin, pmax)); - if (!snd_mixer_selem_get_playback_dB(elem, SND_MIXER_SCHN_MONO, &db)) { - printf(" ["); - print_dB(db); - printf("]"); - } + show_selem_volume(elem, SND_MIXER_SCHN_MONO, 0, pmin, pmax); } if (snd_mixer_selem_has_common_switch(elem)) { snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_MONO, &psw); @@ -891,13 +955,7 @@ static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char if (snd_mixer_selem_has_playback_volume(elem)) { printf(" Playback"); title = 1; - snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_MONO, &pvol); - printf(" %s", get_percent(pvol, pmin, pmax)); - if (!snd_mixer_selem_get_playback_dB(elem, SND_MIXER_SCHN_MONO, &db)) { - printf(" ["); - print_dB(db); - printf("]"); - } + show_selem_volume(elem, SND_MIXER_SCHN_MONO, 0, pmin, pmax); } } if (!snd_mixer_selem_has_common_switch(elem)) { @@ -919,13 +977,7 @@ static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char if (snd_mixer_selem_has_capture_volume(elem)) { printf(" Capture"); title = 1; - snd_mixer_selem_get_capture_volume(elem, SND_MIXER_SCHN_MONO, &cvol); - printf(" %s", get_percent(cvol, cmin, cmax)); - if (!snd_mixer_selem_get_capture_dB(elem, SND_MIXER_SCHN_MONO, &db)) { - printf(" ["); - print_dB(db); - printf("]"); - } + show_selem_volume(elem, SND_MIXER_SCHN_MONO, 1, cmin, cmax); } } if (!snd_mixer_selem_has_common_switch(elem)) { @@ -946,13 +998,7 @@ static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char continue; printf("%s%s:", space, snd_mixer_selem_channel_name(chn)); if (!pmono && !cmono && snd_mixer_selem_has_common_volume(elem)) { - snd_mixer_selem_get_playback_volume(elem, chn, &pvol); - printf(" %s", get_percent(pvol, pmin, pmax)); - if (!snd_mixer_selem_get_playback_dB(elem, chn, &db)) { - printf(" ["); - print_dB(db); - printf("]"); - } + show_selem_volume(elem, chn, 0, pmin, pmax); } if (!pmono && !cmono && snd_mixer_selem_has_common_switch(elem)) { snd_mixer_selem_get_playback_switch(elem, chn, &psw); @@ -964,13 +1010,7 @@ static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char if (snd_mixer_selem_has_playback_volume(elem)) { printf(" Playback"); title = 1; - snd_mixer_selem_get_playback_volume(elem, chn, &pvol); - printf(" %s", get_percent(pvol, pmin, pmax)); - if (!snd_mixer_selem_get_playback_dB(elem, chn, &db)) { - printf(" ["); - print_dB(db); - printf("]"); - } + show_selem_volume(elem, chn, 0, pmin, pmax); } } if (!snd_mixer_selem_has_common_switch(elem)) { @@ -988,13 +1028,7 @@ static int show_selem(snd_mixer_t *handle, snd_mixer_selem_id_t *id, const char if (snd_mixer_selem_has_capture_volume(elem)) { printf(" Capture"); title = 1; - snd_mixer_selem_get_capture_volume(elem, chn, &cvol); - printf(" %s", get_percent(cvol, cmin, cmax)); - if (!snd_mixer_selem_get_capture_dB(elem, chn, &db)) { - printf(" ["); - print_dB(db); - printf("]"); - } + show_selem_volume(elem, chn, 1, cmin, cmax); } } if (!snd_mixer_selem_has_common_switch(elem)) { @@ -1927,6 +1961,8 @@ int main(int argc, char *argv[]) {"version", 0, NULL, 'v'}, {"abstract", 1, NULL, 'a'}, {"stdin", 0, NULL, 's'}, + {"raw-volume", 0, NULL, 'R'}, + {"mapped-volume", 0, NULL, 'M'}, {NULL, 0, NULL, 0}, }; @@ -1934,7 +1970,7 @@ int main(int argc, char *argv[]) while (1) { int c; - if ((c = getopt_long(argc, argv, "hc:D:qidnva:s", long_option, NULL)) < 0) + if ((c = getopt_long(argc, argv, "hc:D:qidnva:sRM", long_option, NULL)) < 0) break; switch (c) { case 'h': @@ -1987,6 +2023,12 @@ int main(int argc, char *argv[]) case 's': read_stdin = 1; break; + case 'R': + std_vol_type = VOL_RAW; + break; + case 'M': + std_vol_type = VOL_MAP; + break; default: fprintf(stderr, "Invalid switch or option needs an argument.\n"); morehelp++;

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[alsa-devel] [PATCH 1/2] ASoC: core: Rename card mutex subclass to better align with usage.
by Liam Girdwood 09 Mar '12

09 Mar '12

Change SND_SOC_CARD_CLASS_PCM to SND_SOC_CARD_CLASS_RUNTIME to better describe all uses for this mutex subclass and align with DAPM too. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 66fd9bc..0989987 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -289,8 +289,8 @@ enum snd_soc_pcm_subclass { }; enum snd_soc_card_subclass { - SND_SOC_CARD_CLASS_INIT = 0, - SND_SOC_CARD_CLASS_PCM = 1, + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_RUNTIME = 1, }; int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: da7210: Remove extra registers from defaults list
by Ashish Chavan 09 Mar '12

09 Mar '12

This patch removes following registers from reg map defaults, - Registers which are currently not used by driver - Non existing registers - Volatile registers Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 88 --------------------------------------------- 1 files changed, 0 insertions(+), 88 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0c23f19..7843711 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -631,9 +631,7 @@ struct da7210_priv { }; static struct reg_default da7210_reg_defaults[] = { - { 0x00, 0x00 }, { 0x01, 0x11 }, - { 0x02, 0x00 }, { 0x03, 0x00 }, { 0x04, 0x00 }, { 0x05, 0x00 }, @@ -676,92 +674,6 @@ static struct reg_default da7210_reg_defaults[] = { { 0x2a, 0x00 }, { 0x2b, 0x30 }, { 0x2c, 0x2A }, - { 0x2d, 0x00 }, - { 0x2e, 0x40 }, - { 0x2f, 0x00 }, - { 0x30, 0x40 }, - { 0x31, 0x00 }, - { 0x32, 0x40 }, - { 0x33, 0x00 }, - { 0x34, 0x40 }, - { 0x35, 0x00 }, - { 0x36, 0x40 }, - { 0x37, 0x00 }, - { 0x38, 0x40 }, - { 0x39, 0x00 }, - { 0x3a, 0x40 }, - { 0x3b, 0x00 }, - { 0x3c, 0x40 }, - { 0x3d, 0x00 }, - { 0x3e, 0x00 }, - { 0x3f, 0x00 }, - { 0x40, 0x00 }, - { 0x41, 0x00 }, - { 0x42, 0x00 }, - { 0x43, 0x00 }, - { 0x44, 0x00 }, - { 0x45, 0x00 }, - { 0x46, 0x00 }, - { 0x47, 0x00 }, - { 0x48, 0x00 }, - { 0x49, 0x00 }, - { 0x4a, 0x00 }, - { 0x4b, 0x00 }, - { 0x4c, 0x00 }, - { 0x4d, 0x00 }, - { 0x4e, 0x00 }, - { 0x4f, 0x00 }, - { 0x50, 0x00 }, - { 0x51, 0x00 }, - { 0x52, 0x00 }, - { 0x53, 0x00 }, - { 0x54, 0x00 }, - { 0x55, 0x00 }, - { 0x56, 0x00 }, - { 0x57, 0x00 }, - { 0x58, 0x00 }, - { 0x59, 0x00 }, - { 0x5a, 0x00 }, - { 0x5b, 0x00 }, - { 0x5c, 0x00 }, - { 0x5d, 0x00 }, - { 0x5e, 0x00 }, - { 0x5f, 0x00 }, - { 0x60, 0x00 }, - { 0x61, 0x00 }, - { 0x62, 0x00 }, - { 0x63, 0x00 }, - { 0x64, 0x00 }, - { 0x65, 0x00 }, - { 0x66, 0x00 }, - { 0x67, 0x00 }, - { 0x68, 0x00 }, - { 0x69, 0x00 }, - { 0x6a, 0x00 }, - { 0x6b, 0x00 }, - { 0x6c, 0x00 }, - { 0x6d, 0x00 }, - { 0x6e, 0x00 }, - { 0x6f, 0x00 }, - { 0x70, 0x00 }, - { 0x71, 0x00 }, - { 0x72, 0x00 }, - { 0x73, 0x00 }, - { 0x74, 0x00 }, - { 0x75, 0x00 }, - { 0x76, 0x00 }, - { 0x77, 0x00 }, - { 0x78, 0x00 }, - { 0x79, 0x00 }, - { 0x7a, 0x00 }, - { 0x7b, 0x00 }, - { 0x7c, 0x00 }, - { 0x7d, 0x54 }, - { 0x7e, 0x54 }, - { 0x7f, 0x00 }, - { 0x80, 0x00 }, - { 0x81, 0x00 }, - { 0x82, 0x2c }, { 0x83, 0x00 }, { 0x84, 0x00 }, { 0x85, 0x00 }, -- 1.7.1

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