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[alsa-devel] [PATCH] ALSA: ASoC: McASP: treat SNDRV_PCM_FORMAT_U24_* as 32 bit words
by Daniel Mack 10 Oct '12

10 Oct '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Mike Looijmans <mike.looijmans(a)topic.nl> Cc: Mike Looijmans <mike.looijmans(a)topic.nl> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 501f8b4..82877a9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -868,12 +868,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - case SNDRV_PCM_FORMAT_U24_LE: - case SNDRV_PCM_FORMAT_S24_LE: dma_params->data_type = 3; word_length = DAVINCI_AUDIO_WORD_24; break; + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; -- 1.7.11.4

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[alsa-devel] [PATCH] ASoC: dmaengine: Correct Makefile when sound is built as module
by Peter Ujfalusi 10 Oct '12

10 Oct '12

soc-dmaengine-pcm library need to be part of the snd-soc-core in order to be able to compile ASoC as modules when dmaengine is enabled on the platform. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index bcbf1d0..99f32f7 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,8 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o -snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o -obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o +ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) +snd-soc-core-objs += soc-dmaengine-pcm.o +endif obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ -- 1.7.12

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Re: [alsa-devel] [GIT PULL] Disintegrate UAPI for sound
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 09 Oct 2012 14:58:41 +0100, David Howells wrote: > > Takashi Iwai <tiwai(a)suse.de> wrote: > > > > Can you merge the following branch into the sound tree please. > > > > The changes look OK, but is it targeted for 3.7 or 3.8 in general? > > I assumed the latter. > > Either will do. There are no dependencies on the branch I gave you. OK, then I'm going to take it once when 3.7-rc1 is released. thanks, Takashi

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[alsa-devel] Complex hw buffer with MMAP_COMPLEX?
by Adrian Knoth 09 Oct '12

09 Oct '12

Hi! I'm about to start working on the driver for the new RME MADI FX card with its 194 input and 196 output channels. I've noticed that the device has a rather uncommon hw buffer layout. Let's stick to the input side for the sake of simplicity: 2ch AES, interleaved 8ch MADI 001-008, interleaved 8ch MADI 009-015, interleaved [..] 8ch MADI 184-192, interleaved On the output side, there is another stereo pair for headphones. That is, we're dealing with 25 (or 26 for output) interleaved streams with two or eight channels each. How could we represent such a layout in ALSA, that is, how are we talking to userspace? Would one create subdevices and rejoin them via plughw? Or use MMAP_COMPLEX and then let every application deal with it on its own? It's certainly not a problem to special-case the MADI FX in jackd1 and jackd2, but OTOH, this would require other ALSA apps to do the same (I'm thinking of proprietary custom code). I wonder if we could afford to rearrange the buffers in the driver and expose a single non-interleaved buffer to userspace. WDYT? TIA

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Re: [alsa-devel] [GIT PULL] Disintegrate UAPI for sound
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 09 Oct 2012 14:30:42 +0100, David Howells wrote: > > Can you merge the following branch into the sound tree please. The changes look OK, but is it targeted for 3.7 or 3.8 in general? I assumed the latter. thanks, Takashi > > This is to complete part of the Userspace API (UAPI) disintegration for which > the preparatory patches were pulled recently. After these patches, userspace > headers will be segregated into: > > include/uapi/linux/.../foo.h > > for the userspace interface stuff, and: > > include/linux/.../foo.h > > for the strictly kernel internal stuff. > > --- > The following changes since commit 9e2d8656f5e8aa214e66b462680cf86b210b74a8: > > Merge branch 'akpm' (Andrew's patch-bomb) (2012-10-09 16:23:15 +0900) > > are available in the git repository at: > > > git://git.infradead.org/users/dhowells/linux-headers.git tags/disintegrate-sound-20121009 > > for you to fetch changes up to 674e95ca446369e9ca40e4e92b214a616a52bcd6: > > UAPI: (Scripted) Disintegrate include/sound (2012-10-09 09:49:13 +0100) > > ---------------------------------------------------------------- > UAPI Disintegration 2012-10-09 > > ---------------------------------------------------------------- > David Howells (1): > UAPI: (Scripted) Disintegrate include/sound > > include/sound/Kbuild | 10 - > include/sound/asequencer.h | 594 +---------------- > include/sound/asound.h | 935 +-------------------------- > include/sound/emu10k1.h | 355 +---------- > include/sound/sb16_csp.h | 104 +-- > include/uapi/sound/Kbuild | 10 + > include/uapi/sound/asequencer.h | 614 ++++++++++++++++++ > include/uapi/sound/asound.h | 957 ++++++++++++++++++++++++++++ > include/{ => uapi}/sound/asound_fm.h | 0 > include/{ => uapi}/sound/compress_offload.h | 0 > include/{ => uapi}/sound/compress_params.h | 0 > include/uapi/sound/emu10k1.h | 373 +++++++++++ > include/{ => uapi}/sound/hdsp.h | 0 > include/{ => uapi}/sound/hdspm.h | 0 > include/uapi/sound/sb16_csp.h | 122 ++++ > include/{ => uapi}/sound/sfnt_info.h | 0 > 16 files changed, 2084 insertions(+), 1990 deletions(-) > create mode 100644 include/uapi/sound/asequencer.h > create mode 100644 include/uapi/sound/asound.h > rename include/{ => uapi}/sound/asound_fm.h (100%) > rename include/{ => uapi}/sound/compress_offload.h (100%) > rename include/{ => uapi}/sound/compress_params.h (100%) > create mode 100644 include/uapi/sound/emu10k1.h > rename include/{ => uapi}/sound/hdsp.h (100%) > rename include/{ => uapi}/sound/hdspm.h (100%) > create mode 100644 include/uapi/sound/sb16_csp.h > rename include/{ => uapi}/sound/sfnt_info.h (100%) > . >

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[alsa-devel] [PATCH v2] ALSA: ASoC: McASP: add support for 24 bit samples
by Mike Looijmans 09 Oct '12

09 Oct '12

Sorry for the lack of quoting, but I onle get the digest. These are wrong: + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: These pack a 24-bit sample value in a 32-bit word. The codec will send 32 bits to the McASP, and you should transfer 32 bits to the user, not just 24. Hence, SNDRV_PCM_FORMAT_S24_LE must be treated just like SNDRV_PCM_FORMAT_S32_LE. I've tested that on a DA850-alike board with several TLV320AIC3256 codecs, treating them as 3-byte samples will reasult in invalid data. Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

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Re: [alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 9 Oct 2012 19:23:56 +0800, Daniel J Blueman wrote: > > On 9 October 2012 18:07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 09 Oct 2012 12:04:08 +0200, > > Takashi Iwai wrote: > >> > >> At Tue, 9 Oct 2012 00:34:09 +0800, > >> Daniel J Blueman wrote: > >> > > >> > On 8 October 2012 20:58, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > > At Tue, 25 Sep 2012 13:20:05 +0800, > >> > > Daniel J Blueman wrote: > >> > >> On my Macbook with a discrete Nvidia GPU, there is a race between > >> > >> selecting the integrated GPU and putting the discrete GPU into D3 [1], > >> > >> reliably causing a kernel oops [2]. > >> > >> > >> > >> Introducing a delay of ~1s between the calls prevents this. When the > >> > >> second 'OFF' write path executes, it looks like struct azx at > >> > >> card->private_data hasn't yet been allocated yet [3], so there is > >> > >> likely some locking missing. > >> > > > >> > > It's rather pci_get_drvdata() returning NULL (i.e. card is NULL, thus > >> > > card->private_data causes Oops). Could you check the patch like below > >> > > and see whether you get a kernel warning (but no Oops) or the problem > >> > > gets fixed by shifting the assignment of pci drvdata? > >> > [...] > >> > > >> > Good patching. Calling pci_set_drvdata later prevents the oops in HDA, > >> > though we see unexpected 0x0 responses in the response ring buffer > >> > [1], which we don't see when there's a >~1.5s delay between IGD and > >> > OFF. > >> > >> If the previous patch fixed, it means that the switching occurred > >> during the device was being probed. Maybe a better approach to > >> register the VGA switcheroo after the proper initialization. > >> > >> The patch below is a revised one. Please give it a try. > > > > Also, it's not clear which card spews the spurious response. > > Apply the patch below in addition. > [...] > > hda-intel: 0000:01:00.1: spurious response 0x0:0x0, last cmd=0x1f0004 > $ lspci -s :1:0.1 > 01:00.1 Audio device: NVIDIA Corporation Device 0e1b (rev ff) > > It's the NVIDIA device which presumably hasn't completed it's > transition to D3 at the time the OFF is executed. OK, then could you try the patch below on the top of previous two patches? Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f09ff6c..676c64d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2639,13 +2640,13 @@ static void azx_vs_set_state(struct pci_dev *pci, disabled ? "Disabling" : "Enabling", pci_name(chip->pci)); if (disabled) { + snd_hda_lock_devices(chip->bus); azx_suspend(&pci->dev); chip->disabled = true; - snd_hda_lock_devices(chip->bus); } else { - snd_hda_unlock_devices(chip->bus); chip->disabled = false; azx_resume(&pci->dev); + snd_hda_unlock_devices(chip->bus); } } }

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[alsa-devel] [PATCH] ALSA: hda - remove "Mic Jack Mode" for headset jacks (Latitude Exx30)
by David Henningsson 09 Oct '12

09 Oct '12

Dell Latitude 5x30 and 6x30 series of machines all have a single 4-pin headset jack. Enabling line in mode for such jack is very confusing (you would only get mono input, and would have to use non-standard adapters), so remove the option by default. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bb6c50e..f60b353 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -104,6 +104,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_MODELS }; @@ -204,6 +205,7 @@ struct sigmatel_spec { unsigned int check_volume_offset:1; unsigned int auto_mic:1; unsigned int linear_tone_beep:1; + unsigned int headset_jack:1; /* 4-pin headset jack (hp + mono mic) */ /* gpio lines */ unsigned int eapd_mask; @@ -1684,6 +1686,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_HP_LED] = "hp-led", [STAC_92HD83XXX_HP_INV_LED] = "hp-inv-led", [STAC_92HD83XXX_HP_MIC_LED] = "hp-mic-led", + [STAC_92HD83XXX_HEADSET_JACK] = "headset-jack", }; static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1694,6 +1697,24 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, "unknown Dell", STAC_DELL_S14), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0532, + "Dell Latitude E6230", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0533, + "Dell Latitude E6330", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0534, + "Dell Latitude E6430", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0535, + "Dell Latitude E6530", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x053c, + "Dell Latitude E5430", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x053d, + "Dell Latitude E5530", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0549, + "Dell Latitude E5430", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x057d, + "Dell Latitude E6430s", STAC_92HD83XXX_HEADSET_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0584, + "Dell Latitude E6430U", STAC_92HD83XXX_HEADSET_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028, "Dell Vostro 3500", STAC_DELL_VOSTRO_3500), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, @@ -2855,6 +2876,9 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, char name[22]; if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { + if (spec->headset_jack && snd_hda_get_input_pin_attr(def_conf) + != INPUT_PIN_ATTR_DOCK) + return 0; if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD && nid == spec->line_switch) control = STAC_CTL_WIDGET_IO_SWITCH; @@ -5636,6 +5660,9 @@ again: case STAC_92HD83XXX_HP_MIC_LED: spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ break; + case STAC_92HD83XXX_HEADSET_JACK: + spec->headset_jack = 1; + break; } if (find_mute_led_cfg(codec, default_polarity)) -- 1.7.9.5

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Re: [alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Takashi Iwai 09 Oct '12

09 Oct '12

At Tue, 9 Oct 2012 00:34:09 +0800, Daniel J Blueman wrote: > > On 8 October 2012 20:58, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 25 Sep 2012 13:20:05 +0800, > > Daniel J Blueman wrote: > >> On my Macbook with a discrete Nvidia GPU, there is a race between > >> selecting the integrated GPU and putting the discrete GPU into D3 [1], > >> reliably causing a kernel oops [2]. > >> > >> Introducing a delay of ~1s between the calls prevents this. When the > >> second 'OFF' write path executes, it looks like struct azx at > >> card->private_data hasn't yet been allocated yet [3], so there is > >> likely some locking missing. > > > > It's rather pci_get_drvdata() returning NULL (i.e. card is NULL, thus > > card->private_data causes Oops). Could you check the patch like below > > and see whether you get a kernel warning (but no Oops) or the problem > > gets fixed by shifting the assignment of pci drvdata? > [...] > > Good patching. Calling pci_set_drvdata later prevents the oops in HDA, > though we see unexpected 0x0 responses in the response ring buffer > [1], which we don't see when there's a >~1.5s delay between IGD and > OFF. If the previous patch fixed, it means that the switching occurred during the device was being probed. Maybe a better approach to register the VGA switcheroo after the proper initialization. The patch below is a revised one. Please give it a try. Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f09ff6c..dcbfd29 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -501,6 +501,7 @@ struct azx { /* VGA-switcheroo setup */ unsigned int use_vga_switcheroo:1; + unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ unsigned int disabled:1; /* disabled by VGA-switcher */ @@ -2684,14 +2685,20 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int __devinit register_vga_switcheroo(struct azx *chip) { + int err; + if (!chip->use_vga_switcheroo) return 0; /* FIXME: currently only handling DIS controller * is there any machine with two switchable HDMI audio controllers? */ - return vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, VGA_SWITCHEROO_DIS, chip->bus != NULL); + if (err < 0) + return err; + chip->vga_switcheroo_registered = 1; + return 0; } #else #define init_vga_switcheroo(chip) /* NOP */ @@ -2713,7 +2720,8 @@ static int azx_free(struct azx *chip) if (use_vga_switcheroo(chip)) { if (chip->disabled && chip->bus) snd_hda_unlock_devices(chip->bus); - vga_switcheroo_unregister_client(chip->pci); + if (chip->vga_switcheroo_registered) + vga_switcheroo_unregister_client(chip->pci); } if (chip->initialized) { @@ -3067,14 +3075,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } ok: - err = register_vga_switcheroo(chip); - if (err < 0) { - snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); - azx_free(chip); - return err; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); @@ -3345,6 +3345,13 @@ static int __devinit azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "Error registering VGA-switcheroo client\n"); + goto out_free; + } + dev++; return 0;

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[alsa-devel] [PATCH v2] ALSA: ASoC: McASP: add support for 24 bit samples
by Daniel Mack 09 Oct '12

09 Oct '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> Cc: Matt Porter <mporter(a)ti.com> --- v2: also enable 24bit in the DAI sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1252bad..76728be 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -839,6 +839,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, word_length = DAVINCI_AUDIO_WORD_16; break; + case SNDRV_PCM_FORMAT_U24_3LE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: + dma_params->data_type = 3; + word_length = DAVINCI_AUDIO_WORD_24; + break; + case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: dma_params->data_type = 4; @@ -917,6 +925,10 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE) -- 1.7.11.4

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[alsa-devel] [PATCH 0/5] McASP changes for more flexibility
by Daniel Mack 09 Oct '12

09 Oct '12

Hi, I've been working on the McASP driver lately to support a board on which an AM33xx SoC is connected to a CS4271 via I2S. The CPU is clock master for both LRCLK and BITCLK, but receives its master clock externally. Some changes were necessary in order to get this playing, as the driver seems to have been used with DSP_B formats only. I couldn't the pateches on any Davinci platform, so I rely on other people to test and see if they break anything. Thanks, Daniel Daniel Mack (5): ALSA: ASoC: enable 192KHz support for McASP ALSA: ASoC: McASP: add support for clock dividers ALSA: ASoC: McASP: add support for 24 bit samples ALSA: ASoC: McASP: make AHCLK direction configurable ALSA: ASoC: McASP: set format parameters in dependence of the DAI format sound/soc/davinci/davinci-evm.c | 5 +++ sound/soc/davinci/davinci-mcasp.c | 87 ++++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-mcasp.h | 2 +- sound/soc/davinci/davinci-pcm.c | 16 ++----- 4 files changed, 86 insertions(+), 24 deletions(-) -- 1.7.11.4

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[alsa-devel] [PATCH v2 1/2] ARM: imx: clk: Split SSI clock into 'ipg' and 'per'
by Fabio Estevam 09 Oct '12

09 Oct '12

Currently imx ssi driver has been using only the ipg clock as only slave mode is supported. For mx27 it is necessay to provide both 'ipg' and 'per' clocks in order to get ssi functional. This issue was found on mx27 by unselecting the mmc driver from the kernel ,which resulted on a system that could not play audio. When the mmc driver is selected the required 'per2' clock is provided and it allows the ssi to work. Add the required 'per' clock to the mx27 ssi and a dummy one for the other SoCs. Reported-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Drop per4 from the camera clok, which belonged to a different patch arch/arm/mach-imx/clk-imx21.c | 11 +++++++---- arch/arm/mach-imx/clk-imx25.c | 6 ++++-- arch/arm/mach-imx/clk-imx27.c | 8 ++++---- arch/arm/mach-imx/clk-imx31.c | 11 +++++++---- arch/arm/mach-imx/clk-imx35.c | 10 +++++++--- arch/arm/mach-imx/clk-imx51-imx53.c | 19 +++++++++++++------ 6 files changed, 42 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-imx/clk-imx21.c b/arch/arm/mach-imx/clk-imx21.c index cf65148..2ed49c8 100644 --- a/arch/arm/mach-imx/clk-imx21.c +++ b/arch/arm/mach-imx/clk-imx21.c @@ -51,8 +51,8 @@ static const char *mpll_sel_clks[] = { "fpm", "ckih", }; static const char *spll_sel_clks[] = { "fpm", "ckih", }; enum imx21_clks { - ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, per1, - per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, + dummy, ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, + per1, per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, uart4_ipg_gate, gpt1_ipg_gate, gpt2_ipg_gate, gpt3_ipg_gate, pwm_ipg_gate, sdhc1_ipg_gate, sdhc2_ipg_gate, lcdc_ipg_gate, lcdc_hclk_gate, cspi3_ipg_gate, cspi2_ipg_gate, cspi1_ipg_gate, @@ -72,6 +72,7 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) { int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckil] = imx_clk_fixed("ckil", lref); clk[ckih] = imx_clk_fixed("ckih", href); clk[fpm] = imx_clk_fixed_factor("fpm", "ckil", 512, 1); @@ -174,8 +175,10 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) clk_register_clkdev(clk[gpio_gate], "gpio", NULL); clk_register_clkdev(clk[rtc_gate], "rtc", NULL); clk_register_clkdev(clk[csi_hclk_gate], "csi", NULL); - clk_register_clkdev(clk[ssi1_gate], "ssi1", NULL); - clk_register_clkdev(clk[ssi2_gate], "ssi2", NULL); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[sdhc1_ipg_gate], "sdhc1", NULL); clk_register_clkdev(clk[sdhc2_ipg_gate], "sdhc2", NULL); diff --git a/arch/arm/mach-imx/clk-imx25.c b/arch/arm/mach-imx/clk-imx25.c index d20d479..c81864d 100644 --- a/arch/arm/mach-imx/clk-imx25.c +++ b/arch/arm/mach-imx/clk-imx25.c @@ -222,8 +222,10 @@ int __init mx25_clocks_init(void) clk_register_clkdev(clk[lcdc_ipg], "ipg", "imx-fb.0"); clk_register_clkdev(clk[lcdc_ahb], "ahb", "imx-fb.0"); clk_register_clkdev(clk[wdt_ipg], NULL, "imx2-wdt.0"); - clk_register_clkdev(clk[ssi1_ipg], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[esdhc1_ipg_per], "per", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ipg], "ipg", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ahb], "ahb", "sdhci-esdhc-imx25.0"); diff --git a/arch/arm/mach-imx/clk-imx27.c b/arch/arm/mach-imx/clk-imx27.c index 3b6b640..8cbb6b5 100644 --- a/arch/arm/mach-imx/clk-imx27.c +++ b/arch/arm/mach-imx/clk-imx27.c @@ -236,8 +236,10 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[usb_div], "per", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ipg_gate], "ipg", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ahb_gate], "ahb", "mxc-ehci.2"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.0"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.1"); clk_register_clkdev(clk[nfc_baud_gate], NULL, "mxc_nand.0"); clk_register_clkdev(clk[vpu_baud_gate], "per", "coda-imx27.0"); clk_register_clkdev(clk[vpu_ahb_gate], "ahb", "coda-imx27.0"); @@ -262,8 +264,6 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[scc_ipg_gate], "scc", NULL); clk_register_clkdev(clk[cpu_div], "cpu", NULL); clk_register_clkdev(clk[emi_ahb_gate], "emi_ahb" , NULL); - clk_register_clkdev(clk[ssi1_baud_gate], "bitrate" , "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_baud_gate], "bitrate" , "imx-ssi.1"); mxc_timer_init(MX27_IO_ADDRESS(MX27_GPT1_BASE_ADDR), MX27_INT_GPT1); diff --git a/arch/arm/mach-imx/clk-imx31.c b/arch/arm/mach-imx/clk-imx31.c index 1253af2..61567dc 100644 --- a/arch/arm/mach-imx/clk-imx31.c +++ b/arch/arm/mach-imx/clk-imx31.c @@ -35,8 +35,8 @@ static const char *csi_sel[] = { "upll", "spll", }; static const char *fir_sel[] = { "mcu_main", "upll", "spll" }; enum mx31_clks { - ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, per_div, - per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, + dummy, ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, + per_div, per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, fir_div_post, sdhc1_gate, sdhc2_gate, gpt_gate, epit1_gate, epit2_gate, iim_gate, ata_gate, sdma_gate, cspi3_gate, rng_gate, uart1_gate, uart2_gate, ssi1_gate, i2c1_gate, i2c2_gate, i2c3_gate, hantro_gate, @@ -53,6 +53,7 @@ int __init mx31_clocks_init(unsigned long fref) void __iomem *base = MX31_IO_ADDRESS(MX31_CCM_BASE_ADDR); int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", fref); clk[ckil] = imx_clk_fixed("ckil", 32768); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MXC_CCM_MPCTL); @@ -161,8 +162,10 @@ int __init mx31_clocks_init(unsigned long fref) clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1.0"); clk_register_clkdev(clk[sdhc1_gate], NULL, "mxc-mmc.0"); clk_register_clkdev(clk[sdhc2_gate], NULL, "mxc-mmc.1"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[firi_gate], "firi", NULL); clk_register_clkdev(clk[ata_gate], NULL, "pata_imx"); clk_register_clkdev(clk[rtic_gate], "rtic", NULL); diff --git a/arch/arm/mach-imx/clk-imx35.c b/arch/arm/mach-imx/clk-imx35.c index 177259b..1ab7467 100644 --- a/arch/arm/mach-imx/clk-imx35.c +++ b/arch/arm/mach-imx/clk-imx35.c @@ -50,7 +50,7 @@ static const char *std_sel[] = {"ppll", "arm"}; static const char *ipg_per_sel[] = {"ahb_per_div", "arm_per_div"}; enum mx35_clks { - ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, + dummy, ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, arm_per_div, ahb_per_div, ipg_per, uart_sel, uart_div, esdhc_sel, esdhc1_div, esdhc2_div, esdhc3_div, spdif_sel, spdif_div_pre, spdif_div_post, ssi_sel, ssi1_div_pre, ssi1_div_post, ssi2_div_pre, @@ -88,6 +88,7 @@ int __init mx35_clocks_init() aad = &clk_consumer[0]; } + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", 24000000); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MX35_CCM_MPCTL); clk[ppll] = imx_clk_pllv1("ppll", "ckih", base + MX35_CCM_PPCTL); @@ -234,8 +235,11 @@ int __init mx35_clocks_init() clk_register_clkdev(clk[kpp_gate], NULL, "imx-keypad"); clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1"); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + /* i.mx35 has the i.mx21 type uart */ clk_register_clkdev(clk[uart1_gate], "per", "imx21-uart.0"); clk_register_clkdev(clk[ipg], "ipg", "imx21-uart.0"); diff --git a/arch/arm/mach-imx/clk-imx51-imx53.c b/arch/arm/mach-imx/clk-imx51-imx53.c index a0bf848..46d679c 100644 --- a/arch/arm/mach-imx/clk-imx51-imx53.c +++ b/arch/arm/mach-imx/clk-imx51-imx53.c @@ -273,9 +273,12 @@ static void __init mx5_clocks_common_init(unsigned long rate_ckil, clk_register_clkdev(clk[usboh3_gate], "ipg", "fsl-usb2-udc"); clk_register_clkdev(clk[usboh3_gate], "ahb", "fsl-usb2-udc"); clk_register_clkdev(clk[nfc_gate], NULL, "mxc_nand"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "imx-ssi.2"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "imx-ssi.2"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.2"); clk_register_clkdev(clk[ssi_ext1_gate], "ssi_ext1", NULL); clk_register_clkdev(clk[ssi_ext2_gate], "ssi_ext2", NULL); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); @@ -366,9 +369,13 @@ int __init mx51_clocks_init(unsigned long rate_ckil, unsigned long rate_osc, clk_register_clkdev(clk[esdhc4_ipg_gate], "ipg", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[dummy], "ahb", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[esdhc4_per_gate], "per", "sdhci-esdhc-imx51.3"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "83fcc000.ssi"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "70014000.ssi"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "83fe8000.ssi"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "83fcc000.ssi"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "70014000.ssi"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "83fe8000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fcc000.ssi"); + clk_register_clkdev(clk[dummy], "per", "70014000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fe8000.ssi"); + clk_register_clkdev(clk[nfc_gate], NULL, "83fdb000.nand"); /* set the usboh3 parent to pll2_sw */ -- 1.7.9.5

3 4

Re: [alsa-devel] issue of shdma-base library
by Guennadi Liakhovetski 09 Oct '12

09 Oct '12

Subject: [PATCH] ASoC: fsi: don't reschedule DMA from an atomic context shdma doesn't support transfer re-scheduling or triggering from callbacks or from atomic context. The fsi driver issues DMA transfers from a tasklet context, which is a bug. To fix it convert tasklet to a work. Reported-by: Do Q.Thang <dq-thang(a)jinso.co.jp> Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Thang-san, Please, check, whether the attached patch fixes your issues. It seems to fix them for me. Mark, please, wait with committing this patch until the guys confirm, it really helps. It then will also have to go to 3.6-stable. Thanks Guennadi On Tue, 2 Oct 2012, Do Q.Thang wrote: > Hi Guennadi-san, > > I'm testing uptream kernel (v3.6-rc3 ~) on the kzm-a9-gt & armadillo-800eva > board. > I found 2 issues with the sh-dma driver while testing audio playback. > > issue 1: this log appeared while playback with the aplay, then aplay was > stopped. > > ------- log ----------------------------- > # aplay audio_s16le_44100.wav > Playing WAVE 'audio_s16le_44100.wav' : Signed 16 bit Little Endian, Rate > 44100 Hz, Stereo > aplay: pcm_write:1710: write error: Input/output error > ----------------------------------------- > > issue 2: this log appeared while playback with the aplay, then aplay was > stopped. > ( this issue isn't appear on the kzm-a9-gt board only ) > > ------- log ----------------------------- > # aplay audio_s16le_44100.wav > Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > BUG: scheduling while atomic: kworker/1:1/273/0x00000103 > aplay: pcm_write:1710: write error: Input/output error > ----------------------------------------- > > I think these issues are introduced by "dmaengine:add an shdma-base library" > commit. > > ---------- commit ----------------------- > commit 9a7b8e002e331d0599127f16613c32f425a14f2c > Author: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > Date: Wed May 9 17:09:13 2012 +0200 > > dmaengine: add an shdma-base library > > This patch extracts code from shdma.c, that does not directly deal with > hardware implementation details and can be re-used with diverse DMA > controller variants, found on SH-based SoCs. > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > Cc: Sascha Hauer <s.hauer(a)pengutronix.de> > Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> > ------------------------------------------ > > About issue 1: I found that > The fsi.c uses fsi_dma_do_tasklet() to setup dma-transfer.This is a function > of a tasklet. > When one dma-transfer is finished shdma_chan_ld_cleanup() is called.In this > function > __ld_cleanup() is called to cleanup the schan->ld_queue. In behavior of > __ld_cleanup() > if descriptor is the last chunk its callback is called, then its node is > cleanup in the > next call of __ld_cleanup().In callback of the last descriptor, > fsi_dma_do_tasklet() is > scheduled then is called after the shdma_chan_ld_cleanup() is finished. > But when this issue appear, shdma_chan_ld_cleanup() is called while > shdma_chan_ld_cleanup() > is not finished.At that time, schan->ld_queue is not empty, so in > shdma_tx_submit() > schan->pm_state is set to SHDMA_PM_PENDING, and then dma-transfer isn't > started in > shdma_issue_pending(). > > About issue 2: > I found that this issue is appear when pm_runtime_barrier() is running. > This issue is not appear when disable CONFIG_PM_RUNTIME config. > > What do you think about these issues? > > Best regards > --- > Thang diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0540408..1bb0d58c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -20,6 +20,7 @@ #include <linux/sh_dma.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/workqueue.h> #include <sound/soc.h> #include <sound/sh_fsi.h> @@ -223,7 +224,7 @@ struct fsi_stream { */ struct dma_chan *chan; struct sh_dmae_slave slave; /* see fsi_handler_init() */ - struct tasklet_struct tasklet; + struct work_struct work; dma_addr_t dma; }; @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) snd_pcm_period_elapsed(io->substream); } -static void fsi_dma_do_tasklet(unsigned long data) +static void fsi_dma_do_work(struct work_struct *work) { - struct fsi_stream *io = (struct fsi_stream *)data; + struct fsi_stream *io = container_of(work, struct fsi_stream, work); struct fsi_priv *fsi = fsi_stream_to_priv(io); struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) * FIXME * * In DMAEngine case, codec and FSI cannot be started simultaneously - * since FSI is using tasklet. + * since FSI is using the scheduler work queue. * Therefore, in capture case, probably FSI FIFO will have got * overflow error in this point. * in that case, DMA cannot start transfer until error was cleared. @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param) static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_schedule(&io->tasklet); + schedule_work(&io->work); return 0; } @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev return fsi_stream_probe(fsi, dev); } - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); + INIT_WORK(&io->work, fsi_dma_do_work); return 0; } static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_kill(&io->tasklet); + cancel_work_sync(&io->work); fsi_stream_stop(fsi, io);

2 1

[alsa-devel] [PATCH] ASoC: fix documentation in soc-jack
by Peter Meerwald 09 Oct '12

09 Oct '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> Signed-off-by: Peter Meerwald <pmeerw(a)pmeerw.net> --- sound/soc/soc-jack.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0c17293..b24cd53 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,7 +22,7 @@ /** * snd_soc_jack_new - Create a new jack - * @card: ASoC card + * @codec: ASoC codec * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack @@ -139,12 +139,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); /** * snd_soc_jack_get_type - Based on the mic bias value, this function returns - * the type of jack from the zones delcared in the jack type + * the type of jack from the zones declared in the jack type * + * @jack: ASoC jack * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in * * Based on the mic bias value passed, this function helps identify - * the type of jack from the already delcared jack zones + * the type of jack from the already declared jack zones */ int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: ASoC: ak4104: handle more sample rates
by Daniel Mack 09 Oct '12

09 Oct '12

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak4104.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 31d4483..d14163f 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -98,14 +98,32 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, val = 0; switch (params_rate(params)) { + case 22050: + val |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + val |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; case 44100: val |= IEC958_AES3_CON_FS_44100; break; case 48000: val |= IEC958_AES3_CON_FS_48000; break; - case 32000: - val |= IEC958_AES3_CON_FS_32000; + case 88200: + val |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + val |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + val |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + val |= IEC958_AES3_CON_FS_192000; break; default: dev_err(codec->dev, "unsupported sampling rate\n"); -- 1.7.11.4

2 2

[alsa-devel] [PATCH] ASoC: wm5102: Write register value corrections after SYSCLK is enabled
by Mark Brown 09 Oct '12

09 Oct '12

Evalation of the WM5102 has identified a number of register values which should be written after SYSCLK is enabled on revision A in order to improve performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 552 ++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 551 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index fdd7f27..d89652e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -43,6 +43,556 @@ static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static const struct reg_default wm5102_sysclk_reva_patch[] = { + { 0x3000, 0x2225 }, + { 0x3001, 0x3a03 }, + { 0x3002, 0x0225 }, + { 0x3003, 0x0801 }, + { 0x3004, 0x6249 }, + { 0x3005, 0x0c04 }, + { 0x3006, 0x0225 }, + { 0x3007, 0x5901 }, + { 0x3008, 0xe249 }, + { 0x3009, 0x030d }, + { 0x300a, 0x0249 }, + { 0x300b, 0x2c01 }, + { 0x300c, 0xe249 }, + { 0x300d, 0x4342 }, + { 0x300e, 0xe249 }, + { 0x300f, 0x73c0 }, + { 0x3010, 0x4249 }, + { 0x3011, 0x0c00 }, + { 0x3012, 0x0225 }, + { 0x3013, 0x1f01 }, + { 0x3014, 0x0225 }, + { 0x3015, 0x1e01 }, + { 0x3016, 0x0225 }, + { 0x3017, 0xfa00 }, + { 0x3018, 0x0000 }, + { 0x3019, 0xf000 }, + { 0x301a, 0x0000 }, + { 0x301b, 0xf000 }, + { 0x301c, 0x0000 }, + { 0x301d, 0xf000 }, + { 0x301e, 0x0000 }, + { 0x301f, 0xf000 }, + { 0x3020, 0x0000 }, + { 0x3021, 0xf000 }, + { 0x3022, 0x0000 }, + { 0x3023, 0xf000 }, + { 0x3024, 0x0000 }, + { 0x3025, 0xf000 }, + { 0x3026, 0x0000 }, + { 0x3027, 0xf000 }, + { 0x3028, 0x0000 }, + { 0x3029, 0xf000 }, + { 0x302a, 0x0000 }, + { 0x302b, 0xf000 }, + { 0x302c, 0x0000 }, + { 0x302d, 0xf000 }, + { 0x302e, 0x0000 }, + { 0x302f, 0xf000 }, + { 0x3030, 0x0225 }, + { 0x3031, 0x1a01 }, + { 0x3032, 0x0225 }, + { 0x3033, 0x1e00 }, + { 0x3034, 0x0225 }, + { 0x3035, 0x1f00 }, + { 0x3036, 0x6225 }, + { 0x3037, 0xf800 }, + { 0x3038, 0x0000 }, + { 0x3039, 0xf000 }, + { 0x303a, 0x0000 }, + { 0x303b, 0xf000 }, + { 0x303c, 0x0000 }, + { 0x303d, 0xf000 }, + { 0x303e, 0x0000 }, + { 0x303f, 0xf000 }, + { 0x3040, 0x2226 }, + { 0x3041, 0x3a03 }, + { 0x3042, 0x0226 }, + { 0x3043, 0x0801 }, + { 0x3044, 0x6249 }, + { 0x3045, 0x0c06 }, + { 0x3046, 0x0226 }, + { 0x3047, 0x5901 }, + { 0x3048, 0xe249 }, + { 0x3049, 0x030d }, + { 0x304a, 0x0249 }, + { 0x304b, 0x2c01 }, + { 0x304c, 0xe249 }, + { 0x304d, 0x4342 }, + { 0x304e, 0xe249 }, + { 0x304f, 0x73c0 }, + { 0x3050, 0x4249 }, + { 0x3051, 0x0c00 }, + { 0x3052, 0x0226 }, + { 0x3053, 0x1f01 }, + { 0x3054, 0x0226 }, + { 0x3055, 0x1e01 }, + { 0x3056, 0x0226 }, + { 0x3057, 0xfa00 }, + { 0x3058, 0x0000 }, + { 0x3059, 0xf000 }, + { 0x305a, 0x0000 }, + { 0x305b, 0xf000 }, + { 0x305c, 0x0000 }, + { 0x305d, 0xf000 }, + { 0x305e, 0x0000 }, + { 0x305f, 0xf000 }, + { 0x3060, 0x0000 }, + { 0x3061, 0xf000 }, + { 0x3062, 0x0000 }, + { 0x3063, 0xf000 }, + { 0x3064, 0x0000 }, + { 0x3065, 0xf000 }, + { 0x3066, 0x0000 }, + { 0x3067, 0xf000 }, + { 0x3068, 0x0000 }, + { 0x3069, 0xf000 }, + { 0x306a, 0x0000 }, + { 0x306b, 0xf000 }, + { 0x306c, 0x0000 }, + { 0x306d, 0xf000 }, + { 0x306e, 0x0000 }, + { 0x306f, 0xf000 }, + { 0x3070, 0x0226 }, + { 0x3071, 0x1a01 }, + { 0x3072, 0x0226 }, + { 0x3073, 0x1e00 }, + { 0x3074, 0x0226 }, + { 0x3075, 0x1f00 }, + { 0x3076, 0x6226 }, + { 0x3077, 0xf800 }, + { 0x3078, 0x0000 }, + { 0x3079, 0xf000 }, + { 0x307a, 0x0000 }, + { 0x307b, 0xf000 }, + { 0x307c, 0x0000 }, + { 0x307d, 0xf000 }, + { 0x307e, 0x0000 }, + { 0x307f, 0xf000 }, + { 0x3080, 0x2227 }, + { 0x3081, 0x3a03 }, + { 0x3082, 0x0227 }, + { 0x3083, 0x0801 }, + { 0x3084, 0x6255 }, + { 0x3085, 0x0c04 }, + { 0x3086, 0x0227 }, + { 0x3087, 0x5901 }, + { 0x3088, 0xe255 }, + { 0x3089, 0x030d }, + { 0x308a, 0x0255 }, + { 0x308b, 0x2c01 }, + { 0x308c, 0xe255 }, + { 0x308d, 0x4342 }, + { 0x308e, 0xe255 }, + { 0x308f, 0x73c0 }, + { 0x3090, 0x4255 }, + { 0x3091, 0x0c00 }, + { 0x3092, 0x0227 }, + { 0x3093, 0x1f01 }, + { 0x3094, 0x0227 }, + { 0x3095, 0x1e01 }, + { 0x3096, 0x0227 }, + { 0x3097, 0xfa00 }, + { 0x3098, 0x0000 }, + { 0x3099, 0xf000 }, + { 0x309a, 0x0000 }, + { 0x309b, 0xf000 }, + { 0x309c, 0x0000 }, + { 0x309d, 0xf000 }, + { 0x309e, 0x0000 }, + { 0x309f, 0xf000 }, + { 0x30a0, 0x0000 }, + { 0x30a1, 0xf000 }, + { 0x30a2, 0x0000 }, + { 0x30a3, 0xf000 }, + { 0x30a4, 0x0000 }, + { 0x30a5, 0xf000 }, + { 0x30a6, 0x0000 }, + { 0x30a7, 0xf000 }, + { 0x30a8, 0x0000 }, + { 0x30a9, 0xf000 }, + { 0x30aa, 0x0000 }, + { 0x30ab, 0xf000 }, + { 0x30ac, 0x0000 }, + { 0x30ad, 0xf000 }, + { 0x30ae, 0x0000 }, + { 0x30af, 0xf000 }, + { 0x30b0, 0x0227 }, + { 0x30b1, 0x1a01 }, + { 0x30b2, 0x0227 }, + { 0x30b3, 0x1e00 }, + { 0x30b4, 0x0227 }, + { 0x30b5, 0x1f00 }, + { 0x30b6, 0x6227 }, + { 0x30b7, 0xf800 }, + { 0x30b8, 0x0000 }, + { 0x30b9, 0xf000 }, + { 0x30ba, 0x0000 }, + { 0x30bb, 0xf000 }, + { 0x30bc, 0x0000 }, + { 0x30bd, 0xf000 }, + { 0x30be, 0x0000 }, + { 0x30bf, 0xf000 }, + { 0x30c0, 0x2228 }, + { 0x30c1, 0x3a03 }, + { 0x30c2, 0x0228 }, + { 0x30c3, 0x0801 }, + { 0x30c4, 0x6255 }, + { 0x30c5, 0x0c06 }, + { 0x30c6, 0x0228 }, + { 0x30c7, 0x5901 }, + { 0x30c8, 0xe255 }, + { 0x30c9, 0x030d }, + { 0x30ca, 0x0255 }, + { 0x30cb, 0x2c01 }, + { 0x30cc, 0xe255 }, + { 0x30cd, 0x4342 }, + { 0x30ce, 0xe255 }, + { 0x30cf, 0x73c0 }, + { 0x30d0, 0x4255 }, + { 0x30d1, 0x0c00 }, + { 0x30d2, 0x0228 }, + { 0x30d3, 0x1f01 }, + { 0x30d4, 0x0228 }, + { 0x30d5, 0x1e01 }, + { 0x30d6, 0x0228 }, + { 0x30d7, 0xfa00 }, + { 0x30d8, 0x0000 }, + { 0x30d9, 0xf000 }, + { 0x30da, 0x0000 }, + { 0x30db, 0xf000 }, + { 0x30dc, 0x0000 }, + { 0x30dd, 0xf000 }, + { 0x30de, 0x0000 }, + { 0x30df, 0xf000 }, + { 0x30e0, 0x0000 }, + { 0x30e1, 0xf000 }, + { 0x30e2, 0x0000 }, + { 0x30e3, 0xf000 }, + { 0x30e4, 0x0000 }, + { 0x30e5, 0xf000 }, + { 0x30e6, 0x0000 }, + { 0x30e7, 0xf000 }, + { 0x30e8, 0x0000 }, + { 0x30e9, 0xf000 }, + { 0x30ea, 0x0000 }, + { 0x30eb, 0xf000 }, + { 0x30ec, 0x0000 }, + { 0x30ed, 0xf000 }, + { 0x30ee, 0x0000 }, + { 0x30ef, 0xf000 }, + { 0x30f0, 0x0228 }, + { 0x30f1, 0x1a01 }, + { 0x30f2, 0x0228 }, + { 0x30f3, 0x1e00 }, + { 0x30f4, 0x0228 }, + { 0x30f5, 0x1f00 }, + { 0x30f6, 0x6228 }, + { 0x30f7, 0xf800 }, + { 0x30f8, 0x0000 }, + { 0x30f9, 0xf000 }, + { 0x30fa, 0x0000 }, + { 0x30fb, 0xf000 }, + { 0x30fc, 0x0000 }, + { 0x30fd, 0xf000 }, + { 0x30fe, 0x0000 }, + { 0x30ff, 0xf000 }, + { 0x3100, 0x222b }, + { 0x3101, 0x3a03 }, + { 0x3102, 0x222b }, + { 0x3103, 0x5803 }, + { 0x3104, 0xe26f }, + { 0x3105, 0x030d }, + { 0x3106, 0x626f }, + { 0x3107, 0x2c01 }, + { 0x3108, 0xe26f }, + { 0x3109, 0x4342 }, + { 0x310a, 0xe26f }, + { 0x310b, 0x73c0 }, + { 0x310c, 0x026f }, + { 0x310d, 0x0c00 }, + { 0x310e, 0x022b }, + { 0x310f, 0x1f01 }, + { 0x3110, 0x022b }, + { 0x3111, 0x1e01 }, + { 0x3112, 0x022b }, + { 0x3113, 0xfa00 }, + { 0x3114, 0x0000 }, + { 0x3115, 0xf000 }, + { 0x3116, 0x0000 }, + { 0x3117, 0xf000 }, + { 0x3118, 0x0000 }, + { 0x3119, 0xf000 }, + { 0x311a, 0x0000 }, + { 0x311b, 0xf000 }, + { 0x311c, 0x0000 }, + { 0x311d, 0xf000 }, + { 0x311e, 0x0000 }, + { 0x311f, 0xf000 }, + { 0x3120, 0x022b }, + { 0x3121, 0x0a01 }, + { 0x3122, 0x022b }, + { 0x3123, 0x1e00 }, + { 0x3124, 0x022b }, + { 0x3125, 0x1f00 }, + { 0x3126, 0x622b }, + { 0x3127, 0xf800 }, + { 0x3128, 0x0000 }, + { 0x3129, 0xf000 }, + { 0x312a, 0x0000 }, + { 0x312b, 0xf000 }, + { 0x312c, 0x0000 }, + { 0x312d, 0xf000 }, + { 0x312e, 0x0000 }, + { 0x312f, 0xf000 }, + { 0x3130, 0x0000 }, + { 0x3131, 0xf000 }, + { 0x3132, 0x0000 }, + { 0x3133, 0xf000 }, + { 0x3134, 0x0000 }, + { 0x3135, 0xf000 }, + { 0x3136, 0x0000 }, + { 0x3137, 0xf000 }, + { 0x3138, 0x0000 }, + { 0x3139, 0xf000 }, + { 0x313a, 0x0000 }, + { 0x313b, 0xf000 }, + { 0x313c, 0x0000 }, + { 0x313d, 0xf000 }, + { 0x313e, 0x0000 }, + { 0x313f, 0xf000 }, + { 0x3140, 0x0000 }, + { 0x3141, 0xf000 }, + { 0x3142, 0x0000 }, + { 0x3143, 0xf000 }, + { 0x3144, 0x0000 }, + { 0x3145, 0xf000 }, + { 0x3146, 0x0000 }, + { 0x3147, 0xf000 }, + { 0x3148, 0x0000 }, + { 0x3149, 0xf000 }, + { 0x314a, 0x0000 }, + { 0x314b, 0xf000 }, + { 0x314c, 0x0000 }, + { 0x314d, 0xf000 }, + { 0x314e, 0x0000 }, + { 0x314f, 0xf000 }, + { 0x3150, 0x0000 }, + { 0x3151, 0xf000 }, + { 0x3152, 0x0000 }, + { 0x3153, 0xf000 }, + { 0x3154, 0x0000 }, + { 0x3155, 0xf000 }, + { 0x3156, 0x0000 }, + { 0x3157, 0xf000 }, + { 0x3158, 0x0000 }, + { 0x3159, 0xf000 }, + { 0x315a, 0x0000 }, + { 0x315b, 0xf000 }, + { 0x315c, 0x0000 }, + { 0x315d, 0xf000 }, + { 0x315e, 0x0000 }, + { 0x315f, 0xf000 }, + { 0x3160, 0x0000 }, + { 0x3161, 0xf000 }, + { 0x3162, 0x0000 }, + { 0x3163, 0xf000 }, + { 0x3164, 0x0000 }, + { 0x3165, 0xf000 }, + { 0x3166, 0x0000 }, + { 0x3167, 0xf000 }, + { 0x3168, 0x0000 }, + { 0x3169, 0xf000 }, + { 0x316a, 0x0000 }, + { 0x316b, 0xf000 }, + { 0x316c, 0x0000 }, + { 0x316d, 0xf000 }, + { 0x316e, 0x0000 }, + { 0x316f, 0xf000 }, + { 0x3170, 0x0000 }, + { 0x3171, 0xf000 }, + { 0x3172, 0x0000 }, + { 0x3173, 0xf000 }, + { 0x3174, 0x0000 }, + { 0x3175, 0xf000 }, + { 0x3176, 0x0000 }, + { 0x3177, 0xf000 }, + { 0x3178, 0x0000 }, + { 0x3179, 0xf000 }, + { 0x317a, 0x0000 }, + { 0x317b, 0xf000 }, + { 0x317c, 0x0000 }, + { 0x317d, 0xf000 }, + { 0x317e, 0x0000 }, + { 0x317f, 0xf000 }, + { 0x3180, 0x2001 }, + { 0x3181, 0xf101 }, + { 0x3182, 0x0000 }, + { 0x3183, 0xf000 }, + { 0x3184, 0x0000 }, + { 0x3185, 0xf000 }, + { 0x3186, 0x0000 }, + { 0x3187, 0xf000 }, + { 0x3188, 0x0000 }, + { 0x3189, 0xf000 }, + { 0x318a, 0x0000 }, + { 0x318b, 0xf000 }, + { 0x318c, 0x0000 }, + { 0x318d, 0xf000 }, + { 0x318e, 0x0000 }, + { 0x318f, 0xf000 }, + { 0x3190, 0x0000 }, + { 0x3191, 0xf000 }, + { 0x3192, 0x0000 }, + { 0x3193, 0xf000 }, + { 0x3194, 0x0000 }, + { 0x3195, 0xf000 }, + { 0x3196, 0x0000 }, + { 0x3197, 0xf000 }, + { 0x3198, 0x0000 }, + { 0x3199, 0xf000 }, + { 0x319a, 0x0000 }, + { 0x319b, 0xf000 }, + { 0x319c, 0x0000 }, + { 0x319d, 0xf000 }, + { 0x319e, 0x0000 }, + { 0x319f, 0xf000 }, + { 0x31a0, 0x0000 }, + { 0x31a1, 0xf000 }, + { 0x31a2, 0x0000 }, + { 0x31a3, 0xf000 }, + { 0x31a4, 0x0000 }, + { 0x31a5, 0xf000 }, + { 0x31a6, 0x0000 }, + { 0x31a7, 0xf000 }, + { 0x31a8, 0x0000 }, + { 0x31a9, 0xf000 }, + { 0x31aa, 0x0000 }, + { 0x31ab, 0xf000 }, + { 0x31ac, 0x0000 }, + { 0x31ad, 0xf000 }, + { 0x31ae, 0x0000 }, + { 0x31af, 0xf000 }, + { 0x31b0, 0x0000 }, + { 0x31b1, 0xf000 }, + { 0x31b2, 0x0000 }, + { 0x31b3, 0xf000 }, + { 0x31b4, 0x0000 }, + { 0x31b5, 0xf000 }, + { 0x31b6, 0x0000 }, + { 0x31b7, 0xf000 }, + { 0x31b8, 0x0000 }, + { 0x31b9, 0xf000 }, + { 0x31ba, 0x0000 }, + { 0x31bb, 0xf000 }, + { 0x31bc, 0x0000 }, + { 0x31bd, 0xf000 }, + { 0x31be, 0x0000 }, + { 0x31bf, 0xf000 }, + { 0x31c0, 0x0000 }, + { 0x31c1, 0xf000 }, + { 0x31c2, 0x0000 }, + { 0x31c3, 0xf000 }, + { 0x31c4, 0x0000 }, + { 0x31c5, 0xf000 }, + { 0x31c6, 0x0000 }, + { 0x31c7, 0xf000 }, + { 0x31c8, 0x0000 }, + { 0x31c9, 0xf000 }, + { 0x31ca, 0x0000 }, + { 0x31cb, 0xf000 }, + { 0x31cc, 0x0000 }, + { 0x31cd, 0xf000 }, + { 0x31ce, 0x0000 }, + { 0x31cf, 0xf000 }, + { 0x31d0, 0x0000 }, + { 0x31d1, 0xf000 }, + { 0x31d2, 0x0000 }, + { 0x31d3, 0xf000 }, + { 0x31d4, 0x0000 }, + { 0x31d5, 0xf000 }, + { 0x31d6, 0x0000 }, + { 0x31d7, 0xf000 }, + { 0x31d8, 0x0000 }, + { 0x31d9, 0xf000 }, + { 0x31da, 0x0000 }, + { 0x31db, 0xf000 }, + { 0x31dc, 0x0000 }, + { 0x31dd, 0xf000 }, + { 0x31de, 0x0000 }, + { 0x31df, 0xf000 }, + { 0x31e0, 0x0000 }, + { 0x31e1, 0xf000 }, + { 0x31e2, 0x0000 }, + { 0x31e3, 0xf000 }, + { 0x31e4, 0x0000 }, + { 0x31e5, 0xf000 }, + { 0x31e6, 0x0000 }, + { 0x31e7, 0xf000 }, + { 0x31e8, 0x0000 }, + { 0x31e9, 0xf000 }, + { 0x31ea, 0x0000 }, + { 0x31eb, 0xf000 }, + { 0x31ec, 0x0000 }, + { 0x31ed, 0xf000 }, + { 0x31ee, 0x0000 }, + { 0x31ef, 0xf000 }, + { 0x31f0, 0x0000 }, + { 0x31f1, 0xf000 }, + { 0x31f2, 0x0000 }, + { 0x31f3, 0xf000 }, + { 0x31f4, 0x0000 }, + { 0x31f5, 0xf000 }, + { 0x31f6, 0x0000 }, + { 0x31f7, 0xf000 }, + { 0x31f8, 0x0000 }, + { 0x31f9, 0xf000 }, + { 0x31fa, 0x0000 }, + { 0x31fb, 0xf000 }, + { 0x31fc, 0x0000 }, + { 0x31fd, 0xf000 }, + { 0x31fe, 0x0000 }, + { 0x31ff, 0xf000 }, + { 0x024d, 0xff50 }, + { 0x0252, 0xff50 }, + { 0x0259, 0x0112 }, + { 0x025e, 0x0112 }, +}; + +static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm5102_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static const struct snd_kcontrol_new wm5102_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, 1, 0), @@ -305,7 +855,7 @@ static const struct snd_kcontrol_new wm5102_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5102_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH 2.6.39-rc3] hda_intel: ALSA HD Audio patch for Intel Panther Point DeviceIDs
by Takashi Iwai 09 Oct '12

09 Oct '12

At Wed, 20 Apr 2011 10:59:57 -0700, Seth Heasley wrote: > > This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > --- linux-2.6.39-rc3/sound/pci/hda/hda_intel.c.orig 2011-04-11 17:21:51.000000000 -0700 > +++ linux-2.6.39-rc3/sound/pci/hda/hda_intel.c 2011-04-14 11:44:40.000000000 -0700 > @@ -126,6 +126,7 @@ > "{Intel, ICH10}," > "{Intel, PCH}," > "{Intel, CPT}," > + "{Intel, PPT}," > "{Intel, PBG}," > "{Intel, SCH}," > "{ATI, SB450}," > @@ -2759,6 +2760,8 @@ > { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, > + /* Panther Point */ > + { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* Generic Intel */ >

3 2

[alsa-devel] [PATCH 1/2] ARM: imx: clk: Split SSI clock into 'ipg' and 'per'
by Fabio Estevam 08 Oct '12

08 Oct '12

Currently imx ssi driver has been using only the ipg clock as only slave mode is supported. For mx27 it is necessay to provide both 'ipg' and 'per' clocks in order to get ssi functional. This issue was found on mx27 by unselecting the mmc driver from the kernel ,which resulted on a system that could not play audio. When the mmc driver is selected the required 'per2' clock is provided and it allows the ssi to work. Add the required 'per' clock to the mx27 ssi and a dummy one for the other SoCs. Reported-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- arch/arm/mach-imx/clk-imx21.c | 11 +++++++---- arch/arm/mach-imx/clk-imx25.c | 6 ++++-- arch/arm/mach-imx/clk-imx27.c | 9 +++++---- arch/arm/mach-imx/clk-imx31.c | 11 +++++++---- arch/arm/mach-imx/clk-imx35.c | 12 ++++++++---- arch/arm/mach-imx/clk-imx51-imx53.c | 19 +++++++++++++------ 6 files changed, 44 insertions(+), 24 deletions(-) diff --git a/arch/arm/mach-imx/clk-imx21.c b/arch/arm/mach-imx/clk-imx21.c index cf65148..2ed49c8 100644 --- a/arch/arm/mach-imx/clk-imx21.c +++ b/arch/arm/mach-imx/clk-imx21.c @@ -51,8 +51,8 @@ static const char *mpll_sel_clks[] = { "fpm", "ckih", }; static const char *spll_sel_clks[] = { "fpm", "ckih", }; enum imx21_clks { - ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, per1, - per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, + dummy, ckil, ckih, fpm, mpll_sel, spll_sel, mpll, spll, fclk, hclk, ipg, + per1, per2, per3, per4, uart1_ipg_gate, uart2_ipg_gate, uart3_ipg_gate, uart4_ipg_gate, gpt1_ipg_gate, gpt2_ipg_gate, gpt3_ipg_gate, pwm_ipg_gate, sdhc1_ipg_gate, sdhc2_ipg_gate, lcdc_ipg_gate, lcdc_hclk_gate, cspi3_ipg_gate, cspi2_ipg_gate, cspi1_ipg_gate, @@ -72,6 +72,7 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) { int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckil] = imx_clk_fixed("ckil", lref); clk[ckih] = imx_clk_fixed("ckih", href); clk[fpm] = imx_clk_fixed_factor("fpm", "ckil", 512, 1); @@ -174,8 +175,10 @@ int __init mx21_clocks_init(unsigned long lref, unsigned long href) clk_register_clkdev(clk[gpio_gate], "gpio", NULL); clk_register_clkdev(clk[rtc_gate], "rtc", NULL); clk_register_clkdev(clk[csi_hclk_gate], "csi", NULL); - clk_register_clkdev(clk[ssi1_gate], "ssi1", NULL); - clk_register_clkdev(clk[ssi2_gate], "ssi2", NULL); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[sdhc1_ipg_gate], "sdhc1", NULL); clk_register_clkdev(clk[sdhc2_ipg_gate], "sdhc2", NULL); diff --git a/arch/arm/mach-imx/clk-imx25.c b/arch/arm/mach-imx/clk-imx25.c index d20d479..c81864d 100644 --- a/arch/arm/mach-imx/clk-imx25.c +++ b/arch/arm/mach-imx/clk-imx25.c @@ -222,8 +222,10 @@ int __init mx25_clocks_init(void) clk_register_clkdev(clk[lcdc_ipg], "ipg", "imx-fb.0"); clk_register_clkdev(clk[lcdc_ahb], "ahb", "imx-fb.0"); clk_register_clkdev(clk[wdt_ipg], NULL, "imx2-wdt.0"); - clk_register_clkdev(clk[ssi1_ipg], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[esdhc1_ipg_per], "per", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ipg], "ipg", "sdhci-esdhc-imx25.0"); clk_register_clkdev(clk[esdhc1_ahb], "ahb", "sdhci-esdhc-imx25.0"); diff --git a/arch/arm/mach-imx/clk-imx27.c b/arch/arm/mach-imx/clk-imx27.c index 3b6b640..453b4f3 100644 --- a/arch/arm/mach-imx/clk-imx27.c +++ b/arch/arm/mach-imx/clk-imx27.c @@ -224,6 +224,7 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[lcdc_ipg_gate], "ipg", "imx-fb.0"); clk_register_clkdev(clk[lcdc_ahb_gate], "ahb", "imx-fb.0"); clk_register_clkdev(clk[csi_ahb_gate], "ahb", "mx2-camera.0"); + clk_register_clkdev(clk[per4_gate], "per", "mx2-camera.0"); clk_register_clkdev(clk[usb_div], "per", "fsl-usb2-udc"); clk_register_clkdev(clk[usb_ipg_gate], "ipg", "fsl-usb2-udc"); clk_register_clkdev(clk[usb_ahb_gate], "ahb", "fsl-usb2-udc"); @@ -236,8 +237,10 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[usb_div], "per", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ipg_gate], "ipg", "mxc-ehci.2"); clk_register_clkdev(clk[usb_ahb_gate], "ahb", "mxc-ehci.2"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.0"); + clk_register_clkdev(clk[per2_gate], "per", "imx-ssi.1"); clk_register_clkdev(clk[nfc_baud_gate], NULL, "mxc_nand.0"); clk_register_clkdev(clk[vpu_baud_gate], "per", "coda-imx27.0"); clk_register_clkdev(clk[vpu_ahb_gate], "ahb", "coda-imx27.0"); @@ -262,8 +265,6 @@ int __init mx27_clocks_init(unsigned long fref) clk_register_clkdev(clk[scc_ipg_gate], "scc", NULL); clk_register_clkdev(clk[cpu_div], "cpu", NULL); clk_register_clkdev(clk[emi_ahb_gate], "emi_ahb" , NULL); - clk_register_clkdev(clk[ssi1_baud_gate], "bitrate" , "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_baud_gate], "bitrate" , "imx-ssi.1"); mxc_timer_init(MX27_IO_ADDRESS(MX27_GPT1_BASE_ADDR), MX27_INT_GPT1); diff --git a/arch/arm/mach-imx/clk-imx31.c b/arch/arm/mach-imx/clk-imx31.c index 1253af2..a653ecf 100644 --- a/arch/arm/mach-imx/clk-imx31.c +++ b/arch/arm/mach-imx/clk-imx31.c @@ -35,8 +35,8 @@ static const char *csi_sel[] = { "upll", "spll", }; static const char *fir_sel[] = { "mcu_main", "upll", "spll" }; enum mx31_clks { - ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, per_div, - per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, + dummy, ckih, ckil, mpll, spll, upll, mcu_main, hsp, ahb, nfc, ipg, + per_div, per, csi, fir, csi_div, usb_div_pre, usb_div_post, fir_div_pre, fir_div_post, sdhc1_gate, sdhc2_gate, gpt_gate, epit1_gate, epit2_gate, iim_gate, ata_gate, sdma_gate, cspi3_gate, rng_gate, uart1_gate, uart2_gate, ssi1_gate, i2c1_gate, i2c2_gate, i2c3_gate, hantro_gate, @@ -53,6 +53,7 @@ int __init mx31_clocks_init(unsigned long fref) void __iomem *base = MX31_IO_ADDRESS(MX31_CCM_BASE_ADDR); int i; + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", fref); clk[ckil] = imx_clk_fixed("ckil", 32768); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MXC_CCM_MPCTL); @@ -161,8 +162,10 @@ int __init mx31_clocks_init(unsigned long fref) clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1.0"); clk_register_clkdev(clk[sdhc1_gate], NULL, "mxc-mmc.0"); clk_register_clkdev(clk[sdhc2_gate], NULL, "mxc-mmc.1"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); clk_register_clkdev(clk[firi_gate], "firi", NULL); clk_register_clkdev(clk[ata_gate], NULL, "pata_imx"); clk_register_clkdev(clk[rtic_gate], "rtic", NULL); diff --git a/arch/arm/mach-imx/clk-imx35.c b/arch/arm/mach-imx/clk-imx35.c index 177259b..0d952c7 100644 --- a/arch/arm/mach-imx/clk-imx35.c +++ b/arch/arm/mach-imx/clk-imx35.c @@ -50,7 +50,7 @@ static const char *std_sel[] = {"ppll", "arm"}; static const char *ipg_per_sel[] = {"ahb_per_div", "arm_per_div"}; enum mx35_clks { - ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, + dummy, ckih, mpll, ppll, mpll_075, arm, hsp, hsp_div, hsp_sel, ahb, ipg, arm_per_div, ahb_per_div, ipg_per, uart_sel, uart_div, esdhc_sel, esdhc1_div, esdhc2_div, esdhc3_div, spdif_sel, spdif_div_pre, spdif_div_post, ssi_sel, ssi1_div_pre, ssi1_div_post, ssi2_div_pre, @@ -87,7 +87,8 @@ int __init mx35_clocks_init() */ aad = &clk_consumer[0]; } - + + clk[dummy] = imx_clk_fixed("dummy", 0); clk[ckih] = imx_clk_fixed("ckih", 24000000); clk[mpll] = imx_clk_pllv1("mpll", "ckih", base + MX35_CCM_MPCTL); clk[ppll] = imx_clk_pllv1("ppll", "ckih", base + MX35_CCM_PPCTL); @@ -234,8 +235,11 @@ int __init mx35_clocks_init() clk_register_clkdev(clk[kpp_gate], NULL, "imx-keypad"); clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1"); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + /* i.mx35 has the i.mx21 type uart */ clk_register_clkdev(clk[uart1_gate], "per", "imx21-uart.0"); clk_register_clkdev(clk[ipg], "ipg", "imx21-uart.0"); diff --git a/arch/arm/mach-imx/clk-imx51-imx53.c b/arch/arm/mach-imx/clk-imx51-imx53.c index a0bf848..46d679c 100644 --- a/arch/arm/mach-imx/clk-imx51-imx53.c +++ b/arch/arm/mach-imx/clk-imx51-imx53.c @@ -273,9 +273,12 @@ static void __init mx5_clocks_common_init(unsigned long rate_ckil, clk_register_clkdev(clk[usboh3_gate], "ipg", "fsl-usb2-udc"); clk_register_clkdev(clk[usboh3_gate], "ahb", "fsl-usb2-udc"); clk_register_clkdev(clk[nfc_gate], NULL, "mxc_nand"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "imx-ssi.2"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "imx-ssi.2"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.0"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.1"); + clk_register_clkdev(clk[dummy], "per", "imx-ssi.2"); clk_register_clkdev(clk[ssi_ext1_gate], "ssi_ext1", NULL); clk_register_clkdev(clk[ssi_ext2_gate], "ssi_ext2", NULL); clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); @@ -366,9 +369,13 @@ int __init mx51_clocks_init(unsigned long rate_ckil, unsigned long rate_osc, clk_register_clkdev(clk[esdhc4_ipg_gate], "ipg", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[dummy], "ahb", "sdhci-esdhc-imx51.3"); clk_register_clkdev(clk[esdhc4_per_gate], "per", "sdhci-esdhc-imx51.3"); - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "83fcc000.ssi"); - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "70014000.ssi"); - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "83fe8000.ssi"); + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "83fcc000.ssi"); + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "70014000.ssi"); + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "83fe8000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fcc000.ssi"); + clk_register_clkdev(clk[dummy], "per", "70014000.ssi"); + clk_register_clkdev(clk[dummy], "per", "83fe8000.ssi"); + clk_register_clkdev(clk[nfc_gate], NULL, "83fdb000.nand"); /* set the usboh3 parent to pll2_sw */ -- 1.7.9.5

1 1

[alsa-devel] [PATCH] pcm: support for audio timestamps
by Pierre-Louis Bossart 08 Oct '12

08 Oct '12

add new snd_pcm_status_get_audio_htstamp() routine to query the audio timestamps provided by the kernel. This change provides applications with better ways to track elapsed time. Before this patch, applications would subtract queued samples (delay) from written samples, resulting in a 1-2 sample error. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++-- src/pcm/pcm.c | 31 ++++++++++++++++++++++ test/audio_time.c | 68 +++++++++++++++++++++++++++++++++++++++++------- 4 files changed, 96 insertions(+), 12 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 290593b..1ce91e7 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -631,6 +631,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -939,6 +940,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index e24d144..16d03e8 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 65c7646..5880057 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3124,6 +3124,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6214,6 +6234,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * diff --git a/test/audio_time.c b/test/audio_time.c index a910783..03817c7 100644 --- a/test/audio_time.c +++ b/test/audio_time.c @@ -33,6 +33,7 @@ long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, snd_htimestamp_t *trigger_timestamp, + snd_htimestamp_t *audio_timestamp, snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) { int err; @@ -45,6 +46,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, } snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); snd_pcm_status_get_htstamp(status, timestamp); + snd_pcm_status_get_audio_htstamp(status, audio_timestamp); *avail = snd_pcm_status_get_avail(status); *delay = snd_pcm_status_get_delay(status); } @@ -53,6 +55,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, #define PCM_LINK /* sync start for playback and capture */ #define TRACK_CAPTURE /* dump capture timing info */ #define TRACK_PLAYBACK /* dump playback timing info */ +#define TRACK_SAMPLE_COUNTS /* show difference between sample counters and audiotimestamps returned by driver */ #define PLAYBACK_BUFFERS 4 @@ -65,9 +68,13 @@ int main(void) snd_pcm_sframes_t frames; snd_htimestamp_t tstamp_c, tstamp_p; snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + snd_htimestamp_t audio_tstamp_c, audio_tstamp_p; unsigned char buffer_p[PERIOD*4*4]; unsigned char buffer_c[PERIOD*4*4]; + snd_pcm_hw_params_t *hwparams_p; + snd_pcm_hw_params_t *hwparams_c; + snd_pcm_sw_params_t *swparams_p; snd_pcm_sw_params_t *swparams_c; @@ -94,6 +101,18 @@ int main(void) goto _exit; } + snd_pcm_hw_params_alloca(&hwparams_p); + /* get the current hwparams */ + err = snd_pcm_hw_params_current(handle_p, hwparams_p); + if (err < 0) { + printf("Unable to determine current hwparams_p: %s\n", snd_strerror(err)); + goto _exit; + } + if (snd_pcm_hw_params_supports_audio_wallclock_ts(hwparams_p)) + printf("Playback relies on audio wallclock timestamps\n"); + else + printf("Playback relies on audio sample counter timestamps\n"); + snd_pcm_sw_params_alloca(&swparams_p); /* get the current swparams */ err = snd_pcm_sw_params_current(handle_p, swparams_p); @@ -131,6 +150,18 @@ int main(void) goto _exit; } + snd_pcm_hw_params_alloca(&hwparams_c); + /* get the current hwparams */ + err = snd_pcm_hw_params_current(handle_c, hwparams_c); + if (err < 0) { + printf("Unable to determine current hwparams_c: %s\n", snd_strerror(err)); + goto _exit; + } + if (snd_pcm_hw_params_supports_audio_wallclock_ts(hwparams_c)) + printf("Capture relies on audio wallclock timestamps\n"); + else + printf("Capture relies on audio sample counter timestamps\n"); + snd_pcm_sw_params_alloca(&swparams_c); /* get the current swparams */ err = snd_pcm_sw_params_current(handle_c, swparams_c); @@ -202,26 +233,43 @@ int main(void) frame_count_p += frames; #if defined(TRACK_PLAYBACK) - gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &audio_tstamp_p, &avail_p, &delay_p); +#if defined(TRACK_SAMPLE_COUNTS) curr_count_p = frame_count_p - delay_p; /* written minus queued */ - printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_p,trigger_tstamp_p), - (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), - timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + printf("playback: curr_count %lli driver count %lli, delta %lli\n", + (long long)curr_count_p * 1000000000LL / 48000 , + timestamp2ns(audio_tstamp_p), + (long long)curr_count_p * 1000000000LL / 48000 - timestamp2ns(audio_tstamp_p) + ); +#endif + + printf("playback: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_p, trigger_tstamp_p), + timestamp2ns(audio_tstamp_p), + timediff(tstamp_p, trigger_tstamp_p) - timestamp2ns(audio_tstamp_p) ); #endif #if defined(TRACK_CAPTURE) - gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &audio_tstamp_c, &avail_c, &delay_c); +#if defined(TRACK_SAMPLE_COUNTS) curr_count_c = frame_count_c + delay_c; /* read plus queued */ - printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_c,trigger_tstamp_c), - (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), - timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + + printf("capture: curr_count %lli driver count %lli, delta %lli\n", + (long long)curr_count_c * 1000000000LL / 48000 , + timestamp2ns(audio_tstamp_c), + (long long)curr_count_c * 1000000000LL / 48000 - timestamp2ns(audio_tstamp_c) + ); +#endif + + printf("\t capture: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_c, trigger_tstamp_c), + timestamp2ns(audio_tstamp_c), + timediff(tstamp_c, trigger_tstamp_c) - timestamp2ns(audio_tstamp_c) ); #endif -- 1.7.9.5

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - Remove dead GPIO code for VIA codec
by David Henningsson 08 Oct '12

08 Oct '12

>From what I can conclude all GPIO handling was removed in 2009. Remove dead code remnants. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 47 --------------------------------------------- 1 file changed, 47 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 5a45a91..430cb33 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -299,7 +299,6 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) #define VIA_JACK_EVENT 0x20 #define VIA_HP_EVENT 0x01 -#define VIA_GPIO_EVENT 0x02 #define VIA_LINE_EVENT 0x03 enum { @@ -1685,50 +1684,6 @@ static void via_hp_automute(struct hda_codec *codec) via_line_automute(codec, present); } -static void via_gpio_control(struct hda_codec *codec) -{ - unsigned int gpio_data; - unsigned int vol_counter; - unsigned int vol; - unsigned int master_vol; - - struct via_spec *spec = codec->spec; - - gpio_data = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0) & 0x03; - - vol_counter = (snd_hda_codec_read(codec, codec->afg, 0, - 0xF84, 0) & 0x3F0000) >> 16; - - vol = vol_counter & 0x1F; - master_vol = snd_hda_codec_read(codec, 0x1A, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT); - - if (gpio_data == 0x02) { - /* unmute line out */ - snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], - PIN_OUT); - if (vol_counter & 0x20) { - /* decrease volume */ - if (vol > master_vol) - vol = master_vol; - snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, - 0, HDA_AMP_VOLMASK, - master_vol-vol); - } else { - /* increase volume */ - snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, 0, - HDA_AMP_VOLMASK, - ((master_vol+vol) > 0x2A) ? 0x2A : - (master_vol+vol)); - } - } else if (!(gpio_data & 0x02)) { - /* mute line out */ - snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0); - } -} - /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1743,8 +1698,6 @@ static void via_unsol_event(struct hda_codec *codec, if (res == VIA_HP_EVENT || res == VIA_LINE_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) - via_gpio_control(codec); snd_hda_jack_report_sync(codec); } -- 1.7.9.5

2 3

[alsa-devel] [3.6-rc7] switcheroo race with Intel HDA...
by Daniel J Blueman 08 Oct '12

08 Oct '12

On my Macbook with a discrete Nvidia GPU, there is a race between selecting the integrated GPU and putting the discrete GPU into D3 [1], reliably causing a kernel oops [2]. Introducing a delay of ~1s between the calls prevents this. When the second 'OFF' write path executes, it looks like struct azx at card->private_data hasn't yet been allocated yet [3], so there is likely some locking missing. I'm happy to perform further testing and debug of course... Thanks, Daniel --- [1] echo IGD > /sys/kernel/debug/vgaswitcheroo/switch echo OFF > /sys/kernel/debug/vgaswitcheroo/switch --- [2] BUG: unable to handle kernel NULL pointer dereference at 0000000000000170 IP: [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] PGD 259c26067 PUD 25a0fd067 PMD 0 Oops: 0000 [#1] SMP DEBUG_PAGEALLOC Modules linked in: snd_hda_codec_hdmi bnep rfcomm b43 joydev nfsd ssb nfs_acl auth_rpcgss binfmt_misc nfs lockd sunrpc uvcvideo bcm5974 videobuf2_core videobuf2_vmalloc videobuf2_memops coretemp kvm_intel snd_hda_codec_cirrus kvm applesmc input_polldev microcode bcma lpc_ich mfd_core mei snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm snd_timer snd snd_page_alloc nls_iso8859_1 apple_gmux mac_hid apple_bl btrfs hid_apple sdhci_pci ghash_clmulni_intel tg3 sdhci i915 nouveau ttm drm_kms_helper hwmon mxm_wmi video CPU 2 Pid: 961, comm: sh Not tainted 3.6.0-rc7 #2 Apple Inc. MacBookPro10,1/Mac-C3EC7CD22292981F RIP: 0010:[<ffffffffa01ba936>] [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] RSP: 0018:ffff880264271e48 EFLAGS: 00010286 RAX: 0000000000000000 RBX: ffff88025a2f5280 RCX: 0000000000000000 RDX: 0000000000000006 RSI: 0000000000000000 RDI: ffff880265479098 RBP: ffff880264271e68 R08: 2222222222222222 R09: 2222222222222222 R10: 0000000000000000 R11: 0000000000000000 R12: ffff880265479098 R13: 0000000000000000 R14: ffff880264271f50 R15: 0000000000000000 FS: 00007fa4fe183700(0000) GS:ffff88026f280000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000170 CR3: 00000002641a7000 CR4: 00000000001407e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process sh (pid: 961, threadinfo ffff880264270000, task ffff880264503a00) Stack: 2222222222222222 ffff88025a2f5280 0000000000000000 ffff880264271e98 ffff880264271e88 ffffffff812e83a7 ffff8802622835c0 0000000000000004 ffff880264271ef8 ffffffff812e89ac ffff88020a46464f ffff880264503a00 Call Trace: [<ffffffff812e83a7>] set_audio_state+0x67/0x70 [<ffffffff812e89ac>] vga_switcheroo_debugfs_write+0xbc/0x380 [<ffffffff81108773>] vfs_write+0xa3/0x160 [<ffffffff81108a75>] sys_write+0x45/0xa0 [<ffffffff815231a6>] system_call_fastpath+0x1a/0x1f Code: 00 00 00 00 00 55 48 89 e5 48 83 ec 20 4c 89 65 f0 4c 8d a7 98 00 00 00 4c 89 e7 48 89 5d e8 4c 89 6d f8 41 89 f5 e8 2a 35 13 e1 <48> 8b 98 70 01 00 00 0f b6 83 55 02 00 00 a8 08 75 34 45 85 ed RIP [<ffffffffa01ba936>] azx_vs_set_state+0x26/0x178 [snd_hda_intel] RSP <ffff880264271e48> CR2: 0000000000000170 --- [3] (gdb) list *(azx_vs_set_state+0x26) 0x2936 is in azx_vs_set_state (sound/pci/hda/hda_intel.c:2505). 2500 2501 static void azx_vs_set_state(struct pci_dev *pci, 2502 enum vga_switcheroo_state state) 2503 { 2504 struct snd_card *card = pci_get_drvdata(pci); 2505 struct azx *chip = card->private_data; 2506 bool disabled; 2507 2508 if (chip->init_failed) 2509 return; -- Daniel J Blueman

2 1

Re: [alsa-devel] Constant noise on HDA ALC275
by Julien Danjou 08 Oct '12

08 Oct '12

On Mon, Oct 08 2012, Takashi Iwai wrote: > At best, please give alsa-info.sh outputs at both working and > non-working cases. In that way, we can see the internal states more > exactly than ambiguous descriptions. Sure! Both attached [in gzip this time to pass the list]. The diff between background-noise and no-background-noise is: --- /tmp/alsa-info-background-noise.txt 2012-10-08 14:41:05.880816525 +0200 +++ /tmp/alsa-info-no-background-noise.txt 2012-10-08 14:42:24.700814459 +0200 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.61 !!################################ -!!Script ran on: Mon Oct 8 12:41:05 UTC 2012 +!!Script ran on: Mon Oct 8 12:42:24 UTC 2012 !!Linux Distribution @@ -198,7 +198,7 @@ Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 - Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] + Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In -- Julien Danjou // Free Software hacker & freelance // http://julien.danjou.info

1 0

[alsa-devel] Constant noise on HDA ALC275
by Julien Danjou 08 Oct '12

08 Oct '12

Hi there, I've an HDA based chipset in a new Sony Vaio Z (Ivy Bridge). As sound as the system starts and the sound module are loaded, there's a constant noise (shhhhhhht) in the speakers. The noise change a bit if I toggle the mute button, but it's still there. This noise stops as sound as I plug something in the jack port. I've tried to play with HDA Analyzer¹, but I've no clue what to look for, so any hint appreciated. Some information: 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) % aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC275 Analog [ALC275 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 % lsmod | grep snd snd_hda_codec_hdmi 30783 1 snd_hda_codec_realtek 50906 1 snd_hda_intel 26504 3 snd_hda_codec 83533 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13186 1 snd_hda_codec snd_pcm 64080 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 12969 2 snd_pcm,snd_hda_intel snd_seq 45130 0 snd_seq_device 13176 1 snd_seq snd_timer 22917 2 snd_pcm,snd_seq snd 53077 15 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 13026 1 snd /proc/asound/card0/codec#0 ========================== /proc/asound/card0/codec#3 ========================== ¹ http://www.alsa-project.org/main/index.php/HDA_Analyzer -- Julien Danjou # Free Software hacker & freelance # http://julien.danjou.info

2 16

Re: [alsa-devel] 3.6.0 regression, distorted sound failure while using jack and hd-audio
by Daniel Mack 08 Oct '12

08 Oct '12

(cc alsa-devel) On 07.10.2012 21:34, Helge Hafting wrote: > My sound setup works fine with 3.5.4, but have serious problems with 3.6.0 > > It is sound through jack that breaks, simply using "aplay" directly with the > alsa device works. > > When I use jack, my setup works fine with 3.5.4. But boot into 3.6.0, > and I get badly distorted sound as if using an extremely overdone reverb effect. > And jack reports tons of xruns, several per second. > > This can be worked around by going from 256 to 512 frames/period in the jack > settings. But that also double the latency, which makes keyboard playing less > pleasant. > > It is not a case of 256 frames/period being (almost) too extreme for the machine. With kernel 3.5.4, > I can use 128 frames/period and get mostly ok sound, with only an occational > xrun (and a correseponding pop in the sound). So I use 256 because that's solid. > But with 3.6.0, not even 256 is enough. And it is not an occational pop, but > awful distortion all the time. > > I don't really know what has changed from 3.5.4 to 3.6.0. The hd-audio driver? > Something in alsa? Or the scheduling? > > Machine: Dell precision M6600 laptop, 8GB, 2.2GHz quad-core i7. > Usually at 800MHz to save power, this is enough for making sound. > Kernels: 64-bit 3.5.4 or 3.6.0 > Distro: funtoo > > Audio according to lspci: > 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04) > > Please tell if there is anything I should try. I can test other kernel versions > or try patches. > > Helge Hafting > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

2 1

[alsa-devel] [PATCH] ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
by David Henningsson 08 Oct '12

08 Oct '12

In case there is one "Headphone Jack" and one "Dock Headphone Jack", one of them will get an index, even though that is not needed. This patch fixes that issue. BugLink: https://bugs.launchpad.net/bugs/1060729 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 49 +++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 12 deletions(-) Alsa-info at: https://launchpadlibrarian.net/118211803/AlsaInfo diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a98e25e..4ec6dc8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -498,6 +498,38 @@ static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins, return channel_sfx[i]; } +static const char *check_output_pfx(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int attr = snd_hda_get_input_pin_attr(def_conf); + + /* check the location */ + switch (attr) { + case INPUT_PIN_ATTR_DOCK: + return "Dock "; + case INPUT_PIN_ATTR_FRONT: + return "Front "; + } + return ""; +} + +static int get_hp_label_index(struct hda_codec *codec, hda_nid_t nid, + const hda_nid_t *pins, int num_pins) +{ + int i, j, idx = 0; + + const char *pfx = check_output_pfx(codec, nid); + + i = find_idx_in_nid_list(nid, pins, num_pins); + if (i < 0) + return -1; + for (j = 0; j < i; j++) + if (pfx == check_output_pfx(codec, pins[j])) + idx++; + + return idx; +} + static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const struct auto_pin_cfg *cfg, const char *name, char *label, int maxlen, @@ -505,20 +537,13 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); int attr = snd_hda_get_input_pin_attr(def_conf); - const char *pfx = "", *sfx = ""; + const char *pfx, *sfx = ""; /* handle as a speaker if it's a fixed line-out */ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; - /* check the location */ - switch (attr) { - case INPUT_PIN_ATTR_DOCK: - pfx = "Dock "; - break; - case INPUT_PIN_ATTR_FRONT: - pfx = "Front "; - break; - } + pfx = check_output_pfx(codec, nid); + if (cfg) { /* try to give a unique suffix if needed */ sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs, @@ -528,8 +553,8 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, indexp); if (!sfx) { /* don't add channel suffix for Headphone controls */ - int idx = find_idx_in_nid_list(nid, cfg->hp_pins, - cfg->hp_outs); + int idx = get_hp_label_index(codec, nid, cfg->hp_pins, + cfg->hp_outs); if (idx >= 0) *indexp = idx; sfx = ""; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: aloop - add locking to timer access
by Omair Mohammed Abdullah 08 Oct '12

08 Oct '12

When the loopback timer handler is running, calling del_timer() (for STOP trigger) will not wait for the handler to complete before deactivating the timer. The timer gets rescheduled in the handler as usual. Then a subsequent START trigger will try to start the timer using add_timer() with a timer pending leading to a kernel panic. Serialize the calls to add_timer() and del_timer() using a spin lock to avoid this. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/drivers/aloop.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 5a34355..0fe6d64 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -120,6 +120,7 @@ struct loopback_pcm { unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; + spinlock_t timer_lock; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -170,6 +171,7 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); + spin_lock(&dpcm->timer_lock); if (rate_shift != dpcm->pcm_rate_shift) { dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); @@ -182,12 +184,15 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; add_timer(&dpcm->timer); + spin_unlock(&dpcm->timer_lock); } static inline void loopback_timer_stop(struct loopback_pcm *dpcm) { + spin_lock(&dpcm->timer_lock); del_timer(&dpcm->timer); dpcm->timer.expires = 0; + spin_unlock(&dpcm->timer_lock); } #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) @@ -667,6 +672,7 @@ static int loopback_open(struct snd_pcm_substream *substream) dpcm->substream = substream; setup_timer(&dpcm->timer, loopback_timer_function, (unsigned long)dpcm); + spin_lock_init(&dpcm->timer_lock); cable = loopback->cables[substream->number][dev]; if (!cable) { -- 1.7.4.1

3 3

[alsa-devel] [PATCH] hda-emu: Add check for unnecessary indices of kcontrols to testsuite
by David Henningsson 08 Oct '12

08 Oct '12

--- tester/runner.py | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/tester/runner.py b/tester/runner.py index 30118ce..1e7ecf9 100644 --- a/tester/runner.py +++ b/tester/runner.py @@ -25,7 +25,9 @@ class ControlInfo(): self.runner = runner carr = list_info.split(":") self.numid = carr[1] - self.name = carr[2] + self.name = carr[2][1:] + self.index, _ = tuple(carr[3].split(" ")) + self.index = int(self.index) def add_info(self, get_info): minmax_regex = re.compile("MIN/MAX: (\d+)/(\d+),\s+VAL:(( \\[\d+\\])+)") @@ -222,13 +224,28 @@ class HdaEmuRunner(): if (int(a) != int(b)): self.add_error("Tried to set " + c.name + " to " + str([int(x) for x in values]) + ", but got " + str(c.values) + " instead", "Error") - def run_kcontrol_test(self): - minmax_regex = re.compile("MIN/MAX: (\d+)/(\d+),\s+VAL:(( \\[\d+\\])+)") - val_regex = re.compile(" \\[(\d+)\\]") + def check_kcontrol_indices(self, cinfo): + d = {} + for c in cinfo: + if not c.name in d: + d[c.name] = [] + d[c.name].append(c.index) + + for k, v in d.iteritems(): + j = 0 + for i in sorted(v): + if i > j: + self.add_error("Unneccesary index on control '" + k + "' (" + str(i) + ")", "Error") +# Commented out - gives false positives for 'Playback Channel Map' +# elif i < j: +# self.add_error("Duplicate control '" + k + "' (" + str(i) + ")", "Error") + j = i+1 + def run_kcontrol_test(self): controls = self.run_command("list") - for control in controls: - c = ControlInfo(self, control) + cinfo = [ControlInfo(self, control) for control in controls] + self.check_kcontrol_indices(cinfo) + for c in cinfo: c.add_info(self.run_command("get " + c.numid)) if "minval" in c.__dict__: minval = c.minval -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Improve directory/path handling for testsuite
by David Henningsson 08 Oct '12

08 Oct '12

There were some shortcomings when running the test scripts from a random directory. Also added functionality to test an entire directory tree at the same time. --- tester/runner.py | 9 ++++++++- tester/summary.py | 28 ++++++++++++++++++++++++---- 2 files changed, 32 insertions(+), 5 deletions(-) With this improvement we can run it over the entire alsa-info database. The initial result is 619 of 5911 (10%) failing, mostly due to invalid verbs sent (no input/output-amp, invalid amp index, setting *_EN without caps, etc). diff --git a/tester/runner.py b/tester/runner.py index b585ad3..30118ce 100644 --- a/tester/runner.py +++ b/tester/runner.py @@ -44,10 +44,13 @@ class ControlInfo(): class HdaEmuRunner(): def __init__(self): - self.child_args = "../hda-emu -M -F" + import os.path + hda_emu_path = os.path.join(os.path.dirname(os.path.realpath(__file__)),"../hda-emu") + self.child_args = '"' + hda_emu_path + '" -M -F' self.alsa_info = "/proc/asound/card0/codec#0" self.errors = 0 self.warnings = 0 + self.fatals = 0 self.print_errors = False self.comm_log_enabled = False self.last_command = None @@ -81,6 +84,8 @@ class HdaEmuRunner(): self.warnings += 1 else: self.errors += 1 + if severity == "Fatal": + self.fatals += 1 if self.print_errors: if not self.last_command_printed: if self.last_command: @@ -89,6 +94,8 @@ class HdaEmuRunner(): print "Encountered during initial parsing:" self.last_command_printed = True print " ", message + if self.fatals > 0: + raise Exception(message) def check_stdout(self): s = os.read(self.child.stdout.fileno(), 65536) diff --git a/tester/summary.py b/tester/summary.py index 6e94c9b..0f9fbf3 100755 --- a/tester/summary.py +++ b/tester/summary.py @@ -18,6 +18,11 @@ # You should have received a copy of the GNU General Public License along # with this program. If not, see <http://www.gnu.org/licenses/>. +def defaultpath(s): + import os.path + q = os.path.dirname(os.path.realpath(__file__)) + return os.path.join(q, s) + def main(): import os import os.path @@ -26,18 +31,28 @@ def main(): import argparse parser = argparse.ArgumentParser(description='Hda-emu automated test wrapper.') parser.add_argument('--verbose', '-v', action='count') + parser.add_argument('--directory', '-d', action='store', default=defaultpath("../codecs/canonical/")) + parser.add_argument('--recursive', '-r', action="store_true", default=False) + parser_dict = parser.parse_args() verbose = parser_dict.verbose + directory = parser_dict.directory - os.chdir(os.path.dirname(os.path.realpath(__file__))) - directory = "../codecs/canonical/" - files = os.listdir(directory) + if parser_dict.recursive: + files = [os.path.join(root, filename) for root, _, filenames in os.walk(directory) for filename in filenames] + else: + files = os.listdir(directory) + files.sort() successes = 0 fails = 0 warnings = 0 errors = 0 + index = 0 for f in files: + index += 1 + if verbose > 2: + print '[{0}/{1}]: Testing {2}'.format(index, len(files), f) try: r = runner.HdaEmuRunner() r.set_alsa_info_file(os.path.join(directory, f)) @@ -51,7 +66,12 @@ def main(): print '{0} errors, {1} warnings. ({2})'.format(r.errors, r.warnings, f) else: successes += 1 - except: + except KeyboardInterrupt: + import sys + sys.exit(1) + except Exception as e: + if verbose > 0: + print 'Fatal error for {0}: {1}'.format(f, e) errors += 1 fails += 1 -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - fix indices on boost volume on Conexant
by David Henningsson 08 Oct '12

08 Oct '12

After the recent patch "ALSA: hda - use both input paths on Conexant auto parser" suddenly we can have more than one "Mic Boost", this happened on Acer Aspire One 722. Therefore we must add the possibility to put an index on this "Mic Boost" just as we do for the other "Mic Boost" earlier in the same function. BugLink: https://bugs.launchpad.net/bugs/1059523 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c03d3b8..c11ad8a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4273,7 +4273,7 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) { spec->imux_info[idx].boost = mux; - return cx_auto_add_volume(codec, label, " Boost", 0, + return cx_auto_add_volume(codec, label, " Boost", cidx, mux, HDA_OUTPUT); } return 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix hang caused by race during suspend.
by Dylan Reid 08 Oct '12

08 Oct '12

There was a race condition when the system suspends while hda_power_work is running in the work queue. If system suspend (snd_hda_suspend) happens after the work queue releases power_lock but before it calls hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the codec to power up for register reads, and hanging when it calls cancel_delayed_work_sync from the running work queue. The call chain from the work queue will look like this: hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend hda_call_codec_suspend hda_set_power_state snd_hda_codec_read codec_exec_verb snd_hda_power_up snd_hda_power_save __snd_hda_power_up cancel_delayed_work_sync <<-- cancelling executing wq Fix this by waiting for the work queue to finish before starting suspend if suspend is not happening on the work queue. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 960800b..c0ab72c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3612,7 +3612,7 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) { unsigned int state; @@ -3620,7 +3620,9 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - cancel_delayed_work(&codec->power_work); + /* Cancel delayed work if we aren't currently running from it. */ + if (!in_wq) + cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); snd_hda_update_power_acct(codec); trace_hda_power_down(codec); @@ -4478,7 +4480,7 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); - state = hda_call_codec_suspend(codec); + state = hda_call_codec_suspend(codec, true); codec->pm_down_notified = 0; if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; @@ -5127,7 +5129,7 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) - hda_call_codec_suspend(codec); + hda_call_codec_suspend(codec, false); } return 0; } -- 1.7.12.146.g16d26b1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix detection of ALC271X codec
by Herton Ronaldo Krzesinski 08 Oct '12

08 Oct '12

In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup"), alc_auto_parse_customize_define was moved after detection of ALC271X. The problem is that detection of ALC271X relies on spec->cdefine.platform_type, and it's set on alc_auto_parse_customize_define. Move the alc_auto_parse_customize_define and its required fixup setup before the block doing the ALC271X and other codec setup. BugLink: https://bugs.launchpad.net/bugs/1006690 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) It looks safe to move the functions before the block "if (codec->vendor_id == 0x10ec0269) {", as nothing on the fixups seems to depend on what's done there. But if wanted we could just move the detection/codec rename of ALC271X below too. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4c4b01..3b187e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6273,6 +6273,12 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; + alc_pick_fixup(codec, alc269_fixup_models, + alc269_fixup_tbl, alc269_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + if (codec->vendor_id == 0x10ec0269) { spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { @@ -6300,12 +6306,6 @@ static int patch_alc269(struct hda_codec *codec) alc269_fill_coef(codec); } - alc_pick_fixup(codec, alc269_fixup_models, - alc269_fixup_tbl, alc269_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - - alc_auto_parse_customize_define(codec); - /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) -- 1.7.10.4

3 3

[alsa-devel] [PATCH 1/1] ALSA: CA0132 - Ensure SCP packet is little-endian format
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Great - good to hear the previous DSP loader patches are now living in the hda-ca0132-dsp branch. This patch is based on that branch, and addresses the SCP packet endianness. >> The firmware does need to be reloaded when the chip loses power, so it >> doesn't survive S3. >OK, we need suspend/resume hooks, too. Do we need to do anything specific to handle this? The firmware is already restored upon S3 by the current caching code. Thanks, - Ian --- The DSP firmware expects the SCP data to be in little-endian format regardless of the host architecture. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..4d7cda4 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <asm/unaligned.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -701,9 +702,13 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val; + put_unaligned_le32(data, &val); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) return res; } while (--retry); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 0/10] ALSA: Add DSP firmware loader (v3)
by Ian Minett 08 Oct '12

08 Oct '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi, thanks for the response. I've resubmitted the patch series with updates to the ones that needed altering based on the feedback. The MODULE_FIRMWARE line has been shifted to the firmware loading patch, DSP download is now guarded by the config switch and code changes that were bundled in with the previous comments patch have moved to a new patch. Also, hda_stream_format has been removed. Regarding the SCP packet endianness, the DSP image requires data to be sent as little-endian, so we may need to swizzle data being sent to the DSP before the transfer. Is there a recommended or standard way of detecting and handling the case of running on big-endian architectures? >Does the firmware on chip survive after S3? The firmware does need to be reloaded when the chip loses power, so it doesn't survive S3. Thanks very much, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's patch: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's code: Pass DMA buffer pointers in calls to setup_bdle(). Add DSP loader callback routines to controller. Add new DSP loader switch to Kconfig to enable DSP firmware loading. 3: - patch_ca0132.c - ca0132_regs.h Add DSP register definitions header file 4 (updated): - patch_ca0132.c Add DSP firmware enums and defs to CA0132 codec MODULE_FIRMWARE not declared in this patch, moved into patch #5 5 (updated): - patch_ca0132.c Add calls to new DSP loader system to transfer firmware binary to the hardware. Add chip read/write routines, DSP I/O, SCP packet format helper functions and transfer DMA management. Add MODULE_FIRMWARE line for DSP firmware binary Protect DSP download using CONFIG_SND_HDA_DSP_LOADER switch 6: - patch_ca0132.c Add DSP firmware caching to CA0132 codec 7 (updated): - patch_ca0132.c Add comments and descriptions to functions. Previous code changes moved to patch #9 8: - hda_codec.h Change return value for load_dsp_prepare to -ENOSYS in case where DSP loader routines are not available. 9 (new): - patch_ca0132.c Merge chipio write address functions and update timeout mechanism in dsp_write_wait(). 10 (new): - patch_ca0132.c Remove unnecessary struct hda_stream_format and references to it.

3 15

[alsa-devel] [PATCH] ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
by Felix Kaechele 08 Oct '12

08 Oct '12

The Lenovo IdeaPad U310 has an internal mic where the right channel is phase inverted. Signed-off-by: Felix Kaechele <felix(a)fetzig.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..784017e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4462,6 +4462,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} }; -- 1.7.11.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - make a generic unsol event handler
by David Henningsson 08 Oct '12

08 Oct '12

Moving towards less duplication of code between codecs - this patch takes some of the common code of unsol event handling and makes it generic. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 32 ++++++++++++++++++++++++++++++-- sound/pci/hda/hda_jack.h | 9 +++++++++ 2 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index c9333c9..5c690cb 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -192,8 +192,9 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect); /** * snd_hda_jack_detect_enable - enable the jack-detection */ -int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, - unsigned char action) +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_new(codec, nid); if (!jack) @@ -203,10 +204,19 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, jack->jack_detect = 1; if (action) jack->action = action; + if (cb) + jack->callback = cb; return snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); } +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable_callback); + +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + unsigned char action) +{ + return snd_hda_jack_detect_enable_callback(codec, nid, action, NULL); +} EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); /** @@ -411,3 +421,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls); + +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct hda_jack_tbl *event; + int tag = (res >> AC_UNSOL_RES_TAG_SHIFT) & 0x7f; + + event = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (!event) + return; + event->jack_dirty = 1; + + if (event->callback) + event->callback(codec, event); + + snd_hda_jack_report_sync(codec); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_unsol_event); + diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index a9803da..af8dd47 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,12 +13,16 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*hda_jack_callback) (struct hda_codec *, struct hda_jack_tbl *); struct hda_jack_tbl { hda_nid_t nid; unsigned char action; /* event action (0 = none) */ unsigned char tag; /* unsol event tag */ unsigned int private_data; /* arbitrary data */ + hda_jack_callback callback; /* jack-detection stuff */ unsigned int pin_sense; /* cached pin-sense value */ unsigned int jack_detect:1; /* capable of jack-detection? */ @@ -61,6 +65,10 @@ void snd_hda_jack_set_dirty_all(struct hda_codec *codec); int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, unsigned char action); +int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, + unsigned char action, + hda_jack_callback cb); + u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); @@ -74,5 +82,6 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, void snd_hda_jack_report_sync(struct hda_codec *codec); +void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res); #endif /* __SOUND_HDA_JACK_H */ -- 1.7.9.5

2 2

Re: [alsa-devel] Internal Microphone in Lenovo P580 not working with alsa 1.0.25
by Raymond Yau 08 Oct '12

08 Oct '12

2012-10-5 下午3:39 於 "mefis" <mefistof(a)gmx.com> 寫道： > > On 10/05/2012 02:09 AM, Raymond Yau wrote: >> >> 2012-10-5 下午2:20 於 "mefis" <mefistof(a)gmx.com> 寫道： >>> >>> Hi, >>> >>> Mic not working in Lenovo P580 with alsa 1.0.25 >>> >>> Alsa Information script output: >>> >>> http://www.alsa-project.org/db/?f=001523ed61bc76fb5b2ad0d3c354c043f89e5a33 >>> >>> Hope it helps. >>> >>> >>> >>> !!################################ >>> !!ALSA Information Script v 0.4.61 >>> !!################################ >>> >>> !!Script ran on: Fri Oct 5 05:56:38 UTC 2012 >>> >>> >>> !!Linux Distribution >>> !!------------------ >>> >>> Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu >> >> 12.04.1 LTS" >>> >>> >>> !!DMI Information >>> !!--------------- >>> >>> Manufacturer: LENOVO >>> Product Name: Lenovo IdeaPad P580 >>> Product Version: Lenovo IdeaPad P580 >>> Firmware Version: 5ECN33WW(V2.03) >>> >>> >>> !!Kernel Information >>> !!------------------ >>> >>> Kernel release: 3.2.0-31-generic >>> Operating System: GNU/Linux >>> Architecture: x86_64 >>> Processor: x86_64 >>> SMP Enabled: Yes >>> >>> >>> !!ALSA Version >>> !!------------ >>> >>> Driver version: 1.0.25 >>> Library version: 1.0.16 >>> Utilities version: 1.0.25 >> >> is this an upgrade since alsa-lib is quite old ? > > sorry, this was an old log. The newest is here: http://www.alsa-project.org/db/?f=08ae62e61ccc89d92a6154d4e5907b4f676c164a > >> >> it is strange that unsolicied event enabled at fixed mic >> instead of mic jack ? >> > > !!################################ > !!ALSA Information Script v 0.4.61 > !!################################ > > !!Script ran on: Fri Oct 5 07:34:06 UTC 2012 > > > > !!Linux Distribution > !!------------------ > > Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 12.04.1 LTS" > > > !!DMI Information > !!--------------- > > Manufacturer: LENOVO > Product Name: Lenovo IdeaPad P580 > Product Version: Lenovo IdeaPad P580 > Firmware Version: 5ECN33WW(V2.03) > > > !!Kernel Information > !!------------------ > > Kernel release: 3.2.0-31-generic > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.25 > Library version: 1.0.25 > > Utilities version: 1.0.25 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [PCH ]: HDA-Intel - HDA Intel PCH > HDA Intel PCH at 0xe0610000 irq 46 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!------------------------------------------------------- > > 00:1b.0 0403: 8086:1e20 (rev 04) > Subsystem: 17aa:3977 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-usb-audio: index=-2 > snd-usb-caiaq: index=-2 > snd-usb-ua101: index=-2 > snd-usb-us122l: index=-2 > snd-usb-usx2y: index=-2 > snd-cmipci: mpu_port=0x330 fm_port=0x388 > snd-pcsp: index=-2 > snd-usb-audio: index=-2 > snd-hda-intel: model=basic try model=auto http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=Do… Conexant 5066 ============= laptopBasic Laptop config (default) hp-laptopHP laptops, e g G60 asusAsus K52JU, Lenovo G560 dell-laptopDell laptops dell-vostroDell Vostro olpc-xo-1_5OLPC XO 1.5 ideapad Lenovo IdeaPad U150 thinkpadLenovo Thinkpad the model "auto" is not listed in the documentation http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=so… static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP]= "laptop", [CXT5066_DELL_LAPTOP]= "dell-laptop", [CXT5066_OLPC_XO_1_5]= "olpc-xo-1_5", [CXT5066_DELL_VOSTRO]= "dell-vostro", [CXT5066_IDEAPAD]= "ideapad", [CXT5066_THINKPAD]= "thinkpad", [CXT5066_ASUS]= "asus", [CXT5066_HP_LAPTOP]= "hp-laptop", [CXT5066_AUTO]= "auto", }; > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Conexant CX20590 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x14f1506e > Subsystem Id: 0x17aac02d > Revision Id: 0x100003 > Node 0x1a [Pin Complex] wcaps 0x400481: Stereo > Pincap 0x00001324: IN Detect > Vref caps: HIZ 50 80 > Pin Default 0x04a11030: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > > Power states: D0 D1 D2 D3 D3cold EPSS > Power: setting=D0, actual=D0 > Node 0x1b [Pin Complex] wcaps 0x400581: Stereo > Pincap 0x00011334: IN OUT EAPD Detect > Vref caps: HIZ 50 80 > EAPD 0x2: EAPD > Pin Default 0x95a70120: [Fixed] Mic at Int Top > Conn = Analog, Color = Unknown > DefAssociation = 0x2, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=38, enabled=1 > Power states: D0 D1 D2 D3 D3cold EPSS > Power: setting=D0, actual=D0 > Connection: 2 > 0x10* 0x11 >

1 0

Re: [alsa-devel] sound/soc/codecs/wm2200.c:2082:2: error: unknown field 'num_ranges' specified in initializer
by Mark Brown 06 Oct '12

06 Oct '12

On Sat, Oct 06, 2012 at 09:23:04AM +0800, Fengguang Wu wrote: > All error/warnings: > sound/soc/codecs/wm2200.c:2082:2: error: unknown field 'num_ranges' specified in initializer Not a real error (nor is anything else like this in my trees right now), all the branches will be rebased after the merge window which will pull in the dependencies.

1 0

Re: [alsa-devel] [PATCH 0/2] Convert davinci ASoC to genalloc SRAM
by Matt Porter 05 Oct '12

05 Oct '12

On Thu, Oct 04, 2012 at 06:15:43AM -0400, Matt Porter wrote: > This series converts davinci ASoC to use genalloc and enables > that support on DA850. It applies on top of the uio_pruss/genalloc > series [1] which allows DaVinci to provide a gen_pool via pdata > for driver use. > > I've tested this on the AM180x EVM. Note that prior to this, > the SRAM paths in the driver were completely unused. I've only > enabled ping-pong buffering on the platform I can test as it's > best to allow those with DM644x and similar platforms to set the > playback/capture sram size to something that's known to work > for them. > > [1] http://www.spinics.net/lists/arm-kernel/msg198854.html Tested again on top of v4 of the uio_pruss/genalloc series http://www.spinics.net/lists/arm-kernel/msg199255.html on next-20121005 since there's some conflicts falling out of of arm-soc and asoc/sound header move patches on mainline atm. -Matt

1 0

[alsa-devel] [PATCH 1/7] ASoC: wm2200: Convert to devm APIs
by Mark Brown 05 Oct '12

05 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 39 ++++++++++++++------------------------- 1 file changed, 14 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index eab64a1..6ca9bcc 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2011,7 +2011,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dev = &i2c->dev; init_completion(&wm2200->fll_lock); - wm2200->regmap = regmap_init_i2c(i2c, &wm2200_regmap); + wm2200->regmap = devm_regmap_init_i2c(i2c, &wm2200_regmap); if (IS_ERR(wm2200->regmap)) { ret = PTR_ERR(wm2200->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2027,8 +2027,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm2200->core_supplies); i++) wm2200->core_supplies[i].supply = wm2200_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2044,8 +2045,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.ldo_ena) { - ret = gpio_request_one(wm2200->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM2200 LDOENA"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, + "WM2200 LDOENA"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", wm2200->pdata.ldo_ena, ret); @@ -2055,8 +2057,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.reset) { - ret = gpio_request_one(wm2200->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM2200 /RESET"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.reset, + GPIOF_OUT_INIT_HIGH, + "WM2200 /RESET"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", wm2200->pdata.reset, ret); @@ -2166,23 +2169,16 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); err_reset: - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } err_ldo: - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); err_core: - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); err_regmap: - regmap_exit(wm2200->regmap); err: return ret; } @@ -2194,17 +2190,10 @@ static __devexit int wm2200_i2c_remove(struct i2c_client *i2c) snd_soc_unregister_codec(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); - regmap_exit(wm2200->regmap); return 0; } -- 1.7.10.4

1 6

[alsa-devel] [PATCH] ASoC: have unique stream_names in codecs
by Peter Meerwald 05 Oct '12

05 Oct '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> for example, there are two 'Capture' stream_names in twl4030, resulting in twl4030-codec twl4030-codec: ASoC: Failed to create Capture debugfs file with debug enabled (fails to create two sysfs entries named 'Capture' under /sys/kernel/debug/asoc/omap3beagle/twl4030-codec/dapm this patch tries to avoid duplicate stream_names Signed-off-by: Peter Meerwald <p.meerwald(a)bct-electronic.com> --- sound/soc/codecs/stac9766.c | 4 ++-- sound/soc/codecs/twl4030.c | 4 ++-- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/uda1380.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- 5 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 33c0f3d..3379119 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -300,14 +300,14 @@ static struct snd_soc_dai_driver stac9766_dai[] = { /* stream cababilities */ .playback = { - .stream_name = "stac9766 analog", + .stream_name = "stac9766 playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { - .stream_name = "stac9766 analog", + .stream_name = "stac9766 capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc..55582db 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2172,7 +2172,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .formats = TWL4030_FORMATS, .sig_bits = 24,}, .capture = { - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, @@ -2189,7 +2189,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c084c54..6f3a9cf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1046,7 +1046,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-ul", .capture = { - .stream_name = "Capture", + .stream_name = "Capture UL", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 2502214..d780fe7 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -671,7 +671,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { { /* playback only - dual interface */ .name = "uda1380-hifi-playback", .playback = { - .stream_name = "Playback", + .stream_name = "Playback Only", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, @@ -682,7 +682,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = { { /* capture only - dual interface*/ .name = "uda1380-hifi-capture", .capture = { - .stream_name = "Capture", + .stream_name = "Capture Only", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 13bff87..783ac20 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1400,7 +1400,7 @@ static struct snd_soc_dai_driver wm8753_dai[] = { .formats = WM8753_FORMATS }, .capture = { /* dummy for fast DAI switching */ - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, @@ -1418,7 +1418,7 @@ static struct snd_soc_dai_driver wm8753_dai[] = { .formats = WM8753_FORMATS, }, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, -- 1.7.9.5

2 1

Re: [alsa-devel] issue of shdma-base library
by Guennadi Liakhovetski 05 Oct '12

05 Oct '12

Mark, could you take this patch or would you like me to resend it with "subject" in the actual mail subject line or does it also work from the mail body? Thanks Guennadi On Thu, 4 Oct 2012, Do Q.Thang wrote: > Hi Guennadi-san > > Thank you for your hard work. > > On 10/03/2012 09:33 PM, Guennadi Liakhovetski wrote: > > Subject: [PATCH] ASoC: fsi: don't reschedule DMA from an atomic context > > > > shdma doesn't support transfer re-scheduling or triggering from callbacks > > or from atomic context. The fsi driver issues DMA transfers from a tasklet > > context, which is a bug. To fix it convert tasklet to a work. > > > > Reported-by: Do Q.Thang <dq-thang(a)jinso.co.jp> > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > --- > > > > Thang-san, Please, check, whether the attached patch fixes your issues. It > > seems to fix them for me. > > This patch also fix them for me. > > Thanks > --- > Thang > > > Mark, please, wait with committing this patch until the guys confirm, it > > really helps. It then will also have to go to 3.6-stable. > > > > Thanks > > Guennadi > > > > On Tue, 2 Oct 2012, Do Q.Thang wrote: > > > > > Hi Guennadi-san, > > > > > > I'm testing uptream kernel (v3.6-rc3 ~) on the kzm-a9-gt & > > > armadillo-800eva > > > board. > > > I found 2 issues with the sh-dma driver while testing audio playback. > > > > > > issue 1: this log appeared while playback with the aplay, then aplay was > > > stopped. > > > > > > ------- log ----------------------------- > > > # aplay audio_s16le_44100.wav > > > Playing WAVE 'audio_s16le_44100.wav' : Signed 16 bit Little Endian, Rate > > > 44100 Hz, Stereo > > > aplay: pcm_write:1710: write error: Input/output error > > > ----------------------------------------- > > > > > > issue 2: this log appeared while playback with the aplay, then aplay was > > > stopped. > > > ( this issue isn't appear on the kzm-a9-gt board only ) > > > > > > ------- log ----------------------------- > > > # aplay audio_s16le_44100.wav > > > Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, > > > Stereo > > > BUG: scheduling while atomic: kworker/1:1/273/0x00000103 > > > aplay: pcm_write:1710: write error: Input/output error > > > ----------------------------------------- > > > > > > I think these issues are introduced by "dmaengine:add an shdma-base > > > library" > > > commit. > > > > > > ---------- commit ----------------------- > > > commit 9a7b8e002e331d0599127f16613c32f425a14f2c > > > Author: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > > Date: Wed May 9 17:09:13 2012 +0200 > > > > > > dmaengine: add an shdma-base library > > > > > > This patch extracts code from shdma.c, that does not directly deal > > > with > > > hardware implementation details and can be re-used with diverse DMA > > > controller variants, found on SH-based SoCs. > > > > > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > > Cc: Sascha Hauer <s.hauer(a)pengutronix.de> > > > Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> > > > ------------------------------------------ > > > > > > About issue 1: I found that > > > The fsi.c uses fsi_dma_do_tasklet() to setup dma-transfer.This is a > > > function > > > of a tasklet. > > > When one dma-transfer is finished shdma_chan_ld_cleanup() is called.In > > > this > > > function > > > __ld_cleanup() is called to cleanup the schan->ld_queue. In behavior of > > > __ld_cleanup() > > > if descriptor is the last chunk its callback is called, then its node is > > > cleanup in the > > > next call of __ld_cleanup().In callback of the last descriptor, > > > fsi_dma_do_tasklet() is > > > scheduled then is called after the shdma_chan_ld_cleanup() is finished. > > > But when this issue appear, shdma_chan_ld_cleanup() is called while > > > shdma_chan_ld_cleanup() > > > is not finished.At that time, schan->ld_queue is not empty, so in > > > shdma_tx_submit() > > > schan->pm_state is set to SHDMA_PM_PENDING, and then dma-transfer isn't > > > started in > > > shdma_issue_pending(). > > > > > > About issue 2: > > > I found that this issue is appear when pm_runtime_barrier() is running. > > > This issue is not appear when disable CONFIG_PM_RUNTIME config. > > > > > > What do you think about these issues? > > > > > > Best regards > > > --- > > > Thang > > diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c > > index 0540408..1bb0d58c 100644 > > --- a/sound/soc/sh/fsi.c > > +++ b/sound/soc/sh/fsi.c > > @@ -20,6 +20,7 @@ > > #include <linux/sh_dma.h> > > #include <linux/slab.h> > > #include <linux/module.h> > > +#include <linux/workqueue.h> > > #include <sound/soc.h> > > #include <sound/sh_fsi.h> > > @@ -223,7 +224,7 @@ struct fsi_stream { > > */ > > struct dma_chan *chan; > > struct sh_dmae_slave slave; /* see fsi_handler_init() */ > > - struct tasklet_struct tasklet; > > + struct work_struct work; > > dma_addr_t dma; > > }; > > @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) > > snd_pcm_period_elapsed(io->substream); > > } > > -static void fsi_dma_do_tasklet(unsigned long data) > > +static void fsi_dma_do_work(struct work_struct *work) > > { > > - struct fsi_stream *io = (struct fsi_stream *)data; > > + struct fsi_stream *io = container_of(work, struct fsi_stream, work); > > struct fsi_priv *fsi = fsi_stream_to_priv(io); > > struct snd_soc_dai *dai; > > struct dma_async_tx_descriptor *desc; > > @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) > > * FIXME > > * > > * In DMAEngine case, codec and FSI cannot be started simultaneously > > - * since FSI is using tasklet. > > + * since FSI is using the scheduler work queue. > > * Therefore, in capture case, probably FSI FIFO will have got > > * overflow error in this point. > > * in that case, DMA cannot start transfer until error was cleared. > > @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void > > *param) > > static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) > > { > > - tasklet_schedule(&io->tasklet); > > + schedule_work(&io->work); > > return 0; > > } > > @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, > > struct fsi_stream *io, struct dev > > return fsi_stream_probe(fsi, dev); > > } > > - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); > > + INIT_WORK(&io->work, fsi_dma_do_work); > > return 0; > > } > > static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) > > { > > - tasklet_kill(&io->tasklet); > > + cancel_work_sync(&io->work); > > fsi_stream_stop(fsi, io); > > > > > --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

2 1

[alsa-devel] Internal Microphone in Lenovo P580 not working with alsa 1.0.25
by mefis 05 Oct '12

05 Oct '12

Hi, Mic not working in Lenovo P580 with alsa 1.0.25 Alsa Information script output: http://www.alsa-project.org/db/?f=001523ed61bc76fb5b2ad0d3c354c043f89e5a33 Hope it helps. !!################################ !!ALSA Information Script v 0.4.61 !!################################ !!Script ran on: Fri Oct 5 05:56:38 UTC 2012 !!Linux Distribution !!------------------ Ubuntu 12.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 12.04.1 LTS" !!DMI Information !!--------------- Manufacturer: LENOVO Product Name: Lenovo IdeaPad P580 Product Version: Lenovo IdeaPad P580 Firmware Version: 5ECN33WW(V2.03) !!Kernel Information !!------------------ Kernel release: 3.2.0-31-generic Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.25 Library version: 1.0.16 Utilities version: 1.0.25 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xe0610000 irq 46 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 17aa:3977 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 snd-hda-intel: model=basic snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : N bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : basic,(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Conexant CX20590 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506e Subsystem Id: 0x17aac02d Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x39 0x39] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3c 0x3c] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x49 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x50 0x50] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17 0x18 0x23* 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17 0x18 0x23 0x24* Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x02 0x02] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x04211040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=37, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=01, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x2: EAPD Pin Default 0x95a70120: [Fixed] Mic at Int Top Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=38, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x49 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Intel PantherPoint HDMI Address: 3 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862806 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x02 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x03 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560030: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---T 1 root audio 116, 7 Oct 5 00:44 /dev/snd/controlC0 crw-rw---T 1 root audio 116, 6 Oct 5 00:44 /dev/snd/hwC0D0 crw-rw---T 1 root audio 116, 5 Oct 5 00:44 /dev/snd/hwC0D3 crw-rw---T 1 root audio 116, 4 Oct 5 00:48 /dev/snd/pcmC0D0c crw-rw---T 1 root audio 116, 3 Oct 5 00:48 /dev/snd/pcmC0D0p crw-rw---T 1 root audio 116, 2 Oct 5 00:44 /dev/snd/pcmC0D3p crw-rw---T 1 root audio 116, 1 Oct 5 00:44 /dev/snd/seq crw-rw---T 1 root audio 116, 33 Oct 5 00:44 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Oct 5 00:44 . drwxr-xr-x 3 root root 220 Oct 5 00:44 .. lrwxrwxrwx 1 root root 12 Oct 5 00:44 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [PCH] Card hw:0 'PCH'/'HDA Intel PCH at 0xe0610000 irq 46' Mixer name : 'Intel PantherPoint HDMI' Components : 'HDA:14f1506e,17aac02d,00100003 HDA:80862806,80860101,00100000' Controls : 15 Simple ctrls : 7 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 57 [77%] [-17.00dB] [on] Front Right: Playback 57 [77%] [-17.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 135 [53%] [-24.00dB] Front Right: Playback 135 [53%] [-24.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 7 Mono: Playback 3 [43%] [-16.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [6.00dB] [on] Front Right: Capture 80 [100%] [6.00dB] [on] Simple mixer control 'Analog Mic Boost',0 Capabilities: cenum Items: '0dB' '10dB' '20dB' '30dB' '40dB' Item0: '20dB' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] !!Alsactl output !!-------------- --startcollapse-- state.PCH { control.1 { iface MIXER name 'Master Playback Volume' value.0 57 value.1 57 comment { access 'read write' type INTEGER count 2 range '0 - 74' dbmin -7400 dbmax 0 dbvalue.0 -1700 dbvalue.1 -1700 } } control.2 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.3 { iface MIXER name 'Analog Mic Boost Capture Enum' value '20dB' comment { access 'read write' type ENUMERATED count 1 item.0 '0dB' item.1 '10dB' item.2 '20dB' item.3 '30dB' item.4 '40dB' } } control.4 { iface MIXER name 'Capture Volume' value.0 80 value.1 80 comment { access 'read write' type INTEGER count 2 range '0 - 80' dbmin -7400 dbmax 600 dbvalue.0 600 dbvalue.1 600 } } control.5 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Beep Playback Volume' value 3 comment { access 'read write' type INTEGER count 1 range '0 - 7' dbmin -2800 dbmax 0 dbvalue.0 -1600 } } control.7 { iface MIXER name 'Beep Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.9 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.12 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.13 { iface PCM device 3 name ELD value '' comment { access read type BYTES count 0 } } control.14 { iface MIXER name 'PCM Playback Volume' value.0 135 value.1 135 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -2400 dbvalue.1 -2400 } } control.15 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module rfcomm bnep bluetooth parport_pc ppdev binfmt_misc snd_hda_codec_hdmi snd_hda_codec_conexant joydev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_timer snd_seq_device snd alx compat arc4 soundcore snd_page_alloc iwlwifi mac80211 uvcvideo ideapad_laptop videodev i915 drm_kms_helper v4l2_compat_ioctl32 mei drm sparse_keymap cfg80211 i2c_algo_bit video psmouse mac_hid serio_raw lp parport usbhid hid !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x04a11030 0x1b 0x95a70120 0x1c 0x400001f0 0x1d 0x400001f0 0x1e 0x400001f0 0x1f 0x92170110 0x20 0x400001f0 0x22 0x400001f0 0x23 0x400001f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D3/init_pin_configs: 0x05 0x18560010 0x06 0x58560020 0x07 0x58560030 /sys/class/sound/hwC0D3/driver_pin_configs: /sys/class/sound/hwC0D3/user_pin_configs: /sys/class/sound/hwC0D3/init_verbs: !!ALSA/HDA dmesg !!-------------- [ 18.438062] mei 0000:00:16.0: irq 45 for MSI/MSI-X [ 18.583957] snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 [ 18.584008] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 18.584036] snd_hda_intel 0000:00:1b.0: setting latency timer to 64 [ 18.650583] psmouse serio1: synaptics: Touchpad model: 1, fw: 8.0, id: 0x1e2b1, caps: 0xd00123/0x840300/0x123c00 [ 18.724639] input: SynPS/2 Synaptics TouchPad as /devices/platform/i8042/serio1/input/input9 [ 19.124515] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input10 [ 19.128227] HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 19.128283] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 [ 19.888422] EXT4-fs (sda5): re-mounted. Opts: errors=remount-ro

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: spdif: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/spdif.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index bc24c7a..5f3b06d 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -397,7 +397,7 @@ static __devinit int spdif_probe(struct platform_device *pdev) ret = -ENOENT; goto err0; } - clk_enable(spdif->pclk); + clk_prepare_enable(spdif->pclk); spdif->sclk = clk_get(&pdev->dev, "sclk_spdif"); if (IS_ERR(spdif->sclk)) { @@ -405,7 +405,7 @@ static __devinit int spdif_probe(struct platform_device *pdev) ret = -ENOENT; goto err1; } - clk_enable(spdif->sclk); + clk_prepare_enable(spdif->sclk); /* Request S/PDIF Register's memory region */ if (!request_mem_region(mem_res->start, @@ -444,10 +444,10 @@ err4: err3: release_mem_region(mem_res->start, resource_size(mem_res)); err2: - clk_disable(spdif->sclk); + clk_disable_unprepare(spdif->sclk); clk_put(spdif->sclk); err1: - clk_disable(spdif->pclk); + clk_disable_unprepare(spdif->pclk); clk_put(spdif->pclk); err0: return ret; @@ -466,9 +466,9 @@ static __devexit int spdif_remove(struct platform_device *pdev) if (mem_res) release_mem_region(mem_res->start, resource_size(mem_res)); - clk_disable(spdif->sclk); + clk_disable_unprepare(spdif->sclk); clk_put(spdif->sclk); - clk_disable(spdif->pclk); + clk_disable_unprepare(spdif->pclk); clk_put(spdif->pclk); return 0; -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: pcm: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/pcm.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index c860819..45f4a75 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -543,7 +543,7 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) ret = PTR_ERR(pcm->cclk); goto err1; } - clk_enable(pcm->cclk); + clk_prepare_enable(pcm->cclk); /* record our pcm structure for later use in the callbacks */ dev_set_drvdata(&pdev->dev, pcm); @@ -568,7 +568,7 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) ret = -ENOENT; goto err4; } - clk_enable(pcm->pclk); + clk_prepare_enable(pcm->pclk); s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + S3C_PCM_RXFIFO; @@ -592,14 +592,14 @@ static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) return 0; err5: - clk_disable(pcm->pclk); + clk_disable_unprepare(pcm->pclk); clk_put(pcm->pclk); err4: iounmap(pcm->regs); err3: release_mem_region(mem_res->start, resource_size(mem_res)); err2: - clk_disable(pcm->cclk); + clk_disable_unprepare(pcm->cclk); clk_put(pcm->cclk); err1: return ret; @@ -619,8 +619,8 @@ static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem_res->start, resource_size(mem_res)); - clk_disable(pcm->cclk); - clk_disable(pcm->pclk); + clk_disable_unprepare(pcm->cclk); + clk_disable_unprepare(pcm->pclk); clk_put(pcm->pclk); clk_put(pcm->cclk); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: ac97: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 14fbcd3..386bab1 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -442,7 +442,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev) ret = -ENODEV; goto err2; } - clk_enable(s3c_ac97.ac97_clk); + clk_prepare_enable(s3c_ac97.ac97_clk); if (ac97_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); @@ -468,7 +468,7 @@ err5: free_irq(irq_res->start, NULL); err4: err3: - clk_disable(s3c_ac97.ac97_clk); + clk_disable_unprepare(s3c_ac97.ac97_clk); clk_put(s3c_ac97.ac97_clk); err2: iounmap(s3c_ac97.regs); @@ -488,7 +488,7 @@ static __devexit int s3c_ac97_remove(struct platform_device *pdev) if (irq_res) free_irq(irq_res->start, NULL); - clk_disable(s3c_ac97.ac97_clk); + clk_disable_unprepare(s3c_ac97.ac97_clk); clk_put(s3c_ac97.ac97_clk); iounmap(s3c_ac97.regs); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: i2s: use clk_prepare_enable and clk_disable_unprepare
by Thomas Abraham 05 Oct '12

05 Oct '12

Convert clk_enable/clk_disable to clk_prepare_enable/clk_disable_unprepare calls as required by common clock framework. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- sound/soc/samsung/i2s.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 40b00a1..547b919 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -423,7 +423,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (i2s->op_clk) { if ((clk_id && !(mod & MOD_IMS_SYSMUX)) || (!clk_id && (mod & MOD_IMS_SYSMUX))) { - clk_disable(i2s->op_clk); + clk_disable_unprepare(i2s->op_clk); clk_put(i2s->op_clk); } else { i2s->rclk_srcrate = @@ -434,7 +434,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, i2s->op_clk = clk_get(&i2s->pdev->dev, i2s->src_clk[clk_id]); - clk_enable(i2s->op_clk); + clk_prepare_enable(i2s->op_clk); i2s->rclk_srcrate = clk_get_rate(i2s->op_clk); /* Over-ride the other's */ @@ -880,7 +880,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) iounmap(i2s->addr); return -ENOENT; } - clk_enable(i2s->clk); + clk_prepare_enable(i2s->clk); if (other) { other->addr = i2s->addr; @@ -922,7 +922,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(0, i2s->addr + I2SCON); - clk_disable(i2s->clk); + clk_disable_unprepare(i2s->clk); clk_put(i2s->clk); iounmap(i2s->addr); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: fsi: use devm_request_irq()
by Kuninori Morimoto 04 Oct '12

04 Oct '12

This patch uses devm_request_irq() instead of request_irq() and removed free_irq() from driver Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5328ae5..05b8b55 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1701,7 +1701,7 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - ret = request_irq(irq, &fsi_interrupt, 0, + ret = devm_request_irq(&pdev->dev, irq, &fsi_interrupt, 0, id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); @@ -1711,7 +1711,7 @@ static int fsi_probe(struct platform_device *pdev) ret = snd_soc_register_platform(&pdev->dev, &fsi_soc_platform); if (ret < 0) { dev_err(&pdev->dev, "cannot snd soc register\n"); - goto exit_free_irq; + goto exit_fsib; } ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai, @@ -1725,8 +1725,6 @@ static int fsi_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(&pdev->dev); -exit_free_irq: - free_irq(irq, master); exit_fsib: pm_runtime_disable(&pdev->dev); fsi_stream_remove(&master->fsib); @@ -1742,7 +1740,6 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); - free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: fsi: fixup channels_min/max
by Kuninori Morimoto 04 Oct '12

04 Oct '12

FSI can support 1 - 8 channel input/output, but current driver is supporting 2 channel format only. This patch fixes channel mismatch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0540408..2b1062b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1497,7 +1497,7 @@ static struct snd_pcm_hardware fsi_pcm_hardware = { .rates = FSI_RATES, .rate_min = 8000, .rate_max = 192000, - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 64 * 1024, .period_bytes_min = 32, @@ -1585,14 +1585,14 @@ static struct snd_soc_dai_driver fsi_soc_dai[] = { .playback = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .capture = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .ops = &fsi_dai_ops, }, @@ -1601,14 +1601,14 @@ static struct snd_soc_dai_driver fsi_soc_dai[] = { .playback = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .capture = { .rates = FSI_RATES, .formats = FSI_FMTS, - .channels_min = 1, - .channels_max = 8, + .channels_min = 2, + .channels_max = 2, }, .ops = &fsi_dai_ops, }, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: ams-delta: fix card initalization failure
by Janusz Krzysztofik 04 Oct '12

04 Oct '12

Since commit 0998d0631001288a5974afc0b2a5f568bcdecb4d, 'device-core: Ensure drvdata = NULL when no driver is bound', the Amstrad Delta sound card no longer initializes correctly due to drvdata reset to NULL by an upper layer before the codec device, required for successful card setup, is registered. Fix this by moving the codec registration bits up, before the card is probed for. Created and tested against linux-3.6-rc3 Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/omap/ams-delta.c | 18 ++++++++++-------- 1 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 7d4fa8e..270de9c 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -590,20 +590,22 @@ static int __init ams_delta_module_init(void) if (!ams_delta_audio_platform_device) return -ENOMEM; - platform_set_drvdata(ams_delta_audio_platform_device, - &ams_delta_audio_card); - - ret = platform_device_add(ams_delta_audio_platform_device); - if (ret) - goto err; - /* * Codec platform device could be registered from elsewhere (board?), * but I do it here as it makes sense only if used with the card. + * Moreover, it must be registered before the card is probed for, + * or the card setup fails due to drvdata reset by upper layers. */ cx20442_platform_device = platform_device_register_simple("cx20442-codec", -1, NULL, 0); - return 0; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_audio_card); + + ret = platform_device_add(ams_delta_audio_platform_device); + if (!ret) + return ret; + err: platform_device_put(ams_delta_audio_platform_device); return ret; -- 1.7.3.4

4 14

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Fix typo of Vibrator
by Peter Ujfalusi 04 Oct '12

04 Oct '12

It is not Vinrator but Vibrator. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-abe-twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index be525df..750d595 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -220,7 +220,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); -- 1.7.12

2 1

[alsa-devel] [PATCH] ASoC: twl6040: Fix Stream DAPM mapping
by Peter Ujfalusi 04 Oct '12

04 Oct '12

Fixes commit: 805238b ASoC: twl6040: Convert to use DAI DAPM widgets where the connection between the stream widgets and the ADC widgets was reversed and because of this on capture the DAPM is not powering up the codec. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam, in the mentioned commit I made a mistake and the DAPM route got reversed for capture. I did the same change for twl4030 and twl6040 at the same time and only tested twl4030 - since it was easier for me. Obviously I did it wrong for twl6040 :( Can you queue this as a fix for 3.7? Thank you, Peter sound/soc/codecs/twl6040.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index e8f97af..00b85cc 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA DAC", NULL, "Vibra Playback"}, /* ADC -> Stream mapping */ - {"ADC Left", NULL, "Legacy Capture"}, - {"ADC Left", NULL, "Capture"}, - {"ADC Right", NULL, "Legacy Capture"}, - {"ADC Right", NULL, "Capture"}, + {"Legacy Capture" , NULL, "ADC Left"}, + {"Capture", NULL, "ADC Left"}, + {"Legacy Capture", NULL, "ADC Right"}, + {"Capture" , NULL, "ADC Right"}, /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, -- 1.7.12

2 1

[alsa-devel] [PATCH v2] ALSA: ASoC: cs4271: add support for AMUTEB=BMUTEC feature
by Daniel Mack 04 Oct '12

04 Oct '12

The CS4271 has a feature to sync its analog mute flags, so one mute circuitry can be used for both channels. Give users access to this feature with a new DT property and a flag in the platform data. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 2 ++ include/sound/cs4271.h | 1 + sound/soc/codecs/cs4271.c | 21 ++++++++++++++++++--- 3 files changed, 21 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index c81b5fd..a850fb9 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -18,6 +18,8 @@ Optional properties: - reset-gpio: a GPIO spec to define which pin is connected to the chip's !RESET pin + - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag + is enabled. Examples: diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 50a059e..6d9e15e 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,6 +19,7 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ + int amutec_eq_bmutec:1; /* flag to enable AMUTEC=BMUTEC */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f994af3..2ac5fe6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,15 +474,25 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; + int amutec_eq_bmutec = 0; #ifdef CONFIG_OF - if (of_match_device(cs4271_dt_ids, codec->dev)) + if (of_match_device(cs4271_dt_ids, codec->dev)) { gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); + + if (!of_get_property(codec->dev->of_node, + "cirrus,amutec-eq-bmutec", NULL)) + amutec_eq_bmutec = 1; + } #endif - if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + + amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + } if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -528,6 +538,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) /* Power-up sequence requires 85 uS */ udelay(85); + if (amutec_eq_bmutec) + snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); + return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); } -- 1.7.11.4

2 1

[alsa-devel] snd_hwdep_open() as non-root user
by Rafal Powierski 04 Oct '12

04 Oct '12

Developers, I would like to access pin verb hda interface using snd_hwdep_open() function. It works fine if I run as root. I hope there is a way to configure my kubuntu so I can do it as non-root user (e.g. debugging) so far, I have been trying changing access flags for devices in /dev/snd/ directory to crwxrwxrwx no plus character. It did not change my privileges for snd_hwdep_open() - still returns –13. I assume that the device-file responsible is /dev/snd/hwC1D0, the same as parameter given to “hda-verb” program. I am developing a new hardware, using HDA codec. Please help

2 2

[alsa-devel] [PATCH] ALSA: ASoC: cs4271: add DT property for AMUTEB=BMUTEC feature
by Daniel Mack 04 Oct '12

04 Oct '12

The CS4271 has a feature to sync its analog mute flags, so one mute circuitry can be used for both channels. Give DT uses access to this feature with a new property. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 2 ++ sound/soc/codecs/cs4271.c | 13 ++++++++++++- 2 files changed, 14 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index c81b5fd..a850fb9 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -18,6 +18,8 @@ Optional properties: - reset-gpio: a GPIO spec to define which pin is connected to the chip's !RESET pin + - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag + is enabled. Examples: diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f994af3..56ec8dd2 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,11 +474,17 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; + int amutec_eq_bmutec = 0; #ifdef CONFIG_OF - if (of_match_device(cs4271_dt_ids, codec->dev)) + if (of_match_device(cs4271_dt_ids, codec->dev)) { gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); + + if (!of_get_property(codec->dev->of_node, + "cirrus,amutec-eq-bmutec", NULL)) + amutec_eq_bmutec = 1; + } #endif if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) @@ -528,6 +534,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) /* Power-up sequence requires 85 uS */ udelay(85); + if (amutec_eq_bmutec) + snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); + return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); } -- 1.7.11.4

2 2

Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 04 Oct '12

04 Oct '12

On 04.10.2012 12:28, Matt Porter wrote: > On Thu, Oct 04, 2012 at 11:57:30AM +0200, Daniel Mack wrote: >> On 04.10.2012 11:38, Porter, Matt wrote: >>> >>> On Oct 4, 2012, at 5:21 AM, Daniel Mack wrote: >>> >>>> >>>> >>>> On 02.10.2012 18:50, Daniel Mack wrote: >>>>> On 02.10.2012 18:41, Matt Porter wrote: >>>>>> On Tue, Oct 02, 2012 at 03:42:47PM +0200, Daniel Mack wrote: >>>>>>> On 02.10.2012 13:06, Sekhar Nori wrote: >>>>>>>> On 10/2/2012 4:03 PM, Daniel Mack wrote: >>>>>>>>> On 02.10.2012 11:37, Mark Brown wrote: >>>>>>>>>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>>>>>>>>> >>>>>>>>>>> I also agree that ifdef is not a good solution. >>>>>>>>>>> It is better to have this information passed as device_data and via DT it can >>>>>>>>>>> be decided based on the compatible property for the device. >>>>>>>>>> >>>>>>>>>> That's not really the problem here - the problem is that the APIs used >>>>>>>>>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>>>>>>>>> or other platforms. The SRAM code needs to move to a standard API. >>>>>>>>> >>>>>>>>> What about following Matt Porter's idea and ignore the SRAM code >>>>>>>>> entirely and port the entire PCM code to generic dmaengine code first? >>>>>>>>> The EDMA driver needs to learn support for cyclic DMA for that, and I >>>>>>>>> might give that a try in near future. >>>>>>>>> >>>>>>>>> Later on, the SRAM ping-pong code can get added back using genalloc >>>>>>>>> functions, as Sekhar proposed. That needs to be done by someone who has >>>>>>>>> access to a Davinci board though, I only have a AM33xx/OMAP here. >>>>>>>> >>>>>>>> We cannot "get rid" of SRAM code and add it back "later". It is required >>>>>>>> for most DaVinci parts. The SRAM code can be converted to use genalloc >>>>>>>> (conversion should be straightforward and I can help test it) and the >>>>>>>> code that uses SRAM can probably keep using the private EDMA API till >>>>>>>> the dmaengine EDMA driver has cyclic DMA support. Matt has already >>>>>>>> posted patches to move private EDMA APIs to a common location. That >>>>>>>> should keep AM335x build from breaking. Is this something that is feasible? >>>>>>> >>>>>>> Yes - by "later" I just meant in a subsequent patch. But you're probably >>>>>>> right, we can also do that first. >>>>>>> >>>>>>> I'm looking at that right now and the problem seems that we don't have a >>>>>>> sane way to dynamically look up gen_pools independently of the actual >>>>>>> run-time platform. All users of gen_pools seem to know which platform >>>>>>> they run on and access a platform-specific pool. So I don't currently >>>>>>> see how we could implement multi-platform code, gen_pools are fine but >>>>>>> don't solve the problem here. >>>>>>> >>>>>>> Would it be an idea to add a char* member to gen_pools and a function >>>>>>> that can be used to dynamically look it up again? If a buffer with a >>>>>>> certain name exists, we can use it and install that ping-pong buffer, >>>>>>> otherwise we just don't. While that would be easy to do, it's a >>>>>>> tree-wide change. >>>>>>> >>>>>>> Is there a better way that I miss? >>>>>> >>>>>> At the high level there's two platform models we have to handle, the >>>>>> boot from board file !DT case, and then the boot from DT case. Since >>>>>> Davinci is just starting DT conversion, we mostly care about the !DT >>>>>> base in which the struct gen_pool * is passed in pdata to the ASoC >>>>>> driver. It is then selectable on a per-platform basis where the decision >>>>>> should be made. >>>>>> >>>>>> Given a separate discussion with Sekhar, we're only going to have one >>>>>> SRAM pool on any DaVinci part right now...this was only a question on >>>>>> L138 anyway. But regardless, the created gen_pool will be passed via >>>>>> pdata. >>>>> >>>>> I thought about this too, as mmp does it that way. >>>>> >>>>>> Since DT conversion is starting and we need to consider that now, >>>>>> the idea there is to use the DT-based generic sram driver [1] such that >>>>>> when we do boot from DT on Davinci, the genpool is provided via phandle >>>>>> and the pointer extracted with the OF helpers that are part of the >>>>>> series. >>>>> >>>>> A phandle is the cleanest way I think, yes. >>>>> >>>>>> That's pretty much it. I'm reworking the backend support as discussed >>>>>> with Sekhar wrt to my uio_pruss series. I can post a standalone series >>>>>> that just replaces sram_* with genalloc for davinci ASoC. >>>>> >>>>> As you can also test it, it would be easiest if you came up with a patch >>>>> for that, yes. I can have a look at the dma bits laters, once my OMAP >>>>> board finally works with the code as it currently stands. I'm still >>>>> fighting with the mcasp driver right now ... >>>> >>>> I quickly prepared two patches to change that, so that topic is out of >>>> the way. But I did only compile-test them on OMAP - could you check on >>>> your Davinci platform? Note that these apply on top of the patch in >>>> discussion here (which isn't applied to the asoc tree yet). >>> >>> I put a series together yesterday, just ran out of time to post >>> last night after testing. I'm posting that now...it's on top of my >>> uio_pruss/genalloc series and only addresss switching davinci-pcm to >>> genalloc (and actually enabling ping-pon from sram). >> >> Ok, I don't care which version makes it in after all :) > > Ok :) > >>> I'll take a look a your OMAP patches. >> >> The patches I just sent out are only for Davinci and change the SRAM >> functions to genalloc in the mcasp driver. They work just fine on OMAP >> of course, as the code is disabled in there. > > Ok, same thing as I did and just sent out. These are tested on AM180x > and hook up ping-pong buffering so that there's actually one user of > all this ping-pong code. It's been sitting idle for a long time. > > I didn't see your patches come by yet but will take a look to see if > we need to combine. I attached them to my mail for a quick review. I wanted to queue them up to a bigger series I'll send out later ... Daniel

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[alsa-devel] HDA-Intel: despite `power_save=1` device hwC0D0 uses 100 % according to Powertop 2.x
by Paul Menzel 04 Oct '12

04 Oct '12

Dear ALSA folks, running Debian Wheezy with Linux 3.2.23 on a T60 with 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xee240000 irq 46 29 [ThinkPadEC ]: ThinkPad EC - ThinkPad Console Audio Control ThinkPad Console Audio Control at EC reg 0x30, fw 79HT50WW-1.07 (attached `alsa-info.sh` output from before running the following command) despite enabling power saving $ echo 1 | sudo tee /sys/module/snd_hda_intel/parameters/power_save 1 and touching the device or playing some music, I am able to hear a pop/click before playing another piece of music, so some power saving mode seems to be enabled, Powertop 2.1 built from Git still shows the device using 100 %. Others seem to have similar problem with other devices though [1]. Thanks, Paul [1] http://mancoosi.org/~abate/audio-codec-hwc0d0-conexant-and-powertop

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Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 04 Oct '12

04 Oct '12

On 04.10.2012 11:38, Porter, Matt wrote: > > On Oct 4, 2012, at 5:21 AM, Daniel Mack wrote: > >> >> >> On 02.10.2012 18:50, Daniel Mack wrote: >>> On 02.10.2012 18:41, Matt Porter wrote: >>>> On Tue, Oct 02, 2012 at 03:42:47PM +0200, Daniel Mack wrote: >>>>> On 02.10.2012 13:06, Sekhar Nori wrote: >>>>>> On 10/2/2012 4:03 PM, Daniel Mack wrote: >>>>>>> On 02.10.2012 11:37, Mark Brown wrote: >>>>>>>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>>>>>>> >>>>>>>>> I also agree that ifdef is not a good solution. >>>>>>>>> It is better to have this information passed as device_data and via DT it can >>>>>>>>> be decided based on the compatible property for the device. >>>>>>>> >>>>>>>> That's not really the problem here - the problem is that the APIs used >>>>>>>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>>>>>>> or other platforms. The SRAM code needs to move to a standard API. >>>>>>> >>>>>>> What about following Matt Porter's idea and ignore the SRAM code >>>>>>> entirely and port the entire PCM code to generic dmaengine code first? >>>>>>> The EDMA driver needs to learn support for cyclic DMA for that, and I >>>>>>> might give that a try in near future. >>>>>>> >>>>>>> Later on, the SRAM ping-pong code can get added back using genalloc >>>>>>> functions, as Sekhar proposed. That needs to be done by someone who has >>>>>>> access to a Davinci board though, I only have a AM33xx/OMAP here. >>>>>> >>>>>> We cannot "get rid" of SRAM code and add it back "later". It is required >>>>>> for most DaVinci parts. The SRAM code can be converted to use genalloc >>>>>> (conversion should be straightforward and I can help test it) and the >>>>>> code that uses SRAM can probably keep using the private EDMA API till >>>>>> the dmaengine EDMA driver has cyclic DMA support. Matt has already >>>>>> posted patches to move private EDMA APIs to a common location. That >>>>>> should keep AM335x build from breaking. Is this something that is feasible? >>>>> >>>>> Yes - by "later" I just meant in a subsequent patch. But you're probably >>>>> right, we can also do that first. >>>>> >>>>> I'm looking at that right now and the problem seems that we don't have a >>>>> sane way to dynamically look up gen_pools independently of the actual >>>>> run-time platform. All users of gen_pools seem to know which platform >>>>> they run on and access a platform-specific pool. So I don't currently >>>>> see how we could implement multi-platform code, gen_pools are fine but >>>>> don't solve the problem here. >>>>> >>>>> Would it be an idea to add a char* member to gen_pools and a function >>>>> that can be used to dynamically look it up again? If a buffer with a >>>>> certain name exists, we can use it and install that ping-pong buffer, >>>>> otherwise we just don't. While that would be easy to do, it's a >>>>> tree-wide change. >>>>> >>>>> Is there a better way that I miss? >>>> >>>> At the high level there's two platform models we have to handle, the >>>> boot from board file !DT case, and then the boot from DT case. Since >>>> Davinci is just starting DT conversion, we mostly care about the !DT >>>> base in which the struct gen_pool * is passed in pdata to the ASoC >>>> driver. It is then selectable on a per-platform basis where the decision >>>> should be made. >>>> >>>> Given a separate discussion with Sekhar, we're only going to have one >>>> SRAM pool on any DaVinci part right now...this was only a question on >>>> L138 anyway. But regardless, the created gen_pool will be passed via >>>> pdata. >>> >>> I thought about this too, as mmp does it that way. >>> >>>> Since DT conversion is starting and we need to consider that now, >>>> the idea there is to use the DT-based generic sram driver [1] such that >>>> when we do boot from DT on Davinci, the genpool is provided via phandle >>>> and the pointer extracted with the OF helpers that are part of the >>>> series. >>> >>> A phandle is the cleanest way I think, yes. >>> >>>> That's pretty much it. I'm reworking the backend support as discussed >>>> with Sekhar wrt to my uio_pruss series. I can post a standalone series >>>> that just replaces sram_* with genalloc for davinci ASoC. >>> >>> As you can also test it, it would be easiest if you came up with a patch >>> for that, yes. I can have a look at the dma bits laters, once my OMAP >>> board finally works with the code as it currently stands. I'm still >>> fighting with the mcasp driver right now ... >> >> I quickly prepared two patches to change that, so that topic is out of >> the way. But I did only compile-test them on OMAP - could you check on >> your Davinci platform? Note that these apply on top of the patch in >> discussion here (which isn't applied to the asoc tree yet). > > I put a series together yesterday, just ran out of time to post > last night after testing. I'm posting that now...it's on top of my > uio_pruss/genalloc series and only addresss switching davinci-pcm to > genalloc (and actually enabling ping-pon from sram). Ok, I don't care which version makes it in after all :) > I'll take a look a your OMAP patches. The patches I just sent out are only for Davinci and change the SRAM functions to genalloc in the mcasp driver. They work just fine on OMAP of course, as the code is disabled in there. Daniel

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Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 04 Oct '12

04 Oct '12

On 02.10.2012 18:41, Matt Porter wrote: > On Tue, Oct 02, 2012 at 03:42:47PM +0200, Daniel Mack wrote: >> On 02.10.2012 13:06, Sekhar Nori wrote: >>> On 10/2/2012 4:03 PM, Daniel Mack wrote: >>>> On 02.10.2012 11:37, Mark Brown wrote: >>>>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>>>> >>>>>> I also agree that ifdef is not a good solution. >>>>>> It is better to have this information passed as device_data and via DT it can >>>>>> be decided based on the compatible property for the device. >>>>> >>>>> That's not really the problem here - the problem is that the APIs used >>>>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>>>> or other platforms. The SRAM code needs to move to a standard API. >>>> >>>> What about following Matt Porter's idea and ignore the SRAM code >>>> entirely and port the entire PCM code to generic dmaengine code first? >>>> The EDMA driver needs to learn support for cyclic DMA for that, and I >>>> might give that a try in near future. >>>> >>>> Later on, the SRAM ping-pong code can get added back using genalloc >>>> functions, as Sekhar proposed. That needs to be done by someone who has >>>> access to a Davinci board though, I only have a AM33xx/OMAP here. >>> >>> We cannot "get rid" of SRAM code and add it back "later". It is required >>> for most DaVinci parts. The SRAM code can be converted to use genalloc >>> (conversion should be straightforward and I can help test it) and the >>> code that uses SRAM can probably keep using the private EDMA API till >>> the dmaengine EDMA driver has cyclic DMA support. Matt has already >>> posted patches to move private EDMA APIs to a common location. That >>> should keep AM335x build from breaking. Is this something that is feasible? >> >> Yes - by "later" I just meant in a subsequent patch. But you're probably >> right, we can also do that first. >> >> I'm looking at that right now and the problem seems that we don't have a >> sane way to dynamically look up gen_pools independently of the actual >> run-time platform. All users of gen_pools seem to know which platform >> they run on and access a platform-specific pool. So I don't currently >> see how we could implement multi-platform code, gen_pools are fine but >> don't solve the problem here. >> >> Would it be an idea to add a char* member to gen_pools and a function >> that can be used to dynamically look it up again? If a buffer with a >> certain name exists, we can use it and install that ping-pong buffer, >> otherwise we just don't. While that would be easy to do, it's a >> tree-wide change. >> >> Is there a better way that I miss? > > At the high level there's two platform models we have to handle, the > boot from board file !DT case, and then the boot from DT case. Since > Davinci is just starting DT conversion, we mostly care about the !DT > base in which the struct gen_pool * is passed in pdata to the ASoC > driver. It is then selectable on a per-platform basis where the decision > should be made. > > Given a separate discussion with Sekhar, we're only going to have one > SRAM pool on any DaVinci part right now...this was only a question on > L138 anyway. But regardless, the created gen_pool will be passed via > pdata. I thought about this too, as mmp does it that way. > Since DT conversion is starting and we need to consider that now, > the idea there is to use the DT-based generic sram driver [1] such that > when we do boot from DT on Davinci, the genpool is provided via phandle > and the pointer extracted with the OF helpers that are part of the > series. A phandle is the cleanest way I think, yes. > That's pretty much it. I'm reworking the backend support as discussed > with Sekhar wrt to my uio_pruss series. I can post a standalone series > that just replaces sram_* with genalloc for davinci ASoC. As you can also test it, it would be easiest if you came up with a patch for that, yes. I can have a look at the dma bits laters, once my OMAP board finally works with the code as it currently stands. I'm still fighting with the mcasp driver right now ... Daniel

1 1

[alsa-devel] [PATCH 0/2] ASoC: omap-mcpdm updates for 3.7
by Peter Ujfalusi 04 Oct '12

04 Oct '12

Hello Mark, Tony, The mcpdm driver no longer needs to include any plat/*.h file, clearing the way for the single kernel image. Regards, Peter --- Peter Ujfalusi (1): ASoC: omap-mcpdm: Remove OMAP revision check Tony Lindgren (1): ASoC: Fix wrong include for McPDM sound/soc/omap/omap-mcpdm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) -- 1.7.12

1 3

[alsa-devel] ASoC: Fix wrong include for McPDM
by Tony Lindgren 04 Oct '12

04 Oct '12

McPDM needs platt/cpu.h for omap_rev and not omap_hwmod.h. Drivers must not include omap_hwmod.h at, it will be private to mach-omap2 soon. Fix the problem before other drivers will also start including omap_hwmod.h. Note that also plat/cpu.h will be going away, so the omap_rev check needs to be replaced with mcpdm-watchdog flag from platform_data or DT. Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- Can you guys please queue this as fix for -rc1? This removes and unnecessary dependency for making omap_hwmod.h local to mach-omap2. Then I'm hoping Peter will remove the plat/cpu.h dependency for v3.8 merge window ;) --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,7 +40,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <plat/omap_hwmod.h> +#include <plat/cpu.h> #include "omap-mcpdm.h" #include "omap-pcm.h"

2 1

[alsa-devel] [PATCH] ASoC: wm2200: Fix non-inverted OUT2 mute control
by Mark Brown 02 Oct '12

02 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 664067c..6ca9bcc 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, - WM2200_SPK1R_MUTE_SHIFT, 1, 0), + WM2200_SPK1R_MUTE_SHIFT, 1, 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH 2/3] ASoC: Davinci: pcm: add support for sram-support-less platforms
by Daniel Mack 02 Oct '12

02 Oct '12

On 02.10.2012 13:06, Sekhar Nori wrote: > On 10/2/2012 4:03 PM, Daniel Mack wrote: >> On 02.10.2012 11:37, Mark Brown wrote: >>> On Tue, Oct 02, 2012 at 10:48:53AM +0300, Peter Ujfalusi wrote: >>> >>>> I also agree that ifdef is not a good solution. >>>> It is better to have this information passed as device_data and via DT it can >>>> be decided based on the compatible property for the device. >>> >>> That's not really the problem here - the problem is that the APIs used >>> to get the SRAM are DaVinci only so it's not possible to build on OMAP >>> or other platforms. The SRAM code needs to move to a standard API. >> >> What about following Matt Porter's idea and ignore the SRAM code >> entirely and port the entire PCM code to generic dmaengine code first? >> The EDMA driver needs to learn support for cyclic DMA for that, and I >> might give that a try in near future. >> >> Later on, the SRAM ping-pong code can get added back using genalloc >> functions, as Sekhar proposed. That needs to be done by someone who has >> access to a Davinci board though, I only have a AM33xx/OMAP here. > > We cannot "get rid" of SRAM code and add it back "later". It is required > for most DaVinci parts. The SRAM code can be converted to use genalloc > (conversion should be straightforward and I can help test it) and the > code that uses SRAM can probably keep using the private EDMA API till > the dmaengine EDMA driver has cyclic DMA support. Matt has already > posted patches to move private EDMA APIs to a common location. That > should keep AM335x build from breaking. Is this something that is feasible? Yes - by "later" I just meant in a subsequent patch. But you're probably right, we can also do that first. I'm looking at that right now and the problem seems that we don't have a sane way to dynamically look up gen_pools independently of the actual run-time platform. All users of gen_pools seem to know which platform they run on and access a platform-specific pool. So I don't currently see how we could implement multi-platform code, gen_pools are fine but don't solve the problem here. Would it be an idea to add a char* member to gen_pools and a function that can be used to dynamically look it up again? If a buffer with a certain name exists, we can use it and install that ping-pong buffer, otherwise we just don't. While that would be easy to do, it's a tree-wide change. Is there a better way that I miss? Daniel

1 0

[alsa-devel] [PATCH] ASoC: wm2200: Convert to devm APIs
by Mark Brown 01 Oct '12

01 Oct '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 39 ++++++++++++++------------------------- 1 file changed, 14 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index efa93dbb..02ea49e 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2011,7 +2011,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, wm2200->dev = &i2c->dev; init_completion(&wm2200->fll_lock); - wm2200->regmap = regmap_init_i2c(i2c, &wm2200_regmap); + wm2200->regmap = devm_regmap_init_i2c(i2c, &wm2200_regmap); if (IS_ERR(wm2200->regmap)) { ret = PTR_ERR(wm2200->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2027,8 +2027,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm2200->core_supplies); i++) wm2200->core_supplies[i].supply = wm2200_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2044,8 +2045,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.ldo_ena) { - ret = gpio_request_one(wm2200->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM2200 LDOENA"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, + "WM2200 LDOENA"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", wm2200->pdata.ldo_ena, ret); @@ -2055,8 +2057,9 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, } if (wm2200->pdata.reset) { - ret = gpio_request_one(wm2200->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM2200 /RESET"); + ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.reset, + GPIOF_OUT_INIT_HIGH, + "WM2200 /RESET"); if (ret < 0) { dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", wm2200->pdata.reset, ret); @@ -2165,23 +2168,16 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); err_reset: - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } err_ldo: - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); err_core: - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); err_regmap: - regmap_exit(wm2200->regmap); err: return ret; } @@ -2193,17 +2189,10 @@ static __devexit int wm2200_i2c_remove(struct i2c_client *i2c) snd_soc_unregister_codec(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); - if (wm2200->pdata.reset) { + if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_free(wm2200->pdata.reset); - } - if (wm2200->pdata.ldo_ena) { + if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); - gpio_free(wm2200->pdata.ldo_ena); - } - regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), - wm2200->core_supplies); - regmap_exit(wm2200->regmap); return 0; } -- 1.7.10.4

1 0

Re: [alsa-devel] 0.2ms latency
by Pavel Roskin 01 Oct '12

01 Oct '12

Quoting Uwaysi Bin Kareem <uwaysi.bin.kareem(a)paradoxuncreated.com>: > Hi. I am aiming for 0.2ms latency with a HDA onboard soundcard. > > I recently did 0.3ms on a firewire card, with realtime streams. (which > is the lowest possible for firewire protocol.) > This does not work with the alsa HDA device. (some clicking) > Is this a weakness software-wise, that can be improved? > I also do notice that the driver (probably) seems to allow lower > latencies at 96k. Hello! This is the first time I got a message from this list. I subscribed to it about a month ago to report a kernel bug. I guess this mailing list is defunct and the discussion is happening elsewhere. -- Regards, Pavel Roskin -- 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

3 2

Re: [alsa-devel] [PATCH] ASoC: Revert 'ASoC: imx-ssi: Remove mono support'
by Mark Brown 01 Oct '12

01 Oct '12

On Mon, Oct 01, 2012 at 11:33:45AM +0200, javier Martin wrote: > Hi Mark, Fabio, > this patch should be applied as a fix to stable 3.6 too. Otherwise, > audio for Visstrim M10 will continue broken in 3.6. You need to tell the stable team about this, then.

1 0

[alsa-devel] ASoC updates for 3.7
by Mark Brown 01 Oct '12

01 Oct '12

The following changes since commit ddfb43f3881edb47aa0083651ad31983cdc42c33: Merge tag 'v3.6-rc6' into for-3.7 (2012-09-22 11:26:27 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.7 for you to fetch changes up to 9911f7f7562a25381eff93fdc660a4a3b4c0f6e0: ASoC: codecs: Add DA9055 codec driver (2012-09-28 14:21:54 +0100) ---------------------------------------------------------------- ASoC: Additional updates for v3.7 A couple more updates for 3.7, enhancements to the ux500 and wm2000 drivers, a new driver for DA9055 and the support for regulator bypass mode. With the exception of the DA9055 this has all had a chance to soak in -next (the driver was added on Friday so should be in -next today). ---------------------------------------------------------------- Ashish Chavan (1): ASoC: codecs: Add DA9055 codec driver Charles Keepax (1): ASoC: wm5110: Adding missing volume update bits Daniel Mack (1): ALSA: ASoC: add DT bindings for CS4271 Fabio Estevam (2): ASoC: cs4270: Remove mono support ASoC: eukrea-tlv320: Convert it to platform driver Lee Jones (7): ASoC: Ux500: Move MSP pinctrl setup into the MSP driver ASoC: Ux500: Enable MOP500 driver for Device Tree ASoC: Ux500: Enable ux500 MSP driver for Device Tree ASoC: codecs: Enable AB8500 CODEC for Device Tree ASoC: Ux500: Minor coding layout changes Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings Documentation: Define the MSP Driver Device Tree bindings Mark Brown (22): regulator: core: Support bypass mode regulator: core: Provide regmap get/set bypass operations regulator: arizona-ldo: Support get/set bypass regulator: arizona-micsupp: Support get/set bypass regulator: wm831x-ldo: Add bypass support extcon: arizona: Use bypass mode for MICVDD ASoC: wm2000: Convert to devm_regmap_init_i2c() ASoC: wm2000: Add regulator support Merge remote-tracking branch 'asoc/topic/ux500' into for-3.7 Merge tag 'bypass' of git://git.kernel.org/.../broonie/regulator into for-3.7 ASoC: dapm: Allow regulators to bypass as well as disable when idle ASoC: wm5102: Enable bypass mode for MICVDD ASoC: wm5110: Enable bypass mode for MICVDD ASoC: wm0010: Allow slow GPIO for reset ASoC: wm0010: Don't check if reset GPIO is defined when removing ASoC: wm0010: Initialise chip state before we register the interrupt ASoC: arizona: Add more DSP options for mixer input muxes ASoC: arizona: Add more clock rates ASoC: wm5110: Rename EPOUT to HPOUT3 ASoC: wm5110: Add AEC loopback support ASoC: wm5110: Add OUT3R support ASoC: wm_hubs: Ensure volume updates are handled during class W startup Documentation/ABI/testing/sysfs-class-regulator | 21 + Documentation/devicetree/bindings/sound/cs4271.txt | 36 + .../devicetree/bindings/sound/ux500-mop500.txt | 39 + .../devicetree/bindings/sound/ux500-msp.txt | 43 + arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 1 + arch/arm/mach-ux500/board-mop500-msp.c | 79 +- arch/arm/mach-ux500/include/mach/msp.h | 2 - drivers/extcon/extcon-arizona.c | 5 + drivers/regulator/arizona-ldo1.c | 4 + drivers/regulator/arizona-micsupp.c | 5 + drivers/regulator/core.c | 126 ++ drivers/regulator/wm831x-ldo.c | 8 + include/linux/mfd/abx500/ab8500-codec.h | 6 +- include/linux/regulator/consumer.h | 8 + include/linux/regulator/driver.h | 14 + include/linux/regulator/machine.h | 2 + include/sound/da9055.h | 33 + include/sound/soc-dapm.h | 3 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ab8500-codec.c | 81 ++ sound/soc/codecs/arizona.c | 42 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/cs4270.c | 4 +- sound/soc/codecs/cs4271.c | 24 +- sound/soc/codecs/da9055.c | 1510 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 18 +- sound/soc/codecs/wm2000.c | 60 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 79 +- sound/soc/codecs/wm_hubs.c | 5 + sound/soc/fsl/eukrea-tlv320.c | 37 +- sound/soc/soc-dapm.c | 23 +- sound/soc/ux500/mop500.c | 47 +- sound/soc/ux500/ux500_msp_dai.c | 6 + sound/soc/ux500/ux500_msp_i2s.c | 89 +- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 40 files changed, 2307 insertions(+), 174 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt create mode 100644 include/sound/da9055.h create mode 100644 sound/soc/codecs/da9055.c

1 0

Re: [alsa-devel] [PATCH] ASoC: fixup simple-card probe order
by Mark Brown 01 Oct '12

01 Oct '12

On Mon, Oct 01, 2012 at 09:44:37AM +0900, Do Q.Thang wrote: > Hi all, > > KZM-A9-GT board and Armadillo-800eva board use the asoc-simple-card. > but probe of this sound card needs the fsi-dai's registered before. > So this message appear in the bootlog. > > " asoc-simple-card asoc-simple-card.0: CPU DAI fsia-dai not registered > platform asoc-simple-card.0: Driver asoc-simple-card requests probe > deferral " > > The asoc-simple-card was probed while fsi-dai probe hasn't run yet. > So asoc-simple-card probe deferral is requested. > I think the simple-card should be probed after the fsi-dai probe. > I think this patch can fix this issue. > > What do you think about this? The above is normal and expected behaviour, there is no issue here.

1 0

[alsa-devel] [PATCH 1/2] ALSA: core: add hooks for audio timestamps
by Pierre-Louis Bossart 30 Sep '12

30 Sep '12

ALSA did not provide any direct means to infer the audio time for A/V sync and system/audio time correlations (eg. PulseAudio). Applications had to track the number of samples read/written and add/subtract the number of samples queued in the ring buffer. This accounting led to small errors, typically several samples, due to the two-step process. Computing the audio time in the kernel is more direct, as all the information is available in the same routines. Also add new .audio_wallclock routine to enable fine-grain synchronization between monotonic system time and audio hardware time. Using the wallclock, if supported in hardware, allows for a much better sub-microsecond precision and a common drift tracking for all devices sharing the same wall clock (master clock). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/asound.h | 7 +++++-- include/sound/pcm.h | 2 ++ sound/core/pcm_compat.c | 13 +++++++++++-- sound/core/pcm_lib.c | 29 +++++++++++++++++++++++++++-- sound/core/pcm_native.c | 2 ++ 5 files changed, 47 insertions(+), 6 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 0876a1e..2d1ba63 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -152,7 +152,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -274,6 +274,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; @@ -422,7 +423,8 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -431,6 +433,7 @@ struct snd_pcm_mmap_status { snd_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ snd_pcm_state_t suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct snd_pcm_mmap_control { diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cdca2ab..35c2bbf 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -71,6 +71,8 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); + int (*wall_clock)(struct snd_pcm_substream *substream, + struct timespec *audio_ts); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 91cdf94..ab4c953 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -190,7 +190,8 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - unsigned char reserved[60]; + struct timespec audio_tstamp; + unsigned char reserved[60-sizeof(struct timespec)]; } __attribute__((packed)); @@ -215,7 +216,9 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail, &src->avail) || put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || - put_user(status.suspended_state, &src->suspended_state)) + put_user(status.suspended_state, &src->suspended_state) || + put_user(status.audio_tstamp.tv_sec, &src->audio_tstamp.tv_sec) || + put_user(status.audio_tstamp.tv_nsec, &src->audio_tstamp.tv_nsec)) return -EFAULT; return err; @@ -364,6 +367,7 @@ struct snd_pcm_mmap_status32 { u32 hw_ptr; struct compat_timespec tstamp; s32 suspended_state; + struct compat_timespec audio_tstamp; } __attribute__((packed)); struct snd_pcm_mmap_control32 { @@ -426,12 +430,17 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, sstatus.hw_ptr = status->hw_ptr % boundary; sstatus.tstamp = status->tstamp; sstatus.suspended_state = status->suspended_state; + sstatus.audio_tstamp = status->audio_tstamp; snd_pcm_stream_unlock_irq(substream); if (put_user(sstatus.state, &src->s.status.state) || put_user(sstatus.hw_ptr, &src->s.status.hw_ptr) || put_user(sstatus.tstamp.tv_sec, &src->s.status.tstamp.tv_sec) || put_user(sstatus.tstamp.tv_nsec, &src->s.status.tstamp.tv_nsec) || put_user(sstatus.suspended_state, &src->s.status.suspended_state) || + put_user(sstatus.audio_tstamp.tv_sec, + &src->s.status.audio_tstamp.tv_sec) || + put_user(sstatus.audio_tstamp.tv_nsec, + &src->s.status.audio_tstamp.tv_nsec) || put_user(scontrol.appl_ptr, &src->c.control.appl_ptr) || put_user(scontrol.avail_min, &src->c.control.avail_min)) return -EFAULT; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7ae6719..ee0c931 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -315,6 +315,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned long jdelta; unsigned long curr_jiffies; struct timespec curr_tstamp; + struct timespec audio_tstamp; old_hw_ptr = runtime->status->hw_ptr; @@ -326,9 +327,14 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, */ pos = substream->ops->pointer(substream); curr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) && + (substream->ops->wall_clock)) + substream->ops->wall_clock(substream, &audio_tstamp); + } + if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; @@ -506,8 +512,27 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = curr_jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { runtime->status->tstamp = curr_tstamp; + if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) { + /* + * no wall clock available, provide audio timestamp + * derived from pointer position+delay + */ + u64 audio_frames, audio_nsecs; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_frames = runtime->status->hw_ptr + - runtime->delay; + else + audio_frames = runtime->status->hw_ptr + + runtime->delay; + audio_nsecs = audio_frames * 1000000000LL / + runtime->rate; + audio_tstamp = ns_to_timespec(audio_nsecs); + } + runtime->status->audio_tstamp = audio_tstamp; + } return snd_pcm_update_state(substream, runtime); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..c5206f4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -594,6 +594,8 @@ int snd_pcm_status(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { status->tstamp = runtime->status->tstamp; + status->audio_tstamp = + runtime->status->audio_tstamp; goto _tstamp_end; } } -- 1.7.9.5

3 8

[alsa-devel] snd_had_intel turning off internal speakers by going to power state D3
by Jeff Robins 29 Sep '12

29 Sep '12

Hello, I have a Toshiba Satellite C655D with Mageia 2 and I have a problem where the internal speakers eventually stop producing sound, but the headphones still work. After doing a bit of research and running alsa-info (attached) I figured out that one of the parts of the sound card had gone into a D3 power state. Removing and then reloading the snd_hda_intel module fixed the problem, but then the sound would turn off again. I was able to find a post in alsa-users where someone else came up with a better solution using hda-verb to reset the power state: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg28769.html I also found a thread in the gentoo forums where other people are having the same problem. One of them has kernel 3.5.0: http://forums.gentoo.org/viewtopic-t-932038.html?sid=2bee6543c63f2342057c71… I'm open to compiling my own alsa and helping out in any way I can. Please let me know what I can do. Mageia bug report for completeness: https://bugs.mageia.org/show_bug.cgi?id=7292 Thank you, Jeff

2 3

[alsa-devel] Problem with AC97 and WM9714 on S3C6410
by Andrey Gusakov 29 Sep '12

29 Sep '12

Hi. I'm working with mini6410 devkit/demoboard. Itry to move forward from old kernel to last samsung kernel from git. I run into interesting problem with ALSA and WM9714 codec. On kernel boot I see some activity on AC97 bus, but then BCLK and SYNC stops. As i figure out clock stops after write to WM9714 register 0x00. As I understand it's a reset comand. But it brokes all communication. No blayback or control regiter read/write posible after. And there is no reset or warm_reset commands after it. If i filter out all writes to register 0x00 in samsung AC97 driver, everything seems to work corect. But it's a dirty fix. I use dump_stack() to collect some aditional info. That what i got: Try to reset codec! (write at 0x00) [<c0014978>] (unwind_backtrace+0x0/0xf4) from [<c035caf0>] (ac97_write+0x4c/0x64) [<c035caf0>] (ac97_write+0x4c/0x64) from [<c034d00c>] (snd_soc_write+0xc/0x10) [<c034d00c>] (snd_soc_write+0xc/0x10) from [<c035355c>] (soc_widget_update_bits_locked+0x114/0x190) [<c035355c>] (soc_widget_update_bits_locked+0x114/0x190) from [<c0353830>] (dapm_seq_run_coalesced+0x114/0x16c) [<c0353830>] (dapm_seq_run_coalesced+0x114/0x16c) from [<c0353b88>] (dapm_seq_run+0x300/0x4a0) [<c0353b88>] (dapm_seq_run+0x300/0x4a0) from [<c0354be0>] (dapm_power_widgets+0x404/0x600) [<c0354be0>] (dapm_power_widgets+0x404/0x600) from [<c03577ec>] (snd_soc_dapm_new_widgets+0x2ac/0x6ac) [<c03577ec>] (snd_soc_dapm_new_widgets+0x2ac/0x6ac) from [<c034f940>] (soc_post_component_init+0x44/0x27c) [<c034f940>] (soc_post_component_init+0x44/0x27c) from [<c03513f4>] (snd_soc_register_card+0xc44/0x1a00) [<c03513f4>] (snd_soc_register_card+0xc44/0x1a00) from [<c03521e8>] (soc_probe+0x38/0x64) [<c03521e8>] (soc_probe+0x38/0x64) from [<c025f9d8>] (platform_drv_probe+0x18/0x1c) [<c025f9d8>] (platform_drv_probe+0x18/0x1c) from [<c025e518>] (driver_probe_device+0x78/0x1fc) [<c025e518>] (driver_probe_device+0x78/0x1fc) from [<c025cd9c>] (bus_for_each_drv+0x60/0x8c) [<c025cd9c>] (bus_for_each_drv+0x60/0x8c) from [<c025e810>] (device_attach+0x80/0x98) [<c025e810>] (device_attach+0x80/0x98) from [<c025d8bc>] (bus_probe_device+0x84/0xac) [<c025d8bc>] (bus_probe_device+0x84/0xac) from [<c025c6ec>] (device_add+0x5b0/0x67c) [<c025c6ec>] (device_add+0x5b0/0x67c) from [<c025ffa8>] (platform_device_add+0xf4/0x20c) [<c025ffa8>] (platform_device_add+0xf4/0x20c) from [<c060b3a4>] (mini6410_init+0x60/0xa0) [<c060b3a4>] (mini6410_init+0x60/0xa0) from [<c00085d0>] (do_one_initcall+0x34/0x178) [<c00085d0>] (do_one_initcall+0x34/0x178) from [<c05f3310>] (kernel_init+0xe0/0x1ac) [<c05f3310>] (kernel_init+0xe0/0x1ac) from [<c000f848>] (kernel_thread_exit+0x0/0x8) I'm begginer in kernel programing and have no enouth experience to figure what's happening. Can any one help with this problem? Thank you.

1 0

[alsa-devel] imx28 pcm issue
by George Stefan 29 Sep '12

29 Sep '12

Hi! I have an issues with a modem connected to the SAIF port of an imx28. I have the following configuration running on the modem: -modem is slave(but still have to fix the clock at 256 khz) -"short frame" sync. PCM data is always formatted as 16-bit uncompressed two’s complement. Also, all PCM data and frame synchronization signals are written to the PCM bus on the rising clock edge and read on the falling edge. - 16 bits liniar sample, MSB first. On the imx side i have the following configuration priv->sysclk = 512 * rate; snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSP_SYS_MCLK, 32); dai_format = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM; snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, priv->sysclk, SND_SOC_CLOCK_OUT); The modem has the 2 options for testing, either outputing a tone or an audio loop. My problem is that i cannot receive anything else but noise. arecord -f S16_LE -c2 -r8000 test.wav hexdump -C test.wav 00000000 52 49 46 46 24 c4 09 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 02 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 10 00 64 61 74 61 00 c4 09 00 00 00 00 00 |....data........| 00000030 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |................| * 00040f10 00 00 00 00 f0 03 00 00 00 00 00 00 00 00 00 00 |................| 00040f20 00 00 0f 00 ff ff ff ff ff ff f0 ff fc 00 00 00 |................| 00040f30 00 00 00 00 00 00 00 00 00 00 ff 01 ff ff ff ff |................| 00040f40 00 fc 03 00 00 e0 00 00 00 00 00 00 00 00 00 00 |................| 00040f50 00 00 ff 1f ff ff ff ff 00 ff ff 1f 00 f8 00 00 |................| 00040f60 00 00 00 00 00 00 00 00 03 00 ff ff ff ff ff ff |................| 00040f70 81 ff ff ff 00 00 00 00 00 00 00 00 00 00 00 00 |................| 00040f80 3f 00 ff ff ff ff 80 ff 00 00 00 00 00 00 00 00 |?...............| When i select the audio loop mode, play a wav sample, and record it at the same time, i hear noise every time i play that wav and silence between them(so something is moving out there) Do you have any suggestions on why this is happening? Regards.

1 0

[alsa-devel] imx28 pcm issues
by George Stefan 29 Sep '12

29 Sep '12

Hi! I have an issues with a modem connected to the SAIF port of an imx28. I have the following configuration running on the modem: -modem is slave(but still have to fix the clock at 256 khz)

1 0

[alsa-devel] Delay calculations for USB audio?
by David Henningsson 28 Sep '12

28 Sep '12

Hi, While researching some crackling audio problem, I noticed that I got these messages in my dmesg: delay: estimated 0, actual 384 delay: estimated 384, actual 768 They come at every start of playback. I want to understand this stuff, so I tried to dig a little. It looks like the endpoint is started in the prepare stage, so between prepare and trigger, silence data will be sent. First question: Isn't this messing up the delay calculations? If you fill some "urb buffer" [1] with silence, depending on the urb buffer size, delay calculation should take this into account for proper lipsync. I mean, the USB audio device would first have to play all the silence data before it could start the audio data, right? (And how big is this buffer anyway?) Second question: This also means that some of the silence packets will go through retire_playback_urb, right? Isn't this also messing with the delay calculation? (Oh, and btw, if you have an idea of the crackling/distorted USB audio, let me know. I'm using a standard Logitech USB Headset, using speaker-test -c 2 -D plughw:Headset -t sine -r 48000 I can make the problem go away using any of these options: * Booting a 3.2 kernel - I've tried 3.5, 3.6-rc7 and 3.2 * Choosing the 44100 sample rate instead of the default of 48000 * Selecting the outer USB port instead of the inner one. Yes, there are two USB ports next to each other, and this only happens on one of them. (!)) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] maybe there's a better name for it?

4 5

[alsa-devel] [GIT PULL] MOP500 Audio patches due for ASoC
by Lee Jones 28 Sep '12

28 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.linaro.org/people/ljones/linux-3.0-ux500.git for-asoc-next for you to fetch changes up to b4cad7af6665743647f28119b689e1552326d4da: Documentation: Define the MSP Driver Device Tree bindings (2012-09-20 09:10:59 +0200) ---------------------------------------------------------------- Lee Jones (7): ASoC: Ux500: Move MSP pinctrl setup into the MSP driver ASoC: Ux500: Enable MOP500 driver for Device Tree ASoC: Ux500: Enable ux500 MSP driver for Device Tree ASoC: codecs: Enable AB8500 CODEC for Device Tree ASoC: Ux500: Minor coding layout changes Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings Documentation: Define the MSP Driver Device Tree bindings .../devicetree/bindings/sound/ux500-mop500.txt | 39 +++++++++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 79 +---------------- arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 ++++++++++++++++++ sound/soc/ux500/mop500.c | 47 +++++++++-- sound/soc/ux500/ux500_msp_dai.c | 6 ++ sound/soc/ux500/ux500_msp_i2s.c | 89 +++++++++++++++----- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 10 files changed, 292 insertions(+), 108 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt

2 12

[alsa-devel] [PATCH v3] ASoC: codecs: Add DA9055 codec driver
by Ashish Chavan 28 Sep '12

28 Sep '12

This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- Changes since v2: - Added platform data support for Mic bias voltage and source - Used DAPM for DAC source selection - Other minor improvements --- Changes since v1: - Made extensive use of DAPM throughout the driver - Removed LDO, DMIC and other non used controls - Corrected names for many controls - Other minor improvements --- include/sound/da9055.h | 33 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1510 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1549 insertions(+), 0 deletions(-) diff --git a/include/sound/da9055.h b/include/sound/da9055.h new file mode 100644 index 0000000..cf1241b --- /dev/null +++ b/include/sound/da9055.h @@ -0,0 +1,33 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * 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. + */ + +#ifndef __SOUND_DA9055_H__ +#define __SOUND_DA9055_H__ + +enum da9055_micbias_voltage { + DA9055_MICBIAS_1_6V = 0, + DA9055_MICBIAS_1_8V = 1, + DA9055_MICBIAS_2_1V = 2, + DA9055_MICBIAS_2_2V = 3, +}; + +struct da9055_platform_data { + /* Selects which of the two MicBias pins acts as the bias source */ + bool micbias_source; + /* Selects the micbias voltage */ + enum da9055_micbias_voltage micbias; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3684255..b92759a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca508b2..9bd4d95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c new file mode 100644 index 0000000..185d8dd --- /dev/null +++ b/sound/soc/codecs/da9055.c @@ -0,0 +1,1510 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * 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. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/da9055.h> + +/* DA9055 register space */ + +/* Status Registers */ +#define DA9055_STATUS1 0x02 +#define DA9055_PLL_STATUS 0x03 +#define DA9055_AUX_L_GAIN_STATUS 0x04 +#define DA9055_AUX_R_GAIN_STATUS 0x05 +#define DA9055_MIC_L_GAIN_STATUS 0x06 +#define DA9055_MIC_R_GAIN_STATUS 0x07 +#define DA9055_MIXIN_L_GAIN_STATUS 0x08 +#define DA9055_MIXIN_R_GAIN_STATUS 0x09 +#define DA9055_ADC_L_GAIN_STATUS 0x0A +#define DA9055_ADC_R_GAIN_STATUS 0x0B +#define DA9055_DAC_L_GAIN_STATUS 0x0C +#define DA9055_DAC_R_GAIN_STATUS 0x0D +#define DA9055_HP_L_GAIN_STATUS 0x0E +#define DA9055_HP_R_GAIN_STATUS 0x0F +#define DA9055_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA9055_CIF_CTRL 0x20 +#define DA9055_DIG_ROUTING_AIF 0X21 +#define DA9055_SR 0x22 +#define DA9055_REFERENCES 0x23 +#define DA9055_PLL_FRAC_TOP 0x24 +#define DA9055_PLL_FRAC_BOT 0x25 +#define DA9055_PLL_INTEGER 0x26 +#define DA9055_PLL_CTRL 0x27 +#define DA9055_AIF_CLK_MODE 0x28 +#define DA9055_AIF_CTRL 0x29 +#define DA9055_DIG_ROUTING_DAC 0x2A +#define DA9055_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA9055_AUX_L_GAIN 0x30 +#define DA9055_AUX_R_GAIN 0x31 +#define DA9055_MIXIN_L_SELECT 0x32 +#define DA9055_MIXIN_R_SELECT 0x33 +#define DA9055_MIXIN_L_GAIN 0x34 +#define DA9055_MIXIN_R_GAIN 0x35 +#define DA9055_ADC_L_GAIN 0x36 +#define DA9055_ADC_R_GAIN 0x37 +#define DA9055_ADC_FILTERS1 0x38 +#define DA9055_MIC_L_GAIN 0x39 +#define DA9055_MIC_R_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA9055_DAC_FILTERS5 0x40 +#define DA9055_DAC_FILTERS2 0x41 +#define DA9055_DAC_FILTERS3 0x42 +#define DA9055_DAC_FILTERS4 0x43 +#define DA9055_DAC_FILTERS1 0x44 +#define DA9055_DAC_L_GAIN 0x45 +#define DA9055_DAC_R_GAIN 0x46 +#define DA9055_CP_CTRL 0x47 +#define DA9055_HP_L_GAIN 0x48 +#define DA9055_HP_R_GAIN 0x49 +#define DA9055_LINE_GAIN 0x4A +#define DA9055_MIXOUT_L_SELECT 0x4B +#define DA9055_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA9055_SYSTEM_MODES_INPUT 0x50 +#define DA9055_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA9055_AUX_L_CTRL 0x60 +#define DA9055_AUX_R_CTRL 0x61 +#define DA9055_MIC_BIAS_CTRL 0x62 +#define DA9055_MIC_L_CTRL 0x63 +#define DA9055_MIC_R_CTRL 0x64 +#define DA9055_MIXIN_L_CTRL 0x65 +#define DA9055_MIXIN_R_CTRL 0x66 +#define DA9055_ADC_L_CTRL 0x67 +#define DA9055_ADC_R_CTRL 0x68 +#define DA9055_DAC_L_CTRL 0x69 +#define DA9055_DAC_R_CTRL 0x6A +#define DA9055_HP_L_CTRL 0x6B +#define DA9055_HP_R_CTRL 0x6C +#define DA9055_LINE_CTRL 0x6D +#define DA9055_MIXOUT_L_CTRL 0x6E +#define DA9055_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA9055_LDO_CTRL 0x90 +#define DA9055_IO_CTRL 0x91 +#define DA9055_GAIN_RAMP_CTRL 0x92 +#define DA9055_MIC_CONFIG 0x93 +#define DA9055_PC_COUNT 0x94 +#define DA9055_CP_VOL_THRESHOLD1 0x95 +#define DA9055_CP_DELAY 0x96 +#define DA9055_CP_DETECTOR 0x97 +#define DA9055_AIF_OFFSET 0x98 +#define DA9055_DIG_CTRL 0x99 +#define DA9055_ALC_CTRL2 0x9A +#define DA9055_ALC_CTRL3 0x9B +#define DA9055_ALC_NOISE 0x9C +#define DA9055_ALC_TARGET_MIN 0x9D +#define DA9055_ALC_TARGET_MAX 0x9E +#define DA9055_ALC_GAIN_LIMITS 0x9F +#define DA9055_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA9055_ALC_ANTICLIP_CTRL 0xA1 +#define DA9055_ALC_ANTICLIP_LEVEL 0xA2 +#define DA9055_ALC_OFFSET_OP2M_L 0xA6 +#define DA9055_ALC_OFFSET_OP2U_L 0xA7 +#define DA9055_ALC_OFFSET_OP2M_R 0xAB +#define DA9055_ALC_OFFSET_OP2U_R 0xAC +#define DA9055_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA9055_ALC_CIC_OP_LVL_DATA 0xAE +#define DA9055_DAC_NG_SETUP_TIME 0xAF +#define DA9055_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA9055_DAC_NG_ON_THRESHOLD 0xB1 +#define DA9055_DAC_NG_CTRL 0xB2 + +/* SR bit fields */ +#define DA9055_SR_8000 (0x1 << 0) +#define DA9055_SR_11025 (0x2 << 0) +#define DA9055_SR_12000 (0x3 << 0) +#define DA9055_SR_16000 (0x5 << 0) +#define DA9055_SR_22050 (0x6 << 0) +#define DA9055_SR_24000 (0x7 << 0) +#define DA9055_SR_32000 (0x9 << 0) +#define DA9055_SR_44100 (0xA << 0) +#define DA9055_SR_48000 (0xB << 0) +#define DA9055_SR_88200 (0xE << 0) +#define DA9055_SR_96000 (0xF << 0) + +/* REFERENCES bit fields */ +#define DA9055_BIAS_EN (1 << 3) +#define DA9055_VMID_EN (1 << 7) + +/* PLL_CTRL bit fields */ +#define DA9055_PLL_INDIV_10_20_MHZ (1 << 2) +#define DA9055_PLL_SRM_EN (1 << 6) +#define DA9055_PLL_EN (1 << 7) + +/* AIF_CLK_MODE bit fields */ +#define DA9055_AIF_BCLKS_PER_WCLK_32 (0 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_64 (1 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_128 (2 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_256 (3 << 0) +#define DA9055_AIF_CLK_EN_SLAVE_MODE (0 << 7) +#define DA9055_AIF_CLK_EN_MASTER_MODE (1 << 7) + +/* AIF_CTRL bit fields */ +#define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) +#define DA9055_AIF_FORMAT_LEFT_J (1 << 0) +#define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_WORD_S16_LE (0 << 2) +#define DA9055_AIF_WORD_S20_3LE (1 << 2) +#define DA9055_AIF_WORD_S24_LE (2 << 2) +#define DA9055_AIF_WORD_S32_LE (3 << 2) + +/* MIXIN_L_CTRL bit fields */ +#define DA9055_MIXIN_L_MIX_EN (1 << 3) + +/* MIXIN_R_CTRL bit fields */ +#define DA9055_MIXIN_R_MIX_EN (1 << 3) + +/* ADC_L_CTRL bit fields */ +#define DA9055_ADC_L_EN (1 << 7) + +/* ADC_R_CTRL bit fields */ +#define DA9055_ADC_R_EN (1 << 7) + +/* DAC_L_CTRL bit fields */ +#define DA9055_DAC_L_MUTE_EN (1 << 6) + +/* DAC_R_CTRL bit fields */ +#define DA9055_DAC_R_MUTE_EN (1 << 6) + +/* HP_L_CTRL bit fields */ +#define DA9055_HP_L_AMP_OE (1 << 3) + +/* HP_R_CTRL bit fields */ +#define DA9055_HP_R_AMP_OE (1 << 3) + +/* LINE_CTRL bit fields */ +#define DA9055_LINE_AMP_OE (1 << 3) + +/* MIXOUT_L_CTRL bit fields */ +#define DA9055_MIXOUT_L_MIX_EN (1 << 3) + +/* MIXOUT_R_CTRL bit fields */ +#define DA9055_MIXOUT_R_MIX_EN (1 << 3) + +/* MIC bias select bit fields */ +#define DA9055_MICBIAS2_EN (1 << 6) + +/* ALC_CIC_OP_LEVEL_CTRL bit fields */ +#define DA9055_ALC_DATA_MIDDLE (2 << 0) +#define DA9055_ALC_DATA_TOP (3 << 0) +#define DA9055_ALC_CIC_OP_CHANNEL_LEFT (0 << 7) +#define DA9055_ALC_CIC_OP_CHANNEL_RIGHT (1 << 7) + +#define DA9055_AIF_BCLK_MASK (3 << 0) +#define DA9055_AIF_CLK_MODE_MASK (1 << 7) +#define DA9055_AIF_FORMAT_MASK (3 << 0) +#define DA9055_AIF_WORD_LENGTH_MASK (3 << 2) +#define DA9055_GAIN_RAMPING_EN (1 << 5) +#define DA9055_MICBIAS_LEVEL_MASK (3 << 4) + +#define DA9055_ALC_OFFSET_15_8 0x00FF00 +#define DA9055_ALC_OFFSET_17_16 0x030000 +#define DA9055_ALC_AVG_ITERATIONS 5 + +struct pll_div { + int fref; + int fout; + u8 frac_top; + u8 frac_bot; + u8 integer; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL divisor table */ +static const struct pll_div da9055_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz and its harmonics */ + {11289600, 2822400, 0x00, 0x00, 0x20, 1}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x03, 0x61, 0x1E, 1}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x0C, 0xCC, 0x1D, 1}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x19, 0x45, 0x1B, 1}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x18, 0x56, 0x1A, 1}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x02, 0xD0, 0x19, 1}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x1A, 0x1C, 0x12, 1}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x0B, 0x6D, 0x12, 1}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x07, 0xDD, 0x12, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz and its harmonics */ + {11289600, 3072000, 0x1A, 0x8E, 0x22, 1}, /* MCLK=11.2896Mhz */ + {12000000, 3072000, 0x18, 0x93, 0x20, 1}, /* MCLK=12Mhz */ + {12288000, 3072000, 0x00, 0x00, 0x20, 1}, /* MCLK=12.288Mhz */ + {13000000, 3072000, 0x07, 0xEA, 0x1E, 1}, /* MCLK=13Mhz */ + {13500000, 3072000, 0x04, 0x11, 0x1D, 1}, /* MCLK=13.5Mhz */ + {14400000, 3072000, 0x09, 0xD0, 0x1B, 1}, /* MCLK=14.4Mhz */ + {19200000, 3072000, 0x0F, 0x5C, 0x14, 1}, /* MCLK=19.2Mhz */ + {19680000, 3072000, 0x1F, 0x60, 0x13, 1}, /* MCLK=19.68Mhz */ + {19800000, 3072000, 0x1B, 0x80, 0x13, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + {11289600, 2822400, 0x0D, 0x47, 0x21, 0}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x0D, 0xFA, 0x1F, 0}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x16, 0x66, 0x1E, 0}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x00, 0x98, 0x1D, 0}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x1E, 0x33, 0x1B, 0}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x06, 0x50, 0x1A, 0}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x14, 0xBC, 0x13, 0}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x05, 0x66, 0x13, 0}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x01, 0xAE, 0x13, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA9055_CLKSRC_MCLK +}; + +/* Gain and Volume */ + +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da9055_hpf_cutoff_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da9055_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +static const struct soc_enum da9055_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da9055_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da9055_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +static const struct soc_enum da9055_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +/* Gain ramping rate value */ +static const char * const da9055_gain_ramping_txt[] = { + "nominal rate", "nominal rate * 4", "nominal rate * 8", + "nominal rate / 8" +}; + +static const struct soc_enum da9055_gain_ramping_rate = + SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); + +/* DAC noise gate setup time value */ +static const char * const da9055_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da9055_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, + da9055_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da9055_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, + da9055_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da9055_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, + da9055_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da9055_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da9055_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, + da9055_dac_soft_mute_rate_txt); + +/* DAC routing select */ +static const char * const da9055_dac_src_txt[] = { + "ADC output left", "ADC output right", "AIF input left", + "AIF input right" +}; + +static const struct soc_enum da9055_dac_l_src = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_src_txt); + +static const struct soc_enum da9055_dac_r_src = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_src_txt); + +/* MIC PGA Left source select */ +static const char * const da9055_mic_l_src_txt[] = { + "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_l_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt); + +/* MIC PGA Right source select */ +static const char * const da9055_mic_r_src_txt[] = { + "MIC2_R_L", "MIC2_R", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_r_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da9055_signal_tracking_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da9055_integ_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, + da9055_signal_tracking_rate_txt); + +static const struct soc_enum da9055_integ_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, + da9055_signal_tracking_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da9055_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da9055_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da9055_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da9055_hold_time = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); + +static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA9055_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA9055_ALC_AVG_ITERATIONS; +} + +static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 reg_val, adc_left, adc_right; + int avg_left_data, avg_right_data, offset_l, offset_r; + + if (ucontrol->value.integer.value[0]) { + /* + * While enabling ALC (or ALC sync mode), calibration of the DC + * offsets must be done first + */ + + /* Save current values from ADC control registers */ + adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); + adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_ADC_L_EN, DA9055_ADC_L_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_ADC_R_EN, DA9055_ADC_R_EN); + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_L, reg_val); + reg_val = (offset_l & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_L, reg_val); + + reg_val = (offset_r & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_R, reg_val); + reg_val = (offset_r & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_R, reg_val); + + /* Restore original values of ADC control registers */ + snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); + snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +static const struct snd_kcontrol_new da9055_snd_controls[] = { + + /* Volume controls */ + SOC_DOUBLE_R_TLV("Mic Volume", + DA9055_MIC_L_GAIN, DA9055_MIC_R_GAIN, + 0, 0x7, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", + DA9055_AUX_L_GAIN, DA9055_AUX_R_GAIN, + 0, 0x3f, 0, aux_vol_tlv), + SOC_DOUBLE_R_TLV("Mixin PGA Volume", + DA9055_MIXIN_L_GAIN, DA9055_MIXIN_R_GAIN, + 0, 0xf, 0, mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", + DA9055_ADC_L_GAIN, DA9055_ADC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Volume", + DA9055_DAC_L_GAIN, DA9055_DAC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", + DA9055_HP_L_GAIN, DA9055_HP_R_GAIN, + 0, 0x3f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA9055_LINE_GAIN, 0, 0x3f, 0, + lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA9055_DAC_FILTERS4, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA9055_DAC_FILTERS2, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA9055_DAC_FILTERS2, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA9055_DAC_FILTERS3, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA9055_DAC_FILTERS3, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA9055_DAC_FILTERS4, 0, 0xf, 0, + eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA9055_ADC_FILTERS1, 7, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da9055_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA9055_ADC_FILTERS1, 3, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da9055_adc_vf_cutoff), + + SOC_SINGLE("DAC HPF Switch", DA9055_DAC_FILTERS1, 7, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da9055_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA9055_DAC_FILTERS1, 3, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da9055_dac_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Switch", DA9055_MIC_L_CTRL, + DA9055_MIC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Aux Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Mixin PGA Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("ADC Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Headphone Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 6, 1, 0), + SOC_SINGLE("Lineout Switch", DA9055_LINE_CTRL, 6, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0), + SOC_ENUM("DAC Soft Mute Rate", da9055_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Headphone ZC Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 4, 1, 0), + SOC_SINGLE("Lineout ZC Switch", DA9055_LINE_CTRL, 4, 1, 0), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA9055_DAC_L_CTRL, + DA9055_DAC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Headphone Gain Ramping Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 5, 1, 0), + SOC_SINGLE("Lineout Gain Ramping Switch", DA9055_LINE_CTRL, 5, 1, 0), + SOC_ENUM("Gain Ramping Rate", da9055_gain_ramping_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Switch", DA9055_DAC_NG_CTRL, 7, 1, 0), + SOC_SINGLE("DAC NG ON Threshold", DA9055_DAC_NG_ON_THRESHOLD, + 0, 0x7, 0), + SOC_SINGLE("DAC NG OFF Threshold", DA9055_DAC_NG_OFF_THRESHOLD, + 0, 0x7, 0), + SOC_ENUM("DAC NG Setup Time", da9055_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da9055_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da9055_dac_ng_rampdown_rate), + + /* DAC Invertion control */ + SOC_SINGLE("DAC Left Invert", DA9055_DIG_CTRL, 3, 1, 0), + SOC_SINGLE("DAC Right Invert", DA9055_DIG_CTRL, 7, 1, 0), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Switch", DA9055_MIXIN_L_SELECT, + DA9055_MIXIN_R_SELECT, 7, 1, 0), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Switch", DA9055_ALC_CTRL1, 3, 7, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE_EXT("ALC Sync Mode Switch", DA9055_ALC_CTRL1, 1, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE("ALC Offset Switch", DA9055_ALC_CTRL1, 0, 1, 0), + SOC_SINGLE("ALC Anticlip Mode Switch", DA9055_ALC_ANTICLIP_CTRL, + 7, 1, 0), + SOC_SINGLE("ALC Anticlip Level", DA9055_ALC_ANTICLIP_LEVEL, + 0, 0x7f, 0), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA9055_ALC_TARGET_MIN, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA9055_ALC_TARGET_MAX, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA9055_ALC_NOISE, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA9055_ALC_GAIN_LIMITS, + 4, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA9055_ALC_GAIN_LIMITS, + 0, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 0, 0x7, 0, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 4, 0x7, 0, alc_analog_gain_tlv), + SOC_ENUM("ALC Attack Rate", da9055_attack_rate), + SOC_ENUM("ALC Release Rate", da9055_release_rate), + SOC_ENUM("ALC Hold Time", da9055_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da9055_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da9055_integ_release_rate), +}; + +/* DAPM Controls */ + +/* Mic PGA Left Source */ +static const struct snd_kcontrol_new da9055_mic_l_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_l_src); + +/* Mic PGA Right Source */ +static const struct snd_kcontrol_new da9055_mic_r_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_r_src); + +/* In Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXIN_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_L_SELECT, 2, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXIN_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXIN_R_SELECT, 3, 1, 0), +}; + +/* DAC Left Source */ +static const struct snd_kcontrol_new da9055_dac_l_mux_controls = +SOC_DAPM_ENUM("Route", da9055_dac_l_src); + +/* DAC Right Source */ +static const struct snd_kcontrol_new da9055_dac_r_mux_controls = +SOC_DAPM_ENUM("Route", da9055_dac_r_src); + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXOUT_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_L_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA9055_MIXOUT_L_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_L_SELECT, + 6, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXOUT_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA9055_MIXOUT_R_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_R_SELECT, + 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { + /* Input Side */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* MUXs for Mic PGA source selection */ + SND_SOC_DAPM_MUX("Mic Left Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_l_mux_controls), + SND_SOC_DAPM_MUX("Mic Right Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_r_mux_controls), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA9055_MIC_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA9055_MIC_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left", DA9055_AUX_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right", DA9055_AUX_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Left", DA9055_MIXIN_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Right", DA9055_MIXIN_R_CTRL, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", DA9055_MIC_BIAS_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF", DA9055_AIF_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", DA9055_CP_CTRL, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinl_controls[0], + ARRAY_SIZE(da9055_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinr_controls[0], + ARRAY_SIZE(da9055_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA9055_ADC_L_CTRL, 7, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA9055_ADC_R_CTRL, 7, 0), + + /* Output Side */ + + /* MUXs for DAC source selection */ + SND_SOC_DAPM_MUX("DAC Left Source", SND_SOC_NOPM, 0, 0, + &da9055_dac_l_mux_controls), + SND_SOC_DAPM_MUX("DAC Right Source", SND_SOC_NOPM, 0, 0, + &da9055_dac_r_mux_controls), + + /* AIF input */ + SND_SOC_DAPM_AIF_IN("AIFIN Left", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN Right", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA9055_DAC_L_CTRL, 7, 0), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA9055_DAC_R_CTRL, 7, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutl_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutr_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("MIXOUT Left", DA9055_MIXOUT_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXOUT Right", DA9055_MIXOUT_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout", DA9055_LINE_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA9055_HP_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA9055_HP_R_CTRL, 7, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da9055_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"Mic Left Source", "MIC1_P_N", "MIC1"}, + {"Mic Left Source", "MIC1_P", "MIC1"}, + {"Mic Left Source", "MIC1_N", "MIC1"}, + {"Mic Left Source", "MIC2_L", "MIC2"}, + + {"Mic Right Source", "MIC2_R_L", "MIC2"}, + {"Mic Right Source", "MIC2_R", "MIC2"}, + {"Mic Right Source", "MIC2_L", "MIC2"}, + + {"Mic Left", NULL, "Mic Left Source"}, + {"Mic Right", NULL, "Mic Right Source"}, + + {"Aux Left", NULL, "AUXL"}, + {"Aux Right", NULL, "AUXR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux Left Switch", "Aux Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux Right Switch", "Aux Right"}, + {"In Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + + {"MIXIN Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "MIXIN Left"}, + + {"MIXIN Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "MIXIN Right"}, + + {"ADC Left", NULL, "AIF"}, + {"ADC Right", NULL, "AIF"}, + + /* Output path */ + {"AIFIN Left", NULL, "AIF"}, + {"AIFIN Right", NULL, "AIF"}, + + {"DAC Left Source", "ADC output left", "ADC Left"}, + {"DAC Left Source", "ADC output right", "ADC Right"}, + {"DAC Left Source", "AIF input left", "AIFIN Left"}, + {"DAC Left Source", "AIF input right", "AIFIN Right"}, + + {"DAC Right Source", "ADC output left", "ADC Left"}, + {"DAC Right Source", "ADC output right", "ADC Right"}, + {"DAC Right Source", "AIF input left", "AIFIN Left"}, + {"DAC Right Source", "AIF input right", "AIFIN Right"}, + + {"DAC Left", NULL, "DAC Left Source"}, + {"DAC Right", NULL, "DAC Right Source"}, + + {"Out Mixer Left", "Aux Left Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Left", "Aux Left Invert Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux Right Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Right", "Aux Right Invert Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"MIXOUT Left", NULL, "Out Mixer Left"}, + {"Headphone Left", NULL, "MIXOUT Left"}, + {"Headphone Left", NULL, "Charge Pump"}, + {"HPL", NULL, "Headphone Left"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Headphone Right", NULL, "MIXOUT Right"}, + {"Headphone Right", NULL, "Charge Pump"}, + {"HPR", NULL, "Headphone Right"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Lineout", NULL, "MIXOUT Right"}, + {"LINE", NULL, "Lineout"}, +}; + +/* Codec private data */ +struct da9055_priv { + struct regmap *regmap; + unsigned int mclk_rate; + int master; + struct da9055_platform_data *pdata; +}; + +static struct reg_default da9055_reg_defaults[] = { + { 0x21, 0x10 }, + { 0x22, 0x0A }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x0C }, + { 0x28, 0x01 }, + { 0x29, 0x08 }, + { 0x2A, 0x32 }, + { 0x2B, 0x00 }, + { 0x30, 0x35 }, + { 0x31, 0x35 }, + { 0x32, 0x00 }, + { 0x33, 0x00 }, + { 0x34, 0x03 }, + { 0x35, 0x03 }, + { 0x36, 0x6F }, + { 0x37, 0x6F }, + { 0x38, 0x80 }, + { 0x39, 0x01 }, + { 0x3A, 0x01 }, + { 0x40, 0x00 }, + { 0x41, 0x88 }, + { 0x42, 0x88 }, + { 0x43, 0x08 }, + { 0x44, 0x80 }, + { 0x45, 0x6F }, + { 0x46, 0x6F }, + { 0x47, 0x61 }, + { 0x48, 0x35 }, + { 0x49, 0x35 }, + { 0x4A, 0x35 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x60, 0x44 }, + { 0x61, 0x44 }, + { 0x62, 0x00 }, + { 0x63, 0x40 }, + { 0x64, 0x40 }, + { 0x65, 0x40 }, + { 0x66, 0x40 }, + { 0x67, 0x40 }, + { 0x68, 0x40 }, + { 0x69, 0x48 }, + { 0x6A, 0x40 }, + { 0x6B, 0x41 }, + { 0x6C, 0x40 }, + { 0x6D, 0x40 }, + { 0x6E, 0x10 }, + { 0x6F, 0x10 }, + { 0x90, 0x80 }, + { 0x92, 0x02 }, + { 0x93, 0x00 }, + { 0x99, 0x00 }, + { 0x9A, 0x00 }, + { 0x9B, 0x00 }, + { 0x9C, 0x3F }, + { 0x9D, 0x00 }, + { 0x9E, 0x3F }, + { 0x9F, 0xFF }, + { 0xA0, 0x71 }, + { 0xA1, 0x00 }, + { 0xA2, 0x00 }, + { 0xA6, 0x00 }, + { 0xA7, 0x00 }, + { 0xAB, 0x00 }, + { 0xAC, 0x00 }, + { 0xAD, 0x00 }, + { 0xAF, 0x08 }, + { 0xB0, 0x00 }, + { 0xB1, 0x00 }, + { 0xB2, 0x00 }, +}; + +static bool da9055_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case DA9055_STATUS1: + case DA9055_PLL_STATUS: + case DA9055_AUX_L_GAIN_STATUS: + case DA9055_AUX_R_GAIN_STATUS: + case DA9055_MIC_L_GAIN_STATUS: + case DA9055_MIC_R_GAIN_STATUS: + case DA9055_MIXIN_L_GAIN_STATUS: + case DA9055_MIXIN_R_GAIN_STATUS: + case DA9055_ADC_L_GAIN_STATUS: + case DA9055_ADC_R_GAIN_STATUS: + case DA9055_DAC_L_GAIN_STATUS: + case DA9055_DAC_R_GAIN_STATUS: + case DA9055_HP_L_GAIN_STATUS: + case DA9055_HP_R_GAIN_STATUS: + case DA9055_LINE_GAIN_STATUS: + case DA9055_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +/* Set DAI word length */ +static int da9055_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_ctrl, fs; + u32 sysclk; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif_ctrl = DA9055_AIF_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif_ctrl = DA9055_AIF_WORD_S20_3LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif_ctrl = DA9055_AIF_WORD_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif_ctrl = DA9055_AIF_WORD_S32_LE; + break; + default: + return -EINVAL; + } + + /* Set AIF format */ + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_WORD_LENGTH_MASK, + aif_ctrl); + + switch (params_rate(params)) { + case 8000: + fs = DA9055_SR_8000; + sysclk = 3072000; + break; + case 11025: + fs = DA9055_SR_11025; + sysclk = 2822400; + break; + case 12000: + fs = DA9055_SR_12000; + sysclk = 3072000; + break; + case 16000: + fs = DA9055_SR_16000; + sysclk = 3072000; + break; + case 22050: + fs = DA9055_SR_22050; + sysclk = 2822400; + break; + case 32000: + fs = DA9055_SR_32000; + sysclk = 3072000; + break; + case 44100: + fs = DA9055_SR_44100; + sysclk = 2822400; + break; + case 48000: + fs = DA9055_SR_48000; + sysclk = 3072000; + break; + case 88200: + fs = DA9055_SR_88200; + sysclk = 2822400; + break; + case 96000: + fs = DA9055_SR_96000; + sysclk = 3072000; + break; + default: + return -EINVAL; + } + + if (da9055->mclk_rate) { + /* PLL Mode, Write actual FS */ + snd_soc_write(codec, DA9055_SR, fs); + } else { + /* + * Non-PLL Mode + * When PLL is bypassed, chip assumes constant MCLK of + * 12.288MHz and uses sample rate value to divide this MCLK + * to derive its sys clk. As sys clk has to be 256 * Fs, we + * need to write constant sample rate i.e. 48KHz. + */ + snd_soc_write(codec, DA9055_SR, DA9055_SR_48000); + } + + if (da9055->mclk_rate && (da9055->mclk_rate != sysclk)) { + /* PLL Mode */ + if (!da9055->master) { + /* PLL slave mode, enable PLL and also SRM */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN | DA9055_PLL_SRM_EN, + DA9055_PLL_EN | DA9055_PLL_SRM_EN); + } else { + /* PLL master mode, only enable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN, DA9055_PLL_EN); + } + } else { + /* Non PLL Mode, disable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + } + + return 0; +} + +/* Set DAI mode and Format */ +static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_clk_mode, aif_ctrl, mode; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* DA9055 in I2S Master Mode */ + mode = 1; + aif_clk_mode = DA9055_AIF_CLK_EN_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* DA9055 in I2S Slave Mode */ + mode = 0; + aif_clk_mode = DA9055_AIF_CLK_EN_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* Don't allow change of mode if PLL is enabled */ + if ((snd_soc_read(codec, DA9055_PLL_CTRL) & DA9055_PLL_EN) && + (da9055->master != mode)) + return -EINVAL; + + da9055->master = mode; + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_ctrl = DA9055_AIF_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_ctrl = DA9055_AIF_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + aif_clk_mode |= DA9055_AIF_BCLKS_PER_WCLK_32; + + snd_soc_update_bits(codec, DA9055_AIF_CLK_MODE, + (DA9055_AIF_CLK_MODE_MASK | DA9055_AIF_BCLK_MASK), + aif_clk_mode); + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_FORMAT_MASK, + aif_ctrl); + return 0; +} + +static int da9055_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, DA9055_DAC_R_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, 0); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, 0); + } + + return 0; +} + +#define DA9055_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da9055_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA9055_CLKSRC_MCLK: + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da9055->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* + * da9055_set_dai_pll : Configure the codec PLL + * @param codec_dai : Pointer to codec DAI + * @param pll_id : da9055 has only one pll, so pll_id is always zero + * @param fref : Input MCLK frequency + * @param fout : FsDM value + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 11.2896MHz, 12MHz, 12.288MHz, + * 13MHz, 13.5MHz, 14.4MHz, 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da9055_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + u8 pll_frac_top, pll_frac_bot, pll_integer, cnt; + + /* Disable PLL before setting the divisors */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + + /* In slave mode, there is only one set of divisors */ + if (!da9055->master && (fout != 2822400)) + goto pll_err; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da9055_pll_div); cnt++) { + /* Check fref, mode and fout */ + if ((fref == da9055_pll_div[cnt].fref) && + (da9055->master == da9055_pll_div[cnt].mode) && + (fout == da9055_pll_div[cnt].fout)) { + /* All match, pick up divisors */ + pll_frac_top = da9055_pll_div[cnt].frac_top; + pll_frac_bot = da9055_pll_div[cnt].frac_bot; + pll_integer = da9055_pll_div[cnt].integer; + break; + } + } + if (cnt >= ARRAY_SIZE(da9055_pll_div)) + goto pll_err; + + /* Write PLL dividers */ + snd_soc_write(codec, DA9055_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA9055_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA9055_PLL_INTEGER, pll_integer); + + return 0; +pll_err: + dev_err(codec_dai->dev, "Error in setting up PLL\n"); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da9055_dai_ops = { + .hw_params = da9055_hw_params, + .set_fmt = da9055_set_dai_fmt, + .set_sysclk = da9055_set_dai_sysclk, + .set_pll = da9055_set_dai_pll, + .digital_mute = da9055_mute, +}; + +static struct snd_soc_dai_driver da9055_dai = { + .name = "da9055-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + .ops = &da9055_dai_ops, + .symmetric_rates = 1, +}; + +static int da9055_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Enable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, + DA9055_VMID_EN | DA9055_BIAS_EN); + } + break; + case SND_SOC_BIAS_OFF: + /* Disable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int da9055_probe(struct snd_soc_codec *codec) +{ + int ret; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = da9055->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_AUX_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + + /* + * There are two separate control bits for input and output mixers as + * well as headphone and line outs. + * One to enable corresponding amplifier and other to enable its + * output. As amplifier bits are related to power control, they are + * being managed by DAPM while other (non power related) bits are + * enabled here + */ + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_L_MIX_EN, DA9055_MIXIN_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_MIXIN_R_MIX_EN, DA9055_MIXIN_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_MIX_EN, DA9055_MIXOUT_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); + + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); + + /* Set this as per your system configuration */ + snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); + + /* Set platform data values */ + if (da9055->pdata) { + /* set mic bias source */ + if (da9055->pdata->micbias_source) { + snd_soc_update_bits(codec, DA9055_MIXIN_R_SELECT, + DA9055_MICBIAS2_EN, + DA9055_MICBIAS2_EN); + } else { + snd_soc_update_bits(codec, DA9055_MIXIN_R_SELECT, + DA9055_MICBIAS2_EN, 0); + } + /* set mic bias voltage */ + switch (da9055->pdata->micbias) { + case DA9055_MICBIAS_2_2V: + case DA9055_MICBIAS_2_1V: + case DA9055_MICBIAS_1_8V: + case DA9055_MICBIAS_1_6V: + snd_soc_update_bits(codec, DA9055_MIC_CONFIG, + DA9055_MICBIAS_LEVEL_MASK, + (da9055->pdata->micbias) << 4); + break; + } + } + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da9055 = { + .probe = da9055_probe, + .set_bias_level = da9055_set_bias_level, + + .controls = da9055_snd_controls, + .num_controls = ARRAY_SIZE(da9055_snd_controls), + + .dapm_widgets = da9055_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da9055_dapm_widgets), + .dapm_routes = da9055_audio_map, + .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), +}; + +static const struct regmap_config da9055_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da9055_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da9055_reg_defaults), + .volatile_reg = da9055_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da9055_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da9055_priv *da9055; + struct da9055_platform_data *pdata = dev_get_platdata(&i2c->dev); + int ret; + + da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + GFP_KERNEL); + if (!da9055) + return -ENOMEM; + + if (pdata) + da9055->pdata = pdata; + + i2c_set_clientdata(i2c, da9055); + + da9055->regmap = devm_regmap_init_i2c(i2c, &da9055_regmap_config); + if (IS_ERR(da9055->regmap)) { + ret = PTR_ERR(da9055->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da9055, &da9055_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + ret); + } + return ret; +} + +static int __devexit da9055_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da9055_i2c_id[] = { + { "da9055", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da9055_i2c_driver = { + .driver = { + .name = "da9055", + .owner = THIS_MODULE, + }, + .probe = da9055_i2c_probe, + .remove = __devexit_p(da9055_remove), + .id_table = da9055_i2c_id, +}; + +module_i2c_driver(da9055_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); +MODULE_AUTHOR("David Chen, Ashish Chavan"); +MODULE_LICENSE("GPL"); -- 1.7.1

2 1

[alsa-devel] [PATCH v2] ASoC: eukrea-tlv320: Convert it to platform driver
by Fabio Estevam 28 Sep '12

28 Sep '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Convert eukrea-tlv320 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - No changes, only CC'ed ARM kernel list as requested by Mark Brown arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 1 + sound/soc/fsl/eukrea-tlv320.c | 37 ++++++++++++------------ 5 files changed, 23 insertions(+), 18 deletions(-) diff --git a/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c b/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c index fd3177f..98aef57 100644 --- a/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c @@ -348,4 +348,5 @@ void __init eukrea_mbimx27_baseboard_init(void) imx27_add_imx_keypad(&eukrea_mbimx27_keymap_data); gpio_led_register_device(-1, &eukrea_mbimx27_gpio_led_info); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c index dfd2da8..0b84666 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c @@ -306,4 +306,5 @@ void __init eukrea_mbimxsd25_baseboard_init(void) platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); gpio_led_register_device(-1, &eukrea_mbimxsd_led_info); imx_add_gpio_keys(&eukrea_mbimxsd_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c index 6e9dd12..c6532a0 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c @@ -315,4 +315,5 @@ void __init eukrea_mbimxsd35_baseboard_init(void) platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); gpio_led_register_device(-1, &eukrea_mbimxsd_led_info); imx_add_gpio_keys(&eukrea_mbimxsd_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c index 96a24b7..8b0de30 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c @@ -228,4 +228,5 @@ void __init eukrea_mbimxsd51_baseboard_init(void) gpio_led_register_device(-1, &eukrea_mbimxsd51_led_info); imx_add_gpio_keys(&eukrea_mbimxsd51_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index efb9ede..267d5b4 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -93,9 +93,7 @@ static struct snd_soc_card eukrea_tlv320 = { .num_links = 1, }; -static struct platform_device *eukrea_tlv320_snd_device; - -static int __init eukrea_tlv320_init(void) +static int __devinit eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; @@ -136,29 +134,32 @@ static int __init eukrea_tlv320_init(void) return 0; } - eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); - if (!eukrea_tlv320_snd_device) - return -ENOMEM; - - platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320); - ret = platform_device_add(eukrea_tlv320_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(eukrea_tlv320_snd_device); - } + eukrea_tlv320.dev = &pdev->dev; + ret = snd_soc_register_card(&eukrea_tlv320); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); return ret; } -static void __exit eukrea_tlv320_exit(void) +static int __devexit eukrea_tlv320_remove(struct platform_device *pdev) { - platform_device_unregister(eukrea_tlv320_snd_device); + snd_soc_unregister_card(&eukrea_tlv320); + + return 0; } -module_init(eukrea_tlv320_init); -module_exit(eukrea_tlv320_exit); +static struct platform_driver eukrea_tlv320_driver = { + .driver = { + .name = "eukrea_tlv320", + .owner = THIS_MODULE, + }, + .probe = eukrea_tlv320_probe, + .remove = __devexit_p(eukrea_tlv320_remove),}; + +module_platform_driver(eukrea_tlv320_driver); MODULE_AUTHOR("Eric Bénard <eric(a)eukrea.com>"); MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:eukrea_tlv320"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: add DT bindings for CS4271
by Daniel Mack 28 Sep '12

28 Sep '12

Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 36 ++++++++++++++++++++++ sound/soc/codecs/cs4271.c | 24 +++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt new file mode 100644 index 0000000..c81b5fd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -0,0 +1,36 @@ +Cirrus Logic CS4271 DT bindings + +This driver supports both the I2C and the SPI bus. + +Required properties: + + - compatible: "cirrus,cs4271" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the i2c address + + +Optional properties: + + - reset-gpio: a GPIO spec to define which pin is connected to the chip's + !RESET pin + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9eb01d7..f994af3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -22,12 +22,14 @@ #include <linux/module.h> #include <linux/slab.h> #include <linux/delay.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/tlv.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> #include <sound/cs4271.h> #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -458,6 +460,14 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) #define cs4271_soc_resume NULL #endif /* CONFIG_PM */ +#ifdef CONFIG_OF +static const struct of_device_id cs4271_dt_ids[] = { + { .compatible = "cirrus,cs4271", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4271_dt_ids); +#endif + static int cs4271_probe(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -465,6 +475,12 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; +#ifdef CONFIG_OF + if (of_match_device(cs4271_dt_ids, codec->dev)) + gpio_nreset = of_get_named_gpio(codec->dev->of_node, + "reset-gpio", 0); +#endif + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -569,6 +585,7 @@ static struct spi_driver cs4271_spi_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .probe = cs4271_spi_probe, .remove = __devexit_p(cs4271_spi_remove), @@ -608,6 +625,7 @@ static struct i2c_driver cs4271_i2c_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .id_table = cs4271_i2c_id, .probe = cs4271_i2c_probe, -- 1.7.11.4

2 1

Re: [alsa-devel] ASoC: dapm: Allow regulators to bypass as well as disable when idle
by Dan Carpenter 28 Sep '12

28 Sep '12

Hi Mark, I had a question about patch c05b84d14b23: "ASoC: dapm: Allow regulators to bypass as well as disable when idle" from Sep 7, 2012. 1022 if (SND_SOC_DAPM_EVENT_ON(event)) { 1023 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 1024 ret = regulator_allow_bypass(w->regulator, true); 1025 if (ret != 0) 1026 dev_warn(w->dapm->dev, 1027 "Failed to bypass %s: %d\n", 1028 w->name, ret); 1029 } 1030 1031 return regulator_enable(w->regulator); 1032 } else { 1033 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Right now ->invert is a single bit bitfield. We mostly test "if (w->invert) {...". Are we planning to add some more bits in there? 1034 ret = regulator_allow_bypass(w->regulator, false); 1035 if (ret != 0) 1036 dev_warn(w->dapm->dev, 1037 "Failed to unbypass %s: %d\n", 1038 w->name, ret); 1039 } regards, dan carpenter

2 1

Re: [alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s
by Stephen Warren 28 Sep '12

28 Sep '12

On 09/27/2012 04:25 AM, Padmavathi Venna wrote: > Add support for device based discovery. > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > +i2s@03830000 { > + compatible = "samsung,samsung-i2s"; > + reg = <0x03830000 0x100>; > + tx-dma-channel-secondary = <&pdma0 8>; > + tx-dma-channel = <&pdma0 10>; > + rx-dma-channel = <&pdma0 9>; > + supports-6ch; > + supports-rstclr; > + supports-secdai; Those 6 properties are custom to this binding so should have a vendor prefix, e.g. "samsung,supports-6ch". Also, I wonder if this shouldn't wait for the forthcoming standardized DT DMA bindings (or perhaps document custom bindings that are identical to the new standardized bindings)? Otherwise, the DMA stuff will all have to be redefined again. Apparently it's hoped the bindings will be in 3.7, or if not there, in 3.8.

2 1

Re: [alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s
by Sangbeom Kim 28 Sep '12

28 Sep '12

Hi! On Thu, Sep 27, 2012 at 07:26 PM, Padmavathi Venna wrote: > Subject: [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s > > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> Please describe the difference between v1 and v2. Thanks, Sangbeom.

2 1

[alsa-devel] [PATCH] pcm: support for audio timestamps
by Pierre-Louis Bossart 28 Sep '12

28 Sep '12

add new snd_pcm_status_get_audio_htstamp() routine to query the audio timestamps provided by the kernel. This change provides applications with better ways to track elapsed time. Before this patch, applications would subtract queued samples (delay) from written samples, resulting in a 1-2 sample error. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++-- src/pcm/pcm.c | 31 +++++++++++++++++++++++++++++++ test/audio_time.c | 27 +++++++++++++++++---------- 4 files changed, 55 insertions(+), 12 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..8d08471 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -550,6 +550,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -858,6 +859,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..109302e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..748e8a6 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3123,6 +3123,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6213,6 +6233,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * diff --git a/test/audio_time.c b/test/audio_time.c index a910783..de8df17 100644 --- a/test/audio_time.c +++ b/test/audio_time.c @@ -33,6 +33,7 @@ long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, snd_htimestamp_t *trigger_timestamp, + snd_htimestamp_t *audio_timestamp, snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) { int err; @@ -45,6 +46,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, } snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); snd_pcm_status_get_htstamp(status, timestamp); + snd_pcm_status_get_audio_htstamp(status, audio_timestamp); *avail = snd_pcm_status_get_avail(status); *delay = snd_pcm_status_get_delay(status); } @@ -65,6 +67,7 @@ int main(void) snd_pcm_sframes_t frames; snd_htimestamp_t tstamp_c, tstamp_p; snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + snd_htimestamp_t audio_tstamp_c, audio_tstamp_p; unsigned char buffer_p[PERIOD*4*4]; unsigned char buffer_c[PERIOD*4*4]; @@ -202,26 +205,30 @@ int main(void) frame_count_p += frames; #if defined(TRACK_PLAYBACK) - gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &audio_tstamp_p, &avail_p, &delay_p); curr_count_p = frame_count_p - delay_p; /* written minus queued */ - printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_p,trigger_tstamp_p), - (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), - timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + /* uncomment next line to see difference between old sample count and audio timestamps. + printf("curr_count %lli driver count %lli\n", (long long)curr_count_p * 1000000000LL / 48000, timestamp2ns(audio_tstamp_p)); + */ + + printf("playback: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_p, trigger_tstamp_p), + timestamp2ns(audio_tstamp_p), + timediff(tstamp_p, trigger_tstamp_p) - timestamp2ns(audio_tstamp_p) ); #endif #if defined(TRACK_CAPTURE) - gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &audio_tstamp_c, &avail_c, &delay_c); curr_count_c = frame_count_c + delay_c; /* read plus queued */ - printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_c,trigger_tstamp_c), - (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), - timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + printf("\t capture: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_c, trigger_tstamp_c), + timestamp2ns(audio_tstamp_c), + timediff(tstamp_c, trigger_tstamp_c) - timestamp2ns(audio_tstamp_c) ); #endif -- 1.7.9.5

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[alsa-devel] setup_dig_out_stream() resets the SPDIF steam causing clicks
by Laurence Darby 28 Sep '12

28 Sep '12

Hi, I'm playing audio via SPDIF, with a Realtek codec, and there's a problem that whenever audio is started, or seeking in every music player I tried, the DAC lost sync briefly, which made it 'click' when it muted and unmuted itself. I recently upgraded the DAC, and this problem only happened with the new DAC, but not the old one, I don't know (or really care) why - the old DAC had much worse problems. Here's what I did to fix this: diff -up linux-3.5.4/sound/pci/hda/hda_codec.c.orig linux-3.5.4/sound/pci/hda/hda_codec.c --- linux-3.5.4/sound/pci/hda/hda_codec.c.orig 2012-09-28 00:13:25.000000000 +0100 +++ linux-3.5.4/sound/pci/hda/hda_codec.c 2012-09-28 00:39:31.000000000 +0100 @@ -4693,9 +4693,11 @@ static void setup_dig_out_stream(struct format); } /* turn on again (if needed) */ - if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) { set_dig_out_convert(codec, nid, spdif->ctls & 0xff, -1); + codec->spdif_status_reset=0; + } } static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) That makes SPDIF turn off and on just once when sound is first played (i.e. it clicks once then, then no more). In patch_realtek.c it's got this: /* FIXME: do we need this for all Realtek codec models? */ codec->spdif_status_reset = 1; If I set that to 0, then the left channel doesn't initialize and is silent, so the reset really does need to happen once (for my realtek model anyway, I don't know how to find out what that would be?) So, is the above an acceptable fix? Regards, Laurence

1 0

[alsa-devel] [PATCH] ASoC: wm5110: Adding missing volume update bits.
by Mark Brown 27 Sep '12

27 Sep '12

From: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> The volume update bits were being set on all but one input and one output. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b99451e..9211e41 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -916,6 +916,8 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_ADC_DIGITAL_VOLUME_3L, ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, @@ -927,6 +929,8 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, + ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, }; static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { -- 1.7.10.4

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[alsa-devel] ASoC updates for 3.6
by Mark Brown 27 Sep '12

27 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40: ASoC: wm2000: Correct register size (2012-09-26 12:06:20 +0100) Takashi is on vacation now so he asked me to send directly, it's a pretty bad bug with low regression risk. ---------------------------------------------------------------- ASoC: Updates for 3.6 One small and obvious driver-specific fix. ---------------------------------------------------------------- Mark Brown (1): ASoC: wm2000: Correct register size sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

1 0

[alsa-devel] ASoC on today's linux-next: dma_alloc_writecombine() failing at least on OMAP platforms
by Peter Ujfalusi 27 Sep '12

27 Sep '12

Hello, I tried today's linux-next on OMAP platform and audio is not probing because dma_alloc_writecombine() fails to get the area. My linux-next from two days ago works OK and I don't see anything directly connected to OMAP which can cause this failure. Can someone try linux-next on non OMAP platform to see if dma_alloc_writecombine() works there? Meanwhile I'll try to find the reason why it fails for me. Thanks, Péter

2 2

[alsa-devel] OMAP4 McBSP AC97 link
by Sebastian 27 Sep '12

27 Sep '12

Hi All, i have to support a 2nd Audio Codec with AC97 Interface on McBSP Port2. Processor: OMAP4460 If any one have any idea on how to go about this. Please let me know. BR, Sebastian

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[alsa-devel] sound/pci/hda/patch_ca0132.c: potential null dereference 'dma_engine'
by Fengguang Wu 27 Sep '12

27 Sep '12

Hi Ian, FYI, there are new smatch warnings show up in commit 0b37d4c7982442730061a737c16939e8cc363861 Author: Ian Minett <ian_minett(a)creativelabs.com> Commit: Takashi Iwai <tiwai(a)suse.de> CommitDate: Fri Sep 21 09:54:29 2012 +0200 + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1963) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Add more clock rates
by Mark Brown 26 Sep '12

26 Sep '12

Some devices support additional clock rates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5764960..c03b65a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -270,6 +270,9 @@ static unsigned int arizona_sysclk_48k_rates[] = { 12288000, 22579200, 49152000, + 73728000, + 98304000, + 147456000, }; static unsigned int arizona_sysclk_44k1_rates[] = { @@ -277,6 +280,9 @@ static unsigned int arizona_sysclk_44k1_rates[] = { 11289600, 24576000, 45158400, + 67737600, + 90316800, + 135475200, }; static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: pcm: Report error code when we fail to init platform
by Mark Brown 26 Sep '12

26 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a67b239..40c4edb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2097,7 +2097,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - dev_err(platform->dev, "pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed: %d\n", + ret); return ret; } } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: wm0010: Allow slow GPIO for reset
by Mark Brown 26 Sep '12

26 Sep '12

We never set the GPIO from atomic context so there's no reason why we can't support a GPIO that needs to sleep when configuring. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f8d6c31..4722acf 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -168,7 +168,8 @@ static void wm0010_halt(struct snd_soc_codec *codec) case WM0010_STAGE2: case WM0010_FIRMWARE: /* Remember to put chip back into reset */ - gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_set_value_cansleep(wm0010->gpio_reset, + wm0010->gpio_reset_value); /* Disable the regulators */ regulator_disable(wm0010->dbvdd); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), @@ -387,7 +388,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) } /* Release reset */ - gpio_set_value(wm0010->gpio_reset, !wm0010->gpio_reset_value); + gpio_set_value_cansleep(wm0010->gpio_reset, !wm0010->gpio_reset_value); spin_lock_irqsave(&wm0010->irq_lock, flags); wm0010->state = WM0010_OUT_OF_RESET; spin_unlock_irqrestore(&wm0010->irq_lock, flags); @@ -918,7 +919,8 @@ static int __devexit wm0010_spi_remove(struct spi_device *spi) if (wm0010->gpio_reset) { /* Remember to put chip back into reset */ - gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_set_value_cansleep(wm0010->gpio_reset, + wm0010->gpio_reset_value); } if (wm0010->irq) -- 1.7.10.4

1 2

[alsa-devel] ASoC fix for 3.6
by Mark Brown 26 Sep '12

26 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40: ASoC: wm2000: Correct register size (2012-09-26 12:06:20 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.6 One small and obvious driver-specific fix. ---------------------------------------------------------------- Mark Brown (1): ASoC: wm2000: Correct register size sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

1 0

[alsa-devel] [PATCH] ASoC: wm2000: Correct register size
by Mark Brown 26 Sep '12

26 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index b723e91..683dc43 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -726,7 +726,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) } static const struct regmap_config wm2000_regmap = { - .reg_bits = 8, + .reg_bits = 16, .val_bits = 8, .max_register = WM2000_REG_IF_CTL, -- 1.7.10.4

1 0

[alsa-devel] Merging alsa-driver and alsa-kmirror to one GIT repo
by Jaroslav Kysela 26 Sep '12

26 Sep '12

Hi all, I believe, it's time to manage only one GIT repository for the alsa-driver package now. The merge has benefits mainly for the end users - bisecting may work, the correlation between kernel sources and our out-of-kernel build framework becomes more strong. I merged the alsa-driver and alsa-kmirror repos with full history. I also added some code changes to enable the "build-in" mirror tree in Makefile and gitcompile from v1.0.18 (but I don't think that it will be used). The structure of new repository is similar, but the kernel mirrored code is located in the mirror/ subdirectory. The paths are exactly same as in the Linux kernel tree, so possible patches can be applied without any path modifications. Of course, we may create a bash script to do proper and full cherry-picks between kernel and mirrored trees (comments, author, dates). New temporary repository has name alsa-driver.new: http://git.alsa-project.org/?p=alsa-driver.new.git;a=summary The another difference is that the releases are now branches not tags. It allows me to do quick fixes out of the standard development or separate the main development in the release time window (to not include some very new code). The previous branches won't contain probably any additional commits, but it's probably better to play with only one reference scheme. I think that the change from alsa-driver.git to alsa-driver.new.git should be quick. Please, report any objections or acks now. Schedule: When accepted, I will rename the original alsa-driver.git repository to alsa-driver.old.git and move alsa-driver.new.git to alsa-driver.git quickly. 1.0.26 release: This was my last big change (with the ALSA server upgrade) which blocked me to do the 1.0.26 release. I'm going to create new release after the repos switch ASAP. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH 1/2] ASoC: fsl_ssi: Let it understand mono
by Fabio Estevam 25 Sep '12

25 Sep '12

Currently there are two variants for the FSL SSI driver: - fsl_ssi.c is the device tree driver version for the FSL SSI driver (used by PowerPC and i.MX) - imx-ssi.c is the non device tree version that runs on some i.MX processors that are not yet converted to dt. Commit 0865a7 (ASoC: imx-ssi: Remove mono support) attempted to fix mono audio playback issue inside imx-ssi.c, which according to 104c22 (ASoC: Revert 'ASoC: imx-ssi: Remove mono support' ) caused problem on another i.MX platform. The original mono playback issue was later fixed inside the mc13783 codec driver. As the long term goal is to move other platforms to dt, then fsl_ssi needs to have the same fix for channels_min as in imx-ssi. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Timur, Can you please try to play a mono track on your PowerPC platform with these series applied? sound/soc/fsl/fsl_ssi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4ed2afd..7768b93 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -559,14 +559,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { /* Template for the CPU dai driver structure */ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: core: Configure pin muxing via pinctrl when registering a DAI
by Peter Ujfalusi 25 Sep '12

25 Sep '12

pinctrl framework now becoming widely available for SoCs to configure pin multiplexing. Instead of adding the same code to all dai drivers the core can take care of this transparently. Do not make too much noise if pinctrl is not provided via DT for the dai but just use dev_info(). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam, I was about to add the pinctrl support for OMAP dais (McBSP/McPDM/DMIC/HDMI) when I realized that I just kept duplicating the same code all over. It is better to leave this to the core. Regards, Peter sound/soc/soc-core.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9a6daf9..37464c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -33,6 +33,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/pinctrl/consumer.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -3686,6 +3687,19 @@ static inline char *fmt_multiple_name(struct device *dev, return kstrdup(dai_drv->name, GFP_KERNEL); } +/* + * Simple function to be used in snd_soc_dai_register_dai/s to set the default + * pinmuxing for the dai. + */ +static inline void snd_soc_configure_dai_pins(struct device *dev) +{ + struct pinctrl *pinctrl; + + pinctrl = devm_pinctrl_get_select_default(dev); + if (IS_ERR(pinctrl)) + dev_info(dev, "pins are not configured for the DAI\n"); +} + /** * snd_soc_register_dai - Register a DAI with the ASoC core * @@ -3734,6 +3748,8 @@ int snd_soc_register_dai(struct device *dev, mutex_unlock(&client_mutex); + snd_soc_configure_dai_pins(dev); + pr_debug("Registered DAI '%s'\n", dai->name); return 0; @@ -3828,6 +3844,8 @@ int snd_soc_register_dais(struct device *dev, pr_debug("Registered DAI '%s'\n", dai->name); } + snd_soc_configure_dai_pins(dev); + return 0; err: -- 1.7.12

4 23

[alsa-devel] [PATCH] ALSA: hda - Remove ignore_misc_bit
by David Henningsson 25 Sep '12

25 Sep '12

The purpose of this flag is unclear. If the problem is that some machines have broken misc/NO_PRESENCE bits, they should be fixed by pin fixups. In addition, this causes jack detection functionality to be flawed on the M31EI, where there are two jacks without jack detection (which is properly marked as NO_PRESENCE), but due to ignore_misc_bit, these jacks are instead being reported as being present but always unplugged. BugLink: https://bugs.launchpad.net/bugs/939161 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 4 ---- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_jack.c | 5 ++--- 3 files changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4f7d2df..9acd5a9 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -141,7 +141,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, memset(sequences_hp, 0, sizeof(sequences_hp)); assoc_line_out = 0; - codec->ignore_misc_bit = true; end_nid = codec->start_nid + codec->num_nodes; for (nid = codec->start_nid; nid < end_nid; nid++) { unsigned int wid_caps = get_wcaps(codec, nid); @@ -157,9 +156,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - codec->ignore_misc_bit = false; conn = get_defcfg_connect(def_conf); if (conn == AC_JACK_PORT_NONE) continue; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8fd9f63..ee4ae8e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,7 +861,6 @@ struct hda_codec { unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ - unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index aaccc02..c9333c9 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -26,9 +26,8 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) return false; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) return false; - if (!codec->ignore_misc_bit && - (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) + if (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE) return false; if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) return false; -- 1.7.9.5

4 20

[alsa-devel] snd_seq_parse_address accesses uninitialised data
by Henning Thielemann 25 Sep '12

25 Sep '12

I think I found a bug. The following small program simply parses an address: $ cat ctest/parse-address.c #include <alsa/asoundlib.h> int main () { snd_seq_t *seq; snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0); snd_seq_addr_t addr = {128,0}; const char *dest_name = "TiMidity"; snd_seq_parse_address(seq, &addr, dest_name); printf("found client %s: %d:%d\n", dest_name, addr.client, addr.port); snd_seq_close(seq); return 0; } $ gcc -Wall -o ctest/parse-address ctest/parse-address.c `pkg-config --libs alsa` $ valgrind ctest/parse-address ==27797== Memcheck, a memory error detector ==27797== Copyright (C) 2002-2011, and GNU GPL'd, by Julian Seward et al. ==27797== Using Valgrind-3.7.0 and LibVEX; rerun with -h for copyright info ==27797== Command: ctest/parse-address ==27797== ==27797== Syscall param ioctl(generic) points to uninitialised byte(s) ==27797== at 0x5209D27: ioctl (syscall-template.S:82) ==27797== by 0x4EC9168: snd_seq_hw_query_next_client (seq_hw.c:367) ==27797== by 0x4ECF3E2: snd_seq_parse_address (seqmid.c:416) ==27797== by 0x40069D: main (in /home/thielema/programming/haskell/alsa-seq/ctest/parse-address) ==27797== Address 0x7fefffd24 is on thread 1's stack ==27797== found client TiMidity: 129:0 ==27797== ==27797== HEAP SUMMARY: ==27797== in use at exit: 75,317 bytes in 1,409 blocks ==27797== total heap usage: 2,054 allocs, 645 frees, 278,456 bytes allocated ==27797== ==27797== LEAK SUMMARY: ==27797== definitely lost: 0 bytes in 0 blocks ==27797== indirectly lost: 0 bytes in 0 blocks ==27797== possibly lost: 42,852 bytes in 1,310 blocks ==27797== still reachable: 32,465 bytes in 99 blocks ==27797== suppressed: 0 bytes in 0 blocks ==27797== Rerun with --leak-check=full to see details of leaked memory ==27797== ==27797== For counts of detected and suppressed errors, rerun with: -v ==27797== Use --track-origins=yes to see where uninitialised values come from ==27797== ERROR SUMMARY: 1 errors from 1 contexts (suppressed: 2 from 2) $ pkg-config --modversion alsa 1.0.25

3 5

[alsa-devel] [PATCH v3 00/15] ASoC: OMAP: Convert to use dmaengine
by Peter Ujfalusi 25 Sep '12

25 Sep '12

Hello, Changes since v2: - As it has been discussed the no_wakeup parameter has been replaced with flags for the dmaengine APIs Changes since v1: - Support for pause/resume for OMAP audio via dmaengine - dmaengine: support for NO_PERIOD_WAKEUP in cyclic mode - OMAP to keep supporting NO_PERIOD_WAKEUP for audio - Other plaforms can also try to enable this mode since we have now generic interface to do so. This series will switch the OMAP audio to use dmaengine. The final patch which does the switch was based on Russell King's earlier patch. The first 10 patch is to prepare the OMAP audio drivers for a smooth change to dmaengine: - sDMA FRAME sync mode is removed and replaced with PACKET mode - dai drivers no longer need to configure sDMA sync mode - dai drivers does not need to specify the DMA word length - with the exception of the omap-hdmi driver which requires 32bit word length regardless of the audio format in use - the McPDM driver used (to my surprise) hackish way of getting the DMA channel and address - via defines from some header files After the conversion OMAP audio support should have the same features as before, no regressions expected. I have tested the series on: - BeagleBoard (audio via McBSP): - aplay/arecord. In element mode and in threshold mode with different period sizes - mplayer -ao alsa: for direct ALSA access - mplayer -ao pulse: via PulseAudio to test NO_PERIOD_WAKEUP feature - OMAP4 Blaze (audio via McPDM and DMIC) - aplay/arecord - mplayer -ao alsa: for direct ALSA access - mplayer -ao pulse: via PulseAudio to test NO_PERIOD_WAKEUP feature The patches has been generated against: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 Janusz: Can you retest this series on OMAP1 to be sure I have not broken it? Ricardo: Can you test the omap-hmdi if it is still working? Regards, Peter --- Peter Ujfalusi (15): dmaengine: omap: Support for element mode in cyclic DMA dmaengine: omap: Add support for pause/resume in cyclic dma mode dmaengine: Add flags parameter to dmaengine_prep_dma_cyclic() dmaengine: Pass flags via device_prep_dma_cyclic() callback dmaengine: omap-dma: Add support to suppress interrupts in cyclic mode ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-dmic: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + drivers/dma/at_hdmac.c | 3 +- drivers/dma/ep93xx_dma.c | 4 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/imx-sdma.c | 2 +- drivers/dma/mmp_tdma.c | 2 +- drivers/dma/mxs-dma.c | 2 +- drivers/dma/omap-dma.c | 45 ++++-- drivers/dma/pl330.c | 2 +- drivers/dma/sa11x0-dma.c | 2 +- drivers/dma/sirf-dma.c | 2 +- drivers/dma/ste_dma40.c | 3 +- drivers/dma/tegra20-apb-dma.c | 2 +- include/linux/dmaengine.h | 7 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 ++- sound/soc/omap/omap-mcbsp.c | 60 +++----- sound/soc/omap/omap-mcpdm.c | 40 +++-- sound/soc/omap/omap-pcm.c | 236 ++++++++--------------------- sound/soc/omap/omap-pcm.h | 4 +- sound/soc/soc-dmaengine-pcm.c | 6 +- 22 files changed, 186 insertions(+), 269 deletions(-) -- 1.7.12

8 30

[alsa-devel] [PATCH] ASoC: Add device tree binding description for pcm030-audio-fabric
by Eric Millbrandt 25 Sep '12

25 Sep '12

Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- .../bindings/sound/pcm030-audio-fabric.txt | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/pcm030-audio-fabric.txt diff --git a/Documentation/devicetree/bindings/sound/pcm030-audio-fabric.txt b/Documentation/devicetree/bindings/sound/pcm030-audio-fabric.txt new file mode 100644 index 0000000..6eacf23 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm030-audio-fabric.txt @@ -0,0 +1,12 @@ +Phytec pcm030 audio + +Required properties: +- compatible : "phytec,pcm030-audio-fabric" +- asoc-platform : The mpc5200 controller's phandle for the ac97 controller + +Example: + +sound { + compatible = "phytec,pcm030-audio-fabric"; + asoc-platform = <&audioplatform>; +}; -- 1.7.2.5

2 1

[alsa-devel] HD Audio with 'mute light'
by Arnout Engelen 24 Sep '12

24 Sep '12

Hi, I've got a laptop with a HD Audio onboard sound card: the CX20590. It seems like this card has a sort of 'hardware mute state': the 'mute button' has a light, and when this light is on, no sound is heard, regardless of whether or not the channel is muted in the ALSA mixer. The hardware mute button switches the 'hardware mute state', and also sends a keyboard event which I can use to also update the ALSA mixer mute. This doesn't, however, work the other way around: I haven't found any way yet to manipulate the 'hardware mute state' from software. Looking with hda_analyzer, I notice this card has a 'Vendor Defined Widget'. Might this be something that could be used to manipulate the hardware mute state? I can use hda-verb but I wouldn't know how to discover which verb to use. Anything else to try? Attached is the codec info. Kind regards, Arnout

3 3

[alsa-devel] Dynamic HDMI PCM creation
by Takashi Iwai 23 Sep '12

23 Sep '12

Hi David, as we discussed at Plumbers, I tried some hack to create/delete HDMI/DP PCM stream per hotplug/unplug. Test patches are found in sound-unstable git tree topic/hdmi-dynamic-pcm branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git The patches are all small and easy, but it's still in a pretty rough cut. It won't handle some cases, e.g. unplug during suspend, or such, I'm afraid. After all, it's just a test. With these patches, the PCM device appears and disappears properly upon HDMI/DP hotplug/unplug on my system. On mine, it appears as /dev/snd/pcmC0D8 as it's an Intel on-board. So far, so good. Now the problem is that PA gets confused when this happens. It can switch to HDMI/DP, but then the analog output disappears from PA's profile. You cannot switch back to analog output after that, even after you unplug HDMI/DP cable. Or, it might be my PA version... I'll check newer one. But it'd be good if you also check in your side. thanks, Takashi

4 8

[alsa-devel] [PATCH 1/2] ASoC: wm2000: Convert to devm_regmap_init_i2c()
by Mark Brown 23 Sep '12

23 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 89cd6fc..6675477 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -760,7 +760,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, dev_set_drvdata(&i2c->dev, wm2000); - wm2000->regmap = regmap_init_i2c(i2c, &wm2000_regmap); + wm2000->regmap = devm_regmap_init_i2c(i2c, &wm2000_regmap); if (IS_ERR(wm2000->regmap)) { ret = PTR_ERR(wm2000->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -777,7 +777,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (id != 0x2000) { dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); ret = -ENODEV; - goto out_regmap_exit; + goto out; } reg = wm2000_read(i2c, WM2000_REG_REVISON); @@ -796,7 +796,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = request_firmware(&fw, filename, &i2c->dev); if (ret != 0) { dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); - goto out_regmap_exit; + goto out; } /* Pre-cook the concatenation of the register address onto the image */ @@ -807,7 +807,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (wm2000->anc_download == NULL) { dev_err(&i2c->dev, "Out of memory\n"); ret = -ENOMEM; - goto out_regmap_exit; + goto out; } wm2000->anc_download[0] = 0x80; @@ -825,8 +825,6 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (!ret) goto out; -out_regmap_exit: - regmap_exit(wm2000->regmap); out: release_firmware(fw); return ret; @@ -834,10 +832,7 @@ out: static __devexit int wm2000_i2c_remove(struct i2c_client *i2c) { - struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); - snd_soc_unregister_codec(&i2c->dev); - regmap_exit(wm2000->regmap); return 0; } -- 1.7.10.4

1 1

[alsa-devel] ASoC updates for 3.7
by Mark Brown 22 Sep '12

22 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.7 for you to fetch changes up to ddfb43f3881edb47aa0083651ad31983cdc42c33: Merge tag 'v3.6-rc6' into for-3.7 (2012-09-22 11:26:27 -0400) ---------------------------------------------------------------- ASoC: Updates for v3.7 Lots and lots of driver specific cleanups and enhancements but the only substantial framework feature this time round is the compressed API binding: - Addition of ASoC bindings for the compressed API, used by the mid-x86 drivers. - Lots of cleanups and API refreshes for CODEC drivers and DaVinci. - Conversion of OMAP to dmaengine. - New machine driver for Wolfson Microelectronics Bells. - New CODEC driver for Wolfson Microelectronics WM0010. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: Rename ep93xx soc directory to cirrus Bo Shen (1): ASoC: wm8904: remove redundant code Chris Rattray (1): ASoC: wm8994: enable mic and short detect debounce. Dan Carpenter (1): ASoC: wm0010: unlock on error path Daniel Mack (2): ASoC: add DT bindings for cs4270 ASoC: Add reset-gpio DT property to cs4270 driver Dimitris Papastamos (1): ASoC: wm0010: Add initial wm0010 DSP driver Emil Goode (1): ASoC: wm0010: Fix warning, use format %zu for type size_t Eric Millbrandt (6): ASoC: fsl: mpc5200 combine psc_dma platform data ASoC: fsl: mpc5200 add missing information to snd_soc_dai_driver ASoC: fsl: cleanup headers in pcm030-audio-fabric ASoC: fsl: add PPC_MPC52xx dependency to SND_POWERPC_SOC ASoC: fsl: pcm030-audio-fabric use snd_soc_register_card ASoC: fsl: register the wm9712-codec Fabio Estevam (5): ASoC: imx-ssi: Use devm functions ASoC: mxs-saif: Use devm_clk_get() ASoC: Revert "ASoC: mc13783: Provide codec->control_data" ASoC: Revert "ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements" Revert "ASoC: AC97 doesn't use regmap by default" Fengguang Wu (1): ASoC: wm0010: Add missing IRQF_ONESHOT Gaëtan Carlier (1): ASoC: imx-mc13783: use defines instead of numerical address of register Hebbar, Gururaja (9): ASoC: Davinci: McASP: remove unused header include ASoC: McASP: Convert driver to use Runtime PM API ASoC: tlv320aic3x: Add device tree bindings ASoC: davinci: davinci-pcm does not need to be a plaform_driver ARM: Davinci: Remove references to davinci pcm ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific ASoC: Davinci: McASP: add device tree support for McASP ASoC: Davinci: evm: Fix typo in cpu dai name ASoC: Davinci: McASP: add support new McASP IP Variant Javier Martin (1): ASoC: Revert 'ASoC: imx-ssi: Remove mono support' Jerry Snitselaar (1): ASoC: core: remove unused variable in soc_probe() in linux-next Kuninori Morimoto (3): ASoC: fsi: fixup pm_runtime_disable() timing on fsi_probe() ASoC: fsi: tidyup: remove un-necessary operation from fsi_probe() ASoC: fsi: convert to devm_xxx() Lars-Peter Clausen (4): ASoC: Remove unused 'saved_value' field from snd_soc_dapm_widget struct ASoC: ad193x: Use managed regmap init ASoC: ad1836: Convert to direct regmap usage. ASoC: Avoid recalculating the bitmask for SOC_ENUM controls Lee Jones (1): ASoC: ux500: Strengthen error checking after memory allocation Mark Brown (79): ASoC: wm8994: Implement support for self-oscillation mode in the FLL ASoC: jack: Always update jack state even for noop changes ASoC: wm_hubs: Move CODEC stored in private data into wm_hubs ASoC: wm8994: Ensure we get a notification on startup for jackdet ASoC: wm_hubs: Factor out DC servo readback code ASoC: wm_hubs: Rename calibrate_dc_servo() ASoC: wm8994: Fix some indentation issues ASoC: ab8500: Remove pointless cast ASoC: wm9712: Fix funky indentation ASoC: wm9712: Fix name of Capture Switch ASoC: sta529: Staticise non-exported codec driver struct ASoC: imx-ssi: Say if we fail to register a second AC'97 bus ASoC: core: Remove pointless error on card registration failure ASoC: arizona: Implement OPCLK support ASoC: wm5102: Enable output clocks ASoC: wm5110: Enable output clocks ASoC: bells: Add machine driver for Wolfson Bells boards Merge remote-tracking branch 'asoc/topic/omap' into for-3.7 ASoC: wm5102: Add AEC routing control ASoC: wm8994: Add bytes controls for DRC Merge remote-tracking branch 'asoc/topic/blackfin' into for-3.7 ASoC: wm9712: Provide TLV information for capture boost controls ASoC: wm_hubs: Use explicit casts for converting to signed ASoC: wm_hubs: Add trace showing semantics of the DCS update ASoC: wm_hubs: Allow configuration of MICBIAS power up delay via pdata Merge branch 'asoc-omap' into for-3.7 Merge branch 'asoc-omap' into for-3.7 ASoC: wm8994: Update for new WM1811 variants ASoC: wm0010: Fix passthrough routing ASoC: samsung: Add hookup of WM0010 on Speyside ASoC: wm0010: Enable 44.1kHz support ASoC: wm0010: Add dummy widget for CLKIN ASoC: wm0010: Set idle_bias_off ASoC: wm0010: Don't double free reset GPIO ASoC: wm0010: Tweak diagnostic output ASoC: wm0010: Move resource acquisition to device probe Merge branch 'asoc-omap' into for-3.7 Merge tag 'v3.6-rc4' into asoc-omap Merge branch 'asoc-omap' into for-3.7 ASoC: dapm: Ensure bypass paths are suspended and resumed ASoC: dapm: Add flags to regulator supplies ASoC: wm8983: Convert to devm_kzalloc() ASoC: wm8983: Convert to direct regmap API usage ASoC: wm8523: Convert to devm_kzalloc() ASoC: wm8523: Move regulator acquisition to I2C probe() ASoC: wm8523: Convert to direct regmap API usage ASoC: wm8523: Move device ID verification and reset to I2C probe ASoC: sta32x: Move regulator acquisition to I2C probe ASoC: sta32x: Convert to regmap ASoC: wm8741: Move regulator acquisition to I2C/SPI probe() ASoC: wm8741: Convert to direct regmap API usage ASoC: wm8960: Convert to direct regmap API usage ASoC: cs4270: Conver to data based control init ASoC: cs4270: Move regulator acquisition to I2C probe() ASoC: cs4270: Convert to direct regmap API usage ASoC: wm8900: Convert to devm_kzalloc() ASoC: wm8900: Convert to direct regmap API usage ASoC: wm8900: Fix typo of name to wm9700 ASoC: wm8510: Convert to devm_kzalloc() ASoC: wm8580: Convert to devm_kzalloc() ASoC: wm8711: Convert to devm_kzalloc() ASoC: wm8728: Convert to devm_kzalloc() ASoC: wm8737: Convert to devm_kzalloc() ASoC: wm8990: Convert to devm_kzalloc() ASoC: wm8991: Convert to devm_kzalloc() ASoC: wm8510: Convert to direct regmap API usage ASoC: wm8580: Convert to direct regmap API usage ASoC: wm8580: Move regulator acquisition to I2C probe ASoC: wm8711: Convert to direct regmap API usage ASoC: wm8728: Convert to direct regmap API usage ASoC: wm8737: Move regulator acquisition to device registration ASoC: wm8737: Convert to direct regmap API usage ASoC: arizona: Clarify logging for FLL lock status interrupt ASoC: core: Mark regmap CODEC register maps as dirty when suspending ASoC: wm8776: Convert to direct regmap API usage ASoC: wm8960: Support shared LRCLK ASoC: wm8961: Convert to direct regmap API usage ASoC: wm8961: Move device identification and reset to I2C probe Merge tag 'v3.6-rc6' into for-3.7 Namarta Kohli (1): ASoC: add compress stream support Peter Senna Tschudin (1): ASoC: core: Remove useless kfree Peter Ujfalusi (54): ASoC: omap-mcpdm: Convert to use devm_* ASoC: omap-abe-twl6040: Add device tree support ASoC: core: Set dapm->idle_bias_off for DAIs not mapped with a codec ASoC: omap-twl4030: Simple machine driver for TI SoC with twl4030 codec ARM: OMAP: twl-common: Add helper function to register the omap-twl4030 audio driver ARM: OMAP3: Switch to use the unified audio driver (omap-twl4030) for selected boards ASoC: Remove obsolete OMAP3 machine drivers ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver ARM: OMAP: mcbsp: Enable FIFO use for OMAP2430 ARM: OMAP: board-am3517evm: Configure McBSP1 CLKR/FSR signal source ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code ASoC: omap-mcbsp: Remove unused defines ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code ARM: OMAP2+: McBSP: Do not create legacy devices when booting with DT data ASoC: omap-mcbsp: Add device tree bindings ASoC: omap-mcbsp: Device tree binding documentation update ARM: OMAP2+: twl-common: Fix compile time error when omap-twl4030 audio is not enabled ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls ASoC: omap-mcbsp: Single macro for st channel volume set/get ASoC: omap-mcbsp: Fix compilation error due to leftover code ASoC: am3517evm: Remove unused cpu_dai from hw_params ASoC: twl4030: Convert to use DAI DAPM widgets ASoC: twl6040: Convert to use DAI DAPM widgets dmaengine: omap: Support for element mode in cyclic DMA dmaengine: omap: Add support for pause/resume in cyclic dma mode dmaengine: Pass flags via device_prep_dma_cyclic() callback dmaengine: omap-dma: Add support to suppress interrupts in cyclic mode ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-dmic: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine mfd: twl4030-audio: Clean up MODULE_* and platform_driver part mfd: twl4030-audio: Convert to use devm_kzalloc mfd: twl4030-audio: Rearange and clean-up the probe function mfd: twl-core: Add API to query the HFCLK rate mfd: twl4030-audio: Get audio MCLK via twl-core API instead of pdata dt: Add empty of_find_node_by_name() function mfd: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/mfd: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel Richard Zhao (3): ASoC: imx-pcm-dma: check kzalloc return value in function snd_imx_open ASoC: imx-pcm-dma: open function failed when snd_dmaengine_pcm_open fail ASoC: imx-audmux: remove null check of audmux_base in audmux_read_file Sachin Kamat (33): ASoC: isabelle: Remove version.h header file inclusion ASoC: tlv320aic32x4: Use module_i2c_driver ASoC: max9877: Use module_i2c_driver ASoC: wm9090: Use module_i2c_driver ASoC: wm8991: Use module_i2c_driver ASoC: cs4270: Use module_i2c_driver ASoC: tlv320aic3x: Use module_i2c_driver ASoC: sta32x: Use module_i2c_driver ASoC: tlv320dac33: Use module_i2c_driver ASoC: adau1701: Use module_i2c_driver ASoC: max9850: Use module_i2c_driver ASoC: wm8971: Use module_i2c_driver ASoC: ak4671: Use module_i2c_driver ASoC: lm4857: Use module_i2c_driver ASoC: wm8978: Use module_i2c_driver ASoC: max98088: Use module_i2c_driver ASoC: wm8955: Use module_i2c_driver ASoC: wm2200: Use module_i2c_driver ASoC: wm2000: Use module_i2c_driver ASoC: wm8940: Use module_i2c_driver ASoC: wm8961: Use module_i2c_driver ASoC: wm8903: Use module_i2c_driver ASoC: adau1373: Use module_i2c_driver ASoC: wm8960: Use module_i2c_driver ASoC: wm8974: Use module_i2c_driver ASoC: max98095: Use module_i2c_driver ASoC: cs42l51: Use module_i2c_driver ASoC: tpa6130a2: Use module_i2c_driver ASoC: cs42l52: Remove duplicate inclusion of slab.h header file ASoC: soc-compress: Remove unused variable ASoC: tlv320aic26: Use module_spi_driver ASoC: wm8770: Use module_spi_driver ASoC: ad1836: Use module_spi_driver Sangsu Park (1): ASoC: SAMSUNG: Change Kconfig to support all SAMSUNG ASoC Scott Jiang (2): ASoC: bf5xx-ad1836: convert to use snd_soc_register_card blackfin: add platform device for ad1836 machine driver Shawn Guo (1): ASoC: mx27vis: retrieve gpio numbers from platform_data Stephen Warren (1): ASoC: tegra: move platform data header Timur Tabi (4): ASoC: fsl: use snd_soc_register_card to register the card ASoC: fsl: remove unnecessary call to dma_unmap_single ASoC: wm8960: add support for big-endian audio samples ASoC: wm8960: remove 'dres' field from platform data structure Vinod Koul (3): ASoC: add definations for compressed operations ASoC: mid-x86 - add support for compressed streams ASoC: compress - fix code alignment .../devicetree/bindings/mfd/twl4030-audio.txt | 46 + Documentation/devicetree/bindings/sound/cs4270.txt | 21 + .../bindings/sound/davinci-mcasp-audio.txt | 45 + .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 ++ .../devicetree/bindings/sound/omap-mcbsp.txt | 37 + .../devicetree/bindings/sound/omap-twl4030.txt | 17 + .../devicetree/bindings/sound/tlv320aic3x.txt | 20 + arch/arm/mach-davinci/asp.h | 49 + arch/arm/mach-davinci/davinci.h | 3 +- arch/arm/mach-davinci/devices-da8xx.c | 8 +- arch/arm/mach-davinci/devices.c | 11 - arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 42 +- arch/arm/mach-omap2/board-am3517evm.c | 13 + arch/arm/mach-omap2/board-cm-t35.c | 1 + arch/arm/mach-omap2/board-devkit8000.c | 1 + arch/arm/mach-omap2/board-igep0020.c | 1 + arch/arm/mach-omap2/board-omap3beagle.c | 1 + arch/arm/mach-omap2/board-omap3evm.c | 1 + arch/arm/mach-omap2/board-overo.c | 1 + arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - arch/arm/mach-omap2/mcbsp.c | 126 +-- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + arch/arm/mach-omap2/twl-common.c | 27 + arch/arm/mach-omap2/twl-common.h | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 2 - arch/blackfin/mach-bf527/boards/ezkit.c | 20 + arch/blackfin/mach-bf533/boards/stamp.c | 20 + arch/blackfin/mach-bf537/boards/stamp.c | 20 + arch/blackfin/mach-bf561/boards/ezkit.c | 20 + arch/blackfin/mach-bf609/boards/ezkit.c | 19 + drivers/dma/at_hdmac.c | 3 +- drivers/dma/ep93xx_dma.c | 4 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/imx-sdma.c | 2 +- drivers/dma/mmp_tdma.c | 2 +- drivers/dma/mxs-dma.c | 2 +- drivers/dma/omap-dma.c | 45 +- drivers/dma/pl330.c | 2 +- drivers/dma/sa11x0-dma.c | 2 +- drivers/dma/sirf-dma.c | 2 +- drivers/dma/ste_dma40.c | 3 +- drivers/dma/tegra20-apb-dma.c | 2 +- drivers/input/misc/twl4030-vibra.c | 18 +- drivers/mfd/twl-core.c | 32 + drivers/mfd/twl4030-audio.c | 105 ++- include/linux/dmaengine.h | 4 +- include/linux/i2c/twl.h | 3 +- include/linux/mfd/wm8994/pdata.h | 4 + include/linux/of.h | 6 + include/linux/platform_data/asoc-mx27vis.h | 11 + .../linux/platform_data/davinci_asp.h | 74 +- include/linux/platform_data/omap-twl4030.h | 32 + include/sound/compress_driver.h | 1 + include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 7 +- include/sound/soc.h | 20 +- .../sound/tegra_wm8903.h | 7 +- include/sound/wm0010.h | 27 + include/sound/wm8960.h | 2 +- include/sound/wm8993.h | 4 + sound/soc/Kconfig | 3 +- sound/soc/Makefile | 4 +- sound/soc/blackfin/bf5xx-ad1836.c | 73 +- sound/soc/{ep93xx => cirrus}/Kconfig | 0 sound/soc/{ep93xx => cirrus}/Makefile | 0 sound/soc/{ep93xx => cirrus}/edb93xx.c | 0 sound/soc/{ep93xx => cirrus}/ep93xx-ac97.c | 0 sound/soc/{ep93xx => cirrus}/ep93xx-i2s.c | 0 sound/soc/{ep93xx => cirrus}/ep93xx-pcm.c | 0 sound/soc/{ep93xx => cirrus}/ep93xx-pcm.h | 0 sound/soc/{ep93xx => cirrus}/simone.c | 0 sound/soc/{ep93xx => cirrus}/snappercl15.c | 0 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ab8500-codec.c | 13 +- sound/soc/codecs/ad1836.c | 88 +- sound/soc/codecs/ad193x.c | 50 +- sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/adau1373.c | 12 +- sound/soc/codecs/adau1701.c | 12 +- sound/soc/codecs/ak4671.c | 12 +- sound/soc/codecs/arizona.c | 68 +- sound/soc/codecs/arizona.h | 6 +- sound/soc/codecs/cs4270.c | 152 ++-- sound/soc/codecs/cs42l51.c | 19 +- sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/isabelle.c | 1 - sound/soc/codecs/lm4857.c | 12 +- sound/soc/codecs/max98088.c | 18 +- sound/soc/codecs/max98095.c | 18 +- sound/soc/codecs/max9850.c | 12 +- sound/soc/codecs/max9877.c | 12 +- sound/soc/codecs/mc13783.c | 60 +- sound/soc/codecs/sta32x.c | 151 ++-- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/stac9766.c | 1 - sound/soc/codecs/tlv320aic26.c | 12 +- sound/soc/codecs/tlv320aic32x4.c | 19 +- sound/soc/codecs/tlv320aic3x.c | 49 +- sound/soc/codecs/tlv320dac33.c | 19 +- sound/soc/codecs/tpa6130a2.c | 13 +- sound/soc/codecs/twl4030.c | 157 +++- sound/soc/codecs/twl6040.c | 43 +- sound/soc/codecs/wm0010.c | 944 ++++++++++++++++++++ sound/soc/codecs/wm2000.c | 12 +- sound/soc/codecs/wm2200.c | 16 +- sound/soc/codecs/wm5100.c | 6 +- sound/soc/codecs/wm5102.c | 52 +- sound/soc/codecs/wm5110.c | 18 +- sound/soc/codecs/wm8510.c | 129 ++- sound/soc/codecs/wm8523.c | 184 ++-- sound/soc/codecs/wm8580.c | 149 ++- sound/soc/codecs/wm8711.c | 65 +- sound/soc/codecs/wm8728.c | 60 +- sound/soc/codecs/wm8737.c | 132 +-- sound/soc/codecs/wm8741.c | 115 ++- sound/soc/codecs/wm8770.c | 19 +- sound/soc/codecs/wm8776.c | 75 +- sound/soc/codecs/wm8900.c | 166 ++-- sound/soc/codecs/wm8903.c | 18 +- sound/soc/codecs/wm8904.c | 2 - sound/soc/codecs/wm8940.c | 18 +- sound/soc/codecs/wm8955.c | 18 +- sound/soc/codecs/wm8958-dsp2.c | 28 +- sound/soc/codecs/wm8960.c | 150 +++- sound/soc/codecs/wm8961.c | 492 ++++------ sound/soc/codecs/wm8971.c | 18 +- sound/soc/codecs/wm8974.c | 18 +- sound/soc/codecs/wm8978.c | 18 +- sound/soc/codecs/wm8983.c | 162 ++-- sound/soc/codecs/wm8990.c | 8 +- sound/soc/codecs/wm8991.c | 25 +- sound/soc/codecs/wm8993.c | 2 + sound/soc/codecs/wm8994.c | 132 ++- sound/soc/codecs/wm8994.h | 12 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9090.c | 12 +- sound/soc/codecs/wm9712.c | 12 +- sound/soc/codecs/wm9713.c | 1 - sound/soc/codecs/wm_hubs.c | 114 ++- sound/soc/codecs/wm_hubs.h | 6 + sound/soc/davinci/davinci-evm.c | 19 +- sound/soc/davinci/davinci-i2s.c | 13 +- sound/soc/davinci/davinci-mcasp.c | 260 +++++- sound/soc/davinci/davinci-mcasp.h | 6 +- sound/soc/davinci/davinci-pcm.c | 24 +- sound/soc/davinci/davinci-pcm.h | 6 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 8 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_dma.c | 6 - sound/soc/fsl/imx-audmux.c | 3 - sound/soc/fsl/imx-pcm-dma.c | 5 +- sound/soc/fsl/imx-ssi.c | 30 +- sound/soc/fsl/mpc5200_dma.c | 24 +- sound/soc/fsl/mpc5200_dma.h | 3 + sound/soc/fsl/mpc5200_psc_ac97.c | 10 + sound/soc/fsl/mpc5200_psc_i2s.c | 8 + sound/soc/fsl/mpc8610_hpcd.c | 32 +- sound/soc/fsl/mx27vis-aic32x4.c | 42 +- sound/soc/fsl/p1022_ds.c | 31 +- sound/soc/fsl/pcm030-audio-fabric.c | 100 ++- sound/soc/mid-x86/mfld_machine.c | 9 + sound/soc/mid-x86/sst_dsp.h | 134 +++ sound/soc/mid-x86/sst_platform.c | 204 ++++- sound/soc/mid-x86/sst_platform.h | 26 +- sound/soc/mxs/mxs-saif.c | 20 +- sound/soc/omap/Kconfig | 51 +- sound/soc/omap/Makefile | 10 +- sound/soc/omap/am3517evm.c | 21 +- sound/soc/omap/igep0020.c | 120 --- sound/soc/omap/mcbsp.c | 54 +- sound/soc/omap/mcbsp.h | 3 - sound/soc/omap/omap-abe-twl6040.c | 145 ++- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 +- sound/soc/omap/omap-mcbsp.c | 230 ++--- sound/soc/omap/omap-mcbsp.h | 20 - sound/soc/omap/omap-mcpdm.c | 92 +- sound/soc/omap/omap-pcm.c | 236 ++--- sound/soc/omap/omap-pcm.h | 4 +- sound/soc/omap/omap-twl4030.c | 188 ++++ sound/soc/omap/omap3beagle.c | 150 ---- sound/soc/omap/omap3evm.c | 118 --- sound/soc/omap/overo.c | 122 --- sound/soc/omap/zoom2.c | 4 - sound/soc/samsung/Kconfig | 11 +- sound/soc/samsung/Makefile | 2 + sound/soc/samsung/bells.c | 346 +++++++ sound/soc/samsung/speyside.c | 42 +- sound/soc/sh/fsi.c | 28 +- sound/soc/soc-compress.c | 294 ++++++ sound/soc/soc-core.c | 100 ++- sound/soc/soc-dapm.c | 44 +- sound/soc/soc-jack.c | 6 - sound/soc/tegra/tegra_wm8903.c | 3 +- sound/soc/ux500/ux500_msp_dai.c | 3 + sound/soc/ux500/ux500_msp_i2s.c | 2 + 204 files changed, 5905 insertions(+), 3245 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cs4270.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt create mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 arch/arm/mach-davinci/asp.h create mode 100644 include/linux/platform_data/asoc-mx27vis.h rename arch/arm/mach-davinci/include/mach/asp.h => include/linux/platform_data/davinci_asp.h (56%) create mode 100644 include/linux/platform_data/omap-twl4030.h rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (89%) create mode 100644 include/sound/wm0010.h rename sound/soc/{ep93xx => cirrus}/Kconfig (100%) rename sound/soc/{ep93xx => cirrus}/Makefile (100%) rename sound/soc/{ep93xx => cirrus}/edb93xx.c (100%) rename sound/soc/{ep93xx => cirrus}/ep93xx-ac97.c (100%) rename sound/soc/{ep93xx => cirrus}/ep93xx-i2s.c (100%) rename sound/soc/{ep93xx => cirrus}/ep93xx-pcm.c (100%) rename sound/soc/{ep93xx => cirrus}/ep93xx-pcm.h (100%) rename sound/soc/{ep93xx => cirrus}/simone.c (100%) rename sound/soc/{ep93xx => cirrus}/snappercl15.c (100%) create mode 100644 sound/soc/codecs/wm0010.c create mode 100644 sound/soc/mid-x86/sst_dsp.h delete mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/omap/omap-twl4030.c delete mode 100644 sound/soc/omap/omap3beagle.c delete mode 100644 sound/soc/omap/omap3evm.c delete mode 100644 sound/soc/omap/overo.c create mode 100644 sound/soc/samsung/bells.c create mode 100644 sound/soc/soc-compress.c

2 1

[alsa-devel] [PATCH v2 00/34] i.MX multi-platform support
by Shawn Guo 22 Sep '12

22 Sep '12

Here is the second post, which should have addressed the comments that reviewers put on v1. It's available on branch below. git://git.linaro.org/people/shawnguo/linux-2.6.git imx/multi-platform-v2 And it's based on the following branches. calxeda/multi-plat arm-soc/multiplatform/platform-data arm-soc/multiplatform/smp_ops arm-soc/imx/cleanup arm-soc/imx/dt sound/for-3.7 Subsystem maintainers, I plan to send the whole series for 3.7 via arm-soc tree. Please let me know if you have problem with that. Thanks. Shawn --- Shawn Guo (34): ARM: imx: include board headers in the same folder ARM: imx: move iomux drivers and headers into mach-imx ARM: imx: remove unnecessary inclusion from device-imx*.h ARM: imx: move platform device code into mach-imx ARM: imx: merge plat-mxc into mach-imx ARM: imx: include common.h rather than mach/common.h ARM: imx: ARM: imx: include cpuidle.h rather than mach/cpuidle.h ARM: imx: include iim.h rather than mach/iim.h ARM: imx: include iram.h rather than mach/iram.h ARM: imx: include ulpi.h rather than mach/ulpi.h media: mx1_camera: remove the driver ARM: imx: remove mach/dma-mx1-mx2.h dma: ipu: rename mach/ipu.h to include/linux/dma/ipu-dma.h dma: imx-sdma: remove unneeded mach/hardware.h inclusion ASoC: imx-ssi: remove unneeded mach/hardware.h inclusion usb: ehci-mxc: remove unneeded mach/hardware.h inclusion video: mx3fb: remove unneeded mach/hardware.h inclusion watchdog: imx2_wdt: remove unneeded mach/hardware.h inclusion i2c: imx: remove cpu_is_xxx by using platform_device_id mtd: mxc_nand: remove cpu_is_xxx by using platform_device_id rtc: mxc_rtc: remove cpu_is_xxx by using platform_device_id dma: imx-dma: use devm_kzalloc and devm_request_irq dma: imx-dma: retrieve MEM and IRQ from resources dma: imx-dma: remove cpu_is_xxx by using platform_device_id media: mx2_camera: remove dead code in mx2_camera_add_device media: mx2_camera: use managed functions to clean up code media: mx2_camera: remove cpu_is_xxx by using platform_device_id mmc: mxcmmc: remove cpu_is_xxx by using platform_device_id video: imxfb: remove cpu_is_xxx by using platform_device_id ARM: imx: move debug macros to include/debug ARM: imx: include hardware.h rather than mach/hardware.h ARM: imx: remove header file mach/irqs.h ARM: imx: call mxc_device_init() in soc specific function ARM: imx: enable multi-platform build .../devicetree/bindings/i2c/fsl-imx-i2c.txt | 4 +- MAINTAINERS | 1 - arch/arm/Kconfig | 15 +- arch/arm/Kconfig.debug | 8 + arch/arm/Makefile | 1 - arch/arm/boot/dts/imx27.dtsi | 4 +- arch/arm/boot/dts/imx51.dtsi | 4 +- arch/arm/boot/dts/imx53.dtsi | 6 +- arch/arm/boot/dts/imx6q.dtsi | 6 +- arch/arm/configs/imx_v4_v5_defconfig | 5 +- arch/arm/configs/imx_v6_v7_defconfig | 3 +- .../mach/debug-macro.S => include/debug/imx.S} | 33 +- arch/arm/{plat-mxc => mach-imx}/3ds_debugboard.c | 2 +- .../include/mach => mach-imx}/3ds_debugboard.h | 0 arch/arm/mach-imx/Kconfig | 89 +- arch/arm/mach-imx/Makefile | 23 +- arch/arm/{plat-mxc => mach-imx}/avic.c | 5 +- .../include/mach => mach-imx}/board-mx31lilly.h | 0 .../include/mach => mach-imx}/board-mx31lite.h | 0 .../include/mach => mach-imx}/board-mx31moboard.h | 0 .../include/mach => mach-imx}/board-pcm038.h | 0 arch/arm/mach-imx/clk-imx1.c | 18 +- arch/arm/mach-imx/clk-imx21.c | 18 +- arch/arm/mach-imx/clk-imx25.c | 26 +- arch/arm/mach-imx/clk-imx27.c | 40 +- arch/arm/mach-imx/clk-imx31.c | 21 +- arch/arm/mach-imx/clk-imx35.c | 13 +- arch/arm/mach-imx/clk-imx51-imx53.c | 15 +- arch/arm/mach-imx/clk-imx6q.c | 3 +- arch/arm/mach-imx/clk-pllv1.c | 4 +- .../{plat-mxc/include/mach => mach-imx}/common.h | 1 + arch/arm/mach-imx/cpu-imx25.c | 5 +- arch/arm/mach-imx/cpu-imx27.c | 2 +- arch/arm/mach-imx/cpu-imx31.c | 7 +- arch/arm/mach-imx/cpu-imx35.c | 5 +- arch/arm/mach-imx/cpu-imx5.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpu.c | 3 +- arch/arm/mach-imx/cpu_op-mx51.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpufreq.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpuidle.c | 0 .../{plat-mxc/include/mach => mach-imx}/cpuidle.h | 0 arch/arm/mach-imx/devices-imx1.h | 3 +- arch/arm/mach-imx/devices-imx21.h | 3 +- arch/arm/mach-imx/devices-imx25.h | 3 +- arch/arm/mach-imx/devices-imx27.h | 3 +- arch/arm/mach-imx/devices-imx31.h | 3 +- arch/arm/mach-imx/devices-imx35.h | 3 +- arch/arm/mach-imx/devices-imx50.h | 3 +- arch/arm/mach-imx/devices-imx51.h | 3 +- arch/arm/{plat-mxc => mach-imx}/devices/Kconfig | 3 - arch/arm/{plat-mxc => mach-imx}/devices/Makefile | 3 +- .../mach => mach-imx/devices}/devices-common.h | 19 +- arch/arm/{plat-mxc => mach-imx/devices}/devices.c | 4 +- .../devices/platform-ahci-imx.c | 5 +- .../{plat-mxc => mach-imx}/devices/platform-fec.c | 5 +- .../devices/platform-flexcan.c | 4 +- .../devices/platform-fsl-usb2-udc.c | 5 +- .../devices/platform-gpio-mxc.c | 2 +- .../devices/platform-gpio_keys.c | 5 +- .../devices/platform-imx-dma.c | 23 +- .../devices/platform-imx-fb.c | 16 +- .../devices/platform-imx-i2c.c | 32 +- .../devices/platform-imx-keypad.c | 4 +- .../devices/platform-imx-ssi.c | 4 +- .../devices/platform-imx-uart.c | 4 +- .../devices/platform-imx2-wdt.c | 5 +- .../devices/platform-imx21-hcd.c | 4 +- .../devices/platform-imx_udc.c | 4 +- .../devices/platform-imxdi_rtc.c | 5 +- .../devices/platform-ipu-core.c | 5 +- .../devices/platform-mx2-camera.c | 16 +- .../devices/platform-mxc-ehci.c | 5 +- .../devices/platform-mxc-mmc.c | 20 +- .../devices/platform-mxc_nand.c | 25 +- .../devices/platform-mxc_pwm.c | 4 +- .../devices/platform-mxc_rnga.c | 4 +- .../devices/platform-mxc_rtc.c | 13 +- .../devices/platform-mxc_w1.c | 4 +- .../devices/platform-pata_imx.c | 4 +- .../devices/platform-sdhci-esdhc-imx.c | 5 +- .../devices/platform-spi_imx.c | 4 +- arch/arm/mach-imx/ehci-imx25.c | 4 +- arch/arm/mach-imx/ehci-imx27.c | 4 +- arch/arm/mach-imx/ehci-imx31.c | 4 +- arch/arm/mach-imx/ehci-imx35.c | 4 +- arch/arm/mach-imx/ehci-imx5.c | 4 +- arch/arm/{plat-mxc => mach-imx}/epit.c | 6 +- .../include/mach => mach-imx}/eukrea-baseboards.h | 0 arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 7 +- arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 8 +- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 7 +- arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 7 +- .../{plat-mxc/include/mach => mach-imx}/hardware.h | 26 +- arch/arm/mach-imx/hotplug.c | 3 +- arch/arm/{plat-mxc/include/mach => mach-imx}/iim.h | 0 arch/arm/mach-imx/imx27-dt.c | 11 +- arch/arm/mach-imx/imx31-dt.c | 5 +- arch/arm/mach-imx/imx51-dt.c | 9 +- arch/arm/mach-imx/include/mach/dma-mx1-mx2.h | 10 - arch/arm/mach-imx/iomux-imx31.c | 5 +- .../include/mach => mach-imx}/iomux-mx1.h | 2 +- .../include/mach => mach-imx}/iomux-mx21.h | 4 +- .../include/mach => mach-imx}/iomux-mx25.h | 2 +- .../include/mach => mach-imx}/iomux-mx27.h | 4 +- .../include/mach => mach-imx}/iomux-mx2x.h | 0 .../include/mach => mach-imx}/iomux-mx3.h | 0 .../include/mach => mach-imx}/iomux-mx35.h | 2 +- .../include/mach => mach-imx}/iomux-mx50.h | 2 +- .../include/mach => mach-imx}/iomux-mx51.h | 2 +- arch/arm/{plat-mxc => mach-imx}/iomux-v1.c | 5 +- .../{plat-mxc/include/mach => mach-imx}/iomux-v1.h | 0 arch/arm/{plat-mxc => mach-imx}/iomux-v3.c | 5 +- .../{plat-mxc/include/mach => mach-imx}/iomux-v3.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/iram.h | 0 arch/arm/{plat-mxc => mach-imx}/iram_alloc.c | 3 +- arch/arm/{plat-mxc => mach-imx}/irq-common.c | 0 arch/arm/{plat-mxc => mach-imx}/irq-common.h | 3 + arch/arm/mach-imx/lluart.c | 3 +- arch/arm/mach-imx/mach-apf9328.c | 7 +- arch/arm/mach-imx/mach-armadillo5x0.c | 9 +- arch/arm/mach-imx/mach-bug.c | 7 +- arch/arm/mach-imx/mach-cpuimx27.c | 11 +- arch/arm/mach-imx/mach-cpuimx35.c | 9 +- arch/arm/mach-imx/mach-cpuimx51sd.c | 9 +- arch/arm/mach-imx/mach-eukrea_cpuimx25.c | 10 +- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 6 +- arch/arm/mach-imx/mach-imx27ipcam.c | 6 +- arch/arm/mach-imx/mach-imx27lite.c | 6 +- arch/arm/mach-imx/mach-imx53.c | 11 +- arch/arm/mach-imx/mach-imx6q.c | 6 +- arch/arm/mach-imx/mach-kzm_arm11_01.c | 7 +- arch/arm/mach-imx/mach-mx1ads.c | 7 +- arch/arm/mach-imx/mach-mx21ads.c | 6 +- arch/arm/mach-imx/mach-mx25_3ds.c | 8 +- arch/arm/mach-imx/mach-mx27_3ds.c | 10 +- arch/arm/mach-imx/mach-mx27ads.c | 6 +- arch/arm/mach-imx/mach-mx31_3ds.c | 12 +- arch/arm/mach-imx/mach-mx31ads.c | 5 +- arch/arm/mach-imx/mach-mx31lilly.c | 11 +- arch/arm/mach-imx/mach-mx31lite.c | 11 +- arch/arm/mach-imx/mach-mx31moboard.c | 14 +- arch/arm/mach-imx/mach-mx35_3ds.c | 8 +- arch/arm/mach-imx/mach-mx50_rdp.c | 7 +- arch/arm/mach-imx/mach-mx51_3ds.c | 9 +- arch/arm/mach-imx/mach-mx51_babbage.c | 7 +- arch/arm/mach-imx/mach-mxt_td60.c | 6 +- arch/arm/mach-imx/mach-pca100.c | 8 +- arch/arm/mach-imx/mach-pcm037.c | 8 +- arch/arm/mach-imx/mach-pcm037_eet.c | 5 +- arch/arm/mach-imx/mach-pcm038.c | 13 +- arch/arm/mach-imx/mach-pcm043.c | 9 +- arch/arm/mach-imx/mach-qong.c | 6 +- arch/arm/mach-imx/mach-scb9328.c | 7 +- arch/arm/mach-imx/mach-vpr200.c | 7 +- arch/arm/mach-imx/mm-imx1.c | 9 +- arch/arm/mach-imx/mm-imx21.c | 14 +- arch/arm/mach-imx/mm-imx25.c | 12 +- arch/arm/mach-imx/mm-imx27.c | 14 +- arch/arm/mach-imx/mm-imx3.c | 13 +- arch/arm/mach-imx/mm-imx5.c | 12 +- arch/arm/mach-imx/mx1-camera-fiq-ksym.c | 18 - arch/arm/mach-imx/mx1-camera-fiq.S | 35 - arch/arm/{plat-mxc/include/mach => mach-imx}/mx1.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx21.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx25.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx27.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx2x.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx31.h | 0 arch/arm/mach-imx/mx31lilly-db.c | 9 +- arch/arm/mach-imx/mx31lite-db.c | 9 +- arch/arm/mach-imx/mx31moboard-devboard.c | 9 +- arch/arm/mach-imx/mx31moboard-marxbot.c | 9 +- arch/arm/mach-imx/mx31moboard-smartbot.c | 11 +- .../arm/{plat-mxc/include/mach => mach-imx}/mx35.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx3x.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx50.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx51.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx53.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx6q.h | 0 arch/arm/{plat-mxc/include/mach => mach-imx}/mxc.h | 0 arch/arm/mach-imx/pcm970-baseboard.c | 7 +- arch/arm/mach-imx/platsmp.c | 5 +- arch/arm/mach-imx/pm-imx27.c | 3 +- arch/arm/mach-imx/pm-imx3.c | 7 +- arch/arm/mach-imx/pm-imx5.c | 7 +- arch/arm/mach-imx/pm-imx6q.c | 5 +- arch/arm/{plat-mxc => mach-imx}/ssi-fiq-ksym.c | 0 arch/arm/{plat-mxc => mach-imx}/ssi-fiq.S | 0 arch/arm/{plat-mxc => mach-imx}/system.c | 5 +- arch/arm/{plat-mxc => mach-imx}/time.c | 5 +- arch/arm/{plat-mxc => mach-imx}/tzic.c | 6 +- arch/arm/{plat-mxc => mach-imx}/ulpi.c | 2 +- .../arm/{plat-mxc/include/mach => mach-imx}/ulpi.h | 0 arch/arm/plat-mxc/Kconfig | 89 -- arch/arm/plat-mxc/Makefile | 24 - arch/arm/plat-mxc/devices/platform-mx1-camera.c | 42 - arch/arm/plat-mxc/include/mach/irqs.h | 21 - arch/arm/plat-mxc/include/mach/timex.h | 22 - arch/arm/plat-mxc/include/mach/uncompress.h | 132 --- drivers/dma/imx-dma.c | 137 +-- drivers/dma/imx-sdma.c | 1 - drivers/dma/ipu/ipu_idmac.c | 3 +- drivers/dma/ipu/ipu_irq.c | 3 +- drivers/i2c/busses/i2c-imx.c | 40 +- drivers/media/video/Kconfig | 12 - drivers/media/video/Makefile | 1 - drivers/media/video/mx1_camera.c | 889 -------------------- drivers/media/video/mx2_camera.c | 246 +++--- drivers/media/video/mx3_camera.c | 2 +- drivers/mmc/host/mxcmmc.c | 31 +- drivers/mtd/nand/mxc_nand.c | 86 +- drivers/rtc/rtc-mxc.c | 34 +- drivers/usb/host/ehci-mxc.c | 1 - drivers/video/imxfb.c | 38 +- drivers/video/mx3fb.c | 3 +- drivers/watchdog/imx2_wdt.c | 1 - .../mach/ipu.h => include/linux/dma/ipu-dma.h | 6 +- include/linux/platform_data/asoc-imx-ssi.h | 2 + include/linux/platform_data/camera-mx1.h | 35 - include/linux/platform_data/dma-imx.h | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 1 - sound/soc/fsl/imx-ssi.c | 1 - 222 files changed, 1115 insertions(+), 2161 deletions(-) rename arch/arm/{plat-mxc/include/mach/debug-macro.S => include/debug/imx.S} (59%) rename arch/arm/{plat-mxc => mach-imx}/3ds_debugboard.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/3ds_debugboard.h (100%) rename arch/arm/{plat-mxc => mach-imx}/avic.c (98%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31lilly.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31lite.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31moboard.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-pcm038.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/common.h (99%) rename arch/arm/{plat-mxc => mach-imx}/cpu.c (97%) rename arch/arm/{plat-mxc => mach-imx}/cpufreq.c (99%) rename arch/arm/{plat-mxc => mach-imx}/cpuidle.c (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/cpuidle.h (100%) rename arch/arm/{plat-mxc => mach-imx}/devices/Kconfig (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/Makefile (96%) rename arch/arm/{plat-mxc/include/mach => mach-imx/devices}/devices-common.h (96%) rename arch/arm/{plat-mxc => mach-imx/devices}/devices.c (92%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-ahci-imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-fec.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-flexcan.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-fsl-usb2-udc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-gpio-mxc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-gpio_keys.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-dma.c (63%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-fb.c (79%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-i2c.c (76%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-keypad.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-ssi.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-uart.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx2-wdt.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx21-hcd.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx_udc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imxdi_rtc.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-ipu-core.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mx2-camera.c (83%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc-ehci.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc-mmc.c (76%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_nand.c (74%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_pwm.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_rnga.c (95%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_rtc.c (77%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_w1.c (95%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-pata_imx.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-sdhci-esdhc-imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-spi_imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/epit.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/eukrea-baseboards.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/hardware.h (94%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iim.h (100%) delete mode 100644 arch/arm/mach-imx/include/mach/dma-mx1-mx2.h rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx1.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx21.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx25.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx27.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx2x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx3.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx35.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx50.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx51.h (99%) rename arch/arm/{plat-mxc => mach-imx}/iomux-v1.c (98%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-v1.h (100%) rename arch/arm/{plat-mxc => mach-imx}/iomux-v3.c (97%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-v3.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iram.h (100%) rename arch/arm/{plat-mxc => mach-imx}/iram_alloc.c (98%) rename arch/arm/{plat-mxc => mach-imx}/irq-common.c (100%) rename arch/arm/{plat-mxc => mach-imx}/irq-common.h (94%) delete mode 100644 arch/arm/mach-imx/mx1-camera-fiq-ksym.c delete mode 100644 arch/arm/mach-imx/mx1-camera-fiq.S rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx1.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx21.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx25.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx27.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx2x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx31.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx35.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx3x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx50.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx51.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx53.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx6q.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mxc.h (100%) rename arch/arm/{plat-mxc => mach-imx}/ssi-fiq-ksym.c (100%) rename arch/arm/{plat-mxc => mach-imx}/ssi-fiq.S (100%) rename arch/arm/{plat-mxc => mach-imx}/system.c (97%) rename arch/arm/{plat-mxc => mach-imx}/time.c (99%) rename arch/arm/{plat-mxc => mach-imx}/tzic.c (98%) rename arch/arm/{plat-mxc => mach-imx}/ulpi.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/ulpi.h (100%) delete mode 100644 arch/arm/plat-mxc/Kconfig delete mode 100644 arch/arm/plat-mxc/Makefile delete mode 100644 arch/arm/plat-mxc/devices/platform-mx1-camera.c delete mode 100644 arch/arm/plat-mxc/include/mach/irqs.h delete mode 100644 arch/arm/plat-mxc/include/mach/timex.h delete mode 100644 arch/arm/plat-mxc/include/mach/uncompress.h delete mode 100644 drivers/media/video/mx1_camera.c rename arch/arm/plat-mxc/include/mach/ipu.h => include/linux/dma/ipu-dma.h (97%) delete mode 100644 include/linux/platform_data/camera-mx1.h -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: eukrea-tlv320: Convert it to platform driver
by Fabio Estevam 22 Sep '12

22 Sep '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Convert eukrea-tlv320 to platform driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 1 + arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 1 + sound/soc/fsl/eukrea-tlv320.c | 37 ++++++++++++------------ 5 files changed, 23 insertions(+), 18 deletions(-) diff --git a/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c b/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c index fd3177f..98aef57 100644 --- a/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimx27-baseboard.c @@ -348,4 +348,5 @@ void __init eukrea_mbimx27_baseboard_init(void) imx27_add_imx_keypad(&eukrea_mbimx27_keymap_data); gpio_led_register_device(-1, &eukrea_mbimx27_gpio_led_info); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c index dfd2da8..0b84666 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c @@ -306,4 +306,5 @@ void __init eukrea_mbimxsd25_baseboard_init(void) platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); gpio_led_register_device(-1, &eukrea_mbimxsd_led_info); imx_add_gpio_keys(&eukrea_mbimxsd_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c index 6e9dd12..c6532a0 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c @@ -315,4 +315,5 @@ void __init eukrea_mbimxsd35_baseboard_init(void) platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); gpio_led_register_device(-1, &eukrea_mbimxsd_led_info); imx_add_gpio_keys(&eukrea_mbimxsd_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c b/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c index 96a24b7..8b0de30 100644 --- a/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c +++ b/arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c @@ -228,4 +228,5 @@ void __init eukrea_mbimxsd51_baseboard_init(void) gpio_led_register_device(-1, &eukrea_mbimxsd51_led_info); imx_add_gpio_keys(&eukrea_mbimxsd51_button_data); + imx_add_platform_device("eukrea_tlv320", 0, NULL, 0, NULL, 0); } diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index efb9ede..267d5b4 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -93,9 +93,7 @@ static struct snd_soc_card eukrea_tlv320 = { .num_links = 1, }; -static struct platform_device *eukrea_tlv320_snd_device; - -static int __init eukrea_tlv320_init(void) +static int __devinit eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; @@ -136,29 +134,32 @@ static int __init eukrea_tlv320_init(void) return 0; } - eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); - if (!eukrea_tlv320_snd_device) - return -ENOMEM; - - platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320); - ret = platform_device_add(eukrea_tlv320_snd_device); - - if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(eukrea_tlv320_snd_device); - } + eukrea_tlv320.dev = &pdev->dev; + ret = snd_soc_register_card(&eukrea_tlv320); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); return ret; } -static void __exit eukrea_tlv320_exit(void) +static int __devexit eukrea_tlv320_remove(struct platform_device *pdev) { - platform_device_unregister(eukrea_tlv320_snd_device); + snd_soc_unregister_card(&eukrea_tlv320); + + return 0; } -module_init(eukrea_tlv320_init); -module_exit(eukrea_tlv320_exit); +static struct platform_driver eukrea_tlv320_driver = { + .driver = { + .name = "eukrea_tlv320", + .owner = THIS_MODULE, + }, + .probe = eukrea_tlv320_probe, + .remove = __devexit_p(eukrea_tlv320_remove),}; + +module_platform_driver(eukrea_tlv320_driver); MODULE_AUTHOR("Eric Bénard <eric(a)eukrea.com>"); MODULE_DESCRIPTION("CPUIMX ALSA SoC driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:eukrea_tlv320"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 22 Sep '12

22 Sep '12

Hello, Generated on top of: git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap Changes since v3: - Added Acked-by from Tony and Dmitry - Removed #ifdef around of_find_node_by_name() in the drivers Changed since v2: - Added commit message to patch 2 (as Tero requested it) - Fixed patch 6 (dt: Add empty...) since the previous patch caused x86_64 build to fail due to missing struct before device_node. Changes since v1: - Get the MCLK frequencey from twl-core driver (via new API) - hs_extmute_disable_level parameter has been removed - empty of_find_node_by_name() in of.h for !CONFIG_OF builds Mark: the extmute GPIO handling (when it is used) remained in the codec driver for now. I can think more on how to add support for this type of mute control. If you can point me to other codec needing this it would help on designing something which is common enough for other users. As for now I have a separate series to add GPIO controlled output amps (hp/spk). I will send that to review soon. Currently this piece of code does not work when booted with device tree since I have done it on n900 which has the GPIO numbers as defines. I need to revisit the aproach on how to do it in a dynamic way. Introl mail from v1: The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform). If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (14): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl-core: Add API to query the HFCLK rate MFD: twl4030-audio: Get audio MCLK via twl-core API instead of pdata dt: Add empty of_find_node_by_name() function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 46 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 18 +++- drivers/mfd/twl-core.c | 32 +++++++ drivers/mfd/twl4030-audio.c | 105 ++++++++++++++------- include/linux/i2c/twl.h | 3 +- include/linux/of.h | 6 ++ sound/soc/codecs/twl4030.c | 101 ++++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 10 files changed, 255 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.12

2 15

[alsa-devel] [PATCH v2 1/3] ASoC: fsl: add PPC_MPC52xx dependency to SND_POWERPC_SOC
by Eric Millbrandt 22 Sep '12

22 Sep '12

mpc52xx socs do not define FSL_SOC but need SND_POWERPC_SOC defined to build ASoC drivers. Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- Changes since v1: Patch was "powerpc/52xx: define FSL_SOC" Changed from defining FSL_SOC for PPC_MPC52xx to adding PPC_MPC52xx as a dependency of SND_POWERPC_SOC as per Anatolij Gustschin's comment. sound/soc/fsl/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d701330..4563b28 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS menuconfig SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" - depends on FSL_SOC + depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. -- 1.7.2.5

2 5

[alsa-devel] [PATCH] ASoC: twl6040: Convert to use DAI DAPM widgets
by Peter Ujfalusi 22 Sep '12

22 Sep '12

Use DAPM mapping for stream events and give unique names for the streams. This change also fixes the following warning: twl6040-codec twl6040-codec: Failed to create Capture debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 43 +++++++++++++++++++++++++++++-------------- 1 file changed, 29 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c084c54..e8f97af 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -727,10 +727,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_MICRCTL, 1, 0, NULL, 0), /* ADCs */ - SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", - TWL6040_REG_MICLCTL, 2, 0), - SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", - TWL6040_REG_MICRCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Left", NULL, TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, TWL6040_REG_MICRCTL, 2, 0), /* Microphone bias */ SND_SOC_DAPM_SUPPLY("Headset Mic Bias", @@ -743,15 +741,12 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0, NULL, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", - TWL6040_REG_HFLCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", - TWL6040_REG_HFRCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", NULL, TWL6040_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", NULL, TWL6040_REG_HFRCTL, 0, 0), /* Virtual DAC for vibra path (DL4 channel) */ - SND_SOC_DAPM_DAC("VIBRA DAC", "Vibra Playback", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("VIBRA DAC", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("Handsfree Left Playback", SND_SOC_NOPM, 0, 0, &hfl_mux_controls), @@ -810,6 +805,26 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + /* Stream -> DAC mapping */ + {"HSDAC Left", NULL, "Legacy Playback"}, + {"HSDAC Left", NULL, "Headset Playback"}, + {"HSDAC Right", NULL, "Legacy Playback"}, + {"HSDAC Right", NULL, "Headset Playback"}, + + {"HFDAC Left", NULL, "Legacy Playback"}, + {"HFDAC Left", NULL, "Handsfree Playback"}, + {"HFDAC Right", NULL, "Legacy Playback"}, + {"HFDAC Right", NULL, "Handsfree Playback"}, + + {"VIBRA DAC", NULL, "Legacy Playback"}, + {"VIBRA DAC", NULL, "Vibra Playback"}, + + /* ADC -> Stream mapping */ + {"ADC Left", NULL, "Legacy Capture"}, + {"ADC Left", NULL, "Capture"}, + {"ADC Right", NULL, "Legacy Capture"}, + {"ADC Right", NULL, "Capture"}, + /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, @@ -1028,14 +1043,14 @@ static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-legacy", .playback = { - .stream_name = "Playback", + .stream_name = "Legacy Playback", .channels_min = 1, .channels_max = 5, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .capture = { - .stream_name = "Capture", + .stream_name = "Legacy Capture", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, -- 1.7.12

2 1

[alsa-devel] [PATCH] ASoC: twl4030: Convert to use DAI DAPM widgets
by Peter Ujfalusi 22 Sep '12

22 Sep '12

Use DAPM mapping for stream events and give unique names for the streams. This change also fixes the following warning: twl4030-codec twl4030-codec: Failed to create Capture debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 48 +++++++++++++++++++++++++++------------------- 1 file changed, 28 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2548f5c..962341d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1237,16 +1237,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"), /* DACs */ - SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Left1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Left2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Voice", NULL, SND_SOC_NOPM, 0, 0), /* Analog bypasses */ SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, @@ -1375,14 +1370,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Introducing four virtual ADC, since TWL4030 have four channel for capture */ - SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right2", NULL, SND_SOC_NOPM, 0, 0), /* Analog/Digital mic path selection. TX1 Left/Right: either analog Left/Right or Digimic0 @@ -1426,6 +1417,23 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + /* Stream -> DAC mapping */ + {"DAC Right1", NULL, "HiFi Playback"}, + {"DAC Left1", NULL, "HiFi Playback"}, + {"DAC Right2", NULL, "HiFi Playback"}, + {"DAC Left2", NULL, "HiFi Playback"}, + {"DAC Voice", NULL, "Voice Playback"}, + + /* ADC -> Stream mapping */ + {"HiFi Capture", NULL, "ADC Virtual Left1"}, + {"HiFi Capture", NULL, "ADC Virtual Right1"}, + {"HiFi Capture", NULL, "ADC Virtual Left2"}, + {"HiFi Capture", NULL, "ADC Virtual Right2"}, + {"Voice Capture", NULL, "ADC Virtual Left1"}, + {"Voice Capture", NULL, "ADC Virtual Right1"}, + {"Voice Capture", NULL, "ADC Virtual Left2"}, + {"Voice Capture", NULL, "ADC Virtual Right2"}, + {"Digital L1 Playback Mixer", NULL, "DAC Left1"}, {"Digital R1 Playback Mixer", NULL, "DAC Right1"}, {"Digital L2 Playback Mixer", NULL, "DAC Left2"}, @@ -2170,7 +2178,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .formats = TWL4030_FORMATS, .sig_bits = 24,}, .capture = { - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, @@ -2187,7 +2195,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, -- 1.7.12

2 1

Re: [alsa-devel] snd_usb changes broke audio
by Daniel Mack 22 Sep '12

22 Sep '12

(Cc: ALSA list, please don't drop it on replies!) On 22.09.2012 10:21, Nate Garner wrote: > output of lsusb -v (assuming you just want the sound card related part. > if you need more let me know): Ok, thanks - more stuff below. > Bus 009 Device 003: ID 041e:3040 Creative Technology, Ltd SoundBlaster > Live! 24-bit External SB0490 > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 1.10 > bDeviceClass 0 (Defined at Interface level) > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 8 > idVendor 0x041e Creative Technology, Ltd > idProduct 0x3040 SoundBlaster Live! 24-bit External SB0490 > bcdDevice 1.00 > iManufacturer 1 Creative Technology > iProduct 2 SB Live! 24-bit External? > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 979 > bNumInterfaces 3 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0x80 > (Bus Powered) > MaxPower 308mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 1 Control Device > bInterfaceProtocol 0 > iInterface 0 > AudioControl Interface Descriptor: > bLength 10 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 76 > bInCollection 2 > baInterfaceNr( 0) 1 > baInterfaceNr( 1) 2 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 1 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bNrChannels 6 > wChannelConfig 0x003f > Left Front (L) > Right Front (R) > Center Front (C) > Low Freqency Enhancement (LFE) > Left Surround (LS) > Right Surround (RS) > iChannelNames 0 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 14 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 2 > bSourceID 1 > bControlSize 1 > bmaControls( 0) 0x03 > Mute Control > Volume Control > bmaControls( 1) 0x00 > bmaControls( 2) 0x00 > bmaControls( 3) 0x00 > bmaControls( 4) 0x00 > bmaControls( 5) 0x00 > bmaControls( 6) 0x00 > iFeature 0 > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 3 > wTerminalType 0x0301 Speaker > bAssocTerminal 0 > bSourceID 2 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 12 > bDescriptorType 36 > bDescriptorSubtype 2 (INPUT_TERMINAL) > bTerminalID 4 > wTerminalType 0x0603 Line Connector > bAssocTerminal 0 > bNrChannels 2 > wChannelConfig 0x0003 > Left Front (L) > Right Front (R) > iChannelNames 0 > iTerminal 0 > AudioControl Interface Descriptor: > bLength 10 > bDescriptorType 36 > bDescriptorSubtype 6 (FEATURE_UNIT) > bUnitID 5 > bSourceID 4 > bControlSize 1 > bmaControls( 0) 0x02 > Volume Control > bmaControls( 1) 0x00 > bmaControls( 2) 0x00 > iFeature 0 > AudioControl Interface Descriptor: > bLength 9 > bDescriptorType 36 > bDescriptorSubtype 3 (OUTPUT_TERMINAL) > bTerminalID 6 > wTerminalType 0x0101 USB Streaming > bAssocTerminal 0 > bSourceID 5 > iTerminal 0 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x83 EP 3 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0008 1x 8 bytes > bInterval 10 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x00b4 1x 180 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x010e 1x 270 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 3 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0168 1x 360 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 4 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x021c 1x 540 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 5 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x021c 1x 540 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 6 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x032a 1x 810 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 7 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x00c0 1x 192 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 8 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0120 1x 288 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 9 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0180 1x 384 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 10 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 4 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0240 1x 576 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 11 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0240 1x 576 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 12 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 6 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0360 1x 864 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 13 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0180 1x 384 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 14 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 1 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 9 > Transfer Type Isochronous > Synch Type Adaptive > Usage Type Data > wMaxPacketSize 0x0240 1x 576 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x00b4 1x 180 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 44100 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x010e 1x 270 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 3 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x00c4 1x 196 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 4 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x0126 1x 294 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 5 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x0184 1x 388 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 6 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 6 > bDelay 1 frames > wFormatTag 1 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 2 > bSubframeSize 3 > bBitResolution 24 > bSamFreqType 1 Discrete > tSamFreq[ 0] 96000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 13 > Transfer Type Isochronous > Synch Type Synchronous > Usage Type Data > wMaxPacketSize 0x0246 1x 582 bytes > bInterval 1 > bRefresh 0 > bSynchAddress 0 > AudioControl Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0 Undefined > Device Status: 0x0000 > (Bus Powered) > > Nothing in dmesg looks relevent. Errors.log contains: > localhost pulseaudio[900]: [pulseaudio] bluetooth-util.c: > org.bluez.Manager.ListAdapters() failed: > org.freedesktop.DBus.Error.Spawn.ChildExited: Launch helper exited with > unknown return code 1 > > nothing else obvious I can see. Are you sure you there's no "not enough bandwidth" message anywhere? And is your device connected to the host directly or via a hub? Thanks, Daniel > > On Wed, Sep 19, 2012 at 11:41 PM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > > (Cc the ALSA list - please don't drop it in replies) > > On 20.09.2012 02:16, Nate Garner wrote: > > I recently updated my kernel from 3.5.3 to 3.5.4. Upon rebooting, I no > > longer got any audio through my external sound card, but my headphones > > worked. I noticed in the kernel change log that you made some > changes to > > snd-usb, and thought that might be the culprit. I hope you're the > right > > person to report this too, if not, feel free to ignore. > > Relevant info: > > Distro: arch linux, x86_64 > > sound card: creative labs model SB0490 (soundblaser Live! 24-bit > > external (link here > > > <http://support.creative.com/Products/ProductDetails.aspx?catID=1&subCatID=2…>) > > alsa version 1.0.26 > > pulseaudio version 2.1 > > Argh! I believed we finally fixed all the regressions. > > Are there any messages in the logs or in dmesg that look releated? Can > you send the output of "lsusb -v", please? > > > Daniel > >

1 0

[alsa-devel] [PATCH] test: add audio_time
by Pierre-Louis Bossart 22 Sep '12

22 Sep '12

Simple test to create playback and capture streams, and check elapsed time vs. sample counts reported by driver. This should be helpful for driver developers and anyone interested in system/audio time drift. tested only on HDAudio TODO: - make period configurable - better output messages - support for wall clock when it's in the mainline Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- test/audio_time.c | 237 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 237 insertions(+) create mode 100644 test/audio_time.c diff --git a/test/audio_time.c b/test/audio_time.c new file mode 100644 index 0000000..a910783 --- /dev/null +++ b/test/audio_time.c @@ -0,0 +1,237 @@ +/* + * This program only tracks the difference between system time + * and audio time, as reported in snd_pcm_status(). It should be + * helpful to verify the information reported by drivers. + */ + +#include "../include/asoundlib.h" +#include <math.h> + +static char *device = "hw:0,0"; + +snd_output_t *output = NULL; + +long long timestamp2ns(snd_htimestamp_t t) +{ + long long nsec; + + nsec = t.tv_sec * 1000000000; + nsec += t.tv_nsec; + + return nsec; +} + +long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) +{ + long long nsec1, nsec2; + + nsec1 = timestamp2ns(t1); + nsec2 = timestamp2ns(t2); + + return nsec1 - nsec2; +} + +void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, + snd_htimestamp_t *trigger_timestamp, + snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) +{ + int err; + snd_pcm_status_t *status; + + snd_pcm_status_alloca(&status); + if ((err = snd_pcm_status(handle, status)) < 0) { + printf("Stream status error: %s\n", snd_strerror(err)); + exit(0); + } + snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); + snd_pcm_status_get_htstamp(status, timestamp); + *avail = snd_pcm_status_get_avail(status); + *delay = snd_pcm_status_get_delay(status); +} + +#define PERIOD 6000 +#define PCM_LINK /* sync start for playback and capture */ +#define TRACK_CAPTURE /* dump capture timing info */ +#define TRACK_PLAYBACK /* dump playback timing info */ +#define PLAYBACK_BUFFERS 4 + + +int main(void) +{ + int err; + unsigned int i; + snd_pcm_t *handle_p = NULL; + snd_pcm_t *handle_c = NULL; + snd_pcm_sframes_t frames; + snd_htimestamp_t tstamp_c, tstamp_p; + snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + unsigned char buffer_p[PERIOD*4*4]; + unsigned char buffer_c[PERIOD*4*4]; + + snd_pcm_sw_params_t *swparams_p; + snd_pcm_sw_params_t *swparams_c; + + snd_pcm_uframes_t curr_count_c; + snd_pcm_uframes_t frame_count_c = 0; + snd_pcm_uframes_t curr_count_p; + snd_pcm_uframes_t frame_count_p = 0; + + snd_pcm_sframes_t delay_p, delay_c; + snd_pcm_uframes_t avail_p, avail_c; + + if ((err = snd_pcm_open(&handle_p, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { + printf("Playback open error: %s\n", snd_strerror(err)); + goto _exit; + } + if ((err = snd_pcm_set_params(handle_p, + SND_PCM_FORMAT_S16, + SND_PCM_ACCESS_RW_INTERLEAVED, + 2, + 48000, + 0, + 500000)) < 0) { /* 0.5sec */ + printf("Playback open error: %s\n", snd_strerror(err)); + goto _exit; + } + + snd_pcm_sw_params_alloca(&swparams_p); + /* get the current swparams */ + err = snd_pcm_sw_params_current(handle_p, swparams_p); + if (err < 0) { + printf("Unable to determine current swparams_p: %s\n", snd_strerror(err)); + goto _exit; + } + + /* enable tstamp */ + err = snd_pcm_sw_params_set_tstamp_mode(handle_p, swparams_p, SND_PCM_TSTAMP_ENABLE); + if (err < 0) { + printf("Unable to set tstamp mode : %s\n", snd_strerror(err)); + goto _exit; + } + + /* write the sw parameters */ + err = snd_pcm_sw_params(handle_p, swparams_p); + if (err < 0) { + printf("Unable to set swparams_p : %s\n", snd_strerror(err)); + goto _exit; + } + + if ((err = snd_pcm_open(&handle_c, device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK)) < 0) { + printf("Capture open error: %s\n", snd_strerror(err)); + goto _exit; + } + if ((err = snd_pcm_set_params(handle_c, + SND_PCM_FORMAT_S16, + SND_PCM_ACCESS_RW_INTERLEAVED, + 2, + 48000, + 0, + 500000)) < 0) { /* 0.5sec */ + printf("Capture open error: %s\n", snd_strerror(err)); + goto _exit; + } + + snd_pcm_sw_params_alloca(&swparams_c); + /* get the current swparams */ + err = snd_pcm_sw_params_current(handle_c, swparams_c); + if (err < 0) { + printf("Unable to determine current swparams_c: %s\n", snd_strerror(err)); + goto _exit; + } + + /* enable tstamp */ + err = snd_pcm_sw_params_set_tstamp_mode(handle_c, swparams_c, SND_PCM_TSTAMP_ENABLE); + if (err < 0) { + printf("Unable to set tstamp mode : %s\n", snd_strerror(err)); + goto _exit; + } + + /* write the sw parameters */ + err = snd_pcm_sw_params(handle_c, swparams_c); + if (err < 0) { + printf("Unable to set swparams_c : %s\n", snd_strerror(err)); + goto _exit; + } + +#ifdef PCM_LINK + if ((err = snd_pcm_link(handle_c, handle_p)) < 0) { + printf("Streams link error: %s\n", snd_strerror(err)); + exit(0); + } +#endif + + i = PLAYBACK_BUFFERS; + while (i--) { + frames = snd_pcm_writei(handle_p, buffer_p, PERIOD); + if (frames < 0) { + printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); + goto _exit; + } + frame_count_p += frames; + } + + if (PLAYBACK_BUFFERS != 4) + snd_pcm_start(handle_p); + +#ifndef PCM_LINK + /* need to start capture explicitly */ + snd_pcm_start(handle_c); +#endif + + while (1) { + + frames = snd_pcm_wait(handle_c, -1); + if (frames < 0) { + printf("snd_pcm_wait failed: %s\n", snd_strerror(frames)); + goto _exit; + } + + frames = snd_pcm_readi(handle_c, buffer_c, PERIOD); + if (frames < 0) { + printf("snd_pcm_readi failed: %s\n", snd_strerror(frames)); + goto _exit; + } + frame_count_c += frames; + + frames = snd_pcm_writei(handle_p, buffer_p, PERIOD); + if (frames < 0) { + printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); + goto _exit; + } + + frame_count_p += frames; + +#if defined(TRACK_PLAYBACK) + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + + curr_count_p = frame_count_p - delay_p; /* written minus queued */ + + printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", + timediff(tstamp_p,trigger_tstamp_p), + (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), + timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + ); +#endif + +#if defined(TRACK_CAPTURE) + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + + curr_count_c = frame_count_c + delay_c; /* read plus queued */ + + printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", + timediff(tstamp_c,trigger_tstamp_c), + (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), + timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + ); +#endif + + } + +_exit: + if (handle_p) + snd_pcm_close(handle_p); + if (handle_c) + snd_pcm_close(handle_c); + + return 0; +} -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ALSA: hda - force use of SSYNC bits
by Pierre-Louis Bossart 22 Sep '12

22 Sep '12

SSYNC bits are typically used to start multiple streams synchronously. It makes sense to use them for a single stream for a more predictable startup sequence. The transfers only start once the DMA and FIFOs are ready. This results in a better correlation between timestamps and number of samples played. Credits to Kar Leong Wang for suggesting this improvement. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 36 ++++++++++++++++-------------------- 1 file changed, 16 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c6..7ba187e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1959,14 +1959,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_lock(&chip->reg_lock); - if (nsync > 1) { - /* first, set SYNC bits of corresponding streams */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) | sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); - } + + /* first, set SYNC bits of corresponding streams */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) | sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); + snd_pcm_group_for_each_entry(s, substream) { if (s->pcm->card != substream->pcm->card) continue; @@ -1984,8 +1984,6 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_unlock(&chip->reg_lock); if (start) { - if (nsync == 1) - return 0; /* wait until all FIFOs get ready */ for (timeout = 5000; timeout; timeout--) { nwait = 0; @@ -2018,16 +2016,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) cpu_relax(); } } - if (nsync > 1) { - spin_lock(&chip->reg_lock); - /* reset SYNC bits */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) & ~sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); - spin_unlock(&chip->reg_lock); - } + spin_lock(&chip->reg_lock); + /* reset SYNC bits */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) & ~sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); + spin_unlock(&chip->reg_lock); return 0; } -- 1.7.9.5

2 3

[alsa-devel] line-out mute on "phantom" HP jack (Via codec)
by Herton Ronaldo Krzesinski 22 Sep '12

22 Sep '12

Hi, On Via codecs reported on https://bugs.launchpad.net/bugs/1052499, while investigating the report, it can be seen that on Headphone pin the plug is not detectable (NO_PRESENCE). It could be that front panel is AC97 wiring or something else, I don't know. But it makes the jack detection handling code to add the headphone jack as "phantom", which makes snd_hda_jack_detect return headphone jack as always present (commit 80c8bfbe768 is applied/cherry-picked on the reported kernel). Thus after jacks are added, the automute code will always detect HP as present, and keep muted all line-outs. The proposed solution following this is using is_jack_detectable inside via_hp_automute to handle this, other codecs seem to use also is_jack_detectable at the moment in similar situations. Other approach could be also to not report presence on phantom jacks with HP or Line Outs. -- []'s Herton

3 3

Re: [alsa-devel] question about realtek ALC269VB
by Takashi Iwai 21 Sep '12

21 Sep '12

At Fri, 21 Sep 2012 19:07:28 +0200, Oleksij Rempel wrote: > > Hallo all, Takashi, > > in attachment is a patch to enable external mic on Asus Zenbook UX32A. > It works fine for me, except there created some mixer/controls which > make no sense on this laptop. For example "Digital" input and IEC958. > How are they detected? By pin descriptor, or some code insight of > patch_realtek.c It must come from the default pin setup. I guess your machine has either a HDMI or a DP output, no? If it has, IEC958 is a mandatory control for the audio output over such a digital interface. "Digital" is likely the volume control created by alsa-lib, so it's no driver issue, and you can safely ignore it. Takashi > amixer and codec dump are attached too. > > > -- > Regards, > Oleksij > [2 0001-ALSA-hda-Add-external-mic-quirk-for-Asus-Zenbook-UX3.patch <text/x-patch (7bit)>] > >From 46472c1fbd9b82ea95cf22f2b49c74d4a7e82360 Mon Sep 17 00:00:00 2001 > From: Oleksij Rempel <bug-track(a)fisher-privat.net> > Date: Fri, 21 Sep 2012 17:44:58 +0200 > Subject: [PATCH] ALSA: hda - Add external mic quirk for Asus Zenbook UX31A > > Signed-off-by: Oleksij Rempel <bug-track(a)fisher-privat.net> > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 1907dda..3d440fc 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -6189,6 +6189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), > SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), > SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), > + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), > SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), > SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), > SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), > -- > 1.7.10.4 > > [3 amixer <text/plain; UTF-8 (base64)>] > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 87 > Mono: Playback 47 [54%] [-30.00dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 87 [100%] [0.00dB] [on] > Front Right: Playback 87 [100%] [0.00dB] [on] > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 87 [100%] [0.00dB] [on] > Front Right: Playback 87 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Mic',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 0 [0%] [-34.50dB] [off] > Front Right: Playback 0 [0%] [-34.50dB] [off] > Simple mixer control 'Mic Boost',0 > Capabilities: volume penum > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'IEC958',0 > Capabilities: pswitch pswitch-joined penum > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 31 > Front Left: Capture 14 [45%] [4.50dB] [on] > Front Right: Capture 14 [45%] [4.50dB] [on] > Simple mixer control 'Auto-Mute Mode',0 > Capabilities: enum > Items: 'Disabled' 'Enabled' > Item0: 'Enabled' > Simple mixer control 'Digital',0 > Capabilities: cvolume penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 120 > Front Left: Capture 120 [100%] [30.00dB] > Front Right: Capture 120 [100%] [30.00dB] > Simple mixer control 'Internal Mic Boost',0 > Capabilities: volume penum > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 2 [67%] [24.00dB] > Front Right: 2 [67%] [24.00dB] > [4 hda_1 <text/plain; UTF-8 (base64)>] > Codec: Realtek ALC269VB > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x10ec0269 > Subsystem Id: 0x10431517 > Revision Id: 0x100100 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x2f 0x2f] > Converter: stream=8, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out > Control: name="Speaker Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x2f 0x2f] > Converter: stream=8, channel=0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x8b 0x8b] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In > Control: name="Capture Switch", index=0, device=0 > Control: name="Capture Volume", index=0, device=0 > Device: name="ALC269VB Analog", type="Audio", device=0 > Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x0e 0x0e] > Converter: stream=4, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Connection: 1 > 0x22 > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Control: name="Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x03 0x0b > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00] [0x80] > Connection: 2 > 0x02 0x0b > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x12 [Pin Complex] wcaps 0x40000b: Stereo Amp-In > Control: name="Internal Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Internal Mic Phantom Jack", index=0, device=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x01 0x01] > Pincap 0x00000020: IN > Pin Default 0x99a3092f: [Fixed] Mic at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x2, Sequence = 0xf > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Speaker Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Speaker Phantom Jack", index=0, device=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00010014: OUT EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI > Conn = ATAPI, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c 0x0d* > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x17 [Pin Complex] wcaps 0x40010c: Mono Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80] > Pincap 0x00000010: OUT > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x0f > Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Control: name="Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Mic Jack", index=0, device=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00001734: IN OUT Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x01a19c20: [Jack] Mic at Ext Rear > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=02, enabled=1 > Connection: 1 > 0x0d > Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00001724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0000003c: IN OUT HP Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x00000034: IN OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Connection: 2 > 0x0c* 0x0d > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x1d [Pin Complex] wcaps 0x400000: Mono > Pincap 0x00000020: IN > Pin Default 0x40079a2d: [N/A] Line Out at Ext N/A > Conn = Analog, Color = Pink > DefAssociation = 0x2, Sequence = 0xd > Pin-ctls: 0x20: IN > Node 0x1e [Pin Complex] wcaps 0x400381: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Connection: 1 > 0x06 > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=25 > Node 0x21 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out > Control: name="Headphone Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Headphone Jack", index=0, device=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0000001c: OUT HP Detect > Pin Default 0x0121401f: [Jack] HP Out at Ext Rear > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Connection: 2 > 0x0c* 0x0d > Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In > Amp-In caps: N/A > Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12* > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 6 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b

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[alsa-devel] Infinite loop in snd_pcm_hw_htimestamp() for capture PCMs?
by Andrew Eikum 21 Sep '12

21 Sep '12

Hi folks, I've found some unexpected behavior in alsa-lib. The attached program uses 100% CPU in an infinite loop, when I'd expect it to return either an error or a valid htimestamp. I believe the loop occurs in <src/pcm/pcm_hw.c:snd_pcm_hw_htimestamp()>. Am I not supposed to call htimestamp() on this type of PCM? Is the PCM still in an invalid state when htimestamp() is called? Card: HDA ATI SB Chip: Realtek ALC889 Happy for any insight. Andrew

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[alsa-devel] [PATCH] ASoC: wm8960: remove 'dres' field from platform data structure
by Timur Tabi 21 Sep '12

21 Sep '12

The 'dres' field (discharge resistance for headphone outputs) is no longer used in the driver, so remove it. It was used in the original version of the driver when entering standby from off, but we stopped using it when we switched from having a single startup sequence to having separate cap and capless sequences. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- include/sound/wm8960.h | 2 -- sound/soc/codecs/wm8960.c | 5 ----- 2 files changed, 0 insertions(+), 7 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index b5a1ab9e..e8ce8ee 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -18,8 +18,6 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ - int dres; /* Discharge resistance for headphone outputs */ - bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 782faa0..f0f6f660 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -962,11 +962,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); } else { - if (pdata->dres > WM8960_DRES_MAX) { - dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); - pdata->dres = 0; - } - if (pdata->capless) wm8960->set_bias_level = wm8960_set_bias_level_capless; } -- 1.7.3.4

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[alsa-devel] HD-audio runtime->delay accounting
by Takashi Iwai 21 Sep '12

21 Sep '12

Hi, recently Pierre questioned about the validity of COMBO positioning mode, and it turned out that the position reporting can be handled better if we take the difference between LPIB and DMAPOS as runtime->delay. (In short, LPIB points the position where the data fed to the codec, and DMAPOS points the data fetched to DMA.) We cooked up a test patch below and it seems working well on a few machines I've tested. I think we might have still chance to put this into 3.7, as this would be a good improvement. The patch below is a combined work of two fixes: fixing the better stream start by always using SSYNC bit, and adjusting the position and runtime->delay. Of course, the biggest question is the test coverage. David, could you check whether this doesn't break so much on (some) machines you can test? Of course, if any other people can test it, it'll be a great help, too. The patch should be applied to the latest sound.git tree master branch, but it should be applicable (and easily resolvable) for older versions. The patch extends this new mode only for recent Intel chips, just to be sure. thanks, Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 66fefd1..b1d99d0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -555,6 +555,7 @@ enum { #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ #define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ +#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -2003,14 +2004,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_lock(&chip->reg_lock); - if (nsync > 1) { - /* first, set SYNC bits of corresponding streams */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) | sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); - } + + /* first, set SYNC bits of corresponding streams */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) | sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); + snd_pcm_group_for_each_entry(s, substream) { if (s->pcm->card != substream->pcm->card) continue; @@ -2028,8 +2029,6 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_unlock(&chip->reg_lock); if (start) { - if (nsync == 1) - return 0; /* wait until all FIFOs get ready */ for (timeout = 5000; timeout; timeout--) { nwait = 0; @@ -2062,16 +2061,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) cpu_relax(); } } - if (nsync > 1) { - spin_lock(&chip->reg_lock); - /* reset SYNC bits */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) & ~sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); - spin_unlock(&chip->reg_lock); - } + spin_lock(&chip->reg_lock); + /* reset SYNC bits */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) & ~sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); + spin_unlock(&chip->reg_lock); return 0; } @@ -2164,6 +2161,28 @@ static unsigned int azx_get_position(struct azx *chip, if (pos >= azx_dev->bufsize) pos = 0; + + /* calculate runtime delay from LPIB */ + if (azx_dev->substream->runtime && + chip->position_fix[stream] == POS_FIX_POSBUF && + (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) { + unsigned int lpib_pos = azx_sd_readl(azx_dev, SD_LPIB); + int delay; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = pos - lpib_pos; + else + delay = lpib_pos - pos; + if (delay < 0) + delay += azx_dev->bufsize; + if (delay >= azx_dev->period_bytes) { + snd_printdd("delay %d > period_bytes %d\n", + delay, azx_dev->period_bytes); + delay = 0; /* something is wrong */ + } + azx_dev->substream->runtime->delay = + bytes_to_frames(azx_dev->substream->runtime, delay); + } + return pos; } @@ -3511,34 +3530,34 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c21), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, { PCI_DEVICE(0x8086, 0x0d0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | @@ -3575,7 +3594,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE }, + .driver_data = AZX_DRIVER_ICH | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* ATI SB 450/600/700/800/900 */ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB },

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Re: [alsa-devel] sound/pci/hda/patch_ca0132.c: potential null dereference 'dma_engine'
by Takashi Iwai 21 Sep '12

21 Sep '12

At Fri, 21 Sep 2012 13:47:54 +0200, walter harms wrote: > > > > Am 21.09.2012 13:46, schrieb Takashi Iwai: > > At Fri, 21 Sep 2012 18:22:42 +0800, > > Fengguang Wu wrote: > >> > >> Hi Ian, > >> > >> FYI, there are new smatch warnings show up in > >> > >> commit 0b37d4c7982442730061a737c16939e8cc363861 > >> Author: Ian Minett <ian_minett(a)creativelabs.com> > >> Commit: Takashi Iwai <tiwai(a)suse.de> > >> CommitDate: Fri Sep 21 09:54:29 2012 +0200 > >> > >> + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) > >> + sound/pci/hda/patch_ca0132.c:2056 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1963) > > > > Fixed with the patch below. Thanks! > > > > > > Takashi > > > > === > > Subject: [PATCH] ALSA: hda - Fix NULL dereference in error path of patch_ca0132.c > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/patch_ca0132.c | 6 ++---- > > 1 file changed, 2 insertions(+), 4 deletions(-) > > > > diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c > > index 5c6a056..03f57c9 100644 > > --- a/sound/pci/hda/patch_ca0132.c > > +++ b/sound/pci/hda/patch_ca0132.c > > @@ -1960,10 +1960,8 @@ static int dspxfr_image(struct hda_codec *codec, > > return -EINVAL; > > > > dma_engine = kzalloc(sizeof(*dma_engine), GFP_KERNEL); > > - if (!dma_engine) { > > - status = -ENOMEM; > > - goto exit; > > - } > > + if (!dma_engine) > > + return -ENOMEM; > > memset((void *)dma_engine, 0, sizeof(*dma_engine)); > > > > Why do you need that memset() ? kzalloc() should fill the buffer with 0. True. Takashi

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[alsa-devel] [RFC PATCH v3 0/5] snd-ice1712: Add Philips PSC724 Ultimate Edge
by Ondrej Zary 20 Sep '12

20 Sep '12

Hello, this is third attempt to add Philips PSC724 Ultimate Edge sound card support to snd-ice1712 driver. Unlike other snd-ice1712 subdrivers, the codec code is splitted into separate files, reusable by other ice1712 subdrivers. Working: all analog outputs (front, rear, center+lfe, headphone) and inputs (front mic, rear mic, line, cd, aux), mixer, headphone jack detection, suspend/resume Currently untested: SPDIF Buggy: switching AGC mode in mixer causes alsamixer to crash - looks like it does not like notifications about control activating/deactivating

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[alsa-devel] [PATCH 08/29] ARM: OMAP2+: Make board-zoom.h local
by Tony Lindgren 20 Sep '12

20 Sep '12

This can be local to mach-omap2 after we rip out the messed up extmute stuff from board-zoom-peripherals.c that's all over the place and probably does not even work because of all the hardcoded GPIO numbers. I also suggest ASoC gusy remove all extmute handling from sound/soc/codecs/twl4030.c unless it's been tested to work and implemented in a saner way. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap2/board-3630sdp.c | 2 +- arch/arm/mach-omap2/board-ldp.c | 2 +- arch/arm/mach-omap2/board-zoom-debugboard.c | 2 +- arch/arm/mach-omap2/board-zoom-display.c | 2 +- arch/arm/mach-omap2/board-zoom-peripherals.c | 16 +--------------- arch/arm/mach-omap2/board-zoom.c | 2 +- arch/arm/mach-omap2/board-zoom.h | 0 sound/soc/omap/zoom2.c | 2 +- 8 files changed, 7 insertions(+), 21 deletions(-) rename arch/arm/mach-omap2/{include/mach/board-zoom.h => board-zoom.h} (100%) diff --git a/arch/arm/mach-omap2/board-3630sdp.c b/arch/arm/mach-omap2/board-3630sdp.c index 8518b13..c7d6ea2 100644 --- a/arch/arm/mach-omap2/board-3630sdp.c +++ b/arch/arm/mach-omap2/board-3630sdp.c @@ -20,7 +20,7 @@ #include <plat/gpmc-smc91x.h> #include <plat/usb.h> -#include <mach/board-zoom.h> +#include "board-zoom.h" #include "board-flash.h" #include "mux.h" diff --git a/arch/arm/mach-omap2/board-ldp.c b/arch/arm/mach-omap2/board-ldp.c index 3f3a552..d0372a6 100644 --- a/arch/arm/mach-omap2/board-ldp.c +++ b/arch/arm/mach-omap2/board-ldp.c @@ -36,7 +36,7 @@ #include <plat/mcspi.h> #include "common.h" #include <plat/gpmc.h> -#include <mach/board-zoom.h> +#include "board-zoom.h" #include <plat/usb.h> #include <plat/gpmc-smsc911x.h> diff --git a/arch/arm/mach-omap2/board-zoom-debugboard.c b/arch/arm/mach-omap2/board-zoom-debugboard.c index 0d8d919..42c97c2 100644 --- a/arch/arm/mach-omap2/board-zoom-debugboard.c +++ b/arch/arm/mach-omap2/board-zoom-debugboard.c @@ -20,7 +20,7 @@ #include <plat/gpmc.h> #include <plat/gpmc-smsc911x.h> -#include <mach/board-zoom.h> +#include "board-zoom.h" #include "soc.h" #include "common.h" diff --git a/arch/arm/mach-omap2/board-zoom-display.c b/arch/arm/mach-omap2/board-zoom-display.c index ea79bc2..6a162fb 100644 --- a/arch/arm/mach-omap2/board-zoom-display.c +++ b/arch/arm/mach-omap2/board-zoom-display.c @@ -16,7 +16,7 @@ #include <linux/spi/spi.h> #include <plat/mcspi.h> #include <video/omapdss.h> -#include <mach/board-zoom.h> +#include "board-zoom.h" #include "common.h" diff --git a/arch/arm/mach-omap2/board-zoom-peripherals.c b/arch/arm/mach-omap2/board-zoom-peripherals.c index 6bcc107..c8377b71 100644 --- a/arch/arm/mach-omap2/board-zoom-peripherals.c +++ b/arch/arm/mach-omap2/board-zoom-peripherals.c @@ -28,7 +28,7 @@ #include "common.h" #include <plat/usb.h> -#include <mach/board-zoom.h> +#include "board-zoom.h" #include "mux.h" #include "hsmmc.h" @@ -245,12 +245,6 @@ static int zoom_twl_gpio_setup(struct device *dev, return ret; } -/* EXTMUTE callback function */ -static void zoom2_set_hs_extmute(int mute) -{ - gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute); -} - static struct twl4030_gpio_platform_data zoom_gpio_data = { .setup = zoom_twl_gpio_setup, }; @@ -271,14 +265,6 @@ static int __init omap_i2c_init(void) TWL_COMMON_PDATA_MADC | TWL_COMMON_PDATA_AUDIO, TWL_COMMON_REGULATOR_VDAC | TWL_COMMON_REGULATOR_VPLL2); - if (machine_is_omap_zoom2()) { - struct twl4030_codec_data *codec_data; - codec_data = zoom_twldata.audio->codec; - - codec_data->ramp_delay_value = 3; /* 161 ms */ - codec_data->hs_extmute = 1; - codec_data->set_hs_extmute = zoom2_set_hs_extmute; - } omap_pmic_init(1, 2400, "twl5030", 7 + OMAP_INTC_START, &zoom_twldata); omap_register_i2c_bus(2, 400, NULL, 0); omap_register_i2c_bus(3, 400, NULL, 0); diff --git a/arch/arm/mach-omap2/board-zoom.c b/arch/arm/mach-omap2/board-zoom.c index 4994438..cbfd9df 100644 --- a/arch/arm/mach-omap2/board-zoom.c +++ b/arch/arm/mach-omap2/board-zoom.c @@ -24,7 +24,7 @@ #include "common.h" #include <plat/usb.h> -#include <mach/board-zoom.h> +#include "board-zoom.h" #include "board-flash.h" #include "mux.h" diff --git a/arch/arm/mach-omap2/include/mach/board-zoom.h b/arch/arm/mach-omap2/board-zoom.h similarity index 100% rename from arch/arm/mach-omap2/include/mach/board-zoom.h rename to arch/arm/mach-omap2/board-zoom.h diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 920e0d9..187cb0c 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -28,7 +28,6 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/board-zoom.h> #include <plat/mcbsp.h> /* Register descriptions for twl4030 codec part */ @@ -39,6 +38,7 @@ #include "omap-pcm.h" #define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) +#define ZOOM2_HEADSET_EXTMUTE_GPIO 153 static int zoom2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params)

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[alsa-devel] [PATCH 04/29] ARM: OMAP1: Move board-ams-delta.h from plat to mach
by Tony Lindgren 20 Sep '12

20 Sep '12

This is only used by omap1. And to fix things properly, this should not be included from the drivers at all. Cc: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Cc: Tomi Valkeinen <tomi.valkeinen(a)ti.com> Cc: linux-fbdev(a)vger.kernel.org Cc: Artem Bityutskiy <Artem.Bityutskiy(a)linux.intel.com> Cc: linux-mtd(a)lists.infradead.org Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: linux-input(a)vger.kernel.org Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap1/ams-delta-fiq-handler.S | 2 +- arch/arm/mach-omap1/ams-delta-fiq.c | 2 +- arch/arm/mach-omap1/board-ams-delta.c | 2 +- arch/arm/mach-omap1/include/mach/board-ams-delta.h | 0 drivers/input/serio/ams_delta_serio.c | 2 +- drivers/mtd/nand/ams-delta.c | 2 +- drivers/video/omap/lcd_ams_delta.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- 8 files changed, 7 insertions(+), 7 deletions(-) rename arch/arm/{plat-omap/include/plat/board-ams-delta.h => mach-omap1/include/mach/board-ams-delta.h} (100%) diff --git a/arch/arm/mach-omap1/ams-delta-fiq-handler.S b/arch/arm/mach-omap1/ams-delta-fiq-handler.S index d2b6acc..3d1e1c2 100644 --- a/arch/arm/mach-omap1/ams-delta-fiq-handler.S +++ b/arch/arm/mach-omap1/ams-delta-fiq-handler.S @@ -16,7 +16,7 @@ #include <linux/linkage.h> #include <asm/assembler.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include <mach/irqs.h> #include <mach/ams-delta-fiq.h> diff --git a/arch/arm/mach-omap1/ams-delta-fiq.c b/arch/arm/mach-omap1/ams-delta-fiq.c index 68e8e56..f12a12a 100644 --- a/arch/arm/mach-omap1/ams-delta-fiq.c +++ b/arch/arm/mach-omap1/ams-delta-fiq.c @@ -19,7 +19,7 @@ #include <linux/module.h> #include <linux/io.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include <asm/fiq.h> diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 05af063..b9929cb 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -35,7 +35,7 @@ #include <asm/mach/arch.h> #include <asm/mach/map.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include <plat/keypad.h> #include <mach/mux.h> diff --git a/arch/arm/plat-omap/include/plat/board-ams-delta.h b/arch/arm/mach-omap1/include/mach/board-ams-delta.h similarity index 100% rename from arch/arm/plat-omap/include/plat/board-ams-delta.h rename to arch/arm/mach-omap1/include/mach/board-ams-delta.h diff --git a/drivers/input/serio/ams_delta_serio.c b/drivers/input/serio/ams_delta_serio.c index f5fbdf9..45887e3 100644 --- a/drivers/input/serio/ams_delta_serio.c +++ b/drivers/input/serio/ams_delta_serio.c @@ -27,7 +27,7 @@ #include <linux/module.h> #include <asm/mach-types.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include <mach/ams-delta-fiq.h> diff --git a/drivers/mtd/nand/ams-delta.c b/drivers/mtd/nand/ams-delta.c index c28eec0..9e7723a 100644 --- a/drivers/mtd/nand/ams-delta.c +++ b/drivers/mtd/nand/ams-delta.c @@ -29,7 +29,7 @@ #include <asm/io.h> #include <asm/sizes.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include <mach/hardware.h> diff --git a/drivers/video/omap/lcd_ams_delta.c b/drivers/video/omap/lcd_ams_delta.c index 5cca6b3..ed4cad8 100644 --- a/drivers/video/omap/lcd_ams_delta.c +++ b/drivers/video/omap/lcd_ams_delta.c @@ -27,7 +27,7 @@ #include <linux/lcd.h> #include <linux/gpio.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include "omapfb.h" diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 7b18b74..98e25d5 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -32,7 +32,7 @@ #include <asm/mach-types.h> -#include <plat/board-ams-delta.h> +#include <mach/board-ams-delta.h> #include <plat/mcbsp.h> #include "omap-mcbsp.h"

3 4

[alsa-devel] [PATCH] ALSA: hda - use both input paths on Conexant auto parser
by David Henningsson 20 Sep '12

20 Sep '12

On the Thinkpad W520 - and probably several other machines with Conexant 506x chips - the Dock Mic and Mic are connected to the same two selector nodes. This patch will make Dock Mic take one selector node and Mic take the other, when possible. Without the patch, both paths would take the first selector, leading to the normal Mic's volume being controlled by "Dock Mic Boost". (On other machines, this could instead fixup similar problems between Mic and Line In, for example.) BugLink: https://bugs.launchpad.net/bugs/1037642 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- I asked for alsa-info on this one but got Ubuntu apport info instead, see e g https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1037642/+attachme… In general I guess this could be sent to stable but it always feels scary when it's generic code :-) what do you think? sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b871b013..c03d3b8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3540,8 +3540,9 @@ static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, hda_nid_t pin, hda_nid_t *srcp, bool do_select, int depth) { + struct conexant_spec *spec = codec->spec; hda_nid_t conn[HDA_MAX_NUM_INPUTS]; - int i, nums; + int startidx, i, nums; switch (get_wcaps_type(get_wcaps(codec, mux))) { case AC_WID_AUD_IN: @@ -3565,14 +3566,25 @@ static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, depth++; if (depth == 2) return -1; + + /* Try to rotate around connections to avoid one boost controlling + another input path as well */ + startidx = 0; + for (i = 0; i < spec->private_imux.num_items; i++) + if (spec->imux_info[i].pin == pin) { + startidx = i; + break; + } + for (i = 0; i < nums; i++) { - int ret = __select_input_connection(codec, conn[i], pin, srcp, + int j = (i + startidx) % nums; + int ret = __select_input_connection(codec, conn[j], pin, srcp, do_select, depth); if (ret >= 0) { if (do_select) snd_hda_codec_write(codec, mux, 0, - AC_VERB_SET_CONNECT_SEL, i); - return i; + AC_VERB_SET_CONNECT_SEL, j); + return j; } } return -1; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe rather than as a device
by Lee Jones 20 Sep '12

20 Sep '12

The PCM is a pseudo-device. It doesn't have any of it's own registers or hardware. It rather acts as a layer of abstraction for DMA transfers. Hence, instead of classifying it as a device in its own right, we call the initialisation from the MSP driver. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/mop500.c | 4 ++-- sound/soc/ux500/ux500_msp_dai.c | 14 +++++++++++++- sound/soc/ux500/ux500_pcm.c | 19 ++++--------------- sound/soc/ux500/ux500_pcm.h | 3 +++ 4 files changed, 22 insertions(+), 18 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 6840df7..fe6223a 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -33,7 +33,7 @@ struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_0", .cpu_dai_name = "ux500-msp-i2s.1", .codec_dai_name = "ab8500-codec-dai.0", - .platform_name = "ux500-pcm.0", + .platform_name = "ux500-msp-i2s.1", .codec_name = "ab8500-codec.0", .init = mop500_ab8500_machine_init, .ops = mop500_ab8500_ops, @@ -43,7 +43,7 @@ struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_1", .cpu_dai_name = "ux500-msp-i2s.3", .codec_dai_name = "ab8500-codec-dai.1", - .platform_name = "ux500-pcm.0", + .platform_name = "ux500-msp-i2s.3", .codec_name = "ab8500-codec.0", .init = NULL, .ops = mop500_ab8500_ops, diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 0f7dd49..3476d6a 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -28,6 +28,7 @@ #include "ux500_msp_i2s.h" #include "ux500_msp_dai.h" +#include "ux500_pcm.h" static int setup_pcm_multichan(struct snd_soc_dai *dai, struct ux500_msp_config *msp_config) @@ -806,11 +807,20 @@ static int __devinit ux500_msp_drv_probe(struct platform_device *pdev) goto err_init_msp; } + ret = ux500_pcm_register_platform(pdev); + if (ret < 0) { + dev_err(&pdev->dev, + "Error: %s: Failed to register PCM platform device!\n", + __func__); + goto err_reg_plat; + } + return 0; +err_reg_plat: + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); err_init_msp: clk_put(drvdata->clk); - err_clk: devm_regulator_put(drvdata->reg_vape); @@ -821,6 +831,8 @@ static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(&pdev->dev); + ux500_pcm_unregister_platform(pdev); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(ux500_msp_dai_drv)); devm_regulator_put(drvdata->reg_vape); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 1a04e24..894c9f4 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -282,7 +282,7 @@ static struct snd_soc_platform_driver ux500_pcm_soc_drv = { .pcm_new = ux500_pcm_new, }; -static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +int __devinit ux500_pcm_register_platform(struct platform_device *pdev) { int ret; @@ -296,23 +296,12 @@ static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) return 0; } +EXPORT_SYMBOL_GPL(ux500_pcm_register_platform); -static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +int __devexit ux500_pcm_unregister_platform(struct platform_device *pdev) { 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), -}; -module_platform_driver(ux500_pcm_driver); - -MODULE_LICENSE("GPL v2"); +EXPORT_SYMBOL_GPL(ux500_pcm_unregister_platform); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h index 77ed44d..76d3444 100644 --- a/sound/soc/ux500/ux500_pcm.h +++ b/sound/soc/ux500/ux500_pcm.h @@ -32,4 +32,7 @@ #define UX500_PLATFORM_PERIODS_MAX 48 #define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) +int ux500_pcm_register_platform(struct platform_device *pdev); +int ux500_pcm_unregister_platform(struct platform_device *pdev); + #endif -- 1.7.9.5

4 5

[alsa-devel] [PATCH 04/32] Documentation: Add the AB8500 CODEC device to the MFD AB8500 doc
by Lee Jones 20 Sep '12

20 Sep '12

Here we document the AB8500 Audio CODEC in the existing MFD AB8500 document. CC: alsa-devel(a)alsa-project.org Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- Documentation/devicetree/bindings/mfd/ab8500.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/Documentation/devicetree/bindings/mfd/ab8500.txt b/Documentation/devicetree/bindings/mfd/ab8500.txt index 69e757a..ce83c8d 100644 --- a/Documentation/devicetree/bindings/mfd/ab8500.txt +++ b/Documentation/devicetree/bindings/mfd/ab8500.txt @@ -23,6 +23,7 @@ Device IRQ Names Supply Names Description ab8500-bm : : : Battery Manager ab8500-btemp : : : Battery Temperature ab8500-charger : : : Battery Charger +ab8500-codec : : : Audio Codec ab8500-fg : : : Fuel Gauge ab8500-gpadc : HW_CONV_END : vddadc : Analogue to Digital Converter SW_CONV_END : : @@ -52,6 +53,14 @@ Optional child device properties: supplied in the interrupts property - <supply_name>-supply : contains a phandle to the regulator supply node in Device Tree +Non-standard child device properties: + - Audio CODEC: + - stericsson,amic[1|2]-type-single-ended : Single-ended Analoge Mic (default: differential) + - stericsson,amic1a-bias-vamic2 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,amic1b-bias-vamic2 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,amic2-bias-vamic1 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,earpeice-cmv : Earpeice voltage (only: 950 | 1100 | 1270 | 1580) + ab8500@5 { compatible = "stericsson,ab8500"; reg = <5>; /* mailbox 5 is i2c */ @@ -110,6 +119,12 @@ ab8500@5 { compatible = "stericsson,ab8500-pwm"; }; + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; + ab8500-regulators { compatible = "stericsson,ab8500-regulator"; -- 1.7.9.5

1 0

[alsa-devel] [GIT PULL] MOP500 Audio patches due for ASoC
by Lee Jones 20 Sep '12

20 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.linaro.org/people/ljones/linux-3.0-ux500.git for-asoc-next for you to fetch changes up to b4cad7af6665743647f28119b689e1552326d4da: Documentation: Define the MSP Driver Device Tree bindings (2012-09-20 09:10:59 +0200) ---------------------------------------------------------------- Lee Jones (7): ASoC: Ux500: Move MSP pinctrl setup into the MSP driver ASoC: Ux500: Enable MOP500 driver for Device Tree ASoC: Ux500: Enable ux500 MSP driver for Device Tree ASoC: codecs: Enable AB8500 CODEC for Device Tree ASoC: Ux500: Minor coding layout changes Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings Documentation: Define the MSP Driver Device Tree bindings .../devicetree/bindings/sound/ux500-mop500.txt | 39 +++++++++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 79 +---------------- arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 ++++++++++++++++++ sound/soc/ux500/mop500.c | 47 +++++++++-- sound/soc/ux500/ux500_msp_dai.c | 6 ++ sound/soc/ux500/ux500_msp_i2s.c | 89 +++++++++++++++----- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 10 files changed, 292 insertions(+), 108 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt -- Lee Jones Linaro ST-Ericsson Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

2 2

[alsa-devel] [PATCH v2] ASoC: codecs: Add DA9055 codec driver
by Ashish Chavan 20 Sep '12

20 Sep '12

This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- Changes since v1: - Made extensive use of DAPM throughout the driver - Removed LDO, DMIC and other non used controls - Corrected names for many controls - Other minor improvements --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1468 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1474 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3684255..b92759a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca508b2..9bd4d95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c new file mode 100644 index 0000000..86661ec --- /dev/null +++ b/sound/soc/codecs/da9055.c @@ -0,0 +1,1468 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * 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. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +/* DA9055 register space */ + +/* Status Registers */ +#define DA9055_STATUS1 0x02 +#define DA9055_PLL_STATUS 0x03 +#define DA9055_AUX_L_GAIN_STATUS 0x04 +#define DA9055_AUX_R_GAIN_STATUS 0x05 +#define DA9055_MIC_L_GAIN_STATUS 0x06 +#define DA9055_MIC_R_GAIN_STATUS 0x07 +#define DA9055_MIXIN_L_GAIN_STATUS 0x08 +#define DA9055_MIXIN_R_GAIN_STATUS 0x09 +#define DA9055_ADC_L_GAIN_STATUS 0x0A +#define DA9055_ADC_R_GAIN_STATUS 0x0B +#define DA9055_DAC_L_GAIN_STATUS 0x0C +#define DA9055_DAC_R_GAIN_STATUS 0x0D +#define DA9055_HP_L_GAIN_STATUS 0x0E +#define DA9055_HP_R_GAIN_STATUS 0x0F +#define DA9055_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA9055_CIF_CTRL 0x20 +#define DA9055_DIG_ROUTING_AIF 0X21 +#define DA9055_SR 0x22 +#define DA9055_REFERENCES 0x23 +#define DA9055_PLL_FRAC_TOP 0x24 +#define DA9055_PLL_FRAC_BOT 0x25 +#define DA9055_PLL_INTEGER 0x26 +#define DA9055_PLL_CTRL 0x27 +#define DA9055_AIF_CLK_MODE 0x28 +#define DA9055_AIF_CTRL 0x29 +#define DA9055_DIG_ROUTING_DAC 0x2A +#define DA9055_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA9055_AUX_L_GAIN 0x30 +#define DA9055_AUX_R_GAIN 0x31 +#define DA9055_MIXIN_L_SELECT 0x32 +#define DA9055_MIXIN_R_SELECT 0x33 +#define DA9055_MIXIN_L_GAIN 0x34 +#define DA9055_MIXIN_R_GAIN 0x35 +#define DA9055_ADC_L_GAIN 0x36 +#define DA9055_ADC_R_GAIN 0x37 +#define DA9055_ADC_FILTERS1 0x38 +#define DA9055_MIC_L_GAIN 0x39 +#define DA9055_MIC_R_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA9055_DAC_FILTERS5 0x40 +#define DA9055_DAC_FILTERS2 0x41 +#define DA9055_DAC_FILTERS3 0x42 +#define DA9055_DAC_FILTERS4 0x43 +#define DA9055_DAC_FILTERS1 0x44 +#define DA9055_DAC_L_GAIN 0x45 +#define DA9055_DAC_R_GAIN 0x46 +#define DA9055_CP_CTRL 0x47 +#define DA9055_HP_L_GAIN 0x48 +#define DA9055_HP_R_GAIN 0x49 +#define DA9055_LINE_GAIN 0x4A +#define DA9055_MIXOUT_L_SELECT 0x4B +#define DA9055_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA9055_SYSTEM_MODES_INPUT 0x50 +#define DA9055_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA9055_AUX_L_CTRL 0x60 +#define DA9055_AUX_R_CTRL 0x61 +#define DA9055_MIC_BIAS_CTRL 0x62 +#define DA9055_MIC_L_CTRL 0x63 +#define DA9055_MIC_R_CTRL 0x64 +#define DA9055_MIXIN_L_CTRL 0x65 +#define DA9055_MIXIN_R_CTRL 0x66 +#define DA9055_ADC_L_CTRL 0x67 +#define DA9055_ADC_R_CTRL 0x68 +#define DA9055_DAC_L_CTRL 0x69 +#define DA9055_DAC_R_CTRL 0x6A +#define DA9055_HP_L_CTRL 0x6B +#define DA9055_HP_R_CTRL 0x6C +#define DA9055_LINE_CTRL 0x6D +#define DA9055_MIXOUT_L_CTRL 0x6E +#define DA9055_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA9055_LDO_CTRL 0x90 +#define DA9055_IO_CTRL 0x91 +#define DA9055_GAIN_RAMP_CTRL 0x92 +#define DA9055_MIC_CONFIG 0x93 +#define DA9055_PC_COUNT 0x94 +#define DA9055_CP_VOL_THRESHOLD1 0x95 +#define DA9055_CP_DELAY 0x96 +#define DA9055_CP_DETECTOR 0x97 +#define DA9055_AIF_OFFSET 0x98 +#define DA9055_DIG_CTRL 0x99 +#define DA9055_ALC_CTRL2 0x9A +#define DA9055_ALC_CTRL3 0x9B +#define DA9055_ALC_NOISE 0x9C +#define DA9055_ALC_TARGET_MIN 0x9D +#define DA9055_ALC_TARGET_MAX 0x9E +#define DA9055_ALC_GAIN_LIMITS 0x9F +#define DA9055_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA9055_ALC_ANTICLIP_CTRL 0xA1 +#define DA9055_ALC_ANTICLIP_LEVEL 0xA2 +#define DA9055_ALC_OFFSET_OP2M_L 0xA6 +#define DA9055_ALC_OFFSET_OP2U_L 0xA7 +#define DA9055_ALC_OFFSET_OP2M_R 0xAB +#define DA9055_ALC_OFFSET_OP2U_R 0xAC +#define DA9055_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA9055_ALC_CIC_OP_LVL_DATA 0xAE +#define DA9055_DAC_NG_SETUP_TIME 0xAF +#define DA9055_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA9055_DAC_NG_ON_THRESHOLD 0xB1 +#define DA9055_DAC_NG_CTRL 0xB2 + +/* SR bit fields */ +#define DA9055_SR_8000 (0x1 << 0) +#define DA9055_SR_11025 (0x2 << 0) +#define DA9055_SR_12000 (0x3 << 0) +#define DA9055_SR_16000 (0x5 << 0) +#define DA9055_SR_22050 (0x6 << 0) +#define DA9055_SR_24000 (0x7 << 0) +#define DA9055_SR_32000 (0x9 << 0) +#define DA9055_SR_44100 (0xA << 0) +#define DA9055_SR_48000 (0xB << 0) +#define DA9055_SR_88200 (0xE << 0) +#define DA9055_SR_96000 (0xF << 0) + +/* REFERENCES bit fields */ +#define DA9055_BIAS_EN (1 << 3) +#define DA9055_VMID_EN (1 << 7) + +/* PLL_CTRL bit fields */ +#define DA9055_PLL_INDIV_10_20_MHZ (1 << 2) +#define DA9055_PLL_SRM_EN (1 << 6) +#define DA9055_PLL_EN (1 << 7) + +/* AIF_CLK_MODE bit fields */ +#define DA9055_AIF_BCLKS_PER_WCLK_32 (0 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_64 (1 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_128 (2 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_256 (3 << 0) +#define DA9055_AIF_CLK_EN_SLAVE_MODE (0 << 7) +#define DA9055_AIF_CLK_EN_MASTER_MODE (1 << 7) + +/* AIF_CTRL bit fields */ +#define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) +#define DA9055_AIF_FORMAT_LEFT_J (1 << 0) +#define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_WORD_S16_LE (0 << 2) +#define DA9055_AIF_WORD_S20_3LE (1 << 2) +#define DA9055_AIF_WORD_S24_LE (2 << 2) +#define DA9055_AIF_WORD_S32_LE (3 << 2) + +/* MIXIN_L_CTRL bit fields */ +#define DA9055_MIXIN_L_MIX_EN (1 << 3) + +/* MIXIN_R_CTRL bit fields */ +#define DA9055_MIXIN_R_MIX_EN (1 << 3) + +/* ADC_L_CTRL bit fields */ +#define DA9055_ADC_L_EN (1 << 7) + +/* ADC_R_CTRL bit fields */ +#define DA9055_ADC_R_EN (1 << 7) + +/* DAC_L_CTRL bit fields */ +#define DA9055_DAC_L_MUTE_EN (1 << 6) + +/* DAC_R_CTRL bit fields */ +#define DA9055_DAC_R_MUTE_EN (1 << 6) + +/* HP_L_CTRL bit fields */ +#define DA9055_HP_L_AMP_OE (1 << 3) + +/* HP_R_CTRL bit fields */ +#define DA9055_HP_R_AMP_OE (1 << 3) + +/* LINE_CTRL bit fields */ +#define DA9055_LINE_AMP_OE (1 << 3) + +/* MIXOUT_L_CTRL bit fields */ +#define DA9055_MIXOUT_L_MIX_EN (1 << 3) + +/* MIXOUT_R_CTRL bit fields */ +#define DA9055_MIXOUT_R_MIX_EN (1 << 3) + +/* ALC_CIC_OP_LEVEL_CTRL bit fields */ +#define DA9055_ALC_DATA_MIDDLE (2 << 0) +#define DA9055_ALC_DATA_TOP (3 << 0) +#define DA9055_ALC_CIC_OP_CHANNEL_LEFT (0 << 7) +#define DA9055_ALC_CIC_OP_CHANNEL_RIGHT (1 << 7) + +#define DA9055_AIF_BCLK_MASK (3 << 0) +#define DA9055_AIF_CLK_MODE_MASK (1 << 7) +#define DA9055_AIF_FORMAT_MASK (3 << 0) +#define DA9055_AIF_WORD_LENGTH_MASK (3 << 2) +#define DA9055_GAIN_RAMPING_EN (1 << 5) + +#define DA9055_ALC_OFFSET_15_8 0x00FF00 +#define DA9055_ALC_OFFSET_17_16 0x030000 +#define DA9055_ALC_AVG_ITERATIONS 5 + +struct pll_div { + int fref; + int fout; + u8 frac_top; + u8 frac_bot; + u8 integer; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL divisor table */ +static const struct pll_div da9055_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz and its harmonics */ + {11289600, 2822400, 0x00, 0x00, 0x20, 1}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x03, 0x61, 0x1E, 1}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x0C, 0xCC, 0x1D, 1}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x19, 0x45, 0x1B, 1}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x18, 0x56, 0x1A, 1}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x02, 0xD0, 0x19, 1}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x1A, 0x1C, 0x12, 1}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x0B, 0x6D, 0x12, 1}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x07, 0xDD, 0x12, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz and its harmonics */ + {11289600, 3072000, 0x1A, 0x8E, 0x22, 1}, /* MCLK=11.2896Mhz */ + {12000000, 3072000, 0x18, 0x93, 0x20, 1}, /* MCLK=12Mhz */ + {12288000, 3072000, 0x00, 0x00, 0x20, 1}, /* MCLK=12.288Mhz */ + {13000000, 3072000, 0x07, 0xEA, 0x1E, 1}, /* MCLK=13Mhz */ + {13500000, 3072000, 0x04, 0x11, 0x1D, 1}, /* MCLK=13.5Mhz */ + {14400000, 3072000, 0x09, 0xD0, 0x1B, 1}, /* MCLK=14.4Mhz */ + {19200000, 3072000, 0x0F, 0x5C, 0x14, 1}, /* MCLK=19.2Mhz */ + {19680000, 3072000, 0x1F, 0x60, 0x13, 1}, /* MCLK=19.68Mhz */ + {19800000, 3072000, 0x1B, 0x80, 0x13, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + {11289600, 2822400, 0x0D, 0x47, 0x21, 0}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x0D, 0xFA, 0x1F, 0}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x16, 0x66, 0x1E, 0}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x00, 0x98, 0x1D, 0}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x1E, 0x33, 0x1B, 0}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x06, 0x50, 0x1A, 0}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x14, 0xBC, 0x13, 0}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x05, 0x66, 0x13, 0}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x01, 0xAE, 0x13, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA9055_CLKSRC_MCLK +}; + +/* Gain and Volume */ + +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da9055_hpf_cutoff_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da9055_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +static const struct soc_enum da9055_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da9055_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da9055_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +static const struct soc_enum da9055_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +/* Gain ramping rate value */ +static const char * const da9055_gain_ramping_txt[] = { + "nominal rate", "nominal rate * 4", "nominal rate * 8", + "nominal rate / 8" +}; + +static const struct soc_enum da9055_gain_ramping_rate = + SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); + +/* DAC noise gate setup time value */ +static const char * const da9055_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da9055_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, + da9055_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da9055_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, + da9055_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da9055_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, + da9055_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da9055_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da9055_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, + da9055_dac_soft_mute_rate_txt); + +/* DAC routing select */ +static const char * const da9055_dac_select_txt[] = { + "ADC left output", "ADC right output", "AIF input left", + "AIF input right" +}; + +static const struct soc_enum da9055_dac_l_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_select_txt); + +static const struct soc_enum da9055_dac_r_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_select_txt); + +/* MIC bias select */ +static const char * const da9055_mic_bias_select_txt[] = { + "MIC Bias 1", "MIC Bias 2" +}; + +static const struct soc_enum da9055_mic_bias_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 6, 2, + da9055_mic_bias_select_txt); + +/* MIC PGA Left source select */ +static const char * const da9055_mic_l_src_txt[] = { + "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_l_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt); + +/* MIC PGA Right source select */ +static const char * const da9055_mic_r_src_txt[] = { + "MIC2_R_L", "MIC2_R", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_r_src = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da9055_signal_tracking_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da9055_integ_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, + da9055_signal_tracking_rate_txt); + +static const struct soc_enum da9055_integ_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, + da9055_signal_tracking_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da9055_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da9055_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da9055_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da9055_hold_time = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); + +static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA9055_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA9055_ALC_AVG_ITERATIONS; +} + +static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 reg_val, adc_left, adc_right; + int avg_left_data, avg_right_data, offset_l, offset_r; + + if (ucontrol->value.integer.value[0]) { + /* + * While enabling ALC (or ALC sync mode), calibration of the DC + * offsets must be done first + */ + + /* Save current values from ADC control registers */ + adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); + adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_ADC_L_EN, DA9055_ADC_L_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_ADC_R_EN, DA9055_ADC_R_EN); + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_L, reg_val); + reg_val = (offset_l & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_L, reg_val); + + reg_val = (offset_r & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_R, reg_val); + reg_val = (offset_r & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_R, reg_val); + + /* Restore original values of ADC control registers */ + snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); + snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +static const struct snd_kcontrol_new da9055_snd_controls[] = { + + /* Volume controls */ + SOC_DOUBLE_R_TLV("Mic Volume", + DA9055_MIC_L_GAIN, DA9055_MIC_R_GAIN, + 0, 0x7, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", + DA9055_AUX_L_GAIN, DA9055_AUX_R_GAIN, + 0, 0x3f, 0, aux_vol_tlv), + SOC_DOUBLE_R_TLV("Mixin PGA Volume", + DA9055_MIXIN_L_GAIN, DA9055_MIXIN_R_GAIN, + 0, 0xf, 0, mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", + DA9055_ADC_L_GAIN, DA9055_ADC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Volume", + DA9055_DAC_L_GAIN, DA9055_DAC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", + DA9055_HP_L_GAIN, DA9055_HP_R_GAIN, + 0, 0x3f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA9055_LINE_GAIN, 0, 0x3f, 0, + lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA9055_DAC_FILTERS4, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA9055_DAC_FILTERS2, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA9055_DAC_FILTERS2, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA9055_DAC_FILTERS3, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA9055_DAC_FILTERS3, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA9055_DAC_FILTERS4, 0, 0xf, 0, + eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA9055_ADC_FILTERS1, 7, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da9055_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA9055_ADC_FILTERS1, 3, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da9055_adc_vf_cutoff), + + SOC_SINGLE("DAC HPF Switch", DA9055_DAC_FILTERS1, 7, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da9055_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA9055_DAC_FILTERS1, 3, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da9055_dac_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Switch", DA9055_MIC_L_CTRL, + DA9055_MIC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Aux Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Mixin PGA Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("ADC Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Headphone Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 6, 1, 0), + SOC_SINGLE("Lineout Switch", DA9055_LINE_CTRL, 6, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0), + SOC_ENUM("DAC Soft Mute Rate", da9055_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Headphone ZC Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 4, 1, 0), + SOC_SINGLE("Lineout ZC Switch", DA9055_LINE_CTRL, 4, 1, 0), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA9055_DAC_L_CTRL, + DA9055_DAC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Headphone Gain Ramping Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 5, 1, 0), + SOC_SINGLE("Lineout Gain Ramping Switch", DA9055_LINE_CTRL, 5, 1, 0), + SOC_ENUM("Gain Ramping Rate", da9055_gain_ramping_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Switch", DA9055_DAC_NG_CTRL, 7, 1, 0), + SOC_SINGLE("DAC NG ON Threshold", DA9055_DAC_NG_ON_THRESHOLD, + 0, 0x7, 0), + SOC_SINGLE("DAC NG OFF Threshold", DA9055_DAC_NG_OFF_THRESHOLD, + 0, 0x7, 0), + SOC_ENUM("DAC NG Setup Time", da9055_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da9055_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da9055_dac_ng_rampdown_rate), + + /* DAC Routing control */ + SOC_ENUM("DAC Left Select", da9055_dac_l_select), + SOC_ENUM("DAC Right Select", da9055_dac_r_select), + + /* DAC Invertion control */ + SOC_SINGLE("DAC Left Invert", DA9055_DIG_CTRL, 3, 1, 0), + SOC_SINGLE("DAC Right Invert", DA9055_DIG_CTRL, 7, 1, 0), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Switch", DA9055_MIXIN_L_SELECT, + DA9055_MIXIN_R_SELECT, 7, 1, 0), + + /* To select if MIC Bias powers MIC 1 or MIC 2 */ + SOC_ENUM("Mic Bias Select", da9055_mic_bias_select), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Switch", DA9055_ALC_CTRL1, 3, 7, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE_EXT("ALC Sync Mode Switch", DA9055_ALC_CTRL1, 1, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE("ALC Offset Switch", DA9055_ALC_CTRL1, 0, 1, 0), + SOC_SINGLE("ALC Anticlip Mode Switch", DA9055_ALC_ANTICLIP_CTRL, + 7, 1, 0), + SOC_SINGLE("ALC Anticlip Level", DA9055_ALC_ANTICLIP_LEVEL, + 0, 0x7f, 0), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA9055_ALC_TARGET_MIN, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA9055_ALC_TARGET_MAX, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA9055_ALC_NOISE, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA9055_ALC_GAIN_LIMITS, + 4, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA9055_ALC_GAIN_LIMITS, + 0, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 0, 0x7, 0, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 4, 0x7, 0, alc_analog_gain_tlv), + SOC_ENUM("ALC Attack Rate", da9055_attack_rate), + SOC_ENUM("ALC Release Rate", da9055_release_rate), + SOC_ENUM("ALC Hold Time", da9055_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da9055_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da9055_integ_release_rate), +}; + +/* DAPM Controls */ + +/* Mic PGA Left Source */ +static const struct snd_kcontrol_new da9055_mic_l_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_l_src); + +/* Mic PGA Right Source */ +static const struct snd_kcontrol_new da9055_mic_r_mux_controls = +SOC_DAPM_ENUM("Route", da9055_mic_r_src); + +/* In Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXIN_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_L_SELECT, 2, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXIN_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXIN_R_SELECT, 3, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXOUT_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_L_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA9055_MIXOUT_L_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_L_SELECT, + 6, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXOUT_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA9055_MIXOUT_R_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_R_SELECT, + 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { + /* Input Side */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* MUXs for Mic PGA source selection */ + SND_SOC_DAPM_MUX("Mic Left Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_l_mux_controls), + SND_SOC_DAPM_MUX("Mic Right Source", SND_SOC_NOPM, 0, 0, + &da9055_mic_r_mux_controls), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA9055_MIC_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA9055_MIC_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left", DA9055_AUX_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right", DA9055_AUX_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Left", DA9055_MIXIN_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Right", DA9055_MIXIN_R_CTRL, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", DA9055_MIC_BIAS_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF", DA9055_AIF_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Charge Pump", DA9055_CP_CTRL, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinl_controls[0], + ARRAY_SIZE(da9055_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinr_controls[0], + ARRAY_SIZE(da9055_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA9055_ADC_L_CTRL, 7, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA9055_ADC_R_CTRL, 7, 0), + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA9055_DAC_L_CTRL, 7, 0), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA9055_DAC_R_CTRL, 7, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutl_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutr_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("MIXOUT Left", DA9055_MIXOUT_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXOUT Right", DA9055_MIXOUT_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout", DA9055_LINE_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA9055_HP_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA9055_HP_R_CTRL, 7, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da9055_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"Mic Left Source", "MIC1_P_N", "MIC1"}, + {"Mic Left Source", "MIC1_P", "MIC1"}, + {"Mic Left Source", "MIC1_N", "MIC1"}, + {"Mic Left Source", "MIC2_L", "MIC2"}, + + {"Mic Right Source", "MIC2_R_L", "MIC2"}, + {"Mic Right Source", "MIC2_R", "MIC2"}, + {"Mic Right Source", "MIC2_L", "MIC2"}, + + {"Mic Left", NULL, "Mic Left Source"}, + {"Mic Right", NULL, "Mic Right Source"}, + + {"Aux Left", NULL, "AUXL"}, + {"Aux Right", NULL, "AUXR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux Left Switch", "Aux Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux Right Switch", "Aux Right"}, + {"In Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + + {"MIXIN Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "MIXIN Left"}, + + {"MIXIN Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "MIXIN Right"}, + + {"ADC Left", NULL, "AIF"}, + {"ADC Right", NULL, "AIF"}, + + /* Output path */ + {"DAC Left", NULL, "AIF"}, + {"DAC Right", NULL, "AIF"}, + + {"Out Mixer Left", "Aux Left Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Left", "Aux Left Invert Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux Right Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Right", "Aux Right Invert Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"MIXOUT Left", NULL, "Out Mixer Left"}, + {"Headphone Left", NULL, "MIXOUT Left"}, + {"Headphone Left", NULL, "Charge Pump"}, + {"HPL", NULL, "Headphone Left"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Headphone Right", NULL, "MIXOUT Right"}, + {"Headphone Right", NULL, "Charge Pump"}, + {"HPR", NULL, "Headphone Right"}, + + {"MIXOUT Right", NULL, "Out Mixer Right"}, + {"Lineout", NULL, "MIXOUT Right"}, + {"LINE", NULL, "Lineout"}, +}; + +/* Codec private data */ +struct da9055_priv { + struct regmap *regmap; + unsigned int mclk_rate; + int master; +}; + +static struct reg_default da9055_reg_defaults[] = { + { 0x21, 0x10 }, + { 0x22, 0x0A }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x0C }, + { 0x28, 0x01 }, + { 0x29, 0x08 }, + { 0x2A, 0x32 }, + { 0x2B, 0x00 }, + { 0x30, 0x35 }, + { 0x31, 0x35 }, + { 0x32, 0x00 }, + { 0x33, 0x00 }, + { 0x34, 0x03 }, + { 0x35, 0x03 }, + { 0x36, 0x6F }, + { 0x37, 0x6F }, + { 0x38, 0x80 }, + { 0x39, 0x01 }, + { 0x3A, 0x01 }, + { 0x40, 0x00 }, + { 0x41, 0x88 }, + { 0x42, 0x88 }, + { 0x43, 0x08 }, + { 0x44, 0x80 }, + { 0x45, 0x6F }, + { 0x46, 0x6F }, + { 0x47, 0x61 }, + { 0x48, 0x35 }, + { 0x49, 0x35 }, + { 0x4A, 0x35 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x60, 0x44 }, + { 0x61, 0x44 }, + { 0x62, 0x00 }, + { 0x63, 0x40 }, + { 0x64, 0x40 }, + { 0x65, 0x40 }, + { 0x66, 0x40 }, + { 0x67, 0x40 }, + { 0x68, 0x40 }, + { 0x69, 0x48 }, + { 0x6A, 0x40 }, + { 0x6B, 0x41 }, + { 0x6C, 0x40 }, + { 0x6D, 0x40 }, + { 0x6E, 0x10 }, + { 0x6F, 0x10 }, + { 0x90, 0x80 }, + { 0x92, 0x02 }, + { 0x93, 0x00 }, + { 0x99, 0x00 }, + { 0x9A, 0x00 }, + { 0x9B, 0x00 }, + { 0x9C, 0x3F }, + { 0x9D, 0x00 }, + { 0x9E, 0x3F }, + { 0x9F, 0xFF }, + { 0xA0, 0x71 }, + { 0xA1, 0x00 }, + { 0xA2, 0x00 }, + { 0xA6, 0x00 }, + { 0xA7, 0x00 }, + { 0xAB, 0x00 }, + { 0xAC, 0x00 }, + { 0xAD, 0x00 }, + { 0xAF, 0x08 }, + { 0xB0, 0x00 }, + { 0xB1, 0x00 }, + { 0xB2, 0x00 }, +}; + +static bool da9055_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case DA9055_STATUS1: + case DA9055_PLL_STATUS: + case DA9055_AUX_L_GAIN_STATUS: + case DA9055_AUX_R_GAIN_STATUS: + case DA9055_MIC_L_GAIN_STATUS: + case DA9055_MIC_R_GAIN_STATUS: + case DA9055_MIXIN_L_GAIN_STATUS: + case DA9055_MIXIN_R_GAIN_STATUS: + case DA9055_ADC_L_GAIN_STATUS: + case DA9055_ADC_R_GAIN_STATUS: + case DA9055_DAC_L_GAIN_STATUS: + case DA9055_DAC_R_GAIN_STATUS: + case DA9055_HP_L_GAIN_STATUS: + case DA9055_HP_R_GAIN_STATUS: + case DA9055_LINE_GAIN_STATUS: + case DA9055_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +/* Set DAI word length */ +static int da9055_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_ctrl, fs; + u32 sysclk; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif_ctrl = DA9055_AIF_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif_ctrl = DA9055_AIF_WORD_S20_3LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif_ctrl = DA9055_AIF_WORD_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif_ctrl = DA9055_AIF_WORD_S32_LE; + break; + default: + return -EINVAL; + } + + /* Set AIF format */ + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_WORD_LENGTH_MASK, + aif_ctrl); + + switch (params_rate(params)) { + case 8000: + fs = DA9055_SR_8000; + sysclk = 3072000; + break; + case 11025: + fs = DA9055_SR_11025; + sysclk = 2822400; + break; + case 12000: + fs = DA9055_SR_12000; + sysclk = 3072000; + break; + case 16000: + fs = DA9055_SR_16000; + sysclk = 3072000; + break; + case 22050: + fs = DA9055_SR_22050; + sysclk = 2822400; + break; + case 32000: + fs = DA9055_SR_32000; + sysclk = 3072000; + break; + case 44100: + fs = DA9055_SR_44100; + sysclk = 2822400; + break; + case 48000: + fs = DA9055_SR_48000; + sysclk = 3072000; + break; + case 88200: + fs = DA9055_SR_88200; + sysclk = 2822400; + break; + case 96000: + fs = DA9055_SR_96000; + sysclk = 3072000; + break; + default: + return -EINVAL; + } + + if (da9055->mclk_rate) { + /* PLL Mode, Write actual FS */ + snd_soc_write(codec, DA9055_SR, fs); + } else { + /* + * Non-PLL Mode + * When PLL is bypassed, chip assumes constant MCLK of + * 12.288MHz and uses sample rate value to divide this MCLK + * to derive its sys clk. As sys clk has to be 256 * Fs, we + * need to write constant sample rate i.e. 48KHz. + */ + snd_soc_write(codec, DA9055_SR, DA9055_SR_48000); + } + + if (da9055->mclk_rate && (da9055->mclk_rate != sysclk)) { + /* PLL Mode */ + if (!da9055->master) { + /* PLL slave mode, enable PLL and also SRM */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN | DA9055_PLL_SRM_EN, + DA9055_PLL_EN | DA9055_PLL_SRM_EN); + } else { + /* PLL master mode, only enable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN, DA9055_PLL_EN); + } + } else { + /* Non PLL Mode, disable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + } + + return 0; +} + +/* Set DAI mode and Format */ +static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_clk_mode, aif_ctrl, mode; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* DA9055 in I2S Master Mode */ + mode = 1; + aif_clk_mode = DA9055_AIF_CLK_EN_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* DA9055 in I2S Slave Mode */ + mode = 0; + aif_clk_mode = DA9055_AIF_CLK_EN_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* Don't allow change of mode if PLL is enabled */ + if ((snd_soc_read(codec, DA9055_PLL_CTRL) & DA9055_PLL_EN) && + (da9055->master != mode)) + return -EINVAL; + + da9055->master = mode; + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_ctrl = DA9055_AIF_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_ctrl = DA9055_AIF_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + aif_clk_mode |= DA9055_AIF_BCLKS_PER_WCLK_32; + + snd_soc_update_bits(codec, DA9055_AIF_CLK_MODE, + (DA9055_AIF_CLK_MODE_MASK | DA9055_AIF_BCLK_MASK), + aif_clk_mode); + snd_soc_update_bits(codec, DA9055_AIF_CTRL, DA9055_AIF_FORMAT_MASK, + aif_ctrl); + return 0; +} + +static int da9055_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, DA9055_DAC_R_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, 0); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, 0); + } + + return 0; +} + +#define DA9055_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da9055_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA9055_CLKSRC_MCLK: + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da9055->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* + * da9055_set_dai_pll : Configure the codec PLL + * @param codec_dai : Pointer to codec DAI + * @param pll_id : da9055 has only one pll, so pll_id is always zero + * @param fref : Input MCLK frequency + * @param fout : FsDM value + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 11.2896MHz, 12MHz, 12.288MHz, + * 13MHz, 13.5MHz, 14.4MHz, 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da9055_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + u8 pll_frac_top, pll_frac_bot, pll_integer, cnt; + + /* Disable PLL before setting the divisors */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + + /* In slave mode, there is only one set of divisors */ + if (!da9055->master && (fout != 2822400)) + goto pll_err; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da9055_pll_div); cnt++) { + /* Check fref, mode and fout */ + if ((fref == da9055_pll_div[cnt].fref) && + (da9055->master == da9055_pll_div[cnt].mode) && + (fout == da9055_pll_div[cnt].fout)) { + /* All match, pick up divisors */ + pll_frac_top = da9055_pll_div[cnt].frac_top; + pll_frac_bot = da9055_pll_div[cnt].frac_bot; + pll_integer = da9055_pll_div[cnt].integer; + break; + } + } + if (cnt >= ARRAY_SIZE(da9055_pll_div)) + goto pll_err; + + /* Write PLL dividers */ + snd_soc_write(codec, DA9055_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA9055_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA9055_PLL_INTEGER, pll_integer); + + return 0; +pll_err: + dev_err(codec_dai->dev, "Error in setting up PLL\n"); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da9055_dai_ops = { + .hw_params = da9055_hw_params, + .set_fmt = da9055_set_dai_fmt, + .set_sysclk = da9055_set_dai_sysclk, + .set_pll = da9055_set_dai_pll, + .digital_mute = da9055_mute, +}; + +static struct snd_soc_dai_driver da9055_dai = { + .name = "da9055-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + .ops = &da9055_dai_ops, + .symmetric_rates = 1, +}; + +static int da9055_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Enable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, + DA9055_VMID_EN | DA9055_BIAS_EN); + } + regcache_sync(da9055->regmap); + break; + case SND_SOC_BIAS_OFF: + /* Disable VMID reference & master bias */ + snd_soc_update_bits(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int da9055_probe(struct snd_soc_codec *codec) +{ + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = da9055->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + da9055->mclk_rate = 0; /* This will be set from set_sysclk() */ + da9055->master = 0; /* This will be set from set_fmt() */ + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_AUX_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + + /* + * There are two separate control bits for input and output mixers as + * well as headphone and line outs. + * One to enable corresponding amplifier and other to enable its + * output. As amplifier bits are related to power control, they are + * being managed by DAPM while other (non power related) bits are + * enabled here + */ + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_L_MIX_EN, DA9055_MIXIN_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_MIXIN_R_MIX_EN, DA9055_MIXIN_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_MIX_EN, DA9055_MIXOUT_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); + + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); + + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); + + /* Set this as per your system configuration */ + snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da9055 = { + .probe = da9055_probe, + .set_bias_level = da9055_set_bias_level, + + .controls = da9055_snd_controls, + .num_controls = ARRAY_SIZE(da9055_snd_controls), + + .dapm_widgets = da9055_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da9055_dapm_widgets), + .dapm_routes = da9055_audio_map, + .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), +}; + +static const struct regmap_config da9055_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da9055_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da9055_reg_defaults), + .volatile_reg = da9055_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da9055_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da9055_priv *da9055; + int ret; + + da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + GFP_KERNEL); + if (!da9055) + return -ENOMEM; + + i2c_set_clientdata(i2c, da9055); + + da9055->regmap = devm_regmap_init_i2c(i2c, &da9055_regmap_config); + if (IS_ERR(da9055->regmap)) { + ret = PTR_ERR(da9055->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da9055, &da9055_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + ret); + } + return ret; +} + +static int __devexit da9055_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da9055_i2c_id[] = { + { "da9055", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da9055_i2c_driver = { + .driver = { + .name = "da9055", + .owner = THIS_MODULE, + }, + .probe = da9055_i2c_probe, + .remove = __devexit_p(da9055_remove), + .id_table = da9055_i2c_id, +}; + +module_i2c_driver(da9055_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); +MODULE_AUTHOR("David Chen, Ashish Chavan"); +MODULE_LICENSE("GPL"); -- 1.7.1

2 3

[alsa-devel] Audio codec volume control
by Rajeev kumar 20 Sep '12

20 Sep '12

Hello Mark, We have three volume control in audio codec. 1. Master volume: control both left and right channel volume 2. left volume: control left channel vol 3. right volume: control right channel vol. So we can provide interfaces for all volume control to the user. Please find below the control I can provide to the user space case 1: 1. Provide master volume control 2. Provide Left channel volume control. 3. Provide Right channel volume control. case 2: 1. Provide master volume control 2. Instead of providing separate control for left and right channel, combined it and provide a single control to user space. so that left and right volume is always in sync. Please let me know which case is better and why? Thank you very much for your support. Best Regards Rajeev

2 5

[alsa-devel] [PATCH] ALSA: usb - disable broken hw volume for Tenx TP6911
by David Henningsson 20 Sep '12

20 Sep '12

While going through Ubuntu bugs, I discovered this patch being posted and a confirmation that the patch works as expected. Finding out how the hw volume really works would be preferrable to just disabling the broken one, but this would be better than nothing. Credit: sndfnsdfin (qawsnews) BugLink: https://bugs.launchpad.net/bugs/559939 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/mixer.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f40ba8..fe56c9d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1267,6 +1267,13 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* disable non-functional volume control */ master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); break; + case USB_ID(0x1130, 0xf211): + snd_printk(KERN_INFO + "usbmixer: volume control quirk for Tenx TP6911 Audio Headset\n"); + /* disable non-functional volume control */ + channels = 0; + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize); -- 1.7.9.5

2 1

Re: [alsa-devel] snd_usb changes broke audio
by Daniel Mack 20 Sep '12

20 Sep '12

(Cc the ALSA list - please don't drop it in replies) On 20.09.2012 02:16, Nate Garner wrote: > I recently updated my kernel from 3.5.3 to 3.5.4. Upon rebooting, I no > longer got any audio through my external sound card, but my headphones > worked. I noticed in the kernel change log that you made some changes to > snd-usb, and thought that might be the culprit. I hope you're the right > person to report this too, if not, feel free to ignore. > Relevant info: > Distro: arch linux, x86_64 > sound card: creative labs model SB0490 (soundblaser Live! 24-bit > external (link here > <http://support.creative.com/Products/ProductDetails.aspx?catID=1&subCatID=2…>) > alsa version 1.0.26 > pulseaudio version 2.1 Argh! I believed we finally fixed all the regressions. Are there any messages in the logs or in dmesg that look releated? Can you send the output of "lsusb -v", please? Daniel

1 0

[alsa-devel] Internal speaker output from Sony Vaio Z (2010 model) lost somewhere between 3.1.0 and 3.2.6
by Adam Williamson 20 Sep '12

20 Sep '12

Welp, pretty much as the topic says. I just noticed today that sound is no longer playing from the internal speakers of my laptop, a 2010 model Sony Vaio Z. Plugging in headphones causes sound to be played fine through those. There are three possible choices for 'Connector' - 'Analog Speakers', 'Analog Output', and 'Analog Headphones' - but none of these seems to make sound come out of the internal speakers. The default is 'Analog Speakers', and when it's set to this, headphone output does work when headphones are plugged in. I've verified that it's broken on Fedora 16 kernels 3.2.6-4, 3.2.8-3 and 3.2.10-1. It's also broken in a very recent Fedora 17 kernel, 3.3.0rc6 or so. It works on a Fedora 16 live image, with kernel 3.1.0-1. Unfortunately the kernels before 3.2.6 have been trashed from Fedora's buildsystem archives, I think, so it's hard to narrow things down any further :/ the alsa-info output is http://www.alsa-project.org/db/?f=368e2359757b537d4daa0f5399830d3942540196 . Thanks! -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | identi.ca: adamwfedora http://www.happyassassin.net

3 22

[alsa-devel] [PATCH 00/34] i.MX multi-platform support
by Shawn Guo 20 Sep '12

20 Sep '12

The series enables multi-platform support for imx. Since the required frameworks (clk, pwm) and spare_irq have already been adopted on imx, the series is all about cleaning up mach/* headers. Along with the changes, arch/arm/plat-mxc gets merged into arch/arm/mach-imx. It's based on a bunch of branches (works from others), Rob's initial multi-platform series, Arnd's platform-data and smp_ops (Marc's) and imx 3.7 material (Sascha and myself). It's available on branch below. git://git.linaro.org/people/shawnguo/linux-2.6.git imx/multi-platform It's been tested on imx5 and imx6, and only compile-tested on imx2 and imx3, so testing on imx2/3 are appreciated. Subsystem maintainers, I plan to send the whole series via arm-soc tree at the end of 3.7 merge window when all dependant bits hit mainline. Please have a look at the patches you get copied and provide ACKs if the changes are good. Thanks. Shawn Guo (34): ARM: imx: include board headers in the same folder ASoC: mx27vis: retrieve gpio numbers from platform_data ARM: imx: move iomux drivers and headers into mach-imx ARM: imx: remove unnecessary inclusion from device-imx*.h ARM: imx: move platform device code into mach-imx ARM: imx: merge plat-mxc into mach-imx ARM: imx: include common.h rather than mach/common.h ARM: imx: ARM: imx: include cpuidle.h rather than mach/cpuidle.h ARM: imx: include iim.h rather than mach/iim.h ARM: imx: include iram.h rather than mach/iram.h ARM: imx: include ulpi.h rather than mach/ulpi.h media: mx1_camera: remove the driver ARM: imx: remove mach/dma-mx1-mx2.h dma: ipu: rename mach/ipu.h to include/linux/dma/ipu-dma.h dma: imx-sdma: remove unneeded mach/hardware.h inclusion ASoC: imx-ssi: remove unneeded mach/hardware.h inclusion usb: ehci-mxc: remove unneeded mach/hardware.h inclusion video: mx3fb: remove unneeded mach/hardware.h inclusion watchdog: imx2_wdt: remove unneeded mach/hardware.h inclusion i2c: imx: remove mach/hardware.h inclusion mtd: mxc_nand: remove mach/hardware.h inclusion rtc: mxc_rtc: remove mach/hardware.h inclusion dma: imx-dma: use devm_kzalloc and devm_request_irq dma: imx-dma: retrieve MEM and IRQ from resources dma: imx-dma: remove mach/hardware.h inclusion media: mx2_camera: remove dead code in mx2_camera_add_device media: mx2_camera: use managed functions to clean up code media: mx2_camera: remove mach/hardware.h inclusion mmc: mxcmmc: remove mach/hardware.h inclusion video: imxfb: remove mach/hardware.h inclusion ARM: imx: move debug macros to include/debug ARM: imx: include hardware.h rather than mach/hardware.h ARM: imx: remove header file mach/irqs.h ARM: imx: enable multi-platform build .../devicetree/bindings/i2c/fsl-imx-i2c.txt | 4 +- arch/arm/Kconfig | 15 +- arch/arm/Kconfig.debug | 8 + arch/arm/Makefile | 1 - arch/arm/boot/dts/imx27.dtsi | 4 +- arch/arm/boot/dts/imx51.dtsi | 4 +- arch/arm/boot/dts/imx53.dtsi | 6 +- arch/arm/boot/dts/imx6q.dtsi | 6 +- .../mach/debug-macro.S => include/debug/imx.S} | 33 +- arch/arm/{plat-mxc => mach-imx}/3ds_debugboard.c | 2 +- .../include/mach => mach-imx}/3ds_debugboard.h | 0 arch/arm/mach-imx/Kconfig | 86 ++ arch/arm/mach-imx/Makefile | 23 +- arch/arm/{plat-mxc => mach-imx}/avic.c | 5 +- .../include/mach => mach-imx}/board-mx31lilly.h | 0 .../include/mach => mach-imx}/board-mx31lite.h | 0 .../include/mach => mach-imx}/board-mx31moboard.h | 0 .../include/mach => mach-imx}/board-pcm038.h | 0 arch/arm/mach-imx/clk-imx1.c | 15 +- arch/arm/mach-imx/clk-imx21.c | 14 +- arch/arm/mach-imx/clk-imx25.c | 26 +- arch/arm/mach-imx/clk-imx27.c | 36 +- arch/arm/mach-imx/clk-imx31.c | 21 +- arch/arm/mach-imx/clk-imx35.c | 13 +- arch/arm/mach-imx/clk-imx51-imx53.c | 15 +- arch/arm/mach-imx/clk-imx6q.c | 3 +- arch/arm/mach-imx/clk-pllv1.c | 4 +- .../{plat-mxc/include/mach => mach-imx}/common.h | 0 arch/arm/mach-imx/cpu-imx25.c | 5 +- arch/arm/mach-imx/cpu-imx27.c | 2 +- arch/arm/mach-imx/cpu-imx31.c | 7 +- arch/arm/mach-imx/cpu-imx35.c | 5 +- arch/arm/mach-imx/cpu-imx5.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpu.c | 3 +- arch/arm/mach-imx/cpu_op-mx51.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpufreq.c | 3 +- arch/arm/{plat-mxc => mach-imx}/cpuidle.c | 0 .../{plat-mxc/include/mach => mach-imx}/cpuidle.h | 0 arch/arm/mach-imx/devices-imx1.h | 3 +- arch/arm/mach-imx/devices-imx21.h | 3 +- arch/arm/mach-imx/devices-imx25.h | 3 +- arch/arm/mach-imx/devices-imx27.h | 3 +- arch/arm/mach-imx/devices-imx31.h | 3 +- arch/arm/mach-imx/devices-imx35.h | 3 +- arch/arm/mach-imx/devices-imx50.h | 3 +- arch/arm/mach-imx/devices-imx51.h | 3 +- arch/arm/{plat-mxc => mach-imx}/devices/Kconfig | 3 - arch/arm/{plat-mxc => mach-imx}/devices/Makefile | 3 +- .../mach => mach-imx/devices}/devices-common.h | 19 +- arch/arm/{plat-mxc => mach-imx/devices}/devices.c | 1 - .../devices/platform-ahci-imx.c | 5 +- .../{plat-mxc => mach-imx}/devices/platform-fec.c | 5 +- .../devices/platform-flexcan.c | 4 +- .../devices/platform-fsl-usb2-udc.c | 5 +- .../devices/platform-gpio-mxc.c | 2 +- .../devices/platform-gpio_keys.c | 5 +- .../devices/platform-imx-dma.c | 23 +- .../devices/platform-imx-fb.c | 16 +- .../devices/platform-imx-i2c.c | 31 +- .../devices/platform-imx-keypad.c | 4 +- .../devices/platform-imx-ssi.c | 4 +- .../devices/platform-imx-uart.c | 4 +- .../devices/platform-imx2-wdt.c | 5 +- .../devices/platform-imx21-hcd.c | 4 +- .../devices/platform-imx_udc.c | 4 +- .../devices/platform-imxdi_rtc.c | 5 +- .../devices/platform-ipu-core.c | 5 +- .../devices/platform-mx2-camera.c | 16 +- .../devices/platform-mxc-ehci.c | 5 +- .../devices/platform-mxc-mmc.c | 20 +- .../devices/platform-mxc_nand.c | 25 +- .../devices/platform-mxc_pwm.c | 4 +- .../devices/platform-mxc_rnga.c | 4 +- .../devices/platform-mxc_rtc.c | 13 +- .../devices/platform-mxc_w1.c | 4 +- .../devices/platform-pata_imx.c | 4 +- .../devices/platform-sdhci-esdhc-imx.c | 5 +- .../devices/platform-spi_imx.c | 4 +- arch/arm/mach-imx/ehci-imx25.c | 4 +- arch/arm/mach-imx/ehci-imx27.c | 4 +- arch/arm/mach-imx/ehci-imx31.c | 4 +- arch/arm/mach-imx/ehci-imx35.c | 4 +- arch/arm/mach-imx/ehci-imx5.c | 4 +- arch/arm/{plat-mxc => mach-imx}/epit.c | 6 +- .../include/mach => mach-imx}/eukrea-baseboards.h | 0 arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 7 +- arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 8 +- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 7 +- arch/arm/mach-imx/eukrea_mbimxsd51-baseboard.c | 7 +- .../{plat-mxc/include/mach => mach-imx}/hardware.h | 26 +- arch/arm/mach-imx/hotplug.c | 3 +- arch/arm/{plat-mxc/include/mach => mach-imx}/iim.h | 0 arch/arm/mach-imx/imx27-dt.c | 11 +- arch/arm/mach-imx/imx31-dt.c | 5 +- arch/arm/mach-imx/imx51-dt.c | 9 +- arch/arm/mach-imx/include/mach/dma-mx1-mx2.h | 10 - arch/arm/mach-imx/iomux-imx31.c | 5 +- .../include/mach => mach-imx}/iomux-mx1.h | 2 +- .../include/mach => mach-imx}/iomux-mx21.h | 4 +- .../include/mach => mach-imx}/iomux-mx25.h | 2 +- .../include/mach => mach-imx}/iomux-mx27.h | 4 +- .../include/mach => mach-imx}/iomux-mx2x.h | 0 .../include/mach => mach-imx}/iomux-mx3.h | 0 .../include/mach => mach-imx}/iomux-mx35.h | 2 +- .../include/mach => mach-imx}/iomux-mx50.h | 2 +- .../include/mach => mach-imx}/iomux-mx51.h | 2 +- arch/arm/{plat-mxc => mach-imx}/iomux-v1.c | 5 +- .../{plat-mxc/include/mach => mach-imx}/iomux-v1.h | 0 arch/arm/{plat-mxc => mach-imx}/iomux-v3.c | 5 +- .../{plat-mxc/include/mach => mach-imx}/iomux-v3.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/iram.h | 0 arch/arm/{plat-mxc => mach-imx}/iram_alloc.c | 3 +- arch/arm/{plat-mxc => mach-imx}/irq-common.c | 0 arch/arm/{plat-mxc => mach-imx}/irq-common.h | 3 + arch/arm/mach-imx/lluart.c | 3 +- arch/arm/mach-imx/mach-apf9328.c | 7 +- arch/arm/mach-imx/mach-armadillo5x0.c | 9 +- arch/arm/mach-imx/mach-bug.c | 7 +- arch/arm/mach-imx/mach-cpuimx27.c | 11 +- arch/arm/mach-imx/mach-cpuimx35.c | 9 +- arch/arm/mach-imx/mach-cpuimx51sd.c | 9 +- arch/arm/mach-imx/mach-eukrea_cpuimx25.c | 10 +- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 48 +- arch/arm/mach-imx/mach-imx27ipcam.c | 6 +- arch/arm/mach-imx/mach-imx27lite.c | 6 +- arch/arm/mach-imx/mach-imx53.c | 11 +- arch/arm/mach-imx/mach-imx6q.c | 6 +- arch/arm/mach-imx/mach-kzm_arm11_01.c | 7 +- arch/arm/mach-imx/mach-mx1ads.c | 7 +- arch/arm/mach-imx/mach-mx21ads.c | 6 +- arch/arm/mach-imx/mach-mx25_3ds.c | 8 +- arch/arm/mach-imx/mach-mx27_3ds.c | 10 +- arch/arm/mach-imx/mach-mx27ads.c | 6 +- arch/arm/mach-imx/mach-mx31_3ds.c | 12 +- arch/arm/mach-imx/mach-mx31ads.c | 5 +- arch/arm/mach-imx/mach-mx31lilly.c | 11 +- arch/arm/mach-imx/mach-mx31lite.c | 11 +- arch/arm/mach-imx/mach-mx31moboard.c | 14 +- arch/arm/mach-imx/mach-mx35_3ds.c | 8 +- arch/arm/mach-imx/mach-mx50_rdp.c | 7 +- arch/arm/mach-imx/mach-mx51_3ds.c | 9 +- arch/arm/mach-imx/mach-mx51_babbage.c | 7 +- arch/arm/mach-imx/mach-mxt_td60.c | 6 +- arch/arm/mach-imx/mach-pca100.c | 8 +- arch/arm/mach-imx/mach-pcm037.c | 8 +- arch/arm/mach-imx/mach-pcm037_eet.c | 5 +- arch/arm/mach-imx/mach-pcm038.c | 13 +- arch/arm/mach-imx/mach-pcm043.c | 9 +- arch/arm/mach-imx/mach-qong.c | 6 +- arch/arm/mach-imx/mach-scb9328.c | 7 +- arch/arm/mach-imx/mach-vpr200.c | 7 +- arch/arm/mach-imx/mm-imx1.c | 9 +- arch/arm/mach-imx/mm-imx21.c | 12 +- arch/arm/mach-imx/mm-imx25.c | 10 +- arch/arm/mach-imx/mm-imx27.c | 12 +- arch/arm/mach-imx/mm-imx3.c | 9 +- arch/arm/mach-imx/mm-imx5.c | 8 +- arch/arm/mach-imx/mx1-camera-fiq-ksym.c | 18 - arch/arm/mach-imx/mx1-camera-fiq.S | 35 - arch/arm/{plat-mxc/include/mach => mach-imx}/mx1.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx21.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx25.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx27.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx2x.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx31.h | 0 arch/arm/mach-imx/mx31lilly-db.c | 9 +- arch/arm/mach-imx/mx31lite-db.c | 9 +- arch/arm/mach-imx/mx31moboard-devboard.c | 9 +- arch/arm/mach-imx/mx31moboard-marxbot.c | 9 +- arch/arm/mach-imx/mx31moboard-smartbot.c | 11 +- .../arm/{plat-mxc/include/mach => mach-imx}/mx35.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx3x.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx50.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx51.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx53.h | 0 .../arm/{plat-mxc/include/mach => mach-imx}/mx6q.h | 0 arch/arm/{plat-mxc/include/mach => mach-imx}/mxc.h | 0 arch/arm/mach-imx/pcm970-baseboard.c | 7 +- arch/arm/mach-imx/platsmp.c | 5 +- arch/arm/mach-imx/pm-imx27.c | 3 +- arch/arm/mach-imx/pm-imx3.c | 7 +- arch/arm/mach-imx/pm-imx5.c | 7 +- arch/arm/mach-imx/pm-imx6q.c | 5 +- arch/arm/{plat-mxc => mach-imx}/ssi-fiq-ksym.c | 0 arch/arm/{plat-mxc => mach-imx}/ssi-fiq.S | 0 arch/arm/{plat-mxc => mach-imx}/system.c | 5 +- arch/arm/{plat-mxc => mach-imx}/time.c | 5 +- arch/arm/{plat-mxc => mach-imx}/tzic.c | 6 +- arch/arm/{plat-mxc => mach-imx}/ulpi.c | 2 +- .../arm/{plat-mxc/include/mach => mach-imx}/ulpi.h | 0 arch/arm/plat-mxc/Kconfig | 89 -- arch/arm/plat-mxc/Makefile | 24 - arch/arm/plat-mxc/devices/platform-mx1-camera.c | 42 - arch/arm/plat-mxc/include/mach/irqs.h | 21 - arch/arm/plat-mxc/include/mach/timex.h | 22 - arch/arm/plat-mxc/include/mach/uncompress.h | 132 --- drivers/dma/imx-dma.c | 137 +-- drivers/dma/imx-sdma.c | 1 - drivers/dma/ipu/ipu_idmac.c | 3 +- drivers/dma/ipu/ipu_irq.c | 3 +- drivers/i2c/busses/i2c-imx.c | 40 +- drivers/media/video/Kconfig | 12 - drivers/media/video/Makefile | 1 - drivers/media/video/mx1_camera.c | 889 -------------------- drivers/media/video/mx2_camera.c | 240 +++--- drivers/media/video/mx3_camera.c | 2 +- drivers/mmc/host/mxcmmc.c | 31 +- drivers/mtd/nand/mxc_nand.c | 86 +- drivers/rtc/rtc-mxc.c | 34 +- drivers/usb/host/ehci-mxc.c | 1 - drivers/video/imxfb.c | 38 +- drivers/video/mx3fb.c | 3 +- drivers/watchdog/imx2_wdt.c | 1 - .../mach/ipu.h => include/linux/dma/ipu-dma.h | 6 +- include/linux/platform_data/asoc-mx27vis.h | 11 + include/linux/platform_data/camera-mx1.h | 35 - sound/soc/fsl/imx-pcm-fiq.c | 3 +- sound/soc/fsl/imx-ssi.c | 1 - sound/soc/fsl/mx27vis-aic32x4.c | 42 +- 219 files changed, 1149 insertions(+), 2170 deletions(-) rename arch/arm/{plat-mxc/include/mach/debug-macro.S => include/debug/imx.S} (59%) rename arch/arm/{plat-mxc => mach-imx}/3ds_debugboard.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/3ds_debugboard.h (100%) rename arch/arm/{plat-mxc => mach-imx}/avic.c (98%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31lilly.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31lite.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-mx31moboard.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/board-pcm038.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/common.h (100%) rename arch/arm/{plat-mxc => mach-imx}/cpu.c (97%) rename arch/arm/{plat-mxc => mach-imx}/cpufreq.c (99%) rename arch/arm/{plat-mxc => mach-imx}/cpuidle.c (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/cpuidle.h (100%) rename arch/arm/{plat-mxc => mach-imx}/devices/Kconfig (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/Makefile (96%) rename arch/arm/{plat-mxc/include/mach => mach-imx/devices}/devices-common.h (96%) rename arch/arm/{plat-mxc => mach-imx/devices}/devices.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-ahci-imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-fec.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-flexcan.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-fsl-usb2-udc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-gpio-mxc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-gpio_keys.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-dma.c (63%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-fb.c (79%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-i2c.c (77%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-keypad.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-ssi.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx-uart.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx2-wdt.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx21-hcd.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imx_udc.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-imxdi_rtc.c (94%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-ipu-core.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mx2-camera.c (83%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc-ehci.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc-mmc.c (76%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_nand.c (74%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_pwm.c (97%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_rnga.c (95%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_rtc.c (77%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-mxc_w1.c (95%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-pata_imx.c (96%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-sdhci-esdhc-imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/devices/platform-spi_imx.c (98%) rename arch/arm/{plat-mxc => mach-imx}/epit.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/eukrea-baseboards.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/hardware.h (94%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iim.h (100%) delete mode 100644 arch/arm/mach-imx/include/mach/dma-mx1-mx2.h rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx1.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx21.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx25.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx27.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx2x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx3.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx35.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx50.h (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-mx51.h (99%) rename arch/arm/{plat-mxc => mach-imx}/iomux-v1.c (98%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-v1.h (100%) rename arch/arm/{plat-mxc => mach-imx}/iomux-v3.c (97%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iomux-v3.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/iram.h (100%) rename arch/arm/{plat-mxc => mach-imx}/iram_alloc.c (98%) rename arch/arm/{plat-mxc => mach-imx}/irq-common.c (100%) rename arch/arm/{plat-mxc => mach-imx}/irq-common.h (94%) delete mode 100644 arch/arm/mach-imx/mx1-camera-fiq-ksym.c delete mode 100644 arch/arm/mach-imx/mx1-camera-fiq.S rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx1.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx21.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx25.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx27.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx2x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx31.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx35.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx3x.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx50.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx51.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx53.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mx6q.h (100%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/mxc.h (100%) rename arch/arm/{plat-mxc => mach-imx}/ssi-fiq-ksym.c (100%) rename arch/arm/{plat-mxc => mach-imx}/ssi-fiq.S (100%) rename arch/arm/{plat-mxc => mach-imx}/system.c (97%) rename arch/arm/{plat-mxc => mach-imx}/time.c (99%) rename arch/arm/{plat-mxc => mach-imx}/tzic.c (98%) rename arch/arm/{plat-mxc => mach-imx}/ulpi.c (99%) rename arch/arm/{plat-mxc/include/mach => mach-imx}/ulpi.h (100%) delete mode 100644 arch/arm/plat-mxc/Kconfig delete mode 100644 arch/arm/plat-mxc/Makefile delete mode 100644 arch/arm/plat-mxc/devices/platform-mx1-camera.c delete mode 100644 arch/arm/plat-mxc/include/mach/irqs.h delete mode 100644 arch/arm/plat-mxc/include/mach/timex.h delete mode 100644 arch/arm/plat-mxc/include/mach/uncompress.h delete mode 100644 drivers/media/video/mx1_camera.c rename arch/arm/plat-mxc/include/mach/ipu.h => include/linux/dma/ipu-dma.h (97%) create mode 100644 include/linux/platform_data/asoc-mx27vis.h delete mode 100644 include/linux/platform_data/camera-mx1.h -- 1.7.9.5

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[alsa-devel] [PATCH v2 1/7] ASoC: fsl: add PPC_MPC52xx dependency to SND_POWERPC_SOC
by "Eric Millbrandt 20 Sep '12

20 Sep '12

From: Eric Millbrandt <emillbrandt(a)dekaresearch.com> mpc52xx socs need to define SND_POWERPC_SOC to build ASoC drivers. Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- Changes since v1: Patch was "powerpc/52xx: define FSL_SOC" Changed from defining FSL_SOC for PPC_MPC52xx to adding PPC_MPC52xx as a dependency of SND_POWERPC_SOC as per Anatolij Gustschin's comment. sound/soc/fsl/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d701330..4563b28 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS menuconfig SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" - depends on FSL_SOC + depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. -- 1.7.2.5

3 3

[alsa-devel] [RFC PATCH v2 0/5] snd-ice1712: Add Philips PSC724 Ultimate Edge
by Ondrej Zary 19 Sep '12

19 Sep '12

Hello, this is second attempt to add Philips PSC724 Ultimate Edge sound card support to snd-ice1712 driver. Unlike other snd-ice1712 subdrivers, the codec code is splitted into separate modules and thus reusable by other ice1712 subdrivers. Working: all analog outputs (front, rear, center+lfe, headphone) and inputs (front mic, rear mic, line, cd, aux), mixer, headphone jack detection, suspend/resume Currently untested: SPDIF Buggy: switching AGC mode in mixer causes alsamixer to crash - looks like it does not like notifications about control activating/deactivating

2 9

[alsa-devel] [RFC 0/7] mpc5200 ASoC and pcm030 board fixes
by Eric Millbrandt 19 Sep '12

19 Sep '12

This series is a respin of "mpc5200 ASoC fixups" The mpc5200 ASoC and pcm030 board code compiled, but did not run after the multi-codec patches. This series add the missing pieces into the mpc5200 ASoC drivers and updates the pcm030 board to the current api. Eric Millbrandt (7): powerpc/52xx: define FSL_SOC ASoC: fsl: mpc5200 combine psc_dma platform data ASoC: fsl: mpc5200 add missing information to snd_soc_dai_driver ASoC: fsl: cleanup headers in pcm030-audio-fabric ASoC: fsl: convert pcm030-audio-fabric to a platform-driver ASoC: fsl: convert pcm030-audio-fabric to use snd_soc_register_card ASoC: fsl: register the wm9712-codec arch/powerpc/platforms/52xx/Kconfig | 1 + sound/soc/fsl/mpc5200_dma.c | 24 ++------- sound/soc/fsl/mpc5200_dma.h | 3 + sound/soc/fsl/mpc5200_psc_ac97.c | 10 ++++ sound/soc/fsl/mpc5200_psc_i2s.c | 8 +++ sound/soc/fsl/pcm030-audio-fabric.c | 100 +++++++++++++++++++++++++--------- 6 files changed, 99 insertions(+), 47 deletions(-) -- 1.7.2.5

2 13

[alsa-devel] [PATCH 1/5] [v3] ASoC: fsl: use snd_soc_register_card to register the card
by Timur Tabi 19 Sep '12

19 Sep '12

Use snd_soc_register_card() instead of platform_device_alloc("soc-audio") to register the sound card from the machine drivers. The use of platform_device_alloc is deprecated. Although several other drivers still use platform_device_alloc(), the Freescale drivers were not using it to pass driver data. Instead of fixing the driver data usage, it's better to replace the deprecated code. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 32 +++++++------------------------- sound/soc/fsl/p1022_ds.c | 31 ++++++------------------------- 2 files changed, 13 insertions(+), 50 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 60bcba1..9ff9318 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -192,7 +192,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; int ret = -ENODEV; const char *sprop; @@ -341,34 +340,22 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; machine_data->card.name = pdev->name; /* The platform driver name */ + machine_data->card.owner = THIS_MODULE; + machine_data->card.dev = &pdev->dev; machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&machine_data->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; + dev_err(&pdev->dev, "could not register card\n"); + goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; -error_sound: - platform_device_put(sound_device); error: kfree(machine_data); error_alloc: @@ -383,17 +370,12 @@ error_alloc: */ static int __devexit mpc8610_hpcd_remove(struct platform_device *pdev) { - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - platform_device_unregister(sound_device); - + snd_soc_unregister_card(card); kfree(machine_data); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); return 0; } diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 50adf40..144d496 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -202,7 +202,6 @@ static int p1022_ds_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct machine_data *mdata; int ret = -ENODEV; const char *sprop; @@ -349,36 +348,23 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->card.probe = p1022_ds_machine_probe; mdata->card.remove = p1022_ds_machine_remove; mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; mdata->card.num_links = 2; mdata->card.dai_link = mdata->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&mdata->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; error: - if (sound_device) - platform_device_put(sound_device); - kfree(mdata); error_put: of_node_put(codec_np); @@ -392,17 +378,12 @@ error_put: */ static int __devexit p1022_ds_remove(struct platform_device *pdev) { - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); struct machine_data *mdata = container_of(card, struct machine_data, card); - platform_device_unregister(sound_device); - + snd_soc_unregister_card(card); kfree(mdata); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- 1.7.3.4

3 14

Re: [alsa-devel] [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe rather than as a device
by Lee Jones 19 Sep '12

19 Sep '12

Mark, > > > I'm sorry but this patch is breaking the design of ASoC. > > > The ASoC-platform is the DMA-block (in combination with the > > > MSP-block), and there should be a platform-driver for the DMA/PCM. > > > The > > > platform-driver then has a DAI which is the MSP. The ASoC > > > DAI-link-struct should have one driver for each of these, so the > > > dummy-driver for PCM should be there. > > > > Just thinking about this now. I converted it to the current format at > > the request of Mark. If this isn't the correct method I'm not quite > > sure what is. If you want it to be registered as a device, then it > > needs to go into the Device Tree, but Mark doesn't want it in there > > because it doesn't actually represent hardware. I've just taken a closer look at this with a view to finding the most suitable solution. My conclusion is that although the PCM doesn't contain any registers, or represent hardware it should be a device and therefore be present in the Device Tree. These are my findings: PCM devices already represented in DTs: fsl,mpc5200-pcm - written by Grant Likely, the author of Device Tree phytec,pcm030 - written by Grant Likely, the author of Device Tree PCM devices which register as actual devices (should be in DT): samsung - samsung-pcm sh - siu-pcm-audio omap - omap-pcm-audio pxa - pxa-pcm-audio jz4740 - jz4740-pcm-audio kirkwood - kirkwood-pcm-audio ep93xx - ep93xx-pcm-audio ... The later was basically every PCM device bar one I think. The Open Firmware used to stipulate that each device represented in the Device Tree to own registers and therefore be an actual hardware device, but that has since been lifted as it didn't make sense. I propose to represent the PCM in the Device Tree again and have it probe just like all the other PCM devices in sound/soc. If you give me the nod, I'll revert this patch, enable the PCM for DT again and resent the patch-set in full. -- Lee Jones Linaro ST-Ericsson Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

2 8

[alsa-devel] [Resend PATCH] ASoC: wm8904: remove redundant code
by Bo Shen 19 Sep '12

19 Sep '12

The core_intercon register two times for wm8904 Remove one Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index dc4262e..7c8df52 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1185,8 +1185,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, core_intercon, - ARRAY_SIZE(core_intercon)); snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); snd_soc_dapm_add_routes(dapm, dac_intercon, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - avoid non-standard "Docking" name in mixers
by David Henningsson 19 Sep '12

19 Sep '12

The standard name (and what PulseAudio picks up) is "Dock Mic", not "Docking Mic" or "Docking-Station". Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_analog.c | 14 +++++++------- sound/pci/hda/patch_conexant.c | 4 ++-- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a82cce..cdd43ea 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1810,7 +1810,7 @@ static const struct hda_input_mux ad1981_hp_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, - { "Docking-Station", 0x1 }, + { "Dock Mic", 0x1 }, { "Mix", 0x2 }, }, }; @@ -1834,8 +1834,8 @@ static const struct snd_kcontrol_new ad1981_hp_mixers[] = { */ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), /* FIXME: does this laptop have analog CD connection? */ @@ -3565,7 +3565,7 @@ static const struct hda_amp_list ad1884_loopbacks[] = { static const char * const ad1884_slave_vols[] = { "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958", + "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958", NULL }; @@ -3626,7 +3626,7 @@ static const struct hda_input_mux ad1984_thinkpad_capture_source = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, { "Mix", 0x3 }, - { "Docking-Station", 0x4 }, + { "Dock Mic", 0x4 }, }, }; @@ -3655,8 +3655,8 @@ static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7d70210..b871b013 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1710,8 +1710,8 @@ static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH v4 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 19 Sep '12

19 Sep '12

Hi Samuel, On 09/19/2012 03:24 AM, Samuel Ortiz wrote: > Hi Peter, > > On Mon, Sep 10, 2012 at 01:46:18PM +0300, Peter Ujfalusi wrote: >> Hello, >> >> Generated on top of: >> git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap > I applied the first 8 patches, but I'd like to get Mark's ACK for the rest of > the serie. > Unless you're expecting Mark to take the whole thing, in which case you can > add my: > > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> > > to the MFD patches. > Please let me know how you'd like to proceed. If Mark is still OK with it I think it would be better to queue this series via ASoC so we can keep them in one place (also I think patch 14 needs patch 6 in order to not break x86/x86_64 allmodconfig). On the other hand if they go separately (1-8 via MFD, 9-14 via ASoC) they will meet in linux-next where this matters the most. Thank you, Péter

1 0

[alsa-devel] Need help with WM8960
by Timur Tabi 19 Sep '12

19 Sep '12

Mark, I'm trying to port some old Linux code to upstream for a third-party board that uses a Freescale P1022 and a Wolfson WM8960. I've got audio working, but it's using a hack from the third-party code, and I don't know how to do implement the change properly, mostly because I don't understand DAPM. The first problem is that I get this: wm8960 0-001a: No platform data supplied What kind of platform data is the driver expecting, and how do I get it that data? I don't seem to need any platform data to make the code work. Second, I uploaded the hack to wm8960.c at http://pastebin.com/RwVy6cQg You can see that he disables the DAPM controls for DACL, DACR, ADCL, and ADCR, and forces those bits set in the wm8960_probe function. There's also a comment that documents some limitations. What is the proper way to implement this change? Third, in order to actually hear audio, I had to unmute the "Left Output Mixer PCM" and "Right Output Mixer PCM". Is this something that the machine driver should do automatically? And if so, how? -- Timur Tabi Linux kernel developer at Freescale

3 7

[alsa-devel] [PATCH] ASoC: imx-audmux: remove null check of audmux_base in audmux_read_file
by Richard Zhao 19 Sep '12

19 Sep '12

When audmux_read_file is called, it means the driver is already initialised successfully, so we don't need to check audmux_base. It also fix smatch warning: sound/soc/fsl/imx-audmux.c:78 audmux_read_file() warn: possible memory leak of 'buf' Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e7c800e..524ce62 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -74,9 +74,6 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (!audmux_base) - return -ENOSYS; - if (audmux_clk) clk_prepare_enable(audmux_clk); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: imx-pcm-dma: check kzalloc return value in function snd_imx_open
by Richard Zhao 19 Sep '12

19 Sep '12

It fixed smatch warning: sound/soc/fsl/imx-pcm-dma.c:112 snd_imx_open() error: potential null dereference 'dma_data'. (kzalloc returns null) Signed-off-by: Richard Zhao <richard.zhao(a)freescale.com> --- sound/soc/fsl/imx-pcm-dma.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 48f9d88..a23505a 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,6 +109,9 @@ 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); + if (!dma_data) + return -ENOMEM; + dma_data->peripheral_type = dma_params->shared_peripheral ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: Avoid recalculating the bitmask for SOC_ENUM controls
by Lars-Peter Clausen 19 Sep '12

19 Sep '12

For ENUM controls the bitmask is calculated based on the number of items. Currently this is done each time the control is accessed. And while the performance impact of this should be negligible we can easily do better. The roundup_pow_of_two macro performs the same calculation which is currently done manually, but it is also possible to use this macro with compile time constants and so it can be used to initialize static data. So we can use it to initialize the mask field of a ENUM control during its declaration. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 +++- sound/soc/codecs/twl4030.c | 8 +++----- sound/soc/soc-core.c | 16 ++++++---------- sound/soc/soc-dapm.c | 22 ++++++++-------------- 4 files changed, 20 insertions(+), 30 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 313b766..91244a0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -20,6 +20,7 @@ #include <linux/interrupt.h> #include <linux/kernel.h> #include <linux/regmap.h> +#include <linux/log2.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/compress_driver.h> @@ -160,7 +161,8 @@ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .max = xmax, .texts = xtexts } + .max = xmax, .texts = xtexts, \ + .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0} #define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts) #define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc..2548f5c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -999,7 +999,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; - unsigned short mask, bitmask; + unsigned short mask; if (twl4030->configured) { dev_err(codec->dev, @@ -1007,18 +1007,16 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits(codec, e->reg, mask, val); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5b0713..9a6daf9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2413,16 +2413,14 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] - = (val >> e->shift_l) & (bitmask - 1); + = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2443,19 +2441,17 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; - unsigned int mask, bitmask; + unsigned int mask; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits_locked(codec, e->reg, mask, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3b8ddb4..873e6e7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -358,12 +358,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrol_news[i].private_value; - int val, item, bitmask; + int val, item; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = soc_widget_read(w, e->reg); - item = (val >> e->shift_l) & (bitmask - 1); + item = (val >> e->shift_l) & e->mask; p->connect = 0; for (i = 0; i < e->max; i++) { @@ -2680,15 +2678,13 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(widget->codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2712,22 +2708,20 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; - unsigned int mask, bitmask; + unsigned int mask; struct snd_soc_dapm_update update; int wi; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); -- 1.7.10.4

3 2

[alsa-devel] [PATCH v2 0/3] Allow suspend/resume with USB audio.
by Dylan Reid 18 Sep '12

18 Sep '12

Problem being addressed: shell one$ aplay -Dhw:<usbcard> file.wav shell two$ echo mem > /sys/power/state After resume, aplay attempts resume, then restart, then after a timeout gets a write error and no audio will be played until it is restarted. During suspend/resume, if the USB port remains powered the audio device will still be present upon resume. The interface and endpoint have to be reconfigured however. These patches move the configuration from hw_params to prepare causing re-configuration during resume. Is this the right way to fix this? Or is there an easier way I am missing? Thanks for looking. v2: fix unused variable warnings. Dylan Reid (3): ALSA: usb-audio: set period_bytes in substream. ALSA: usb-audio: Don't require hw_params in endpoint. ALSA: usb-audio: Move configuration to prepare. sound/usb/card.h | 2 + sound/usb/endpoint.c | 29 +++++++----- sound/usb/endpoint.h | 5 ++- sound/usb/pcm.c | 121 +++++++++++++++++++++++++++++++-------------------- 4 files changed, 97 insertions(+), 60 deletions(-) -- 1.7.12.146.g16d26b1

2 5

[alsa-devel] [PATCH 0/3] Allow suspend/resume with USB audio.
by Dylan Reid 18 Sep '12

18 Sep '12

Problem being addressed: shell one$ aplay -Dhw:<usbcard> file.wav shell two$ echo mem > /sys/power/state After resume, aplay attempts resume, then restart, then after a timeout gets a write error and no audio will be played until it is restarted. During suspend/resume, if the USB port remains powered the audio device will still be present upon resume. The interface and endpoint have to be reconfigured however. These patches move the configuration from hw_params to prepare causing re-configuration during resume. Is this the right way to fix this? Or is there an easier way I am missing? Thanks for looking. Dylan Reid (3): ALSA: usb-audio: set period_bytes in substream. ALSA: usb-audio: Don't require hw_params in endpoint. ALSA: usb-audio: Move configuration to prepare. sound/usb/card.h | 2 + sound/usb/endpoint.c | 29 +++++++----- sound/usb/endpoint.h | 5 ++- sound/usb/pcm.c | 121 +++++++++++++++++++++++++++++++-------------------- 4 files changed, 97 insertions(+), 60 deletions(-) -- 1.7.12.146.g16d26b1

2 8

[alsa-devel] what is wrong?
by Eva Lovi 18 Sep '12

18 Sep '12

Hello, Does Alsa support this soundcard? card 0: I82801AAICH [Intel 82801AA-ICH], device 0: Intel ICH [Intel 82801AA-ICH] I tried aplay, but I don't hear anything. In amixer I increased the volume, but I still don't hear anything. What's the problem? Eva Lovi

1 1

[alsa-devel] [PATCH] ALSA: hda - limit internal mic boost for Asus X202E
by David Henningsson 18 Sep '12

18 Sep '12

When the input gain for the internal mic is set to its maximum level, the background noise becomes so high - and any relevant signal clipped - that the setting becomes unusable. It is better to limit the amplification. BugLink: https://bugs.launchpad.net/bugs/1052460 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Feel free to add cc to stable if you think it makes sense to do so. Alsa info available at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1052460/+attachment/33… sound/pci/hda/patch_via.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5f36a1..74fb6fd 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3666,6 +3666,32 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) update_power_state(codec, 0x21, AC_PWRST_D3); } +/* + * pin fix-up + */ +enum { + VIA_FIXUP_INTMIC_BOOST, +}; + +static void via_fixup_intmic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + override_mic_boost(codec, 0x30, 0, 2, 40); +} + +static const struct hda_fixup via_fixups[] = { + [VIA_FIXUP_INTMIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = via_fixup_intmic_boost, + }, +}; + +static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), + {} +}; + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3682,6 +3708,9 @@ static int patch_vt2002P(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) { -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 00/15] ASoC: OMAP: Convert to use dmaengine
by Peter Ujfalusi 18 Sep '12

18 Sep '12

Hello, Changelog: - Support for pause/resume for OMAP audio via dmaengine - dmaengine: support for NO_PERIOD_WAKEUP in cyclic mode - OMAP to keep supporting NO_PERIOD_WAKEUP for audio - Other plaforms can also try to enable this mode since we have now generic interface to do so. This series will switch the OMAP audio to use dmaengine. The final patch which does the switch was based on Russell King's earlier patch. The first 10 patch is to prepare the OMAP audio drivers for a smooth change to dmaengine: - sDMA FRAME sync mode is removed and replaced with PACKET mode - dai drivers no longer need to configure sDMA sync mode - dai drivers does not need to specify the DMA word length - with the exception of the omap-hdmi driver which requires 32bit word length regardless of the audio format in use - the McPDM driver used (to my surprise) hackish way of getting the DMA channel and address - via defines from some header files After the conversion OMAP audio support should have the same features as before, no regressions expected. I have tested the series on: - BeagleBoard (audio via McBSP): - aplay/arecord. In element mode and in threshold mode with different period sizes - mplayer -ao alsa: for direct ALSA access - mplayer -ao pulse: via PulseAudio to test NO_PERIOD_WAKEUP feature - OMAP4 Blaze (audio via McPDM and DMIC) - aplay/arecord - mplayer -ao alsa: for direct ALSA access - mplayer -ao pulse: via PulseAudio to test NO_PERIOD_WAKEUP feature The patches has been generated against: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 Janusz: Can you retest this series on OMAP1 to be sure I have not broken it? Ricardo: Can you test the omap-hmdi if it is still working? Regards, Peter --- Peter Ujfalusi (15): dmaengine: omap: Support for element mode in cyclic DMA dmaengine: omap: Add support for pause/resume in cyclic dma mode dmaengine: Add no_wakeup parameter to dmaengine_prep_dma_cyclic() dmaengine: Pass no_wakeup parameter via device_prep_dma_cyclic() callback dmaengine: omap-dma: Add support for no_wakeup in cyclic mode ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-dmic: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + drivers/dma/at_hdmac.c | 3 +- drivers/dma/ep93xx_dma.c | 4 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/imx-sdma.c | 2 +- drivers/dma/mmp_tdma.c | 2 +- drivers/dma/mxs-dma.c | 2 +- drivers/dma/omap-dma.c | 47 ++++-- drivers/dma/pl330.c | 2 +- drivers/dma/sa11x0-dma.c | 2 +- drivers/dma/sirf-dma.c | 2 +- drivers/dma/ste_dma40.c | 3 +- drivers/dma/tegra20-apb-dma.c | 2 +- include/linux/dmaengine.h | 8 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 ++- sound/soc/omap/omap-mcbsp.c | 60 +++----- sound/soc/omap/omap-mcpdm.c | 40 +++-- sound/soc/omap/omap-pcm.c | 236 ++++++++--------------------- sound/soc/omap/omap-pcm.h | 4 +- sound/soc/soc-dmaengine-pcm.c | 3 +- 22 files changed, 186 insertions(+), 269 deletions(-) -- 1.7.12

7 36

[alsa-devel] Channel map API patches: to be 3.7 or not to be?
by Takashi Iwai 17 Sep '12

17 Sep '12

Hi, does anyone have concern if I push the current channel map API patches (in sound-unstable git tree topic/tlv-chmap branch) for linux-next, i.e. for inclusion to 3.7 kernel? I don't worry too much about the kernel API. This can be refactored later. But the kernel ABI must retain, so we won't change in an incompatible way any longer once after it's merged to the upstream. So, if anyone sees a flaw in the kernel ABI definition using control elements, let me know. (But note that the API can't be perfect for all generic purposes. It's designed to be "good enough" for a big lack for multi-channel streaming. Of course, this doesn't discourage to use the API for any other purposes.) thanks, Takashi

3 12

[alsa-devel] [PATCH] hda-emu: Add support for channel mapping API
by David Henningsson 17 Sep '12

17 Sep '12

--- Hi Takashi, I tried to make hda-emu compile with the latest changes to the HDA driver. This is the result. It compiles, but when running any codec, I get the error "Control element Playback Channel Map:0 already exists!". I'm unsure whether this is a problem in the driver or in my simple copy-paste attempt to make this work. Can you offer some insight? Thanks. include/sound/asound.h | 57 +++++++++++++ include/sound/pcm.h | 58 +++++++++++++ include/sound/tlv.h | 8 ++ snd-wrapper.c | 214 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 337 insertions(+) diff --git a/include/sound/asound.h b/include/sound/asound.h index 25e90ed..61f1440 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -164,3 +164,60 @@ struct snd_ctl_event { unsigned char data8[60]; } data; }; + +typedef int __bitwise snd_pcm_state_t; +#define SNDRV_PCM_STATE_OPEN ((snd_pcm_state_t) 0) /* stream is open */ +#define SNDRV_PCM_STATE_SETUP ((snd_pcm_state_t) 1) /* stream has a setup */ +#define SNDRV_PCM_STATE_PREPARED ((snd_pcm_state_t) 2) /* stream is ready to start */ +#define SNDRV_PCM_STATE_RUNNING ((snd_pcm_state_t) 3) /* stream is running */ +#define SNDRV_PCM_STATE_XRUN ((snd_pcm_state_t) 4) /* stream reached an xrun */ +#define SNDRV_PCM_STATE_DRAINING ((snd_pcm_state_t) 5) /* stream is draining */ +#define SNDRV_PCM_STATE_PAUSED ((snd_pcm_state_t) 6) /* stream is paused */ +#define SNDRV_PCM_STATE_SUSPENDED ((snd_pcm_state_t) 7) /* hardware is suspended */ +#define SNDRV_PCM_STATE_DISCONNECTED ((snd_pcm_state_t) 8) /* hardware is disconnected */ +#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_DISCONNECTED + +struct snd_pcm_mmap_status { + snd_pcm_state_t state; /* RO: state - SNDRV_PCM_STATE_XXXX */ +}; + + +/* channel positions */ +enum { + SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ + SNDRV_CHMAP_FL, /* front left */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_RC, /* rear center */ + /* new definitions */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FLW, /* front left wide */ + SNDRV_CHMAP_FRW, /* front right wide */ + SNDRV_CHMAP_FLH, /* front left high */ + SNDRV_CHMAP_FCH, /* front center high */ + SNDRV_CHMAP_FRH, /* front right high */ + SNDRV_CHMAP_TC, /* top center */ + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, +}; + +#define SNDRV_CHMAP_POSITION_MASK 0xffff +#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) +#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f1719cb..4980c69 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -45,6 +45,8 @@ struct snd_pcm_runtime { unsigned int info; unsigned int rate_num; unsigned int rate_den; + /* -- mmap -- */ + struct snd_pcm_mmap_status *status; void *private_data; void (*private_free)(struct snd_pcm_runtime *runtime); @@ -62,12 +64,21 @@ struct snd_pcm_substream { struct snd_pcm_ops *ops; /* -- runtime information -- */ struct snd_pcm_runtime *runtime; + /* -- next substream -- */ + struct snd_pcm_substream *next; void *file; int ref_count; + }; struct snd_pcm_str { + /* -- substreams -- */ unsigned int substream_opened; + unsigned int substream_count; + struct snd_pcm_substream *substream; + + struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ + }; struct snd_pcm { @@ -241,5 +252,52 @@ static inline int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *run, int snd_pcm_format_width(int fmt); +/* + * PCM channel-mapping control API + */ +/* array element of channel maps */ +struct snd_pcm_chmap_elem { + unsigned char channels; + unsigned char map[15]; +}; + +/* channel map information; retrieved via snd_kcontrol_chip() */ +struct snd_pcm_chmap { + struct snd_pcm *pcm; /* assigned PCM instance */ + int stream; /* PLAYBACK or CAPTURE */ + struct snd_kcontrol *kctl; + const struct snd_pcm_chmap_elem *chmap; + unsigned int max_channels; + unsigned int channel_mask; /* optional: active channels bitmask */ + void *private_data; /* optional: private data pointer */ +}; + +/* get the PCM substream assigned to the given chmap info */ +static inline struct snd_pcm_substream * +snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx) +{ + struct snd_pcm_substream *s; + for (s = info->pcm->streams[info->stream].substream; s; s = s->next) + if (s->number == idx) + return s; + return NULL; +} + +/* ALSA-standard channel maps (RL/RR prior to C/LFE) */ +extern const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[]; +/* Other world's standard channel maps (C/LFE prior to RL/RR) */ +extern const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[]; + +/* bit masks to be passed to snd_pcm_chmap.channel_mask field */ +#define SND_PCM_CHMAP_MASK_24 ((1U << 2) | (1U << 4)) +#define SND_PCM_CHMAP_MASK_246 (SND_PCM_CHMAP_MASK_24 | (1U << 6)) +#define SND_PCM_CHMAP_MASK_2468 (SND_PCM_CHMAP_MASK_246 | (1U << 8)) + +int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_chmap_elem *chmap, + int max_channels, + unsigned long private_value, + struct snd_pcm_chmap **info_ret); + #endif /* __SOUND_PCM_H */ diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 7067e2d..de36aaa 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -73,4 +73,12 @@ #define TLV_DB_GAIN_MUTE -9999999 +/* + * channel-mapping TLV items + * TLV length must match with num_channels + */ +#define SNDRV_CTL_TLVT_CHMAP_FIXED 0x101 /* fixed channel position */ +#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */ +#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */ + #endif /* __SOUND_TLV_H */ diff --git a/snd-wrapper.c b/snd-wrapper.c index aaa2467..7716e67 100644 --- a/snd-wrapper.c +++ b/snd-wrapper.c @@ -257,3 +257,217 @@ void mylock_unlock(int *lock, const char *file, int line) break; } } + +/* + * standard channel mapping helpers + */ +#include <sound/tlv.h> + +/* default channel maps for multi-channel playbacks, up to 8 channels */ +const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_MONO } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { .channels = 8, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; +EXPORT_SYMBOL_GPL(snd_pcm_std_chmaps); + +/* alternative channel maps with CLFE <-> surround swapped for 6/8 channels */ +const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_MONO } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 8, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; +EXPORT_SYMBOL_GPL(snd_pcm_alt_chmaps); + +static bool valid_chmap_channels(const struct snd_pcm_chmap *info, int ch) +{ + if (ch > info->max_channels) + return false; + return !info->channel_mask || (info->channel_mask & (1U << ch)); +} + +static int pcm_chmap_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 0; + uinfo->count = info->max_channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SNDRV_CHMAP_LAST; + return 0; +} + +/* get callback for channel map ctl element + * stores the channel position firstly matching with the current channels + */ +static int pcm_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct snd_pcm_substream *substream; + const struct snd_pcm_chmap_elem *map; + + if (snd_BUG_ON(!info->chmap)) + return -EINVAL; + substream = snd_pcm_chmap_substream(info, idx); + if (!substream) + return -ENODEV; + memset(ucontrol->value.integer.value, 0, + sizeof(ucontrol->value.integer.value)); + if (!substream->runtime) + return 0; /* no channels set */ + for (map = info->chmap; map->channels; map++) { + int i; + if (map->channels == substream->runtime->channels && + valid_chmap_channels(info, map->channels)) { + for (i = 0; i < map->channels; i++) + ucontrol->value.integer.value[i] = map->map[i]; + return 0; + } + } + return -EINVAL; +} + +/* tlv callback for channel map ctl element + * expands the pre-defined channel maps in a form of TLV + */ +static int pcm_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + const struct snd_pcm_chmap_elem *map; + unsigned int __user *dst; + int c, count = 0; + + if (snd_BUG_ON(!info->chmap)) + return -EINVAL; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CONTAINER, tlv)) + return -EFAULT; + size -= 8; + dst = tlv + 2; + for (map = info->chmap; map->channels; map++) { + int chs_bytes = map->channels * 4; + if (!valid_chmap_channels(info, map->channels)) + continue; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CHMAP_FIXED, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + size -= chs_bytes; + count += chs_bytes; + for (c = 0; c < map->channels; c++) { + if (put_user(map->map[c], dst)) + return -EFAULT; + dst++; + } + } + if (put_user(count, tlv + 1)) + return -EFAULT; + return 0; +} + +static void pcm_chmap_ctl_private_free(struct snd_kcontrol *kcontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + info->pcm->streams[info->stream].chmap_kctl = NULL; + kfree(info); +} + +/** + * snd_pcm_add_chmap_ctls - create channel-mapping control elements + * @pcm: the assigned PCM instance + * @stream: stream direction + * @chmap: channel map elements (for query) + * @max_channels: the max number of channels for the stream + * @private_value: the value passed to each kcontrol's private_value field + * @info_ret: store struct snd_pcm_chmap instance if non-NULL + * + * Create channel-mapping control elements assigned to the given PCM stream(s). + * Returns zero if succeed, or a negative error value. + */ +int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_chmap_elem *chmap, + int max_channels, + unsigned long private_value, + struct snd_pcm_chmap **info_ret) +{ + struct snd_pcm_chmap *info; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = pcm_chmap_ctl_info, + .get = pcm_chmap_ctl_get, + .tlv.c = pcm_chmap_ctl_tlv, + }; + int err; + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + info->pcm = pcm; + info->stream = stream; + info->chmap = chmap; + info->max_channels = max_channels; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + knew.name = "Playback Channel Map"; + else + knew.name = "Capture Channel Map"; + knew.device = pcm->device; + knew.count = pcm->streams[stream].substream_count; + knew.private_value = private_value; + info->kctl = snd_ctl_new1(&knew, info); + if (!info->kctl) { + kfree(info); + return -ENOMEM; + } + info->kctl->private_free = pcm_chmap_ctl_private_free; + err = snd_ctl_add(pcm->card, info->kctl); + if (err < 0) + return err; + pcm->streams[stream].chmap_kctl = info->kctl; + if (info_ret) + *info_ret = info; + return 0; +} +EXPORT_SYMBOL_GPL(snd_pcm_add_chmap_ctls); -- 1.7.9.5

2 3

[alsa-devel] period events in sw_params
by Prasant J 17 Sep '12

17 Sep '12

Hi, What is period event? (when used in snd_pcm_sw_params_set_period_event for playback) -Prasant

1 0

[alsa-devel] sw_params configuration clarification
by Prasant J 17 Sep '12

17 Sep '12

Hi, I'm using ALSA API (alsa-lib 1.0.25) for writing an audio application for an embedded product. This is my understanding of the sw_params configuration api: (Can someone please confirm if the explanation is correct?) snd_pcm_sw_params_set_start_threshold -- sets the minimum no. of frames before which playback will not start. snd_pcm_sw_params_stop_threshold -- sets the minimum no. of frames when the playback stops (not completely clear) snd_pcm_sw_params_set_silence_threshold -- sets the no. of frames for inserting silence (equal to set by set_silence_size) when frames is less than the value set. Any info will be of help. -Prasant

1 0

[alsa-devel] [PATCH RESEND 1/5] ARM: tegra: config: enable dmaengine based APB DMA driver
by Stephen Warren 17 Sep '12

17 Sep '12

From: Laxman Dewangan <ldewangan(a)nvidia.com> Enable config for dmaengine based Tegra APB DMA driver and disable the legacy APB DMA driver (SYSTEM_DMA). Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- I'm looking for Ack's on patches 4 and 5 from the SPI and ASoC maintainers respectively, so that I can take this series through the Tegra tree for 3.7. When this series was first posted, there was a bug in the Tegra ASoC PCM driver which prevented it from working. A fix for this has since been applied to ASoC's for-3.6 branch and hence should show up in 3.6. arch/arm/configs/tegra_defconfig | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/arm/configs/tegra_defconfig b/arch/arm/configs/tegra_defconfig index adcf3c2..2990396 100644 --- a/arch/arm/configs/tegra_defconfig +++ b/arch/arm/configs/tegra_defconfig @@ -29,6 +29,7 @@ CONFIG_ARCH_TEGRA_2x_SOC=y CONFIG_ARCH_TEGRA_3x_SOC=y CONFIG_TEGRA_PCI=y CONFIG_TEGRA_DEBUG_UART_AUTO_ODMDATA=y +# CONFIG_TEGRA_SYSTEM_DMA is not set CONFIG_TEGRA_EMC_SCALING_ENABLE=y CONFIG_SMP=y CONFIG_PREEMPT=y @@ -175,6 +176,8 @@ CONFIG_RTC_DRV_MAX8907=y CONFIG_RTC_DRV_TPS65910=y CONFIG_RTC_DRV_EM3027=y CONFIG_RTC_DRV_TEGRA=y +CONFIG_DMADEVICES=y +CONFIG_TEGRA20_APB_DMA=y CONFIG_STAGING=y CONFIG_SENSORS_ISL29018=y CONFIG_SENSORS_ISL29028=y -- 1.7.0.4

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Re: [alsa-devel] [PATCH] ALSA: hda - Remove ignore_misc_bit
by Raymond Yau 17 Sep '12

17 Sep '12

>>>>>> But I still wonder why PulseAudio cares the headphone jack state even >>>>>> though this has only one output at all? >>>>> >>>>> >>>>> When seeing the system as a whole, there can be other outputs on other >>>>> cards - HDMI, USB etc. If somebody e g plugs a USB headset in it will be >>>>> simpler for the user if PulseAudio does not also show the unplugged 3.5 >>>>> mm jack phantom port of internal speaker or internal Mic is ok but how about those jack spdif, CD , Aux which does not have jack detect circuit Is it feasible for driver implement a read/ write kcontrol for those port and allow the user to change the state of those port by using special application (e.g. qashctl) ? > > That said, it's not super difficult to comment out a few lines in > /usr/share/pulseaudio/alsa-mixer/paths/*.conf, and also, most mixer UIs (e g pavucontrol) still allows you to route audio to an unavailable port. For ad1988 which still not support auto mute nor hda jack detect. The user still able to use the card in gnome but not in unity. https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/995169 And ca0132 which does not implement auto mute ? https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1043621 And the other ad198x which support auto mute but not implement hda jack detect https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1051494 The other cases are those on-board ac97 controller (e.g. Intel8x0, atiixp,...) with ac97 codecs which have the headphone playback volume control. What is the usage of those "Headphone Jack Sense" control of AC97 codecs ? Some of them seem have inverted control https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1033996

1 0

[alsa-devel] [PATCH] ALSA: hda - Add another pci id for Haswell board
by Wang Xingchao 17 Sep '12

17 Sep '12

A new PCI id 0x0d0c for Haswell HDA Controller. [root@SKBM04SDP ~]# lspci |grep Audio 00:03.0 Audio device: Intel Corporation Device 0d0c (rev 02) 00:1b.0 Audio device: Intel Corporation Lynx Point HD Audio Controller Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c9c779..e54673d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3442,6 +3442,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + { PCI_DEVICE(0x8086, 0x0d0c), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: compress - move the buffer check
by Vinod Koul 17 Sep '12

17 Sep '12

Commit ALSA: compress_core: integer overflow in snd_compr_allocate_buffer() added a new error check for input params. this add new routine for input checks and moves buffer overflow check to this new routine. This allows the error value to be propogated to user space Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/core/compress_offload.c | 20 ++++++++++++++++---- 1 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 68fe02c..bd7f28e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -407,10 +407,6 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, unsigned int buffer_size; void *buffer; - if (params->buffer.fragment_size == 0 || - params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) - return -EINVAL; - buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { buffer = NULL; @@ -429,6 +425,16 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, return 0; } +static int snd_compress_check_input(struct snd_compr_params *params) +{ + /* first let's check the buffer parameter's */ + if (params->buffer.fragment_size == 0 || + params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + return -EINVAL; + + return 0; +} + static int snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) { @@ -447,11 +453,17 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) retval = -EFAULT; goto out; } + + retval = snd_compress_check_input(params); + if (retval) + goto out; + retval = snd_compr_allocate_buffer(stream, params); if (retval) { retval = -ENOMEM; goto out; } + retval = stream->ops->set_params(stream, params); if (retval) goto out; -- 1.7.0.4

2 3

[alsa-devel] ucm: hardware resource arbitration
by Patrick Lai 16 Sep '12

16 Sep '12

Hi, I have two use cases which can acquire same CODEC path but they cannot run concurrently through mixing. When such scenario does arise, arbitration is required based on priority of use cases. On the other hand, same use cases on different platform may use different CODEC paths so arbitration is not required. In order to decide whether arbitration is required, I think the best place to do so is in UCM. I looked at existing UCM APIs but I do not believe existing APIs have concept of arbitration built in. If I am wrong, how can I handle arbitration with existing UCM APIs. If I am correct, is there anyone looking to expand UCM APIs for similar scenario? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] ASoC updates for 3.6
by Mark Brown 15 Sep '12

15 Sep '12

The following changes since commit d9875690d9b89a866022ff49e3fcea892345ad92: Linux 3.6-rc2 (2012-08-16 14:51:24 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 985b11fa8064d55d0d5a84e68667434598911bb2: ASoC: wm8904: correct the index (2012-09-14 18:02:38 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.6 A bigger set of updates than I'm entirely comfortable with - things backed up a bit due to travel. As ever the majority of these are small, focused updates for specific drivers though there are a couple of core changes. There's been good exposure in -next. The AT91 patch fixes a build break. ---------------------------------------------------------------- Bo Shen (1): ASoC: wm8904: correct the index Dylan Reid (1): ASoC: samsung dma - Don't indicate support for pause/resume. Fabio Estevam (1): ASoC: mc13783: Remove mono support Heather Lomond (1): ASoC: arizona: Fix typo in 44.1kHz rates Joachim Eastwood (1): ASoC: atmel-ssc: include linux/io.h for raw io Julia Lawall (3): ASoC: imx-sgtl5000: fix error return code ASoC: ux500_msp_i2s: better use devm functions and fix error return code ASoC: am3517evm: fix error return code Mark Brown (2): ASoC: dapm: Make sure we update the bias level for CODECs with no op ASoC: dapm: Don't force card bias level to be updated Prasad Joshi (1): ASoC: spear: correct the check for NULL dma_buffer pointer Stephen Warren (2): sound: tegra_alc5632: remove HP detect GPIO inversion ASoC: tegra: fix maxburst settings in dmaengine code include/linux/atmel-ssc.h | 1 + sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/mc13783.c | 8 ++++---- sound/soc/codecs/wm8904.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/omap/am3517evm.c | 2 +- sound/soc/samsung/dma.c | 8 +------- sound/soc/soc-dapm.c | 5 ++++- sound/soc/spear/spear_pcm.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 1 - sound/soc/tegra/tegra_pcm.c | 4 ++-- sound/soc/ux500/ux500_msp_i2s.c | 25 +++++-------------------- 12 files changed, 22 insertions(+), 40 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: wm8904: correct the index
by Bo Shen 14 Sep '12

14 Sep '12

Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d..13e8222 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = { { 14, 0x0000 }, /* R14 - Power Management 2 */ { 15, 0x0000 }, /* R15 - Power Management 3 */ { 18, 0x0000 }, /* R18 - Power Management 6 */ - { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 20, 0x945E }, /* R20 - Clock Rates 0 */ { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ { 22, 0x0006 }, /* R22 - Clock Rates 2 */ { 24, 0x0050 }, /* R24 - Audio Interface 0 */ -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 5/5] ASoC: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the I2S FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to I2S FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Please note that I've applied this to Tegra's for-3.7/dmaengine branch, even though it hasn't been ack'd by an ASoC maintainer.

2 3

Re: [alsa-devel] [PATCH 5/5] ASoC: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the I2S FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to I2S FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Mark, Liam, I'd like to take this patch through the Tegra tree, since it relies on the previous Tegra patches to convert to dmaengine. Could I please get an ack if that's OK? Thanks.

2 2

Re: [alsa-devel] [PATCH 4/7] mfd: ab8500-core: Apply the AB8500 CODEC's compatible string to its MFD cell
by Mark Brown 14 Sep '12

14 Sep '12

On Fri, Jul 27, 2012 at 03:36:33PM +0200, Samuel Ortiz wrote: > On Fri, Jul 27, 2012 at 01:38:51PM +0100, Lee Jones wrote: > > Provide a compatible string for the AB8500 CODEC to aid in > > configuration property look-up from its associated Device Tree > > node. > Applied as well. I remain *deeply* unconvinced that this is tasteful design for the device tree bindings, and it's certainly not needed urgently for v3.6 as there's no DT bindings on the CODEC side.

3 2

Re: [alsa-devel] [PATCH 4/5] spi: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the SPI FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to SPI FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Grant, Mark, I'd like to take this patch through the Tegra tree, since it relies on the previous Tegra patches to convert to dmaengine. Could I please get an ack if that's OK? Thanks.

2 3

[alsa-devel] [PATCH] ASoC: wm8960: Support shared LRCLK
by Mark Brown 14 Sep '12

14 Sep '12

If the LRCLK is shared and the WM8960 is clock master then we should enable the LRCM bit to tell the device that it should drive LRCLK when either ADC or DAC is enabled rather than separately driving the two LRCLKs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm8960.h | 2 ++ sound/soc/codecs/wm8960.c | 11 +++++++++++ 2 files changed, 13 insertions(+) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index 74e9a95..b5a1ab9e 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -19,6 +19,8 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ int dres; /* Discharge resistance for headphone outputs */ + + bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ }; #endif diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7cb0d07c..8740423 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1029,6 +1029,7 @@ static const struct regmap_config wm8960_regmap = { static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1041,6 +1042,16 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata && pdata->shared_lrclk) { + ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, + 0x4, 0x4); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", + ret); + return ret; + } + } + i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.10.4

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Re: [alsa-devel] [PATCH 1/7] powerpc/52xx: define FSL_SOC
by Eric Millbrandt 14 Sep '12

14 Sep '12

On 2012-09-14 agust(a)denx.de wrote: > Eric Millbrandt <emillbrandt(a)dekaresearch.com> wrote: > ... > > Better use: > > diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index > d701330..4563b28 100644 > --- a/sound/soc/fsl/Kconfig > +++ b/sound/soc/fsl/Kconfig > @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS > > menuconfig SND_POWERPC_SOC > tristate "SoC Audio for Freescale PowerPC CPUs" > - depends on FSL_SOC > + depends on FSL_SOC || PPC_MPC52xx > help > Say Y or M if you want to add support for codecs attached to > the PowerPC CPUs. > Thanks, > > Anatolij > Looks good to me, I will do this instead. Thanks. Eric -DISCLAIMER: an automatically appended disclaimer may follow. By posting- -to a public e-mail mailing list I hereby grant permission to distribute- -and copy this message.- This e-mail and the information, including any attachments, it contains are intended to be a confidential communication only to the person or entity to whom it is addressed and may contain information that is privileged. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please immediately notify the sender and destroy the original message. Thank you. Please consider the environment before printing this email.

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Re: [alsa-devel] FW: [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe rather than as a device
by Ola Lilja 14 Sep '12

14 Sep '12

On 08/23/2012 09:29 PM, Ola Lilja wrote: > > >> -----Original Message----- >> From: Lee Jones [mailto:lee.jones@linaro.org] >> Sent: den 23 augusti 2012 16:59 >> To: Mark Brown >> Cc: Ola Lilja; 'Linus Walleij'; roger.xr.nilsson(a)stericsson.com; linux- >> arm-kernel(a)lists.infradead.org; alsa-devel(a)alsa-project.org >> Subject: Re: [PATCH 07/22] ASoC: Ux500: Initialise PCM from MSP probe >> rather than as a device >> >> On Thu, Aug 23, 2012 at 03:37:57PM +0100, Mark Brown wrote: >> > On Thu, Aug 23, 2012 at 02:26:19PM +0100, Lee Jones wrote: >> > >> > > > I think Ola is suggesting probing the DMA driver from the machine >> > > > which will also work though I'm not 100% sure if I'm parsing the >> > > > above correctly. The issue in DT terms is that if the DMA >> > > > controller is shared with a bunch of other IPs then it should >> have >> > > > one node shared between them all and not a bunch of shim nodes >> > > > inserted in the middle which only exists due to the way Linux >> instantiates stuff. I haven't followed everything in this discussion, but what I meant in an earlier mail was that using the device for platform-DAI (MSP_I2S) also as platform int the DAI-link struct is not anything I can see fit (as in the patch Lee first submitted). I can see a few acceptable solutions: 1) Use it as it is! (A virtual device for PCM is created in arch/arm and the probe is invoked from the machine-driver). 2) Since the actual platform is the DMA-block, maybe there is some way of making the device "non-virtual" by having some sub-device in the DMA-driver that is instead triggered in the same way as in 1). 3) Move it to DT, but it seems that it is not possible when it is now a virtual device. Any of these three would be fine with me. >> > >> > > When you say 'machine', do you mean from arch/<arch>/mach-*? If so, >> > > I'm keen for that not to happen. >> > >> > No, sound/soc/ux500/snowball.c or whatever. At least that's my >> guess. >> >> Ah, I see. Maybe the mop500.c file then. >> >> > > > They instantiate the PCM driver dynamically from the DAI when >> it's >> > > > probed which is pretty much what you're patch is doing. >> > >> > > So they do it in the same why I have with this patch? Do you know >> > > why Ola might think this is a bad idea? >> > >> > I'm not 100% sure, I'm guessing it might be down to the fact that you >> > end up with multiple PCM drivers. We could avoid that with >> > refcounting but nobody's really worried about it. >> >> I think I'll wait for Ola to get back, as he's the expert on this >> stuff. >> >> I'll attempt to re-jig the patch-set, as this is a blocker atm. >> >> -- >> Lee Jones >> Linaro ST-Ericsson Landing Team Lead >> Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook >> | Twitter | Blog >

3 2

[alsa-devel] [PATCH 00/17] MOP500 Audio fix-ups and DT enablement
by Lee Jones 14 Sep '12

14 Sep '12

This patch-set provides the MOP500 Audio driver with a few small fix-ups along with Device Tree enablement for MOP500 Core and the MSPs. You'll notice that we've left the PCM out for the time being and continue to register that using platform code. This is because we're still in the middle of discussions pertaining to how this pseudo-device should be handled. We also pass pinctrl responsibility to the MSP driver instead of setting it up in platform code, which is an unnecessary over-head. Documentation/devicetree/bindings/mfd/ab8500.txt | 15 ++ .../devicetree/bindings/sound/ux500-mop500.txt | 39 +++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++ arch/arm/boot/dts/dbx5x0.dtsi | 39 +++ arch/arm/boot/dts/snowball.dts | 16 ++ arch/arm/mach-ux500/Makefile | 2 +- arch/arm/mach-ux500/board-mop500-audio.c | 199 +++++++++++++++ arch/arm/mach-ux500/board-mop500-msp.c | 267 -------------------- arch/arm/mach-ux500/board-mop500-msp.h | 14 - arch/arm/mach-ux500/board-mop500.c | 22 +- arch/arm/mach-ux500/board-mop500.h | 8 + arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 ++++++ sound/soc/ux500/mop500.c | 47 +++- sound/soc/ux500/ux500_msp_dai.c | 6 + sound/soc/ux500/ux500_msp_i2s.c | 89 +++++-- sound/soc/ux500/ux500_msp_i2s.h | 8 +- 18 files changed, 584 insertions(+), 319 deletions(-)

5 35

[alsa-devel] [patch] ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()
by Dan Carpenter 14 Sep '12

14 Sep '12

These are 32 bit values that come from the user, we need to check for integer overflows or we could end up allocating a smaller buffer than expected. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d..5a733e7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -409,6 +409,10 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, unsigned int buffer_size; void *buffer; + if (params->buffer.fragment_size == 0 || + params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + return -EINVAL; + buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { buffer = NULL;

3 10

[alsa-devel] [PATCH 0/8] ALSA: Add DSP firmware loader (resub)
by Ian Minett 14 Sep '12

14 Sep '12

From: Ian Minett <ian_minett(a)creativelabs.com> >Thanks for the patches, and sorry for the late response, No problem, and sorry for ours too- we've been working on fixing the issues you raised last time. Hopefully this submission will sort out many of those items. Macros have been taken out, comments added, and firmware is now cached. >Keep the git commit author as me. Ok. I ensured you were set as the git commit author for the first two patchfiles. Is there anything else I need to do to properly retain your authorship on those commits? >- Why hda_stream_format is defined? hda_stream_format is defined to store the stream format information used for the DSP download. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's patch: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's code: Pass DMA buffer pointers in calls to setup_bdle(). Add DSP loader callback routines to controller. Add new DSP loader switch to Kconfig to enable DSP firmware loading. 3: - patch_ca0132.c - ca0132_regs.h Add DSP register definitions header file 4: - patch_ca0132.c Add DSP firmware enums and defs to CA0132 codec 5: - patch_ca0132.c Add calls to new DSP loader system to transfer firmware binary to the hardware. Add chip read/write routines, DSP I/O, SCP packet format helper functions and transfer DMA management. 6: - patch_ca0132.c Add DSP firmware caching to CA0132 codec 7: - patch_ca0132.c Add comments and descriptions to functions. Merge chipio write address functions and add fix to dsp_write_wait(). 8: - hda_codec.h Change return value for load_dsp_prepare to -ENOSYS in case where DSP loader routines are not available.

2 15

Re: [alsa-devel] [PATCH 4/5] spi: tegra: remove support of legacy DMA driver based access
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Remove the support code which uses the legacy APB DMA driver > for accessing the SPI FIFO. > The driver will use the dmaengine based APB DMA driver for > accessing reqding/writing to SPI FIFO. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Please note that I've applied this to Tegra's for-3.7/dmaengine branch, even though it hasn't been ack'd by an SPI maintainer.

1 0

Re: [alsa-devel] [PATCH 0/5] ARM: tegra: move all APB DMA client to dmaengine based driver
by Stephen Warren 14 Sep '12

14 Sep '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > This patch series enable the dmaengine based Tegra APB DMA driver > and remove the support code for the legacy dma driver from different > APB DMA clients. > Patch 1 enable the dmaengine based dma drie and disable the legacy dma driver > in config file. > Patch 2 remove the legacy dma driver. > Patch 3,4, 5 remove the support code from different APB DMA clients to support > legacy dma driver. > > Please note that this series will depends on patch > dma: tegra: enable/disable dma clock > which is not in 3.6-rc1 but will be available in next RC (rc2) of 3.6 as > part of fixes. I have applied this series to Tegra's for-3.7/dmaengine branch. Right now, that branch is based directly on Mark Brown's ASoC for-3.6 branch in order to pick up the fix it depends on. However, I'll rebase on the 3.6 pull request tag he sends for that branch as soon as it's available.

1 0

Re: [alsa-devel] [PATCH v2] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Takashi Iwai 13 Sep '12

13 Sep '12

At Thu, 13 Sep 2012 16:18:08 +0200, Daniel Vetter wrote: > > On Thu, Sep 13, 2012 at 4:00 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > But, this makes me wonder why this patch can't go into 3.6. It's a > > clear bug and the fix is trivial. > > > > Daniel, isn't it better to pick the fix (for 3.6) to drm-intel-fixes, > > then merge back to drm-intel-next? > > Well, on a quick read-through it didn't sound serious (just a bit of > noise in dmesg and a few spurious reconfigs), For the graphics side, yes. But for the sound driver side, you'll miss the jack-detection unplug event without this fix. > and we're rather late in > the -rc cycle. Yes, but a bug is a bug is a bug :) And the fix is two-liners. If there were no conflict, would you mind it? I don't think so. > Applying this on top of -fixes will cause a conflict, so didn't looked > like I should bother. Yeah, that's bad. > Also, I _want_ the patch for -next so that git > blame shows the commit and not some random conflicting merge. With parallel development, you can't always get straight lines, as you know well... Takashi

1 0

[alsa-devel] [PATCH v2] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_enable_hdmi(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_disable_hdmi(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 5d02aad..229897f 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -674,10 +674,7 @@ static void intel_disable_hdmi(struct intel_encoder *encoder) struct drm_i915_private *dev_priv = dev->dev_private; struct intel_hdmi *intel_hdmi = enc_to_intel_hdmi(&encoder->base); u32 temp; - u32 enable_bits = SDVO_ENABLE; - - if (intel_hdmi->has_audio) - enable_bits |= SDVO_AUDIO_ENABLE; + u32 enable_bits = SDVO_ENABLE | SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

3 3

[alsa-devel] [PATCH] ASoC: codecs: Add DA9055 codec driver
by Ashish Chavan 13 Sep '12

13 Sep '12

This patch adds support for Dialog semiconductor's DA9055 audio codec. This has been tested on DA9055 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da9055.c | 1507 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1513 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3684255..b92759a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DA732X if I2C + select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -239,6 +240,9 @@ config SND_SOC_DA7210 config SND_SOC_DA732X tristate +config SND_SOC_DA9055 + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca508b2..9bd4d95 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da732x-objs := da732x.o +snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -144,6 +145,7 @@ obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o +obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c new file mode 100644 index 0000000..be5c9e6 --- /dev/null +++ b/sound/soc/codecs/da9055.c @@ -0,0 +1,1507 @@ +/* + * DA9055 ALSA Soc codec driver + * + * Copyright (c) 2012 Dialog Semiconductor + * + * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C + * Written by David Chen <david.chen(a)diasemi.com> and + * Ashish Chavan <ashish.chavan(a)kpitcummins.com> + * + * 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. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +/* DA9055 register space */ + +/* Status Registers */ +#define DA9055_STATUS1 0x02 +#define DA9055_PLL_STATUS 0x03 +#define DA9055_AUX_L_GAIN_STATUS 0x04 +#define DA9055_AUX_R_GAIN_STATUS 0x05 +#define DA9055_MIC_L_GAIN_STATUS 0x06 +#define DA9055_MIC_R_GAIN_STATUS 0x07 +#define DA9055_MIXIN_L_GAIN_STATUS 0x08 +#define DA9055_MIXIN_R_GAIN_STATUS 0x09 +#define DA9055_ADC_L_GAIN_STATUS 0x0A +#define DA9055_ADC_R_GAIN_STATUS 0x0B +#define DA9055_DAC_L_GAIN_STATUS 0x0C +#define DA9055_DAC_R_GAIN_STATUS 0x0D +#define DA9055_HP_L_GAIN_STATUS 0x0E +#define DA9055_HP_R_GAIN_STATUS 0x0F +#define DA9055_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA9055_CIF_CTRL 0x20 +#define DA9055_DIG_ROUTING_AIF 0X21 +#define DA9055_SR 0x22 +#define DA9055_REFERENCES 0x23 +#define DA9055_PLL_FRAC_TOP 0x24 +#define DA9055_PLL_FRAC_BOT 0x25 +#define DA9055_PLL_INTEGER 0x26 +#define DA9055_PLL_CTRL 0x27 +#define DA9055_AIF_CLK_MODE 0x28 +#define DA9055_AIF_CTRL 0x29 +#define DA9055_DIG_ROUTING_DAC 0x2A +#define DA9055_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA9055_AUX_L_GAIN 0x30 +#define DA9055_AUX_R_GAIN 0x31 +#define DA9055_MIXIN_L_SELECT 0x32 +#define DA9055_MIXIN_R_SELECT 0x33 +#define DA9055_MIXIN_L_GAIN 0x34 +#define DA9055_MIXIN_R_GAIN 0x35 +#define DA9055_ADC_L_GAIN 0x36 +#define DA9055_ADC_R_GAIN 0x37 +#define DA9055_ADC_FILTERS1 0x38 +#define DA9055_MIC_L_GAIN 0x39 +#define DA9055_MIC_R_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA9055_DAC_FILTERS5 0x40 +#define DA9055_DAC_FILTERS2 0x41 +#define DA9055_DAC_FILTERS3 0x42 +#define DA9055_DAC_FILTERS4 0x43 +#define DA9055_DAC_FILTERS1 0x44 +#define DA9055_DAC_L_GAIN 0x45 +#define DA9055_DAC_R_GAIN 0x46 +#define DA9055_CP_CTRL 0x47 +#define DA9055_HP_L_GAIN 0x48 +#define DA9055_HP_R_GAIN 0x49 +#define DA9055_LINE_GAIN 0x4A +#define DA9055_MIXOUT_L_SELECT 0x4B +#define DA9055_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA9055_SYSTEM_MODES_INPUT 0x50 +#define DA9055_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA9055_AUX_L_CTRL 0x60 +#define DA9055_AUX_R_CTRL 0x61 +#define DA9055_MIC_BIAS_CTRL 0x62 +#define DA9055_MIC_L_CTRL 0x63 +#define DA9055_MIC_R_CTRL 0x64 +#define DA9055_MIXIN_L_CTRL 0x65 +#define DA9055_MIXIN_R_CTRL 0x66 +#define DA9055_ADC_L_CTRL 0x67 +#define DA9055_ADC_R_CTRL 0x68 +#define DA9055_DAC_L_CTRL 0x69 +#define DA9055_DAC_R_CTRL 0x6A +#define DA9055_HP_L_CTRL 0x6B +#define DA9055_HP_R_CTRL 0x6C +#define DA9055_LINE_CTRL 0x6D +#define DA9055_MIXOUT_L_CTRL 0x6E +#define DA9055_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA9055_LDO_CTRL 0x90 +#define DA9055_IO_CTRL 0x91 +#define DA9055_GAIN_RAMP_CTRL 0x92 +#define DA9055_MIC_CONFIG 0x93 +#define DA9055_PC_COUNT 0x94 +#define DA9055_CP_VOL_THRESHOLD1 0x95 +#define DA9055_CP_DELAY 0x96 +#define DA9055_CP_DETECTOR 0x97 +#define DA9055_AIF_OFFSET 0x98 +#define DA9055_DIG_CTRL 0x99 +#define DA9055_ALC_CTRL2 0x9A +#define DA9055_ALC_CTRL3 0x9B +#define DA9055_ALC_NOISE 0x9C +#define DA9055_ALC_TARGET_MIN 0x9D +#define DA9055_ALC_TARGET_MAX 0x9E +#define DA9055_ALC_GAIN_LIMITS 0x9F +#define DA9055_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA9055_ALC_ANTICLIP_CTRL 0xA1 +#define DA9055_ALC_ANTICLIP_LEVEL 0xA2 +#define DA9055_ALC_OFFSET_OP2M_L 0xA6 +#define DA9055_ALC_OFFSET_OP2U_L 0xA7 +#define DA9055_ALC_OFFSET_OP2M_R 0xAB +#define DA9055_ALC_OFFSET_OP2U_R 0xAC +#define DA9055_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA9055_ALC_CIC_OP_LVL_DATA 0xAE +#define DA9055_DAC_NG_SETUP_TIME 0xAF +#define DA9055_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA9055_DAC_NG_ON_THRESHOLD 0xB1 +#define DA9055_DAC_NG_CTRL 0xB2 + +/* DIG_ROUTING_AIF bit fields */ +#define DA9055_AIF_L_SRC (0 << 0) +#define DA9055_AIF_R_SRC (1 << 4) + +/* SR bit fields */ +#define DA9055_SR_8000 (0x1 << 0) +#define DA9055_SR_11025 (0x2 << 0) +#define DA9055_SR_12000 (0x3 << 0) +#define DA9055_SR_16000 (0x5 << 0) +#define DA9055_SR_22050 (0x6 << 0) +#define DA9055_SR_24000 (0x7 << 0) +#define DA9055_SR_32000 (0x9 << 0) +#define DA9055_SR_44100 (0xA << 0) +#define DA9055_SR_48000 (0xB << 0) +#define DA9055_SR_88200 (0xE << 0) +#define DA9055_SR_96000 (0xF << 0) + +/* REFERENCES bit fields */ +#define DA9055_BIAS_EN (1 << 3) +#define DA9055_VMID_EN (1 << 7) + +/* PLL_CTRL bit fields */ +#define DA9055_PLL_INDIV_10_20_MHZ (1 << 2) +#define DA9055_PLL_SRM_EN (1 << 6) +#define DA9055_PLL_EN (1 << 7) + +/* AIF_CLK_MODE bit fields */ +#define DA9055_AIF_BCLKS_PER_WCLK_32 (0 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_64 (1 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_128 (2 << 0) +#define DA9055_AIF_BCLKS_PER_WCLK_256 (3 << 0) +#define DA9055_AIF_CLK_EN_SLAVE_MODE (0 << 7) +#define DA9055_AIF_CLK_EN_MASTER_MODE (1 << 7) + +/* AIF_CTRL bit fields */ +#define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) +#define DA9055_AIF_FORMAT_LEFT_J (1 << 0) +#define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_WORD_S16_LE (0 << 2) +#define DA9055_AIF_WORD_S20_3LE (1 << 2) +#define DA9055_AIF_WORD_S24_LE (2 << 2) +#define DA9055_AIF_WORD_S32_LE (3 << 2) +#define DA9055_AIF_OE (1 << 6) +#define DA9055_AIF_EN (1 << 7) + +/* CP_CTRL bit fields */ +#define DA9055_CP_MCHANGE (3 << 4) +#define DA9055_CP_EN (1 << 7) + +/* MIC_BIAS_CTRL bit fields */ +#define DA9055_MIC_BIAS_EN (1 << 7) + +/* MIXIN_L_CTRL bit fields */ +#define DA9055_MIXIN_L_MIX_EN (1 << 3) + +/* MIXIN_R_CTRL bit fields */ +#define DA9055_MIXIN_R_MIX_EN (1 << 3) + +/* ADC_L_CTRL bit fields */ +#define DA9055_ADC_L_EN (1 << 7) + +/* ADC_R_CTRL bit fields */ +#define DA9055_ADC_R_EN (1 << 7) + +/* DAC_L_CTRL bit fields */ +#define DA9055_DAC_L_MUTE_EN (1 << 6) + +/* DAC_R_CTRL bit fields */ +#define DA9055_DAC_R_MUTE_EN (1 << 6) + +/* HP_L_CTRL bit fields */ +#define DA9055_HP_L_AMP_OE (1 << 3) + +/* HP_R_CTRL bit fields */ +#define DA9055_HP_R_AMP_OE (1 << 3) + +/* LINE_CTRL bit fields */ +#define DA9055_LINE_AMP_OE (1 << 3) + +/* MIXOUT_L_CTRL bit fields */ +#define DA9055_MIXOUT_L_MIX_EN (1 << 3) +#define DA9055_MIXOUT_L_AMP_EN (1 << 7) + +/* MIXOUT_R_CTRL bit fields */ +#define DA9055_MIXOUT_R_MIX_EN (1 << 3) +#define DA9055_MIXOUT_R_AMP_EN (1 << 7) + +/* ALC_CIC_OP_LEVEL_CTRL bit fields */ +#define DA9055_ALC_CAPTURE_DATA (0 << 0) +#define DA9055_ALC_DATA_BOT (1 << 0) +#define DA9055_ALC_DATA_MIDDLE (2 << 0) +#define DA9055_ALC_DATA_TOP (3 << 0) +#define DA9055_ALC_CIC_OP_CHANNEL_LEFT (0 << 7) +#define DA9055_ALC_CIC_OP_CHANNEL_RIGHT (1 << 7) + +#define DA9055_GAIN_RAMPING_EN (1 << 5) + +#define DA9055_ALC_OFFSET_15_8 0x00FF00 +#define DA9055_ALC_OFFSET_17_16 0x030000 +#define DA9055_ALC_AVG_ITERATIONS 5 + +struct pll_div { + int fref; + int fout; + u8 frac_top; + u8 frac_bot; + u8 integer; + u8 mode; /* 0 = slave, 1 = master */ +}; + +/* PLL divisor table */ +static const struct pll_div da9055_pll_div[] = { + /* for MASTER mode, fs = 44.1Khz and its harmonics */ + {11289600, 2822400, 0x00, 0x00, 0x20, 1}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x03, 0x61, 0x1E, 1}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x0C, 0xCC, 0x1D, 1}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x19, 0x45, 0x1B, 1}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x18, 0x56, 0x1A, 1}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x02, 0xD0, 0x19, 1}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x1A, 0x1C, 0x12, 1}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x0B, 0x6D, 0x12, 1}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x07, 0xDD, 0x12, 1}, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz and its harmonics */ + {11289600, 3072000, 0x1A, 0x8E, 0x22, 1}, /* MCLK=11.2896Mhz */ + {12000000, 3072000, 0x18, 0x93, 0x20, 1}, /* MCLK=12Mhz */ + {12288000, 3072000, 0x00, 0x00, 0x20, 1}, /* MCLK=12.288Mhz */ + {13000000, 3072000, 0x07, 0xEA, 0x1E, 1}, /* MCLK=13Mhz */ + {13500000, 3072000, 0x04, 0x11, 0x1D, 1}, /* MCLK=13.5Mhz */ + {14400000, 3072000, 0x09, 0xD0, 0x1B, 1}, /* MCLK=14.4Mhz */ + {19200000, 3072000, 0x0F, 0x5C, 0x14, 1}, /* MCLK=19.2Mhz */ + {19680000, 3072000, 0x1F, 0x60, 0x13, 1}, /* MCLK=19.68Mhz */ + {19800000, 3072000, 0x1B, 0x80, 0x13, 1}, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + {11289600, 2822400, 0x0D, 0x47, 0x21, 0}, /* MCLK=11.2896Mhz */ + {12000000, 2822400, 0x0D, 0xFA, 0x1F, 0}, /* MCLK=12Mhz */ + {12288000, 2822400, 0x16, 0x66, 0x1E, 0}, /* MCLK=12.288Mhz */ + {13000000, 2822400, 0x00, 0x98, 0x1D, 0}, /* MCLK=13Mhz */ + {13500000, 2822400, 0x1E, 0x33, 0x1B, 0}, /* MCLK=13.5Mhz */ + {14400000, 2822400, 0x06, 0x50, 0x1A, 0}, /* MCLK=14.4Mhz */ + {19200000, 2822400, 0x14, 0xBC, 0x13, 0}, /* MCLK=19.2Mhz */ + {19680000, 2822400, 0x05, 0x66, 0x13, 0}, /* MCLK=19.68Mhz */ + {19800000, 2822400, 0x01, 0xAE, 0x13, 0}, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA9055_CLKSRC_MCLK +}; + +/* Gain and Volume */ + +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da9055_hpf_cutoff_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da9055_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +static const struct soc_enum da9055_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da9055_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da9055_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +static const struct soc_enum da9055_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); + +/* Gain ramping rate value */ +static const char * const da9055_gain_ramping_txt[] = { + "nominal rate", "nominal rate * 4", "nominal rate * 8", + "nominal rate / 8" +}; + +static const struct soc_enum da9055_gain_ramping_rate = + SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); + +/* DAC noise gate setup time value */ +static const char * const da9055_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da9055_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, + da9055_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da9055_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, + da9055_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da9055_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da9055_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, + da9055_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da9055_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da9055_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, + da9055_dac_soft_mute_rate_txt); + +/* LDO voltage level value */ +static const char * const da9055_ldo_lvl_select_txt[] = { + "1.05V", "1.1V", "1.2V", "1.4V" +}; + +static const struct soc_enum da9055_ldo_lvl_select = + SOC_ENUM_SINGLE(DA9055_LDO_CTRL, 4, 4, da9055_ldo_lvl_select_txt); + +/* AIF routing select */ +static const char * const da9055_aif_select_txt[] = { + "ADC left", "ADC right", "AIF input left", "AIF input right" +}; + +static const struct soc_enum da9055_aif_l_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_AIF, 0, 4, da9055_aif_select_txt); + +static const struct soc_enum da9055_aif_r_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_AIF, 4, 4, da9055_aif_select_txt); + +/* DAC routing select */ +static const char * const da9055_dac_select_txt[] = { + "ADC left output", "ADC right output", "AIF input left", + "AIF input right" +}; + +static const struct soc_enum da9055_dac_l_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_select_txt); + +static const struct soc_enum da9055_dac_r_select = + SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_select_txt); + +/* Digital MIC clock rate select */ +static const char * const da9055_dmic_clk_rate_txt[] = { + "3MHz", "1.5MHz" +}; + +static const struct soc_enum da9055_dmic_clk_rate = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 2, 2, da9055_dmic_clk_rate_txt); + +/* Digital MIC sample phase select */ +static const char * const da9055_dmic_phase_txt[] = { + "Sample on DMICCLK edges", "Sample between DMICCLK edges" +}; + +static const struct soc_enum da9055_dmic_phase = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 1, 2, da9055_dmic_phase_txt); + +/* Digital MIC channel select */ +static const char * const da9055_dmic_channel_select_txt[] = { + "Rising Left Falling Right", "Falling Left Rising Right" +}; + +static const struct soc_enum da9055_dmic_channel_select = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 0, 2, + da9055_dmic_channel_select_txt); + +/* MIC bias select */ +static const char * const da9055_mic_bias_select_txt[] = { + "MIC Bias 1", "MIC Bias 2" +}; + +static const struct soc_enum da9055_mic_bias_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 6, 2, + da9055_mic_bias_select_txt); + +/* MIC bias voltage level select */ +static const char * const da9055_mic_bias_level_txt[] = { + "1.6V", "1.8V", "2.1V", "2.2V" +}; + +static const struct soc_enum da9055_mic_bias_level = + SOC_ENUM_SINGLE(DA9055_MIC_CONFIG, 4, 4, da9055_mic_bias_level_txt); + +/* MIC Left Amp routing select */ +static const char * const da9055_mic_l_select_txt[] = { + "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_l_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_select_txt); + +/* MIC Right Amp routing select */ +static const char * const da9055_mic_r_select_txt[] = { + "MIC2_R_L", "MIC2_R", "MIC2_L" +}; + +static const struct soc_enum da9055_mic_r_select = + SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_select_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da9055_signal_tracking_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da9055_integ_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, + da9055_signal_tracking_rate_txt); + +static const struct soc_enum da9055_integ_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, + da9055_signal_tracking_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da9055_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_attack_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da9055_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da9055_release_rate = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da9055_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da9055_hold_time = + SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); + +static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA9055_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA9055_ALC_CIC_OP_LVL_CTRL, + reg_val | DA9055_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA9055_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA9055_ALC_AVG_ITERATIONS; +} + +static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 reg_val, adc_left, adc_right; + int avg_left_data, avg_right_data, offset_l, offset_r; + + if (ucontrol->value.integer.value[0]) { + /* + * While enabling ALC (or ALC sync mode), calibration of the DC + * offsets must be done first + */ + + /* Save current values from ADC control registers */ + adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); + adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_ADC_L_EN, DA9055_ADC_L_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_ADC_R_EN, DA9055_ADC_R_EN); + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da9055_get_alc_data(codec, + DA9055_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_L, reg_val); + reg_val = (offset_l & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_L, reg_val); + + reg_val = (offset_r & DA9055_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2M_R, reg_val); + reg_val = (offset_r & DA9055_ALC_OFFSET_17_16) >> 16; + snd_soc_write(codec, DA9055_ALC_OFFSET_OP2U_R, reg_val); + + /* Restore original values of ADC control registers */ + snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); + snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +static const struct snd_kcontrol_new da9055_snd_controls[] = { + + /* Volume controls */ + SOC_DOUBLE_R_TLV("Mic Volume", + DA9055_MIC_L_GAIN, DA9055_MIC_R_GAIN, + 0, 0x7, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", + DA9055_AUX_L_GAIN, DA9055_AUX_R_GAIN, + 0, 0x3f, 0, aux_vol_tlv), + SOC_DOUBLE_R_TLV("Mixin PGA Volume", + DA9055_MIXIN_L_GAIN, DA9055_MIXIN_R_GAIN, + 0, 0xf, 0, mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", + DA9055_ADC_L_GAIN, DA9055_ADC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Volume", + DA9055_DAC_L_GAIN, DA9055_DAC_R_GAIN, + 0, 0x7f, 0, digital_gain_tlv), + SOC_DOUBLE_R_TLV("HeadPhone Volume", + DA9055_HP_L_GAIN, DA9055_HP_R_GAIN, + 0, 0x3f, 0, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA9055_LINE_GAIN, 0, 0x3f, 0, + lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA9055_DAC_FILTERS4, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA9055_DAC_FILTERS2, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA9055_DAC_FILTERS2, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA9055_DAC_FILTERS3, 0, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA9055_DAC_FILTERS3, 4, 0xf, 0, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA9055_DAC_FILTERS4, 0, 0xf, 0, + eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA9055_ADC_FILTERS1, 7, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da9055_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA9055_ADC_FILTERS1, 3, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da9055_adc_vf_cutoff), + + SOC_SINGLE("DAC HPF Switch", DA9055_DAC_FILTERS1, 7, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da9055_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA9055_DAC_FILTERS1, 3, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da9055_dac_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Mute Switch", DA9055_MIC_L_CTRL, + DA9055_MIC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Aux Mute Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("Mixin PGA Mute Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("ADC Mute Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 6, 1, 0), + SOC_DOUBLE_R("HeadPhone Mute Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 6, 1, 0), + SOC_SINGLE("Lineout Mute Switch", DA9055_LINE_CTRL, 6, 1, 0), + SOC_SINGLE("DAC Soft Mute Switch", DA9055_DAC_FILTERS5, 7, 1, 0), + SOC_ENUM("DAC Soft Mute Rate", da9055_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 4, 1, 0), + SOC_DOUBLE_R("HeadPhone ZC Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 4, 1, 0), + SOC_SINGLE("Lineout ZC Switch", DA9055_LINE_CTRL, 4, 1, 0), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA9055_AUX_L_CTRL, + DA9055_AUX_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA9055_ADC_L_CTRL, + DA9055_ADC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA9055_DAC_L_CTRL, + DA9055_DAC_R_CTRL, 5, 1, 0), + SOC_DOUBLE_R("HeadPhone Gain Ramping Switch", DA9055_HP_L_CTRL, + DA9055_HP_R_CTRL, 5, 1, 0), + SOC_SINGLE("Lineout Gain Ramping Switch", DA9055_LINE_CTRL, 5, 1, 0), + SOC_ENUM("Gain Ramping Rate", da9055_gain_ramping_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Enable Switch", DA9055_DAC_NG_CTRL, 7, 1, 0), + SOC_SINGLE("DAC NG ON Threshold", DA9055_DAC_NG_ON_THRESHOLD, + 0, 0x7, 0), + SOC_SINGLE("DAC NG OFF Threshold", DA9055_DAC_NG_OFF_THRESHOLD, + 0, 0x7, 0), + SOC_ENUM("DAC NG Setup Time", da9055_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da9055_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da9055_dac_ng_rampdown_rate), + + /* LDO control */ + SOC_SINGLE("LDO Enable", DA9055_LDO_CTRL, 7, 1, 0), + SOC_ENUM("LDO Level Select", da9055_ldo_lvl_select), + + /* AIF Routing control */ + SOC_ENUM("AIF Left Select", da9055_aif_l_select), + SOC_ENUM("AIF Right Select", da9055_aif_r_select), + + /* DAC Routing control */ + SOC_ENUM("DAC Left Select", da9055_dac_l_select), + SOC_ENUM("DAC Right Select", da9055_dac_r_select), + + /* DAC Invertion control */ + SOC_SINGLE("DAC Left Invert", DA9055_DIG_CTRL, 3, 1, 0), + SOC_SINGLE("DAC Right Invert", DA9055_DIG_CTRL, 7, 1, 0), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Enable", DA9055_MIXIN_L_SELECT, + DA9055_MIXIN_R_SELECT, 7, 1, 0), + SOC_ENUM("DMIC Clock Rate", da9055_dmic_clk_rate), + SOC_ENUM("DMIC Sample Phase", da9055_dmic_phase), + SOC_ENUM("DMIC Channel Select", da9055_dmic_channel_select), + + /* MIC PGA input selection controls */ + SOC_ENUM("Mic Left Input Select", da9055_mic_l_select), + SOC_ENUM("Mic Right Input Select", da9055_mic_r_select), + + /* To select if MIC Bias powers MIC 1 or MIC 2 */ + SOC_ENUM("Mic Bias Select", da9055_mic_bias_select), + + /* MIC Bias voltage select control */ + SOC_ENUM("Mic Bias Level", da9055_mic_bias_level), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Enable", DA9055_ALC_CTRL1, 3, 7, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE_EXT("ALC Sync Mode Enable", DA9055_ALC_CTRL1, 1, 1, 0, + snd_soc_get_volsw, da9055_put_alc_sw), + SOC_SINGLE("ALC Offset Enable", DA9055_ALC_CTRL1, 0, 1, 0), + SOC_SINGLE("ALC Anticlip Mode Enable", DA9055_ALC_ANTICLIP_CTRL, + 7, 1, 0), + SOC_SINGLE("ALC Anticlip Level", DA9055_ALC_ANTICLIP_LEVEL, + 0, 0x7f, 0), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA9055_ALC_TARGET_MIN, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA9055_ALC_TARGET_MAX, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA9055_ALC_NOISE, + 0, 0x3f, 1, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA9055_ALC_GAIN_LIMITS, + 4, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA9055_ALC_GAIN_LIMITS, + 0, 0xf, 0, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 0, 0x7, 0, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", + DA9055_ALC_ANA_GAIN_LIMITS, + 4, 0x7, 0, alc_analog_gain_tlv), + SOC_ENUM("ALC Attack Rate", da9055_attack_rate), + SOC_ENUM("ALC Release Rate", da9055_release_rate), + SOC_ENUM("ALC Hold Time", da9055_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da9055_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da9055_integ_release_rate), +}; + +/* DAPM Controls */ + +/* In Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXIN_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_L_SELECT, 2, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXIN_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA9055_MIXIN_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA9055_MIXIN_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXIN_R_SELECT, 3, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da9055_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA9055_MIXOUT_L_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_L_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_L_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA9055_MIXOUT_L_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_L_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_L_SELECT, + 6, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da9055_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA9055_MIXOUT_R_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Switch", DA9055_MIXOUT_R_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Switch", DA9055_MIXOUT_R_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA9055_MIXOUT_R_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 4, 1, 0), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA9055_MIXOUT_R_SELECT, + 5, 1, 0), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA9055_MIXOUT_R_SELECT, + 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da9055_dapm_widgets[] = { + /* Input Side */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2L"), + SND_SOC_DAPM_INPUT("MIC2R"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA9055_MIC_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA9055_MIC_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left", DA9055_AUX_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right", DA9055_AUX_R_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Left", DA9055_MIXIN_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIXIN Right", DA9055_MIXIN_R_CTRL, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinl_controls[0], + ARRAY_SIZE(da9055_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixinr_controls[0], + ARRAY_SIZE(da9055_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA9055_ADC_L_CTRL, 7, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA9055_ADC_R_CTRL, 7, 0), + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA9055_DAC_L_CTRL, 7, 0), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA9055_DAC_R_CTRL, 7, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutl_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da9055_dapm_mixoutr_controls[0], + ARRAY_SIZE(da9055_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Lineout", DA9055_LINE_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA9055_HP_L_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA9055_HP_R_CTRL, 7, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da9055_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"Mic Left", NULL, "MIC1"}, + {"Mic Left", NULL, "MIC2L"}, + {"Mic Right", NULL, "MIC2L"}, + {"Mic Right", NULL, "MIC2R"}, + {"Aux Left", NULL, "AUXL"}, + {"Aux Right", NULL, "AUXR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux Left Switch", "Aux Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux Right Switch", "Aux Right"}, + {"In Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + + {"MIXIN Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "MIXIN Left"}, + + {"MIXIN Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "MIXIN Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux Left Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Left", "Aux Left Invert Switch", "Aux Left"}, + {"Out Mixer Left", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Left", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux Right Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Switch", "MIXIN Left"}, + {"Out Mixer Right", "Aux Right Invert Switch", "Aux Right"}, + {"Out Mixer Right", "Mixin Right Invert Switch", "MIXIN Right"}, + {"Out Mixer Right", "Mixin Left Invert Switch", "MIXIN Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Headphone Left", NULL, "Out Mixer Left"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "Out Mixer Right"}, + {"HPR", NULL, "Headphone Right"}, + + {"Lineout", NULL, "Out Mixer Right"}, + {"LINE", NULL, "Lineout"}, +}; + +/* Codec private data */ +struct da9055_priv { + struct regmap *regmap; + unsigned int mclk_rate; + int master; +}; + +static struct reg_default da9055_reg_defaults[] = { + { 0x21, 0x10 }, + { 0x22, 0x0A }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x0C }, + { 0x28, 0x01 }, + { 0x29, 0x08 }, + { 0x2A, 0x32 }, + { 0x2B, 0x00 }, + { 0x30, 0x35 }, + { 0x31, 0x35 }, + { 0x32, 0x00 }, + { 0x33, 0x00 }, + { 0x34, 0x03 }, + { 0x35, 0x03 }, + { 0x36, 0x6F }, + { 0x37, 0x6F }, + { 0x38, 0x80 }, + { 0x39, 0x01 }, + { 0x3A, 0x01 }, + { 0x40, 0x00 }, + { 0x41, 0x88 }, + { 0x42, 0x88 }, + { 0x43, 0x08 }, + { 0x44, 0x80 }, + { 0x45, 0x6F }, + { 0x46, 0x6F }, + { 0x47, 0x61 }, + { 0x48, 0x35 }, + { 0x49, 0x35 }, + { 0x4A, 0x35 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x60, 0x44 }, + { 0x61, 0x44 }, + { 0x62, 0x00 }, + { 0x63, 0x40 }, + { 0x64, 0x40 }, + { 0x65, 0x40 }, + { 0x66, 0x40 }, + { 0x67, 0x40 }, + { 0x68, 0x40 }, + { 0x69, 0x48 }, + { 0x6A, 0x40 }, + { 0x6B, 0x41 }, + { 0x6C, 0x40 }, + { 0x6D, 0x40 }, + { 0x6E, 0x10 }, + { 0x6F, 0x10 }, + { 0x90, 0x80 }, + { 0x92, 0x02 }, + { 0x93, 0x00 }, + { 0x99, 0x00 }, + { 0x9A, 0x00 }, + { 0x9B, 0x00 }, + { 0x9C, 0x3F }, + { 0x9D, 0x00 }, + { 0x9E, 0x3F }, + { 0x9F, 0xFF }, + { 0xA0, 0x71 }, + { 0xA1, 0x00 }, + { 0xA2, 0x00 }, + { 0xA6, 0x00 }, + { 0xA7, 0x00 }, + { 0xAB, 0x00 }, + { 0xAC, 0x00 }, + { 0xAD, 0x00 }, + { 0xAF, 0x08 }, + { 0xB0, 0x00 }, + { 0xB1, 0x00 }, + { 0xB2, 0x00 }, +}; + +static bool da9055_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case DA9055_STATUS1: + case DA9055_PLL_STATUS: + case DA9055_AUX_L_GAIN_STATUS: + case DA9055_AUX_R_GAIN_STATUS: + case DA9055_MIC_L_GAIN_STATUS: + case DA9055_MIC_R_GAIN_STATUS: + case DA9055_MIXIN_L_GAIN_STATUS: + case DA9055_MIXIN_R_GAIN_STATUS: + case DA9055_ADC_L_GAIN_STATUS: + case DA9055_ADC_R_GAIN_STATUS: + case DA9055_DAC_L_GAIN_STATUS: + case DA9055_DAC_R_GAIN_STATUS: + case DA9055_HP_L_GAIN_STATUS: + case DA9055_HP_R_GAIN_STATUS: + case DA9055_LINE_GAIN_STATUS: + case DA9055_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +/* Set DAI word length */ +static int da9055_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_ctrl, fs; + u32 sysclk; + + /* Set AIF source to Left and Right ADC */ + snd_soc_write(codec, DA9055_DIG_ROUTING_AIF, + DA9055_AIF_L_SRC | DA9055_AIF_R_SRC); + + aif_ctrl = snd_soc_read(codec, DA9055_AIF_CTRL) & 0xf3; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif_ctrl |= DA9055_AIF_WORD_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif_ctrl |= DA9055_AIF_WORD_S20_3LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif_ctrl |= DA9055_AIF_WORD_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif_ctrl |= DA9055_AIF_WORD_S32_LE; + break; + default: + return -EINVAL; + } + + aif_ctrl |= (DA9055_AIF_OE | DA9055_AIF_EN); + + /* Enable AIF */ + snd_soc_write(codec, DA9055_AIF_CTRL, aif_ctrl); + + switch (params_rate(params)) { + case 8000: + fs = DA9055_SR_8000; + sysclk = 3072000; + break; + case 11025: + fs = DA9055_SR_11025; + sysclk = 2822400; + break; + case 12000: + fs = DA9055_SR_12000; + sysclk = 3072000; + break; + case 16000: + fs = DA9055_SR_16000; + sysclk = 3072000; + break; + case 22050: + fs = DA9055_SR_22050; + sysclk = 2822400; + break; + case 32000: + fs = DA9055_SR_32000; + sysclk = 3072000; + break; + case 44100: + fs = DA9055_SR_44100; + sysclk = 2822400; + break; + case 48000: + fs = DA9055_SR_48000; + sysclk = 3072000; + break; + case 88200: + fs = DA9055_SR_88200; + sysclk = 2822400; + break; + case 96000: + fs = DA9055_SR_96000; + sysclk = 3072000; + break; + default: + return -EINVAL; + } + + if (da9055->mclk_rate) { + /* PLL Mode, Write actual FS */ + snd_soc_write(codec, DA9055_SR, fs); + } else { + /* + * Non-PLL Mode + * When PLL is bypassed, chip assumes constant MCLK of + * 12.288MHz and uses sample rate value to divide this MCLK + * to derive its sys clk. As sys clk has to be 256 * Fs, we + * need to write constant sample rate i.e. 48KHz. + */ + snd_soc_write(codec, DA9055_SR, DA9055_SR_48000); + } + + if (da9055->mclk_rate && (da9055->mclk_rate != sysclk)) { + /* PLL Mode */ + if (!da9055->master) { + /* PLL slave mode, enable PLL and also SRM */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN | DA9055_PLL_SRM_EN, + DA9055_PLL_EN | DA9055_PLL_SRM_EN); + } else { + /* PLL master mode, only enable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, + DA9055_PLL_EN, DA9055_PLL_EN); + } + } else { + /* Non PLL Mode, disable PLL */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + } + + return 0; +} + +/* Set DAI mode and Format */ +static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + u8 aif_clk_mode, aif_ctrl; + + aif_clk_mode = snd_soc_read(codec, DA9055_AIF_CLK_MODE) & 0x7c; + aif_ctrl = snd_soc_read(codec, DA9055_AIF_CTRL) & 0xfc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* DA9055 in I2S Master Mode */ + da9055->master = 1; + aif_clk_mode |= DA9055_AIF_CLK_EN_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* DA9055 in I2S Slave Mode */ + da9055->master = 0; + aif_clk_mode |= DA9055_AIF_CLK_EN_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_ctrl |= DA9055_AIF_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_ctrl |= DA9055_AIF_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif_ctrl |= DA9055_AIF_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + aif_clk_mode |= DA9055_AIF_BCLKS_PER_WCLK_32; + + snd_soc_write(codec, DA9055_AIF_CLK_MODE, aif_clk_mode); + snd_soc_write(codec, DA9055_AIF_CTRL, aif_ctrl); + + return 0; +} + +static int da9055_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, DA9055_DAC_R_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_DAC_L_MUTE_EN, 0); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_DAC_R_MUTE_EN, 0); + } + + return 0; +} + +#define DA9055_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da9055_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA9055_CLKSRC_MCLK: + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da9055->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* + * da9055_set_dai_pll : Configure the codec PLL + * @param codec_dai : Pointer to codec DAI + * @param pll_id : da9055 has only one pll, so pll_id is always zero + * @param fref : Input MCLK frequency + * @param fout : FsDM value + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 11.2896MHz, 12MHz, 12.288MHz, + * 13MHz, 13.5MHz, 14.4MHz, 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da9055_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + + u8 pll_frac_top, pll_frac_bot, pll_integer, cnt; + + /* Disable PLL before setting the divisors */ + snd_soc_update_bits(codec, DA9055_PLL_CTRL, DA9055_PLL_EN, 0); + + /* In slave mode, there is only one set of divisors */ + if (!da9055->master) + fout = 2822400; + + /* Search pll div array for correct divisors */ + for (cnt = 0; cnt < ARRAY_SIZE(da9055_pll_div); cnt++) { + /* Check fref, mode and fout */ + if ((fref == da9055_pll_div[cnt].fref) && + (da9055->master == da9055_pll_div[cnt].mode) && + (fout == da9055_pll_div[cnt].fout)) { + /* All match, pick up divisors */ + pll_frac_top = da9055_pll_div[cnt].frac_top; + pll_frac_bot = da9055_pll_div[cnt].frac_bot; + pll_integer = da9055_pll_div[cnt].integer; + break; + } + } + if (cnt >= ARRAY_SIZE(da9055_pll_div)) + goto pll_err; + + /* Write PLL dividers */ + snd_soc_write(codec, DA9055_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA9055_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA9055_PLL_INTEGER, pll_integer); + + return 0; +pll_err: + dev_err(codec_dai->dev, "Error in setting up PLL\n"); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da9055_dai_ops = { + .hw_params = da9055_hw_params, + .set_fmt = da9055_set_dai_fmt, + .set_sysclk = da9055_set_dai_sysclk, + .set_pll = da9055_set_dai_pll, + .digital_mute = da9055_mute, +}; + +static struct snd_soc_dai_driver da9055_dai = { + .name = "da9055-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA9055_FORMATS, + }, + .ops = &da9055_dai_ops, + .symmetric_rates = 1, +}; + +static int da9055_probe(struct snd_soc_codec *codec) +{ + struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = da9055->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + da9055->mclk_rate = 0; /* This will be set from set_sysclk() */ + da9055->master = 0; /* This will be set from set_fmt() */ + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_AUX_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_ADC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_DAC_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); + + /* Input side settings */ + + /* Enable VMID reference & master bias */ + snd_soc_write(codec, DA9055_REFERENCES, + DA9055_VMID_EN | DA9055_BIAS_EN); + + /* Enable Mic Bias */ + snd_soc_write(codec, DA9055_MIC_BIAS_CTRL, DA9055_MIC_BIAS_EN); + + /* Enable Left and Right input mixers */ + snd_soc_update_bits(codec, DA9055_MIXIN_L_CTRL, + DA9055_MIXIN_L_MIX_EN, DA9055_MIXIN_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXIN_R_CTRL, + DA9055_MIXIN_R_MIX_EN, DA9055_MIXIN_R_MIX_EN); + + /* Output side settings */ + + /* Enable Left and Right output mixers */ + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_MIX_EN, DA9055_MIXOUT_L_MIX_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_MIX_EN, DA9055_MIXOUT_R_MIX_EN); + + /* Enable Left and Right output Amps */ + snd_soc_update_bits(codec, DA9055_MIXOUT_L_CTRL, + DA9055_MIXOUT_L_AMP_EN, DA9055_MIXOUT_L_AMP_EN); + snd_soc_update_bits(codec, DA9055_MIXOUT_R_CTRL, + DA9055_MIXOUT_R_AMP_EN, DA9055_MIXOUT_R_AMP_EN); + + /* Enable the charge pump and set mode */ + snd_soc_update_bits(codec, DA9055_CP_CTRL, + DA9055_CP_EN | DA9055_CP_MCHANGE, + DA9055_CP_EN | DA9055_CP_MCHANGE); + + /* Enable Left and Right HeadPhone Output */ + snd_soc_update_bits(codec, DA9055_HP_L_CTRL, + DA9055_HP_L_AMP_OE, DA9055_HP_L_AMP_OE); + snd_soc_update_bits(codec, DA9055_HP_R_CTRL, + DA9055_HP_R_AMP_OE, DA9055_HP_R_AMP_OE); + + /* Enable Lineout Output Enable */ + snd_soc_update_bits(codec, DA9055_LINE_CTRL, + DA9055_LINE_AMP_OE, DA9055_LINE_AMP_OE); + + /* Set this as per your system configuration */ + snd_soc_write(codec, DA9055_PLL_CTRL, DA9055_PLL_INDIV_10_20_MHZ); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da9055 = { + .probe = da9055_probe, + + .controls = da9055_snd_controls, + .num_controls = ARRAY_SIZE(da9055_snd_controls), + + .dapm_widgets = da9055_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da9055_dapm_widgets), + .dapm_routes = da9055_audio_map, + .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), +}; + +static const struct regmap_config da9055_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da9055_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da9055_reg_defaults), + .volatile_reg = da9055_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit da9055_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da9055_priv *da9055; + int ret; + + da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + GFP_KERNEL); + if (!da9055) + return -ENOMEM; + + i2c_set_clientdata(i2c, da9055); + + da9055->regmap = regmap_init_i2c(i2c, &da9055_regmap_config); + if (IS_ERR(da9055->regmap)) { + ret = PTR_ERR(da9055->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da9055, &da9055_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + ret); + regmap_exit(da9055->regmap); + } + return ret; +} + +static int __devexit da9055_remove(struct i2c_client *client) +{ + struct da9055_priv *da9055 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(da9055->regmap); + return 0; +} + +static const struct i2c_device_id da9055_i2c_id[] = { + { "da9055", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da9055_i2c_driver = { + .driver = { + .name = "da9055", + .owner = THIS_MODULE, + }, + .probe = da9055_i2c_probe, + .remove = __devexit_p(da9055_remove), + .id_table = da9055_i2c_id, +}; + +module_i2c_driver(da9055_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); +MODULE_AUTHOR("David Chen, Ashish Chavan"); +MODULE_LICENSE("GPL"); -- 1.7.1

3 6

[alsa-devel] [PATCH] ASoC: ad1836: Convert to direct regmap usage.
by Lars-Peter Clausen 13 Sep '12

13 Sep '12

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 76 +++++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ae1eb51..dce6ebe 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -19,6 +19,8 @@ #include <sound/soc.h> #include <sound/tlv.h> #include <linux/spi/spi.h> +#include <linux/regmap.h> + #include "ad1836.h" enum ad1836_type { @@ -30,6 +32,7 @@ enum ad1836_type { /* codec private data */ struct ad1836_priv { enum ad1836_type type; + struct regmap *regmap; }; /* @@ -161,8 +164,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(dai->codec); int word_len = 0; - struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { @@ -178,10 +181,12 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, + AD1836_DAC_WORD_LEN_MASK, word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, + AD1836_ADC_WORD_LEN_MASK, word_len << AD1836_ADC_WORD_OFFSET); return 0; @@ -223,15 +228,17 @@ static struct snd_soc_dai_driver ad183x_dais[] = { #ifdef CONFIG_PM static int ad1836_suspend(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* reset clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, 0); } static int ad1836_resume(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* restore clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX); } #else @@ -250,37 +257,30 @@ static int ad1836_probe(struct snd_soc_codec *codec) num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2; num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2; - ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); - return ret; - } - /* default setting for ad1836 */ /* de-emphasis: 48kHz, power-on dac */ - snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); + regmap_write(ad1836->regmap, AD1836_DAC_CTRL1, 0x300); /* unmute dac channels */ - snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); + regmap_write(ad1836->regmap, AD1836_DAC_CTRL2, 0x0); /* high-pass filter enable, power-on adc */ - snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ - snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL2, 0x180); /* volume */ for (i = 1; i <= num_dacs; ++i) { - snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF); - snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF); + regmap_write(ad1836->regmap, AD1836_DAC_L_VOL(i), 0x3FF); + regmap_write(ad1836->regmap, AD1836_DAC_R_VOL(i), 0x3FF); } if (ad1836->type == AD1836) { /* left/right diff:PGA/MUX */ - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL3, 0x3A); ret = snd_soc_add_codec_controls(codec, ad1836_controls, ARRAY_SIZE(ad1836_controls)); if (ret) return ret; } else { - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL3, 0x00); } ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2); @@ -313,8 +313,9 @@ static int ad1836_probe(struct snd_soc_codec *codec) /* power down chip */ static int ad1836_remove(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* reset clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, 0); } @@ -323,8 +324,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .remove = ad1836_remove, .suspend = ad1836_suspend, .resume = ad1836_resume, - .reg_cache_size = AD1836_NUM_REGS, - .reg_word_size = sizeof(u16), .controls = ad183x_controls, .num_controls = ARRAY_SIZE(ad183x_controls), @@ -334,6 +333,33 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes), }; +static const struct reg_default ad1836_reg_defaults[] = { + { AD1836_DAC_CTRL1, 0x0000 }, + { AD1836_DAC_CTRL2, 0x0000 }, + { AD1836_DAC_L_VOL(0), 0x0000 }, + { AD1836_DAC_R_VOL(0), 0x0000 }, + { AD1836_DAC_L_VOL(1), 0x0000 }, + { AD1836_DAC_R_VOL(1), 0x0000 }, + { AD1836_DAC_L_VOL(2), 0x0000 }, + { AD1836_DAC_R_VOL(2), 0x0000 }, + { AD1836_DAC_L_VOL(3), 0x0000 }, + { AD1836_DAC_R_VOL(3), 0x0000 }, + { AD1836_ADC_CTRL1, 0x0000 }, + { AD1836_ADC_CTRL2, 0x0000 }, + { AD1836_ADC_CTRL3, 0x0000 }, +}; + +static const struct regmap_config ad1836_regmap_config = { + .val_bits = 12, + .reg_bits = 4, + .read_flag_mask = 0x08, + + .max_register = AD1836_ADC_CTRL3, + .reg_defaults = ad1836_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ad1836_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct ad1836_priv *ad1836; @@ -344,6 +370,10 @@ static int __devinit ad1836_spi_probe(struct spi_device *spi) if (ad1836 == NULL) return -ENOMEM; + ad1836->regmap = devm_regmap_init_spi(spi, &ad1836_regmap_config); + if (IS_ERR(ad1836->regmap)) + return PTR_ERR(ad1836->regmap); + ad1836->type = spi_get_device_id(spi)->driver_data; spi_set_drvdata(spi, ad1836); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: ad193x: Use managed regmap init
by Lars-Peter Clausen 13 Sep '12

13 Sep '12

Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 50 +++++++++------------------------------------ 1 file changed, 10 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 13e62be..2f75266 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -381,40 +381,25 @@ static const struct regmap_config ad193x_spi_regmap_config = { static int __devinit ad193x_spi_probe(struct spi_device *spi) { struct ad193x_priv *ad193x; - int ret; ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); - if (IS_ERR(ad193x->regmap)) { - ret = PTR_ERR(ad193x->regmap); - goto err_out; - } + ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config); + if (IS_ERR(ad193x->regmap)) + return PTR_ERR(ad193x->regmap); spi_set_drvdata(spi, ad193x); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ad193x, &ad193x_dai, 1); - if (ret < 0) - goto err_regmap_exit; - - return 0; - -err_regmap_exit: - regmap_exit(ad193x->regmap); -err_out: - return ret; + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } static int __devexit ad193x_spi_remove(struct spi_device *spi) { - struct ad193x_priv *ad193x = spi_get_drvdata(spi); - snd_soc_unregister_codec(&spi->dev); - regmap_exit(ad193x->regmap); return 0; } @@ -449,40 +434,25 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct ad193x_priv *ad193x; - int ret; ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); - if (IS_ERR(ad193x->regmap)) { - ret = PTR_ERR(ad193x->regmap); - goto err_out; - } + ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config); + if (IS_ERR(ad193x->regmap)) + return PTR_ERR(ad193x->regmap); i2c_set_clientdata(client, ad193x); - ret = snd_soc_register_codec(&client->dev, - &soc_codec_dev_ad193x, &ad193x_dai, 1); - if (ret < 0) - goto err_regmap_exit; - - return 0; - -err_regmap_exit: - regmap_exit(ad193x->regmap); -err_out: - return ret; + return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } static int __devexit ad193x_i2c_remove(struct i2c_client *client) { - struct ad193x_priv *ad193x = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(ad193x->regmap); return 0; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH 00/11] ASoC: OMAP: Convert to use dmaengine
by Peter Ujfalusi 13 Sep '12

13 Sep '12

Hello, This series will switch the OMAP audio to use dmaengine. The final patch which does the switch was based on Russell King's earlier patch. The first 10 patch is to prepare the OMAP audio drivers for a smooth change to dmaengine: - sDMA FRAME sync mode is removed and replaced with PACKET mode - dai drivers no longer need to configure sDMA sync mode - dai drivers does not need to specify the DMA word length - with the exception of the omap-hdmi driver which requires 32bit word length regardless of the audio format in use - the McPDM driver used (to my surprise) hackish way of getting the DMA channel and address - via defines from some header files I have tested the series on: - BeagleBoard (audio via McBSP) with aplay/arecord. In element mode and in threshold mode with different period sizes - OMAP4 Blaze (audio via McPDM and DMIC) With this conversion the NO_PERIOD_WAKEUP is not supported at the moment. I'll be looking at the dmaengine core and omap parts to add this feature back. The patches has been generated against: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 Janusz: Can you retest this series on OMAP1 to be sure I have not broken it? Ricardo: Can you test the omap-hmdi if it is still working? Regards, Peter --- Peter Ujfalusi (11): dmaengine: omap: Support for element mode in cyclic DMA ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode ASoC: omap-pcm: Select sDMA synchronization based on packet_size ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges ASoC: omap-mcpdm: Use platform_get_resource_* to get resources ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type ASoC: omap-pcm, omap-hdmi: Change the use of omap_pcm_dma_data->data_type ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time ASoC: omap-pcm: Convert to use dmaengine arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + drivers/dma/omap-dma.c | 5 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-hdmi.c | 17 +- sound/soc/omap/omap-mcbsp.c | 60 +++---- sound/soc/omap/omap-mcpdm.c | 41 +++-- sound/soc/omap/omap-pcm.c | 246 ++++++++--------------------- sound/soc/omap/omap-pcm.h | 4 +- 9 files changed, 136 insertions(+), 251 deletions(-) -- 1.7.12

4 17

[alsa-devel] Invitation to connect on LinkedIn
by Vitaly Chernooky via LinkedIn 13 Sep '12

13 Sep '12

LinkedIn ------------ Vitaly Chernooky requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Vitaly Accept invitation from Vitaly Chernooky http://www.linkedin.com/e/jpxkk3-h71n762j-6i/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Vitaly Chernooky http://www.linkedin.com/e/jpxkk3-h71n762j-6i/rso/18329483/HTJa/name/1380363… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-h71n762j-6i/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [sound-unstable:master 118/143] sound/pci/hda/patch_cirrus.c:1414:10: sparse: Initializer entry defined twice
by Fengguang Wu 13 Sep '12

13 Sep '12

Hi Takashi, FYI, there are new sparse warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 82cb521360dfd32f4f03ceaa74ef6b83ae983593 commit: ef596a57b4d7d8b258beb570ed309ef85bf24dd1 [118/143] ALSA: hda - Add support for MacBook Pro 10,1 All sparse warnings: + sound/pci/hda/patch_cirrus.c:1414:10: sparse: Initializer entry defined twice sound/pci/hda/patch_cirrus.c:1418:10: also defined here The root cause is, CS420X_MBP101 == CS420X_GPIO_23: @@ -85,6 +85,8 @@ enum { => CS420X_GPIO_23, CS420X_IMAC27_122 = CS420X_GPIO_23, CS420X_APPLE = CS420X_GPIO_13, => + CS420X_MBP101, + CS420X_MBP101_COEF, CS420X_AUTO, }; vim +1414 sound/pci/hda/patch_cirrus.c 1404 [CS420X_IMAC27] = { 1405 .type = HDA_FIXUP_PINS, 1406 .v.pins = imac27_pincfgs, 1407 .chained = true, 1408 .chain_id = CS420X_GPIO_13, 1409 }, 1410 [CS420X_GPIO_13] = { 1411 .type = HDA_FIXUP_FUNC, 1412 .v.func = cs420x_fixup_gpio_13, 1413 }, > 1414 [CS420X_GPIO_23] = { 1415 .type = HDA_FIXUP_FUNC, 1416 .v.func = cs420x_fixup_gpio_23, 1417 }, 1418 [CS420X_MBP101] = { 1419 .type = HDA_FIXUP_PINS, 1420 .v.pins = mbp101_pincfgs, 1421 .chained = true, 1422 .chain_id = CS420X_MBP101_COEF, 1423 }, 1424 [CS420X_MBP101_COEF] = { --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8961: Convert to direct regmap API usage
by a＠opensource.wolfsonmicro.com 13 Sep '12

13 Sep '12

From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8961.c | 426 +++++++++++++++++---------------------------- 1 file changed, 161 insertions(+), 265 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 719fb69..4ea64d6e 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -19,6 +19,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -31,277 +32,158 @@ #define WM8961_MAX_REGISTER 0xFC -static u16 wm8961_reg_defaults[] = { - 0x009F, /* R0 - Left Input volume */ - 0x009F, /* R1 - Right Input volume */ - 0x0000, /* R2 - LOUT1 volume */ - 0x0000, /* R3 - ROUT1 volume */ - 0x0020, /* R4 - Clocking1 */ - 0x0008, /* R5 - ADC & DAC Control 1 */ - 0x0000, /* R6 - ADC & DAC Control 2 */ - 0x000A, /* R7 - Audio Interface 0 */ - 0x01F4, /* R8 - Clocking2 */ - 0x0000, /* R9 - Audio Interface 1 */ - 0x00FF, /* R10 - Left DAC volume */ - 0x00FF, /* R11 - Right DAC volume */ - 0x0000, /* R12 */ - 0x0000, /* R13 */ - 0x0040, /* R14 - Audio Interface 2 */ - 0x0000, /* R15 - Software Reset */ - 0x0000, /* R16 */ - 0x007B, /* R17 - ALC1 */ - 0x0000, /* R18 - ALC2 */ - 0x0032, /* R19 - ALC3 */ - 0x0000, /* R20 - Noise Gate */ - 0x00C0, /* R21 - Left ADC volume */ - 0x00C0, /* R22 - Right ADC volume */ - 0x0120, /* R23 - Additional control(1) */ - 0x0000, /* R24 - Additional control(2) */ - 0x0000, /* R25 - Pwr Mgmt (1) */ - 0x0000, /* R26 - Pwr Mgmt (2) */ - 0x0000, /* R27 - Additional Control (3) */ - 0x0000, /* R28 - Anti-pop */ - 0x0000, /* R29 */ - 0x005F, /* R30 - Clocking 3 */ - 0x0000, /* R31 */ - 0x0000, /* R32 - ADCL signal path */ - 0x0000, /* R33 - ADCR signal path */ - 0x0000, /* R34 */ - 0x0000, /* R35 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 */ - 0x0000, /* R39 */ - 0x0000, /* R40 - LOUT2 volume */ - 0x0000, /* R41 - ROUT2 volume */ - 0x0000, /* R42 */ - 0x0000, /* R43 */ - 0x0000, /* R44 */ - 0x0000, /* R45 */ - 0x0000, /* R46 */ - 0x0000, /* R47 - Pwr Mgmt (3) */ - 0x0023, /* R48 - Additional Control (4) */ - 0x0000, /* R49 - Class D Control 1 */ - 0x0000, /* R50 */ - 0x0003, /* R51 - Class D Control 2 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 */ - 0x0000, /* R55 */ - 0x0106, /* R56 - Clocking 4 */ - 0x0000, /* R57 - DSP Sidetone 0 */ - 0x0000, /* R58 - DSP Sidetone 1 */ - 0x0000, /* R59 */ - 0x0000, /* R60 - DC Servo 0 */ - 0x0000, /* R61 - DC Servo 1 */ - 0x0000, /* R62 */ - 0x015E, /* R63 - DC Servo 3 */ - 0x0010, /* R64 */ - 0x0010, /* R65 - DC Servo 5 */ - 0x0000, /* R66 */ - 0x0001, /* R67 */ - 0x0003, /* R68 - Analogue PGA Bias */ - 0x0000, /* R69 - Analogue HP 0 */ - 0x0060, /* R70 */ - 0x01FB, /* R71 - Analogue HP 2 */ - 0x0000, /* R72 - Charge Pump 1 */ - 0x0065, /* R73 */ - 0x005F, /* R74 */ - 0x0059, /* R75 */ - 0x006B, /* R76 */ - 0x0038, /* R77 */ - 0x000C, /* R78 */ - 0x000A, /* R79 */ - 0x006B, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 - Charge Pump B */ - 0x0087, /* R83 */ - 0x0000, /* R84 */ - 0x005C, /* R85 */ - 0x0000, /* R86 */ - 0x0000, /* R87 - Write Sequencer 1 */ - 0x0000, /* R88 - Write Sequencer 2 */ - 0x0000, /* R89 - Write Sequencer 3 */ - 0x0000, /* R90 - Write Sequencer 4 */ - 0x0000, /* R91 - Write Sequencer 5 */ - 0x0000, /* R92 - Write Sequencer 6 */ - 0x0000, /* R93 - Write Sequencer 7 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0000, /* R96 */ - 0x0000, /* R97 */ - 0x0000, /* R98 */ - 0x0000, /* R99 */ - 0x0000, /* R100 */ - 0x0000, /* R101 */ - 0x0000, /* R102 */ - 0x0000, /* R103 */ - 0x0000, /* R104 */ - 0x0000, /* R105 */ - 0x0000, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 */ - 0x0000, /* R109 */ - 0x0000, /* R110 */ - 0x0000, /* R111 */ - 0x0000, /* R112 */ - 0x0000, /* R113 */ - 0x0000, /* R114 */ - 0x0000, /* R115 */ - 0x0000, /* R116 */ - 0x0000, /* R117 */ - 0x0000, /* R118 */ - 0x0000, /* R119 */ - 0x0000, /* R120 */ - 0x0000, /* R121 */ - 0x0000, /* R122 */ - 0x0000, /* R123 */ - 0x0000, /* R124 */ - 0x0000, /* R125 */ - 0x0000, /* R126 */ - 0x0000, /* R127 */ - 0x0000, /* R128 */ - 0x0000, /* R129 */ - 0x0000, /* R130 */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 */ - 0x0000, /* R135 */ - 0x0000, /* R136 */ - 0x0000, /* R137 */ - 0x0000, /* R138 */ - 0x0000, /* R139 */ - 0x0000, /* R140 */ - 0x0000, /* R141 */ - 0x0000, /* R142 */ - 0x0000, /* R143 */ - 0x0000, /* R144 */ - 0x0000, /* R145 */ - 0x0000, /* R146 */ - 0x0000, /* R147 */ - 0x0000, /* R148 */ - 0x0000, /* R149 */ - 0x0000, /* R150 */ - 0x0000, /* R151 */ - 0x0000, /* R152 */ - 0x0000, /* R153 */ - 0x0000, /* R154 */ - 0x0000, /* R155 */ - 0x0000, /* R156 */ - 0x0000, /* R157 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0000, /* R164 */ - 0x0000, /* R165 */ - 0x0000, /* R166 */ - 0x0000, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 */ - 0x0000, /* R173 */ - 0x0000, /* R174 */ - 0x0000, /* R175 */ - 0x0000, /* R176 */ - 0x0000, /* R177 */ - 0x0000, /* R178 */ - 0x0000, /* R179 */ - 0x0000, /* R180 */ - 0x0000, /* R181 */ - 0x0000, /* R182 */ - 0x0000, /* R183 */ - 0x0000, /* R184 */ - 0x0000, /* R185 */ - 0x0000, /* R186 */ - 0x0000, /* R187 */ - 0x0000, /* R188 */ - 0x0000, /* R189 */ - 0x0000, /* R190 */ - 0x0000, /* R191 */ - 0x0000, /* R192 */ - 0x0000, /* R193 */ - 0x0000, /* R194 */ - 0x0000, /* R195 */ - 0x0030, /* R196 */ - 0x0006, /* R197 */ - 0x0000, /* R198 */ - 0x0060, /* R199 */ - 0x0000, /* R200 */ - 0x003F, /* R201 */ - 0x0000, /* R202 */ - 0x0000, /* R203 */ - 0x0000, /* R204 */ - 0x0001, /* R205 */ - 0x0000, /* R206 */ - 0x0181, /* R207 */ - 0x0005, /* R208 */ - 0x0008, /* R209 */ - 0x0008, /* R210 */ - 0x0000, /* R211 */ - 0x013B, /* R212 */ - 0x0000, /* R213 */ - 0x0000, /* R214 */ - 0x0000, /* R215 */ - 0x0000, /* R216 */ - 0x0070, /* R217 */ - 0x0000, /* R218 */ - 0x0000, /* R219 */ - 0x0000, /* R220 */ - 0x0000, /* R221 */ - 0x0000, /* R222 */ - 0x0003, /* R223 */ - 0x0000, /* R224 */ - 0x0000, /* R225 */ - 0x0001, /* R226 */ - 0x0008, /* R227 */ - 0x0000, /* R228 */ - 0x0000, /* R229 */ - 0x0000, /* R230 */ - 0x0000, /* R231 */ - 0x0004, /* R232 */ - 0x0000, /* R233 */ - 0x0000, /* R234 */ - 0x0000, /* R235 */ - 0x0000, /* R236 */ - 0x0000, /* R237 */ - 0x0080, /* R238 */ - 0x0000, /* R239 */ - 0x0000, /* R240 */ - 0x0000, /* R241 */ - 0x0000, /* R242 */ - 0x0000, /* R243 */ - 0x0000, /* R244 */ - 0x0052, /* R245 */ - 0x0110, /* R246 */ - 0x0040, /* R247 */ - 0x0000, /* R248 */ - 0x0030, /* R249 */ - 0x0000, /* R250 */ - 0x0000, /* R251 */ - 0x0001, /* R252 - General test 1 */ +static const struct reg_default wm8961_reg_defaults[] = { + { 0, 0x009F }, /* R0 - Left Input volume */ + { 1, 0x009F }, /* R1 - Right Input volume */ + { 2, 0x0000 }, /* R2 - LOUT1 volume */ + { 3, 0x0000 }, /* R3 - ROUT1 volume */ + { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0008 }, /* R5 - ADC & DAC Control 1 */ + { 6, 0x0000 }, /* R6 - ADC & DAC Control 2 */ + { 7, 0x000A }, /* R7 - Audio Interface 0 */ + { 8, 0x01F4 }, /* R8 - Clocking2 */ + { 9, 0x0000 }, /* R9 - Audio Interface 1 */ + { 10, 0x00FF }, /* R10 - Left DAC volume */ + { 11, 0x00FF }, /* R11 - Right DAC volume */ + + { 14, 0x0040 }, /* R14 - Audio Interface 2 */ + + { 17, 0x007B }, /* R17 - ALC1 */ + { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x0032 }, /* R19 - ALC3 */ + { 20, 0x0000 }, /* R20 - Noise Gate */ + { 21, 0x00C0 }, /* R21 - Left ADC volume */ + { 22, 0x00C0 }, /* R22 - Right ADC volume */ + { 23, 0x0120 }, /* R23 - Additional control(1) */ + { 24, 0x0000 }, /* R24 - Additional control(2) */ + { 25, 0x0000 }, /* R25 - Pwr Mgmt (1) */ + { 26, 0x0000 }, /* R26 - Pwr Mgmt (2) */ + { 27, 0x0000 }, /* R27 - Additional Control (3) */ + { 28, 0x0000 }, /* R28 - Anti-pop */ + + { 30, 0x005F }, /* R30 - Clocking 3 */ + + { 32, 0x0000 }, /* R32 - ADCL signal path */ + { 33, 0x0000 }, /* R33 - ADCR signal path */ + + { 40, 0x0000 }, /* R40 - LOUT2 volume */ + { 41, 0x0000 }, /* R41 - ROUT2 volume */ + + { 47, 0x0000 }, /* R47 - Pwr Mgmt (3) */ + { 48, 0x0023 }, /* R48 - Additional Control (4) */ + { 49, 0x0000 }, /* R49 - Class D Control 1 */ + + { 51, 0x0003 }, /* R51 - Class D Control 2 */ + + { 56, 0x0106 }, /* R56 - Clocking 4 */ + { 57, 0x0000 }, /* R57 - DSP Sidetone 0 */ + { 58, 0x0000 }, /* R58 - DSP Sidetone 1 */ + + { 60, 0x0000 }, /* R60 - DC Servo 0 */ + { 61, 0x0000 }, /* R61 - DC Servo 1 */ + + { 63, 0x015E }, /* R63 - DC Servo 3 */ + + { 65, 0x0010 }, /* R65 - DC Servo 5 */ + + { 68, 0x0003 }, /* R68 - Analogue PGA Bias */ + { 69, 0x0000 }, /* R69 - Analogue HP 0 */ + + { 71, 0x01FB }, /* R71 - Analogue HP 2 */ + { 72, 0x0000 }, /* R72 - Charge Pump 1 */ + + { 82, 0x0000 }, /* R82 - Charge Pump B */ + + { 87, 0x0000 }, /* R87 - Write Sequencer 1 */ + { 88, 0x0000 }, /* R88 - Write Sequencer 2 */ + { 89, 0x0000 }, /* R89 - Write Sequencer 3 */ + { 90, 0x0000 }, /* R90 - Write Sequencer 4 */ + { 91, 0x0000 }, /* R91 - Write Sequencer 5 */ + { 92, 0x0000 }, /* R92 - Write Sequencer 6 */ + { 93, 0x0000 }, /* R93 - Write Sequencer 7 */ + + { 252, 0x0001 }, /* R252 - General test 1 */ }; struct wm8961_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; -static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8961_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: case WM8961_WRITE_SEQUENCER_7: case WM8961_DC_SERVO_1: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8961_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8961_LEFT_INPUT_VOLUME: + case WM8961_RIGHT_INPUT_VOLUME: + case WM8961_LOUT1_VOLUME: + case WM8961_ROUT1_VOLUME: + case WM8961_CLOCKING1: + case WM8961_ADC_DAC_CONTROL_1: + case WM8961_ADC_DAC_CONTROL_2: + case WM8961_AUDIO_INTERFACE_0: + case WM8961_CLOCKING2: + case WM8961_AUDIO_INTERFACE_1: + case WM8961_LEFT_DAC_VOLUME: + case WM8961_RIGHT_DAC_VOLUME: + case WM8961_AUDIO_INTERFACE_2: + case WM8961_SOFTWARE_RESET: + case WM8961_ALC1: + case WM8961_ALC2: + case WM8961_ALC3: + case WM8961_NOISE_GATE: + case WM8961_LEFT_ADC_VOLUME: + case WM8961_RIGHT_ADC_VOLUME: + case WM8961_ADDITIONAL_CONTROL_1: + case WM8961_ADDITIONAL_CONTROL_2: + case WM8961_PWR_MGMT_1: + case WM8961_PWR_MGMT_2: + case WM8961_ADDITIONAL_CONTROL_3: + case WM8961_ANTI_POP: + case WM8961_CLOCKING_3: + case WM8961_ADCL_SIGNAL_PATH: + case WM8961_ADCR_SIGNAL_PATH: + case WM8961_LOUT2_VOLUME: + case WM8961_ROUT2_VOLUME: + case WM8961_PWR_MGMT_3: + case WM8961_ADDITIONAL_CONTROL_4: + case WM8961_CLASS_D_CONTROL_1: + case WM8961_CLASS_D_CONTROL_2: + case WM8961_CLOCKING_4: + case WM8961_DSP_SIDETONE_0: + case WM8961_DSP_SIDETONE_1: + case WM8961_DC_SERVO_0: + case WM8961_DC_SERVO_1: + case WM8961_DC_SERVO_3: + case WM8961_DC_SERVO_5: + case WM8961_ANALOGUE_PGA_BIAS: + case WM8961_ANALOGUE_HP_0: + case WM8961_ANALOGUE_HP_2: + case WM8961_CHARGE_PUMP_1: + case WM8961_CHARGE_PUMP_B: + case WM8961_WRITE_SEQUENCER_1: + case WM8961_WRITE_SEQUENCER_2: + case WM8961_WRITE_SEQUENCER_3: + case WM8961_WRITE_SEQUENCER_4: + case WM8961_WRITE_SEQUENCER_5: + case WM8961_WRITE_SEQUENCER_6: + case WM8961_WRITE_SEQUENCER_7: + case WM8961_GENERAL_TEST_1: + return true; + default: + return false; } } @@ -958,11 +840,12 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { + struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -975,9 +858,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) } /* This isn't volatile - readback doesn't correspond to write */ - codec->cache_bypass = 1; + regcache_cache_bypass(wm8961->regmap, true); reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); - codec->cache_bypass = 0; + regcache_cache_bypass(wm8961->regmap, false); dev_info(codec->dev, "WM8961 family %d revision %c\n", (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) @@ -1066,10 +949,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8961_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8961_reg_defaults, - .volatile_register = wm8961_volatile_register, +}; + +static const struct regmap_config wm8961_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = WM8961_MAX_REGISTER, + + .reg_defaults = wm8961_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8961_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8961_volatile, + .readable_reg = wm8961_readable, }; static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, @@ -1083,6 +975,10 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, if (wm8961 == NULL) return -ENOMEM; + wm8961->regmap = devm_regmap_init_i2c(i2c, &wm8961_regmap); + if (IS_ERR(wm8961->regmap)) + return PTR_ERR(wm8961->regmap); + i2c_set_clientdata(i2c, wm8961); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.10.4

1 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa-mixer: Don't use the mono path in the analog-stereo mapping.
by Raymond Yau 13 Sep '12

13 Sep '12

2012-9-12 下午9:04 於 "Tanu Kaskinen" <tanuk(a)iki.fi> 寫道： > > As far as I can see, having a mono path in a stereo mapping doesn't > make any sense. It also causes breakage: if the Master Mono mixer > element has two volume channels, Because ac97 codecs have an analog mixer which can downmix stereo to the mono out pin. Some hda codecs also have the mono out pin which connected to a mixer widget for downmixing stereo from stereo DAC to mono The master mono playback volume control of ac97 has only one channel , so it is a bug of vm 's ac97 codec register emulation if the control has two volume channels Either removed the master mono register emulation from vm or pulseaudio ignore the master mono switch of vm's ac97 codec > the analog-output path gets removed > due to being a subset of analog-output-mono, and that in turn causes > the Master element getting muted. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… The above patch try to assign the headphone path for the USB headsets which have speaker playback volume control since speaker playback volume control is not in pulseaudio ' s headphone path. >Users generally don't like that. > > BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=54673 It look like a bug in output volume calculation of ac97 codec emulation in those virtual machine if the emulated ac97 codec only has 16 bits DAC resolution or pulseaudio does not unmute pcm playback switch on those ac97 codec after user mute the sound in sound preference and pulseaudio mute both master playback switch and pcm playback switch Some tablet with alc203 use the master mono output for the mono speaker http://git.alsa-project.org/?p=alsa-kernel.git;a=commitdiff;h=8f4f4ef6fed55… For als4000 the master mono is only enum control to change the channel map of the mono out Simple mixer control 'Master Mono',0 Capabilities: pswitch pswitch-joined cenum Items: 'L chan only' 'R chan only' 'L ch/2 + R ch/2' Item0: 'L chan only' https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/827393

1 0

Re: [alsa-devel] [PATCH 8/9] sound/soc/soc-core.c: Remove useless kfree
by Mark Brown 13 Sep '12

13 Sep '12

On Wed, Sep 12, 2012 at 05:06:46PM +0200, Peter Senna Tschudin wrote: > From: Peter Senna Tschudin <peter.senna(a)gmail.com> > > Remove useless kfree() and clean up code related to the removal. > > The semantic patch that finds this problem is as follows: > (http://coccinelle.lip6.fr/) Applied, thanks.

1 0

[alsa-devel] [RFC PATCH 0/5] mpc5200 asoc fixups
by Eric Millbrandt 13 Sep '12

13 Sep '12

The mpc5200 asoc code did not run after the multi-codec patches. This series fixes the existing code and adds a generic platform driver that allows the audio layout to be described in the device tree. Eric Millbrandt (5): ASoC: fsl: mpc5200 multi-codec fixups ASoC: fsl: mpc5200 combine psc_dma platform data ASoC: fsl: mpc5200-soc-audio driver arch/powerpc/boot/dts pcm030 add mpc5200-soc-audio node ASoC: fsl: mpc5200 remove pcm030 and efika audio fabric .../devicetree/bindings/powerpc/fsl/mpc5200.txt | 17 ++ arch/powerpc/boot/dts/pcm030.dts | 23 ++- arch/powerpc/platforms/52xx/Kconfig | 1 + sound/soc/fsl/Kconfig | 20 +-- sound/soc/fsl/Makefile | 3 +- sound/soc/fsl/efika-audio-fabric.c | 91 -------- sound/soc/fsl/mpc5200_dma.c | 28 +-- sound/soc/fsl/mpc5200_dma.h | 3 + sound/soc/fsl/mpc5200_psc_ac97.c | 13 +- sound/soc/fsl/mpc5200_psc_i2s.c | 8 + sound/soc/fsl/mpc5200_soc_audio.c | 232 ++++++++++++++++++++ sound/soc/fsl/pcm030-audio-fabric.c | 91 -------- 12 files changed, 306 insertions(+), 224 deletions(-) delete mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_soc_audio.c delete mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- 1.7.2.5

3 16

[alsa-devel] [PATCH 1/2] [v2] ASoC: fsl: use snd_soc_register_card to register the card
by Timur Tabi 13 Sep '12

13 Sep '12

Use snd_soc_register_card() instead of platform_device_alloc("soc-audio") to register the sound card from the machine drivers. Although platform_device_alloc is officially deprecated, it is still used by several other drivers. Unfortunately, something is broken somewhere and the Freescale drivers don't register properly when using it. Since we need to transition to snd_soc_register_card() anyway, this fixes the problem and updates our code at the same time. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 23 +++++------------------ sound/soc/fsl/p1022_ds.c | 22 ++++------------------ 2 files changed, 9 insertions(+), 36 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 60bcba1..5479882 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -192,7 +192,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; int ret = -ENODEV; const char *sprop; @@ -341,34 +340,22 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; machine_data->card.name = pdev->name; /* The platform driver name */ + machine_data->card.owner = THIS_MODULE; + machine_data->card.dev = &pdev->dev; machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&machine_data->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; + dev_err(&pdev->dev, "could not register card\n"); + goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; -error_sound: - platform_device_put(sound_device); error: kfree(machine_data); error_alloc: diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 50adf40..a44a8c3 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -202,7 +202,6 @@ static int p1022_ds_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct machine_data *mdata; int ret = -ENODEV; const char *sprop; @@ -349,36 +348,23 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->card.probe = p1022_ds_machine_probe; mdata->card.remove = p1022_ds_machine_remove; mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; mdata->card.num_links = 2; mdata->card.dai_link = mdata->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&mdata->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; error: - if (sound_device) - platform_device_put(sound_device); - kfree(mdata); error_put: of_node_put(codec_np); -- 1.7.3.4

2 5

[alsa-devel] [PATCH] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug unconditionally
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_enable_hdmi(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_disable_hdmi(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 5d02aad..d39ed58 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -676,8 +676,7 @@ static void intel_disable_hdmi(struct intel_encoder *encoder) u32 temp; u32 enable_bits = SDVO_ENABLE; - if (intel_hdmi->has_audio) - enable_bits |= SDVO_AUDIO_ENABLE; + enable_bits |= SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

3 3

[alsa-devel] [PATCH] drm/i915: HDMI - Clear Audio Enable bit for Hot Plug
by Wang Xingchao 13 Sep '12

13 Sep '12

Clear Audio Enable bit to trigger unsolicated event to notify Audio Driver part the HDMI hot plug change. The patch fixed the bug when remove HDMI cable the bit was not cleared correctly. In intel_hdmi_dpms(), if intel_hdmi->has_audio been true, the "Audio enable bit" will be set to trigger unsolicated event to notify Alsa driver the change. intel_hdmi->has_audio will be reset to false from intel_hdmi_detect() after remove the hdmi cable, here's debug log: [ 187.494153] [drm:output_poll_execute], [CONNECTOR:17:HDMI-A-1] status updated from 1 to 2 [ 187.525349] [drm:intel_hdmi_detect], HDMI: has_audio = 0 so when comes back to intel_hdmi_dpms(), the "Audio enable bit" will not be cleared. And this cause the eld infomation and pin presence doesnot update accordingly in alsa driver side. This patch will also trigger unsolicated event to alsa driver to notify the hot plug event: [ 187.853159] ALSA sound/pci/hda/patch_hdmi.c:772 HDMI hot plug event: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=1 [ 187.853268] ALSA sound/pci/hda/patch_hdmi.c:990 HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- drivers/gpu/drm/i915/intel_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_hdmi.c b/drivers/gpu/drm/i915/intel_hdmi.c index 98f6024..12dc330 100644 --- a/drivers/gpu/drm/i915/intel_hdmi.c +++ b/drivers/gpu/drm/i915/intel_hdmi.c @@ -609,7 +609,7 @@ static void intel_hdmi_dpms(struct drm_encoder *encoder, int mode) u32 temp; u32 enable_bits = SDVO_ENABLE; - if (intel_hdmi->has_audio) + if (intel_hdmi->has_audio || mode != DRM_MODE_DPMS_ON) enable_bits |= SDVO_AUDIO_ENABLE; temp = I915_READ(intel_hdmi->sdvox_reg); -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: core: Mark regmap CODEC register maps as dirty when suspending
by Mark Brown 12 Sep '12

12 Sep '12

The core has for a long time had support for marking the register maps of devices dirty when suspending so that they are resynced on resume. Also implement this feature for CODECs using regmap. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0c7e477..b94ce32 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -635,6 +635,8 @@ int snd_soc_suspend(struct device *dev) codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; + if (codec->using_regmap) + regcache_mark_dirty(codec->control_data); break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH - snd_ice1724 1/1] Use linear scale for AK4396 volume control.
by Takashi Iwai 12 Sep '12

12 Sep '12

At Wed, 12 Sep 2012 10:12:06 -0400, athena(a)fftw.org wrote: > > From: Matteo Frigo <athena(a)fftw.org> > > The AK4396 DAC has a linear-scale attentuator, but > sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is > not quite right. This patch restores the correct scale, borrowing > from the ak4396 code in sound/pci/oxygen/oxygen.c. > > Signed-off-by: Matteo Frigo <athena(a)fftw.org> Applied now. Thanks. Takashi > > diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c > index 7f2b63f..7bf093c 100644 > --- a/sound/pci/ice1712/prodigy_hifi.c > +++ b/sound/pci/ice1712/prodigy_hifi.c > @@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem > } > > static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); > +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); > > static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { > { > @@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { > .info = ak4396_dac_vol_info, > .get = ak4396_dac_vol_get, > .put = ak4396_dac_vol_put, > - .tlv = { .p = db_scale_wm_dac }, > + .tlv = { .p = ak4396_db_scale }, > }, > }; > > -- > 1.7.10.4 >

1 0

[alsa-devel] [PATCH] ASoC: wm8776: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8776.c | 75 ++++++++++++++++++++++++++++++++++++--------- 1 file changed, 60 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 879c356..c32249d 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/of_device.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <sound/core.h> @@ -37,18 +38,46 @@ enum wm8776_chip_type { /* codec private data */ struct wm8776_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk[2]; }; -static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { - 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ - 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ - 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ - 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ - 0xa6, 0x01, 0x01 +static const struct reg_default wm8776_reg_defaults[] = { + { 0, 0x79 }, + { 1, 0x79 }, + { 2, 0x79 }, + { 3, 0xff }, + { 4, 0xff }, + { 5, 0xff }, + { 6, 0x00 }, + { 7, 0x90 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x22 }, + { 11, 0x22 }, + { 12, 0x22 }, + { 13, 0x08 }, + { 14, 0xcf }, + { 15, 0xcf }, + { 16, 0x7b }, + { 17, 0x00 }, + { 18, 0x32 }, + { 19, 0x00 }, + { 20, 0xa6 }, + { 21, 0x01 }, + { 22, 0x01 }, }; +static bool wm8776_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8776_RESET: + return true; + default: + return false; + } +} + static int wm8776_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8776_RESET, 0); @@ -306,6 +335,8 @@ static int wm8776_set_sysclk(struct snd_soc_dai *dai, static int wm8776_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -313,7 +344,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8776->regmap); /* Disable the global powerdown; DAPM does the rest */ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); @@ -396,10 +427,9 @@ static int wm8776_resume(struct snd_soc_codec *codec) static int wm8776_probe(struct snd_soc_codec *codec) { - struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -434,9 +464,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .suspend = wm8776_suspend, .resume = wm8776_resume, .set_bias_level = wm8776_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8776_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8776_reg, .controls = wm8776_snd_controls, .num_controls = ARRAY_SIZE(wm8776_snd_controls), @@ -452,6 +479,18 @@ static const struct of_device_id wm8776_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8776_of_match); +static const struct regmap_config wm8776_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8776_RESET, + + .reg_defaults = wm8776_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8776_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8776_volatile, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8776_spi_probe(struct spi_device *spi) { @@ -463,7 +502,10 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) if (wm8776 == NULL) return -ENOMEM; - wm8776->control_type = SND_SOC_SPI; + wm8776->regmap = devm_regmap_init_spi(spi, &wm8776_regmap); + if (IS_ERR(wm8776->regmap)) + return PTR_ERR(wm8776->regmap); + spi_set_drvdata(spi, wm8776); ret = snd_soc_register_codec(&spi->dev, @@ -501,8 +543,11 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, if (wm8776 == NULL) return -ENOMEM; + wm8776->regmap = devm_regmap_init_i2c(i2c, &wm8776_regmap); + if (IS_ERR(wm8776->regmap)) + return PTR_ERR(wm8776->regmap); + i2c_set_clientdata(i2c, wm8776); - wm8776->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); -- 1.7.10.4

1 0

[alsa-devel] ASOC: dapm event errors not reported
by Pierre Peiffer 12 Sep '12

12 Sep '12

Hi, I have observed that in dapm, most of the errors of widget event calls are reported in kernel log but not returned to the caller. This leads in some cases to a situation where the error is not reported to the userspace caller, letting it continue, even if it can't and following calls from userspace will fail as well. Is there a reason for this ? Thanks & BR -- Pierre

1 0

[alsa-devel] Amixer: suppressing errors when called from stdin
by Henrik Fasth 12 Sep '12

12 Sep '12

Hi, We have some problems with amixer always returning 0 when called from stdin (exec_stdin). The global variable ignore_error is always set to 1 in function exec_stdin which leads to sset and cset always returning 0 even if an error occured. Are there any specific reasons for suppressing error messages and error return values like this? Br Henrik

1 0

[alsa-devel] [PATCH V3] ASoC: Davinci: machine: Add device tree binding
by Hebbar, Gururaja 12 Sep '12

12 Sep '12

Device tree support for Davinci Machine driver When the board boots with device tree, the driver will receive card, codec, dai interface details (like the card name, DAPM routing map, phandle for the audio components described in the dts file, codec mclk speed). The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. When here, code indentation and comment style is also fixed Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- Changes from V1: - Change DT parameter from "ti,codec-clock" to "ti,codec-clock-rate" - add more explanation to DT parameters Changes from V2: - Update Documentation details - Remove irrelevant commit message :000000 100644 0000000... b248014... A Documentation/devicetree/bindings/sound/davinci-evm-audio.txt :100644 100644 ab0ad45... ca2a547... M sound/soc/davinci/davinci-evm.c .../bindings/sound/davinci-evm-audio.txt | 66 +++++++ sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++-- 2 files changed, 231 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 0000000..b248014 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,66 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,davinci-evm-audio" +- ti,model : The user-visible name of this sound complex. +- ti,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the codec's pins, and the jacks on the board: + + Codec pins: + + * MIC3L + * MIC3R + * LINE1L + * LINE2L + * LINE1R + * LINE2R + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + +- ti,mcasp-controller : The phandle of the McASP controller +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec + +- dai-data : A list of DAI data used by SOC code to register + DAI, Codecs platform. + The string index "should" be as shown below. +dai-data = +"<DAI Name>", "<DAI Stream Name>", +"<CODEC DAI Name>", "<true if evm_aic3x_init is required, else false>", +"<evm ops required (evm_ops or evm_spdif_ops)>"; + +Here fields +"<DAI Name>" : used to indicate the DAI Name +"<DAI Stream Name>" : used to indicate the Stream Name +"<CODEC DAI Name>" : used to bind the link between Codec DAI and ASOC DAI + +Machine related options +"<true/false>" : Whether the machine specific initialization + : evm_aic3x_init() is required + +"<evm ops required>" : Which hardware ops function is to be used. + : (evm_ops or evm_spdif_ops) + : use evm-spdif-ops if DAI is working in DIT mode + : else use evm-ops. These ops setup hw param callbacks + : which are used to setup CODEC/cpu DAI configuration + : and codec system clock. + +Example: + +sound { + compatible = "ti,davinci-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + dai-data = + "TLV320AIC3X", "AIC3X", + "tlv320aic3x-hifi", "true", + "evm-ops"; +}; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index ab0ad45..ca2a547 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -34,27 +34,38 @@ static int evm_hw_params(struct snd_pcm_substream *substream, 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; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *soc_card = codec->card; + struct device_node *np = soc_card->dev->of_node; int ret = 0; unsigned sysclk; - /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || - machine_is_davinci_dm365_evm()) - sysclk = 27000000; - - /* ASP0 in DM6446 EVM is clocked by U55, as configured by - * board-dm644x-evm.c using GPIOs from U18. There are six - * options; here we "know" we use a 48 KHz sample rate. - */ - else if (machine_is_davinci_evm()) - sysclk = 12288000; - - else if (machine_is_davinci_da830_evm() || - machine_is_davinci_da850_evm()) - sysclk = 24576000; - - else - return -EINVAL; + if (np) { + ret = of_property_read_u32(np, "ti,codec-clock-rate", &sysclk); + if (ret < 0) + return ret; + } else { + /* ASP1 on DM355 EVM is clocked by an external oscillator */ + if (machine_is_davinci_dm355_evm() || + machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) + sysclk = 27000000; + + /* + * ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ + else if (machine_is_davinci_evm()) + sysclk = 12288000; + + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) + sysclk = 24576000; + + else + return -EINVAL; + } /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); @@ -127,13 +138,22 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + if (np) { + ret = snd_soc_of_parse_audio_routing(codec->card, + "ti,audio-routing"); + if (ret) + return ret; + } else { + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + } /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); @@ -282,6 +302,113 @@ static struct snd_soc_card da850_snd_soc_card = { .num_links = 1, }; +#if defined(CONFIG_OF) +/* + * This struct is just used as place holder. It will be filled with + * data from dt node + */ +static struct snd_soc_dai_link evm_dai = { +}; + +/* davinci evm audio machine driver */ +static struct snd_soc_card evm_soc_card = { + .owner = THIS_MODULE, + .dai_link = &evm_dai, + .num_links = 1, +}; + +static int davinci_evm_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + const char *stringPtr, *propname; + u32 val; + int ret = 0; + + propname = "dai-data"; + val = of_property_count_strings(np, propname); + if (val < 0) + return val; + + ret = of_property_read_string_index(np, propname, 0, &evm_dai.name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 1, + &evm_dai.stream_name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 2, + &evm_dai.codec_dai_name); + if (ret < 0) + return ret; + + ret = of_property_read_string_index(np, propname, 3, &stringPtr); + if (ret < 0) + return ret; + else + if (strcasecmp(stringPtr, "true") == 0) + evm_dai.init = evm_aic3x_init; + + ret = of_property_read_string_index(np, propname, 4, &stringPtr); + if (ret < 0) { + return ret; + } else { + if (strcasecmp(stringPtr, "evm-ops") == 0) + evm_dai.ops = &evm_ops; + else if (strcasecmp(stringPtr, "evm-spdif-ops") == 0) + evm_dai.ops = &evm_spdif_ops; + } + + evm_dai.codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); + if (!evm_dai.codec_of_node) + return -EINVAL; + + evm_dai.cpu_dai_of_node = of_parse_phandle(np, + "ti,mcasp-controller", 0); + if (!evm_dai.cpu_dai_of_node) + return -EINVAL; + + evm_dai.platform_of_node = evm_dai.cpu_dai_of_node; + + evm_soc_card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); + if (ret) + return ret; + + ret = snd_soc_register_card(&evm_soc_card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + + return ret; +} + +static int __devexit davinci_evm_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id davinci_evm_dt_ids[] = { + { .compatible = "ti,davinci-evm-audio", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, davinci_mcasp_dt_ids); + +static struct platform_driver davinci_evm_driver = { + .probe = davinci_evm_probe, + .remove = __devexit_p(davinci_evm_remove), + .driver = { + .name = "davinci_evm", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(davinci_evm_dt_ids), + }, +}; +#endif + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -289,6 +416,14 @@ static int __init evm_init(void) struct snd_soc_card *evm_snd_dev_data; int index; int ret; +#if defined(CONFIG_OF) + struct device_node *np; + + np = of_find_compatible_node(NULL, NULL, "ti,davinci-evm-audio"); + if (np) { + return platform_driver_register(&davinci_evm_driver); + } +#endif if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; @@ -325,6 +460,16 @@ static int __init evm_init(void) static void __exit evm_exit(void) { +#if defined(CONFIG_OF) + struct device_node *np; + + np = of_find_compatible_node(NULL, NULL, "ti,davinci-evm-audio"); + if (np) { + platform_driver_unregister(&davinci_evm_driver); + return; + } +#endif + platform_device_unregister(evm_snd_device); } -- 1.7.1

2 6

[alsa-devel] [PATCH] ALSA: hda_intel: add position_fix quirk for Asus K53E
by Catalin Iacob 12 Sep '12

12 Sep '12

Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob <iacobcatalin(a)gmail.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c6..228cdf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), -- 1.7.11.5

2 4

[alsa-devel] [PATCH 1/2] ASoC: wm8737: Move regulator acquisition to device registration
by Mark Brown 12 Sep '12

12 Sep '12

This is better style as we acquire resources we will need before we go into the ASoC card probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8737.c | 41 +++++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 8c25442..3bf5bc7 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -557,7 +557,7 @@ static int wm8737_resume(struct snd_soc_codec *codec) static int wm8737_probe(struct snd_soc_codec *codec) { struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - int ret, i; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8737->control_type); if (ret != 0) { @@ -565,16 +565,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) - wm8737->supplies[i].supply = wm8737_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8737->supplies), - wm8737->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { @@ -607,17 +597,12 @@ static int wm8737_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - return ret; } static int wm8737_remove(struct snd_soc_codec *codec) { - struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); return 0; } @@ -645,13 +630,23 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&i2c->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8737); wm8737->control_type = SND_SOC_I2C; @@ -691,13 +686,23 @@ static struct i2c_driver wm8737_i2c_driver = { static int __devinit wm8737_spi_probe(struct spi_device *spi) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&spi->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&spi->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + wm8737->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8737); -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm8728: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8728.c | 45 ++++++++++++++++++++++++++++----------------- 1 file changed, 28 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f274c25..00a12a0 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -17,6 +17,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -35,16 +36,16 @@ * the volume update bits, mute the output and enable infinite zero * detect. */ -static const u16 wm8728_reg_defaults[] = { - 0x1ff, - 0x1ff, - 0x001, - 0x100, +static const struct reg_default wm8728_reg_defaults[] = { + { 0, 0x1ff }, + { 1, 0x1ff }, + { 2, 0x001 }, + { 3, 0x100 }, }; /* codec private data */ struct wm8728_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); @@ -162,8 +163,8 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8728_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); u16 reg; - int i; switch (level) { case SND_SOC_BIAS_ON: @@ -175,9 +176,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ - for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - snd_soc_write(codec, i, - snd_soc_read(codec, i)); + regcache_sync(wm8728->regmap); } break; @@ -229,10 +228,9 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8728->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", ret); @@ -257,9 +255,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .suspend = wm8728_suspend, .resume = wm8728_resume, .set_bias_level = wm8728_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8728_reg_defaults, .controls = wm8728_snd_controls, .num_controls = ARRAY_SIZE(wm8728_snd_controls), .dapm_widgets = wm8728_dapm_widgets, @@ -274,6 +269,16 @@ static const struct of_device_id wm8728_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8728_of_match); +static const struct regmap_config wm8728_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8728_IFCTL, + + .reg_defaults = wm8728_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8728_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -285,7 +290,10 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) if (wm8728 == NULL) return -ENOMEM; - wm8728->control_type = SND_SOC_SPI; + wm8728->regmap = devm_regmap_init_spi(spi, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + spi_set_drvdata(spi, wm8728); ret = snd_soc_register_codec(&spi->dev, @@ -324,8 +332,11 @@ static __devinit int wm8728_i2c_probe(struct i2c_client *i2c, if (wm8728 == NULL) return -ENOMEM; + wm8728->regmap = devm_regmap_init_i2c(i2c, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + i2c_set_clientdata(i2c, wm8728); - wm8728->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8728, &wm8728_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8711: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8711.c | 38 ++++++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 1b1209a..5c165d0 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -32,7 +33,7 @@ /* codec private data */ struct wm8711_priv { - enum snd_soc_control_type bus_type; + struct regmap *regmap; unsigned int sysclk; }; @@ -42,9 +43,9 @@ struct wm8711_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { - 0x0079, 0x0079, 0x000a, 0x0008, - 0x009f, 0x000a, 0x0000, 0x0000 +static const struct reg_default wm8711_reg_defaults[] = { + { 0, 0x0079 }, { 1, 0x0079 }, { 2, 0x000a }, { 3, 0x0008 }, + { 4, 0x009f }, { 5, 0x000a }, { 6, 0x0000 }, { 7, 0x0000 }, }; #define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) @@ -289,6 +290,7 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; switch (level) { @@ -299,7 +301,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - snd_soc_cache_sync(codec); + regcache_sync(wm8711->regmap); snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; @@ -353,10 +355,9 @@ static int wm8711_resume(struct snd_soc_codec *codec) static int wm8711_probe(struct snd_soc_codec *codec) { - struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -391,9 +392,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .suspend = wm8711_suspend, .resume = wm8711_resume, .set_bias_level = wm8711_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8711_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8711_reg, .controls = wm8711_snd_controls, .num_controls = ARRAY_SIZE(wm8711_snd_controls), .dapm_widgets = wm8711_dapm_widgets, @@ -408,6 +406,16 @@ static const struct of_device_id wm8711_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8711_of_match); +static const struct regmap_config wm8711_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8711_RESET, + + .reg_defaults = wm8711_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8711_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -419,8 +427,11 @@ static int __devinit wm8711_spi_probe(struct spi_device *spi) if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_spi(spi, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + spi_set_drvdata(spi, wm8711); - wm8711->bus_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); @@ -458,8 +469,11 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *client, if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_i2c(client, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + i2c_set_clientdata(client, wm8711); - wm8711->bus_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8580: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 104 +++++++++++++++++++++++++++++++++------------ 1 file changed, 78 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cc198df..2a07c70 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008-11 Wolfson Microelectronics PLC. + * Copyright 2008-12 Wolfson Microelectronics PLC. * * 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 @@ -23,6 +23,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -157,21 +158,60 @@ * We can't read the WM8580 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8580_reg[] = { - 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ - 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ - 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ - 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R27*/ - 0x01f0, 0x0040, 0x0000, 0x0000, /*R31(0x1F)*/ - 0x0000, 0x0000, 0x0031, 0x000b, /*R35*/ - 0x0039, 0x0000, 0x0010, 0x0032, /*R39*/ - 0x0054, 0x0076, 0x0098, 0x0000, /*R43(0x2B)*/ - 0x0000, 0x0000, 0x0000, 0x0000, /*R47*/ - 0x0000, 0x0000, 0x005e, 0x003e, /*R51(0x33)*/ - 0x0000, 0x0000 /*R53*/ +static const struct reg_default wm8580_reg_defaults[] = { + { 0, 0x0121 }, + { 1, 0x017e }, + { 2, 0x007d }, + { 3, 0x0014 }, + { 4, 0x0121 }, + { 5, 0x017e }, + { 6, 0x007d }, + { 7, 0x0194 }, + { 8, 0x0010 }, + { 9, 0x0002 }, + { 10, 0x0002 }, + { 11, 0x00c2 }, + { 12, 0x0182 }, + { 13, 0x0082 }, + { 14, 0x000a }, + { 15, 0x0024 }, + { 16, 0x0009 }, + { 17, 0x0000 }, + { 18, 0x00ff }, + { 19, 0x0000 }, + { 20, 0x00ff }, + { 21, 0x00ff }, + { 22, 0x00ff }, + { 23, 0x00ff }, + { 24, 0x00ff }, + { 25, 0x00ff }, + { 26, 0x00ff }, + { 27, 0x00ff }, + { 28, 0x01f0 }, + { 29, 0x0040 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0031 }, + { 35, 0x000b }, + { 36, 0x0039 }, + { 37, 0x0000 }, + { 38, 0x0010 }, + { 39, 0x0032 }, + { 40, 0x0054 }, + { 41, 0x0076 }, + { 42, 0x0098 }, + { 43, 0x0000 }, + { 44, 0x0000 }, + { 45, 0x0000 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0000 }, + { 50, 0x005e }, + { 51, 0x003e }, + { 52, 0x0000 }, }; struct pll_state { @@ -188,7 +228,7 @@ static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { /* codec private data */ struct wm8580_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; struct pll_state a; struct pll_state b; @@ -203,14 +243,16 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; + struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); unsigned int reg = mc->reg; unsigned int reg2 = mc->rreg; int ret; - /* Clear the register cache so we write without VU set */ - reg_cache[reg] = 0; - reg_cache[reg2] = 0; + /* Clear the register cache VU so we write without VU set */ + regcache_cache_only(wm8580->regmap, true); + regmap_update_bits(wm8580->regmap, reg, 0x100, 0x000); + regmap_update_bits(wm8580->regmap, reg2, 0x100, 0x000); + regcache_cache_only(wm8580->regmap, false); ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) @@ -817,7 +859,7 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0,i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8580->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -875,9 +917,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8580_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8580_reg, .controls = wm8580_snd_controls, .num_controls = ARRAY_SIZE(wm8580_snd_controls), @@ -892,6 +931,16 @@ static const struct of_device_id wm8580_of_match[] = { { }, }; +static const struct regmap_config wm8580_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8580_MAX_REGISTER, + + .reg_defaults = wm8580_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8580_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -904,8 +953,11 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, if (wm8580 == NULL) return -ENOMEM; + wm8580->regmap = devm_regmap_init_i2c(i2c, &wm8580_regmap); + if (IS_ERR(wm8580->regmap)) + return PTR_ERR(wm8580->regmap); + i2c_set_clientdata(i2c, wm8580); - wm8580->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8580, wm8580_dai, ARRAY_SIZE(wm8580_dai)); -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm8510: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 105 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 80 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 98f28a0..d9a3ca6 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -33,22 +34,63 @@ * We can't read the WM8510 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0050, 0x0000, 0x0140, 0x0000, - 0x0000, 0x0000, 0x0000, 0x00ff, - 0x0000, 0x0000, 0x0100, 0x00ff, - 0x0000, 0x0000, 0x012c, 0x002c, - 0x002c, 0x002c, 0x002c, 0x0000, - 0x0032, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0038, 0x000b, 0x0032, 0x0000, - 0x0008, 0x000c, 0x0093, 0x00e9, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0003, 0x0010, 0x0000, 0x0000, - 0x0000, 0x0002, 0x0001, 0x0000, - 0x0000, 0x0000, 0x0039, 0x0000, - 0x0001, +static const struct reg_default wm8510_reg_defaults[] = { + { 1, 0x0000 }, + { 2, 0x0000 }, + { 3, 0x0000 }, + { 4, 0x0050 }, + { 5, 0x0000 }, + { 6, 0x0140 }, + { 7, 0x0000 }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x0000 }, + { 11, 0x00ff }, + { 12, 0x0000 }, + { 13, 0x0000 }, + { 14, 0x0100 }, + { 15, 0x00ff }, + { 16, 0x0000 }, + { 17, 0x0000 }, + { 18, 0x012c }, + { 19, 0x002c }, + { 20, 0x002c }, + { 21, 0x002c }, + { 22, 0x002c }, + { 23, 0x0000 }, + { 24, 0x0032 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0038 }, + { 33, 0x000b }, + { 34, 0x0032 }, + { 35, 0x0000 }, + { 36, 0x0008 }, + { 37, 0x000c }, + { 38, 0x0093 }, + { 39, 0x00e9 }, + { 40, 0x0000 }, + { 41, 0x0000 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0003 }, + { 45, 0x0010 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0002 }, + { 50, 0x0001 }, + { 51, 0x0000 }, + { 52, 0x0000 }, + { 53, 0x0000 }, + { 54, 0x0039 }, + { 55, 0x0000 }, + { 56, 0x0001 }, }; #define WM8510_POWER1_BIASEN 0x08 @@ -58,7 +100,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { /* codec private data */ struct wm8510_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; @@ -454,6 +496,7 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { @@ -467,7 +510,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8510->regmap); /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); @@ -536,10 +579,9 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8510->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); return ret; @@ -569,9 +611,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .suspend = wm8510_suspend, .resume = wm8510_resume, .set_bias_level = wm8510_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8510_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default =wm8510_reg, .controls = wm8510_snd_controls, .num_controls = ARRAY_SIZE(wm8510_snd_controls), @@ -586,6 +625,16 @@ static const struct of_device_id wm8510_of_match[] = { { }, }; +static const struct regmap_config wm8510_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8510_MONOMIX, + + .reg_defaults = wm8510_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8510_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -597,7 +646,10 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) if (wm8510 == NULL) return -ENOMEM; - wm8510->control_type = SND_SOC_SPI; + wm8510->regmap = devm_regmap_init_spi(spi, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + spi_set_drvdata(spi, wm8510); ret = snd_soc_register_codec(&spi->dev, @@ -635,8 +687,11 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, if (wm8510 == NULL) return -ENOMEM; + wm8510->regmap = devm_regmap_init_i2c(i2c, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + i2c_set_clientdata(i2c, wm8510); - wm8510->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); -- 1.7.10.4

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[alsa-devel] [PATCH 1/7] ASoC: wm8510: Convert to devm_kzalloc()
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 56a0495..98f28a0 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -592,7 +592,8 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&spi->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -601,8 +602,7 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } @@ -630,7 +630,8 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&i2c->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -639,8 +640,7 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } -- 1.7.10.4

1 6

[alsa-devel] [PATCH] ALSA: hda - Avoid BDL position workaround when no_period_wakeup is set
by Takashi Iwai 11 Sep '12

11 Sep '12

Originally the bogus period at BDL head was introduced as a workaround for the mismatching position update at the period boundary, typically seen on dmix. However, for applications like PulseAudio that don't require period wake ups, this workaround is just superfluous. Thus better to disable it when no_period_wakeup is given in hw_params. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6a19f6a..76ec242 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,7 @@ struct azx_dev { */ unsigned int insufficient :1; unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; }; /* CORB/RIRB */ @@ -1419,7 +1420,7 @@ static int azx_setup_periods(struct azx *chip, ofs = 0; azx_dev->frags = 0; pos_adj = bdl_pos_adj[chip->dev_index]; - if (pos_adj > 0) { + if (!azx_dev->no_period_wakeup && pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; int pos_align = pos_adj; pos_adj = (pos_adj * runtime->rate + 47999) / 48000; @@ -1435,8 +1436,7 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, - &bdl, ofs, pos_adj, - !substream->runtime->no_period_wakeup); + &bdl, ofs, pos_adj, true); if (ofs < 0) goto error; } @@ -1449,7 +1449,7 @@ static int azx_setup_periods(struct azx *chip, else ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes, - !substream->runtime->no_period_wakeup); + !azx_dev->no_period_wakeup); if (ofs < 0) goto error; } @@ -1922,10 +1922,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || - format_val != azx_dev->format_val) { + format_val != azx_dev->format_val || + runtime->no_period_wakeup != azx_dev->no_period_wakeup) { azx_dev->bufsize = bufsize; azx_dev->period_bytes = period_bytes; azx_dev->format_val = format_val; + azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) return err; -- 1.7.11.5

1 0