December 2014 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Laptop Asus Z99He internal speaker not working until I enable EAPD with HDA-Analizer
by Takashi Iwai 10 Dec '14

10 Dec '14

At Wed, 10 Dec 2014 15:28:09 +0300, Dmitry V. Zimin wrote: > > > Yes, it works. Thank you so much! Good to hear, the patch is queued now. Takashi > > > Tue, 09 Dec 2014 20:01:36 +0100 от Takashi Iwai <tiwai(a)suse.de>: > >At Tue, 09 Dec 2014 18:01:44 +0300, > >Dmitry V. Zimin wrote: > >> > >> Hello! > >> > >> I have laptop Asus Z99He and internal speaker doesn't work until I enable EAPD through HDA-Analyzer by hand. After connect and disconnect external headphones EAPD setting reset again. > >> How to enable it permanently? > >> model=eapd, laptop-eapd or many other does not change anything. > >> http://www.alsa-project.org/db/?f=df83e22e0c9a047db3778f394c1aa73376dfaf39 > >> > >> Also I try old Live CD with OpenELEC. He is working properly . http://www.alsa-project.org/db/?f=a5e75412c8230afe9516f0ebf6404afb18d9040c > > > >Could you try the patch below? > > > > > >Takashi > > > >-- 8< -- > >From: Takashi Iwai < tiwai(a)suse.de > > >Subject: [PATCH] ALSA: hda - Add EAPD fixup for ASUS Z99He laptop > > > >The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A > >codec like another ASUS machine. > > > >Reported-by: Dmitry V. Zimin < pfzim(a)mail.ru > > >Cc: < stable(a)vger.kernel.org > > >Signed-off-by: Takashi Iwai < tiwai(a)suse.de > > >--- > > sound/pci/hda/patch_analog.c | 1 + > > 1 file changed, 1 insertion(+) > > > >diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c > >index 06275f8807a8..28815efeab06 100644 > >--- a/sound/pci/hda/patch_analog.c > >+++ b/sound/pci/hda/patch_analog.c > >@@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { > > > > static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { > > SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), > >+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), > > SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), > > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), > > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), > >-- > >2.2.0 > > > > > > [2 <text/html; utf-8 (base64)>] >

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[alsa-devel] [patch] ALSA: pcxhr: NULL dereference on probe failure
by Dan Carpenter 10 Dec '14

10 Dec '14

"card" is NULL if snd_card_new() fails. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index a602930..c6092e4 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1638,7 +1638,7 @@ static int pcxhr_probe(struct pci_dev *pci, 0, &card); if (err < 0) { - dev_err(card->dev, "cannot allocate the card %d\n", i); + dev_err(&pci->dev, "cannot allocate the card %d\n", i); pcxhr_free(mgr); return err; }

2 1

[alsa-devel] [patch] ALSA: lola: NULL dereference on probe failure
by Dan Carpenter 10 Dec '14

10 Dec '14

"card" is NULL if snd_card_new() fails. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index a75c8dc..4cf4be5 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -719,7 +719,7 @@ static int lola_probe(struct pci_dev *pci, err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - dev_err(card->dev, "Error creating card!\n"); + dev_err(&pci->dev, "Error creating card!\n"); return err; }

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[alsa-devel] [PATCH 00/15 v5] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 10 Dec '14

10 Dec '14

This patchset is the rest of patches to enable ALSA Dice and OXFW driver to handle more models. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/30 v4] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085092.h… - use the same functions for each of MIDI .open/.close callback (dice) - fix a comment for AV/C stream format information command (oxfw) - revert shortname template, just use model name (oxfw) I note that OXFW driver uses the shortname for MIDI substream name, which has 32 bytes, thus the string in shortname should not be so long. == Supplemental patch for userspace library This patchset adds Hwdep functionality for OXFW driver, thus userspace library needs a patch for entries of snd_hwdep_iface_t. Please apply below patch. [alsa-devel] [PATCH alsa-lib] hwdep: add OXFW driver support http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084824.h… Takashi Sakamoto (15): ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 10 +- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 157 ++++++++ sound/firewire/dice/dice-pcm.c | 251 +++++++++---- sound/firewire/dice/dice-stream.c | 332 ++++++++++++---- sound/firewire/dice/dice-transaction.c | 5 - sound/firewire/dice/dice.c | 37 +- sound/firewire/dice/dice.h | 33 +- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-command.c | 153 ++++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++++++++ sound/firewire/oxfw/oxfw-pcm.c | 425 +++++++++++++++------ sound/firewire/oxfw/oxfw-proc.c | 113 ++++++ sound/firewire/oxfw/oxfw-stream.c | 667 +++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 179 +++++++-- sound/firewire/oxfw/oxfw.h | 104 ++++- 19 files changed, 2499 insertions(+), 377 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c -- 2.1.0

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[alsa-devel] ALSA driver installation issue in Ubuntu 10.04 OS
by Bruce Teng 10 Dec '14

10 Dec '14

Hi Sir, I've encountered the following error when run the "alsa-utils-alsa-compile.sh" on Ubuntu 10.04 with 2.6.32 kernel OS: _____ /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 127: warning: initialization from incompatible pointer type /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 132: warning: initialization from incompatible pointer type /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: In function 'set_pages_array_wc': /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 700: error: implicit declaration of function 'set_pages_wc' /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: At top level: /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 703: error: conflicting types for 'set_pages_array_wb' /usr/src/linux-headers-2.6.32-21-generic/arch/x86/include/asm/cacheflush.h:1 44: note: previous declaration of 'set_pages_array_wb' was here /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: In function 'param_set_xint': /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.c: 2937: warning: passing argument 2 of 'param_set_int' discards qualifiers from pointer target type include/linux/moduleparam.h:177: note: expected 'struct kernel_param *' but argument is of type 'const struct kernel_param *' make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda/hda_intel.o ] Error 1 make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci/hda] Error 2 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/pci] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.32-21-generic' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. _____ Could you help to look into this issue? Thanks! Bruce

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[alsa-devel] Question on arm64 unaligned faults during playback
by Abhilash Kesavan 10 Dec '14

10 Dec '14

Hi, I am working on a 64-bit ARM SoC (Samsung's Exynos7) and have observed unaligned faults while testing certain sound streams with aplay. Playing a 44.1k sampling rate stream gives the following crash while 48k streams work fine. [root@(none) ~]# aplay test2_16bit_44k.wav Playing WAVE 'test2_16bit_44k.wav' : Signed 1[ 24.500724] sound: 2ch, 44100Hz, 88208bytes 6 bit Little Endian, Rate 44100 Hz, Stereo [ 24.520738] snd_pcm_lib_write_transfer:sound/core/pcm_lib.c hwbuf is ffffff8000080000, runtime->dma_area is ffffff8000080000, hwoff is 0, frames_to_bytes is 0, frames is 5513 [ 24.535661] snd_pcm_lib_write_transfer:sound/core/pcm_lib.c hwbuf is ffffff8000085624, runtime->dma_area is ffffff8000080000, hwoff is 5513, frames_to_bytes is 22052, frames is 5513 [ 24.551244] Unhandled fault: alignment fault (0x96000061) at 0xffffff8000085624 [ 24.558448] Internal error: : 96000061 [#1] PREEMPT SMP [ 24.563449] Modules linked in: [ 24.566489] CPU: 0 PID: 928 Comm: aplay Not tainted 3.16.0+ #232 [ 24.572473] task: ffffffc0a934c380 ti: ffffffc0038d4000 task.ti: ffffffc0038d4000 [ 24.579944] PC is at __copy_from_user+0x14/0x60 [ 24.584450] LR is at snd_pcm_lib_write_transfer+0xe4/0x104 [ 24.589912] pc : [<ffffffc0002d4784>] lr : [<ffffffc0004b58f0>] pstate: 20000145 [ 24.597285] sp : ffffffc0038d7c90 [ 24.600583] x29: ffffffc0038d7c90 x28: ffffffc0038c1c00 [ 24.605876] x27: ffffffc0a93eae00 x26: 0000000000001589 [ 24.611168] x25: 0000000000000000 x24: 0000000000000000 [ 24.616461] x23: 0000000000000008 x22: ffffff8000085624 [ 24.621753] x21: 0000000000005624 x20: ffffffc0038c1c00 [ 24.627046] x19: 0000000011d69620 x18: 0000007fd2e8b1c0 [ 24.632339] x17: 0000007fa8057370 x16: ffffffc000196108 [ 24.637632] x15: 232b226b00000000 x14: 617266202c333135 [ 24.642925] x13: 352073692066666f x12: 7768202c30303030 [ 24.648218] x11: 3830303030386666 x10: 6666666620736920 [ 24.653510] x9 : 616572615f616d64 x8 : 000000000000026f [ 24.658803] x7 : ffffffc000998790 x6 : 0000000000000001 [ 24.664095] x5 : 0000000000000000 x4 : 0000000011d6ec44 [ 24.669388] x3 : f459f780f2b3f6d4 x2 : 0000000000005614 [ 24.674681] x1 : 0000000011d69628 x0 : ffffff8000085624 [ 24.679973] [ 24.681451] Process aplay (pid: 928, stack limit = 0xffffffc0038d4058) [ 24.687959] Stack: (0xffffffc0038d7c90 to 0xffffffc0038d8000) [ 24.693688] 7c80: 038d7cd0 ffffffc0 004b7578 ffffffc0 [ 24.701846] 7ca0: 00001589 00000000 a93eaf50 ffffffc0 009ef000 ffffffc0 11d69620 00000000 [ 24.710001] 7cc0: 004b580c ffffffc0 004b755c ffffffc0 038d7d70 ffffffc0 004b7758 ffffffc0 [ 24.718158] 7ce0: a93eae00 ffffffc0 d2e8b438 0000007f a91eb858 ffffffc0 d2e8b438 0000007f [ 24.726313] 7d00: 40184150 00000000 d2e8b438 0000007f 00000115 00000000 0000001d 00000000 [ 24.734470] 7d20: 00960000 ffffffc0 038d4000 ffffffc0 0000000a 00000000 038d7d68 ffffffc0 [ 24.742626] 7d40: 00001589 00000000 00001589 00000000 00000000 00000000 009ef898 ffffffc0 [ 24.750782] 7d60: a8b6d800 ffffffc0 0000409b 00000000 038d7da0 ffffffc0 004b2dd0 ffffffc0 [ 24.758938] 7d80: a93eae00 ffffffc0 11d69620 00000000 11d69620 00000000 00001589 00000000 [ 24.767094] 7da0: 038d7df0 ffffffc0 004b2e10 ffffffc0 a8b6d900 ffffffc0 d2e8b438 0000007f [ 24.775249] 7dc0: a91eb858 ffffffc0 00000004 00000000 00000001 00000000 00000000 00000000 [ 24.783406] 7de0: 11d69620 00000000 00001589 00000000 038d7e00 ffffffc0 00195ecc ffffffc0 [ 24.791563] 7e00: 038d7e90 ffffffc0 0019618c ffffffc0 00000000 00000000 a8b6d900 ffffffc0 [ 24.799718] 7e20: a8b6d900 ffffffc0 00000004 00000000 40184150 00000000 00185888 ffffffc0 [ 24.807875] 7e40: a8b6d800 ffffffc0 a8b6d800 ffffffc0 ffffffff ffffffff a80f0588 0000007f [ 24.816030] 7e60: 80000000 00000000 00000015 00000000 038d7e80 ffffffc0 001a0340 ffffffc0 [ 24.824187] 7e80: 038d7e90 ffffffc0 00196148 ffffffc0 d2e8b410 0000007f 0008425c ffffffc0 [ 24.832343] 7ea0: 00000000 00000000 11d500e0 00000000 ffffffff ffffffff a805737c 0000007f [ 24.840499] 7ec0: 60000000 00000000 00000015 00000000 00000004 00000000 40184150 00000000 [ 24.848655] 7ee0: d2e8b438 0000007f a8223fbc 0000007f 00000003 00000000 00000000 00000000 [ 24.856811] 7f00: a82d46f0 0000007f 00000000 00000000 0000001d 00000000 00000120 00000000 [ 24.864967] 7f20: 01010101 01010101 00000016 00000000 00000003 00000000 11d510d0 00000000 [ 24.873123] 7f40: 00000000 00000000 00000000 232b226b a8299b80 0000007f a8057370 0000007f [ 24.881279] 7f60: d2e8b1c0 0000007f 11d691f0 00000000 11d500e0 00000000 00001589 00000000 [ 24.889435] 7f80: 11d69620 00000000 00000000 00000000 0041f7d8 00000000 00000003 00000000 [ 24.897591] 7fa0: 00000000 00000000 00000004 00000000 00000003 00000000 d2e8b410 0000007f [ 24.905748] 7fc0: a8223ff0 0000007f d2e8b410 0000007f a805737c 0000007f 60000000 00000000 [ 24.913904] 7fe0: 00000004 00000000 0000001d 00000000 d10d0579 f13ea086 786e62ff 1f2437e3 [ 24.922054] Call trace: [ 24.924488] [<ffffffc0002d4784>] __copy_from_user+0x14/0x60 [ 24.930040] [<ffffffc0004b7574>] snd_pcm_lib_write1+0x1fc/0x384 [ 24.935940] [<ffffffc0004b7754>] snd_pcm_lib_write+0x58/0x80 [ 24.941580] [<ffffffc0004b2dcc>] snd_pcm_playback_ioctl1+0x2e8/0x300 [ 24.947913] [<ffffffc0004b2e0c>] snd_pcm_playback_ioctl+0x28/0x44 [ 24.953990] [<ffffffc000195ec8>] do_vfs_ioctl+0x370/0x5b0 [ 24.959367] [<ffffffc000196188>] SyS_ioctl+0x80/0x98 [ 24.964314] Code: f1002042 540000a4 f8408423 f1002042 (f8008403) [ 24.976808] ---[ end trace 35415135d4341fc9 ]--- We are using the internal sram available for sound, for DMA buffer allocation, using the generic SRAM driver. From the above log, the buffer address offset is not 8-byte aligned and as we are using the SRAM driver which maps the memory as device memory we are getting an unaligned fault. Is it incorrect to use the generic sram driver for arm64 or am I missing something ? Regards, Abhilash

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[alsa-devel] Laptop Asus Z99He internal speaker not working until I enable EAPD with HDA-Analizer
by Dmitry V. Zimin 09 Dec '14

09 Dec '14

Hello! I have laptop Asus Z99He and internal speaker doesn't work until I enable EAPD through HDA-Analyzer by hand. After connect and disconnect external headphones EAPD setting reset again. How to enable it permanently? model=eapd, laptop-eapd or many other does not change anything. http://www.alsa-project.org/db/?f=df83e22e0c9a047db3778f394c1aa73376dfaf39 Also I try old Live CD with OpenELEC. He is working properly . http://www.alsa-project.org/db/?f=a5e75412c8230afe9516f0ebf6404afb18d9040c

2 1

[alsa-devel] [PATCH] sound / PM: Replace CONFIG_PM_RUNTIME with CONFIG_PM
by Rafael J. Wysocki 09 Dec '14

09 Dec '14

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> After commit b2b49ccbdd54 (PM: Kconfig: Set PM_RUNTIME if PM_SLEEP is selected) PM_RUNTIME is always set if PM is set, so #ifdef blocks depending on CONFIG_PM_RUNTIME may now be changed to depend on CONFIG_PM. Replace CONFIG_PM_RUNTIME with CONFIG_PM everywhere under sound/. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- Note: This depends on commit b2b49ccbdd54 (PM: Kconfig: Set PM_RUNTIME if PM_SLEEP is selected) which is only in linux-next at the moment (via the linux-pm tree). Please let me know if it is OK to take this one into linux-pm. --- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 5 +---- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/tas2552.c | 4 ++-- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 ++-- sound/soc/samsung/i2s.c | 4 ++-- 12 files changed, 15 insertions(+), 18 deletions(-) Index: linux-pm/sound/pci/hda/hda_controller.c =================================================================== --- linux-pm.orig/sound/pci/hda/hda_controller.c +++ linux-pm/sound/pci/hda/hda_controller.c @@ -1676,7 +1676,7 @@ irqreturn_t azx_interrupt(int irq, void u8 sd_status; int i; -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; Index: linux-pm/sound/pci/hda/hda_intel.c =================================================================== --- linux-pm.orig/sound/pci/hda/hda_intel.c +++ linux-pm/sound/pci/hda/hda_intel.c @@ -856,7 +856,7 @@ static int azx_resume(struct device *dev } #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -954,9 +954,6 @@ static int azx_runtime_idle(struct devic return 0; } -#endif /* CONFIG_PM_RUNTIME */ - -#ifdef CONFIG_PM static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) Index: linux-pm/sound/soc/codecs/cs35l32.c =================================================================== --- linux-pm.orig/sound/soc/codecs/cs35l32.c +++ linux-pm/sound/soc/codecs/cs35l32.c @@ -550,7 +550,7 @@ static int cs35l32_i2c_remove(struct i2c return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int cs35l32_runtime_suspend(struct device *dev) { struct cs35l32_private *cs35l32 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/cs42xx8.c =================================================================== --- linux-pm.orig/sound/soc/codecs/cs42xx8.c +++ linux-pm/sound/soc/codecs/cs42xx8.c @@ -537,7 +537,7 @@ err_enable: } EXPORT_SYMBOL_GPL(cs42xx8_probe); -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int cs42xx8_runtime_resume(struct device *dev) { struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/max98090.c =================================================================== --- linux-pm.orig/sound/soc/codecs/max98090.c +++ linux-pm/sound/soc/codecs/max98090.c @@ -2492,7 +2492,7 @@ static int max98090_i2c_remove(struct i2 return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/pcm512x.c =================================================================== --- linux-pm.orig/sound/soc/codecs/pcm512x.c +++ linux-pm/sound/soc/codecs/pcm512x.c @@ -517,7 +517,7 @@ void pcm512x_remove(struct device *dev) } EXPORT_SYMBOL_GPL(pcm512x_remove); -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int pcm512x_suspend(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/tas2552.c =================================================================== --- linux-pm.orig/sound/soc/codecs/tas2552.c +++ linux-pm/sound/soc/codecs/tas2552.c @@ -115,7 +115,7 @@ static const struct snd_soc_dapm_route t {"ClassD", NULL, "PLL"}, }; -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) { u8 cfg1_reg; @@ -264,7 +264,7 @@ static int tas2552_mute(struct snd_soc_d return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int tas2552_runtime_suspend(struct device *dev) { struct tas2552_data *tas2552 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/wm2200.c =================================================================== --- linux-pm.orig/sound/soc/codecs/wm2200.c +++ linux-pm/sound/soc/codecs/wm2200.c @@ -2440,7 +2440,7 @@ static int wm2200_i2c_remove(struct i2c_ return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int wm2200_runtime_suspend(struct device *dev) { struct wm2200_priv *wm2200 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/wm5100.c =================================================================== --- linux-pm.orig/sound/soc/codecs/wm5100.c +++ linux-pm/sound/soc/codecs/wm5100.c @@ -2664,7 +2664,7 @@ static int wm5100_i2c_remove(struct i2c_ return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int wm5100_runtime_suspend(struct device *dev) { struct wm5100_priv *wm5100 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/codecs/wm8962.c =================================================================== --- linux-pm.orig/sound/soc/codecs/wm8962.c +++ linux-pm/sound/soc/codecs/wm8962.c @@ -3782,7 +3782,7 @@ static int wm8962_i2c_remove(struct i2c_ return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); Index: linux-pm/sound/soc/fsl/fsl_asrc.c =================================================================== --- linux-pm.orig/sound/soc/fsl/fsl_asrc.c +++ linux-pm/sound/soc/fsl/fsl_asrc.c @@ -928,7 +928,7 @@ static int fsl_asrc_probe(struct platfor return 0; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); @@ -954,7 +954,7 @@ static int fsl_asrc_runtime_suspend(stru return 0; } -#endif /* CONFIG_PM_RUNTIME */ +#endif /* CONFIG_PM */ #ifdef CONFIG_PM_SLEEP static int fsl_asrc_suspend(struct device *dev) Index: linux-pm/sound/soc/samsung/i2s.c =================================================================== --- linux-pm.orig/sound/soc/samsung/i2s.c +++ linux-pm/sound/soc/samsung/i2s.c @@ -1113,7 +1113,7 @@ static inline const struct samsung_i2s_d platform_get_device_id(pdev)->driver_data; } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int i2s_runtime_suspend(struct device *dev) { struct i2s_dai *i2s = dev_get_drvdata(dev); @@ -1131,7 +1131,7 @@ static int i2s_runtime_resume(struct dev return 0; } -#endif /* CONFIG_PM_RUNTIME */ +#endif /* CONFIG_PM */ static int samsung_i2s_probe(struct platform_device *pdev) {

5 5

[alsa-devel] [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 09 Dec '14

09 Dec '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a Codec DAI driver that controls the SOC external pins - a CPU DAI driver for controlling the LPASS-LPAIF block - a PCM MI2S platform driver - a machine driver that ties the three drivers together Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. == Updates from my previous post: [PATCH 00/9] ASoC: QCOM: Add support for ipq806x SOC http://thread.gmane.org/gmane.linux.ports.arm.msm/10701 - remove the native LPAIF driver, and move its functionality to the CPU DAI driver - add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver) - use devm_* and dev_* where possible - ISR only handles relevant DMA channel now - update device tree documentation to reflect changes - general code cleanup == Additional patch series upon which this patch series has device-tree dependencies (lcc) [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 Kenneth Westfield (11): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Add device tree binding docs ASoC: ipq806x: add LPAIF header file ASoC: codec: Add Maxim codec driver ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: ipq806x: Add machine driver for IPQ806X SOC ASoC: codec: Add ability to build QCOM codec ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../bindings/sound/qcom,ipq806x-snd-card.txt | 25 ++ .../bindings/sound/qcom,lpass-cpu-dai.txt | 32 ++ .../bindings/sound/qcom,lpass-pcm-mi2s.txt | 12 + .../bindings/sound/qcom,max98357a-codec.txt | 23 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 33 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 267 ++++++++++++ sound/soc/qcom/Kconfig | 25 ++ sound/soc/qcom/Makefile | 11 + sound/soc/qcom/ipq806x.c | 127 ++++++ sound/soc/qcom/lpass-cpu-mi2s.c | 374 +++++++++++++++++ sound/soc/qcom/lpass-cpu-mi2s.h | 48 +++ sound/soc/qcom/lpass-lpaif.h | 154 +++++++ sound/soc/qcom/lpass-pcm-mi2s.c | 452 +++++++++++++++++++++ 18 files changed, 1598 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,ipq806x-snd-card.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-dai.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-pcm-mi2s.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,max98357a-codec.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/ipq806x.c create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.c create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.h create mode 100644 sound/soc/qcom/lpass-lpaif.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c -- 1.8.2.1

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[alsa-devel] I2S frame configuration
by thomas chen 09 Dec '14

09 Dec '14

I am working on a ALSA interface to a particular codec over I2S the audio stream format is a bit peculiar... (CODEC being MASTER) there are 24 BCLK cycle between transition of FSYNC... howver, there are only 16 bit that are valid bit 0: ignore bit 1-16: valid pcm data (MSB....LSB) bit 17-23: ignore Can anyone suggest how to configure the DAI format ??? Thanks

3 5

[alsa-devel] [PATCH] ASoC: samsung: i2s: Add missing assignment of variant_regs
by Sylwester Nawrocki 09 Dec '14

09 Dec '14

Add assignment of the variant_regs field which is missing in commit a5a56871f804edac93a53b5e871c0e9818fb9033 ("ASoC: samsung: add support for exynos7 I2S controller"). Without this attempting to probe the secondary DAI fails with an error like: [ 1.763026] Unable to handle kernel NULL pointer dereference at virtual address 0000000c [ 1.780895] pgd = c0004000 [ 1.783606] [0000000c] *pgd=00000000 [ 1.838255] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.843514] Modules linked in: [ 1.846558] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.18.0-rc1-00009-g5dcb01e-dirty #1521 [ 1.854887] task: ee00a800 ti: ee088000 task.ti: ee088000 [ 1.860284] PC is at i2s_txctrl+0x40/0x2d4 [ 1.864350] LR is at i2s_txctrl+0x28/0x2d4 [ 1.868428] pc : [<c036ffd4>] lr : [<c036ffbc>] psr: 60000153 [ 1.868428] sp : ee089dc0 ip : 00000000 fp : ee21f000 [ 1.879883] r10: 00000000 r9 : ee21fb00 r8 : c06406c4 [ 1.885091] r7 : ee21fb00 r6 : 00000000 r5 : f00f6000 r4 : ed943410 [ 1.891601] r3 : 0000016c r2 : c0464550 r1 : c055cef8 r0 : ed943610 [ 1.898113] Flags: nZCv IRQs on FIQs off Mode SVC_32 ISA ARM Segment kernel [ 1.905490] Control: 10c5387d Table: 4000404a DAC: 00000015 [ 1.911218] Process swapper/0 (pid: 1, stack limit = 0xee088240) [ 1.917208] Stack: (0xee089dc0 to 0xee08a000) ... [ 2.068431] [<c036ffd4>] (i2s_txctrl) from [<c03719fc>] (samsung_i2s_dai_probe+0xb8/0x450) [ 2.076676] [<c03719fc>] (samsung_i2s_dai_probe) from [<c03607e0>] (snd_soc_register_card+0xd98/0x1348) [ 2.086044] [<c03607e0>] (snd_soc_register_card) from [<c03726e4>] (odroidx2_audio_probe+0xa8/0x11c) [ 2.095160] [<c03726e4>] (odroidx2_audio_probe) from [<c0249dd0>] (platform_drv_probe+0x48/0xa4) [ 2.103922] [<c0249dd0>] (platform_drv_probe) from [<c0248988>] (driver_probe_device+0x10c/0x22c) [ 2.112773] [<c0248988>] (driver_probe_device) from [<c0248b34>] (__driver_attach+0x8c/0x90) [ 2.121192] [<c0248b34>] (__driver_attach) from [<c02471c8>] (bus_for_each_dev+0x54/0x88) [ 2.129352] [<c02471c8>] (bus_for_each_dev) from [<c0248188>] (bus_add_driver+0xd4/0x1d0) [ 2.137510] [<c0248188>] (bus_add_driver) from [<c024915c>] (driver_register+0x78/0xf4) [ 2.145499] [<c024915c>] (driver_register) from [<c0008924>] (do_one_initcall+0x80/0x1b8) [ 2.153670] [<c0008924>] (do_one_initcall) from [<c05b7d40>] (kernel_init_freeable+0xfc/0x1c8) [ 2.162260] [<c05b7d40>] (kernel_init_freeable) from [<c04146c0>] (kernel_init+0x8/0xec) [ 2.170330] [<c04146c0>] (kernel_init) from [<c000e7f8>] (ret_from_fork+0x14/0x3c) [ 2.177873] Code: e5940000 e59f128c e59f228c e2800010 (e59c700c) Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index c7aafcd..eca17dc 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1261,6 +1261,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } + + sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.ch_name = "tx-sec"; -- 1.7.9.5

3 2

[alsa-devel] [PATCH v2] ASoC: rt5645: Fix potential crash in jd function
by Bard Liao 09 Dec '14

09 Dec '14

If no one defined the rt5645->pdata.hp_det_gpio in coreboot/bios. It will cause kernel to reboot because rt5645->pdata.hp_det_gpio is 0. So it is worth to add a check in rt5645_jack_detect. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- * Return an error if gpio number is invalid --- sound/soc/codecs/rt5645.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a7789a8..27141e2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2209,6 +2209,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) int gpio_state, jack_type = 0; unsigned int val; + if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + dev_err(codec->dev, "invalid gpio\n"); + return -EINVAL; + } gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH RFC 0/4] alsa-lib: audio timestamping evolutions
by Pierre-Louis Bossart 08 Dec '14

08 Dec '14

alsa-lib patches corresponding to the kernel ones just posted. Tested with HDAudio and USB audio class. Pierre-Louis Bossart (4): sync asound.h with kernel version add support for get/set_audio_htstamp_config pcm: add helper functions to query timestamping capabilities test: fix audio_time with new get/set audio_tstamp_config include/pcm.h | 43 ++++- include/sound/asound.h | 20 ++- src/pcm/pcm.c | 82 ++++++++- test/audio_time.c | 473 +++++++++++++++++++++++++++++++------------------ 4 files changed, 436 insertions(+), 182 deletions(-) -- 2.1.0

1 4

Re: [alsa-devel] [PATCH 9/9] ARM: dts: Model IPQ LPASS audio hardware
by Kenneth Westfield 08 Dec '14

08 Dec '14

On Wed, November 19, 2014 2:54 pm, Courtney Cavin wrote: > On Wed, Nov 19, 2014 at 07:52:49PM +0100, Kenneth Westfield wrote: >> --- a/arch/arm/boot/dts/qcom-ipq8064.dtsi >> +++ b/arch/arm/boot/dts/qcom-ipq8064.dtsi >> @@ -2,6 +2,7 @@ >> #include "skeleton.dtsi" >> #include <dt-bindings/clock/qcom,gcc-ipq806x.h> >> +#include <dt-bindings/clock/qcom,lcc-ipq806x.h> > Neither this file nor an associated clock controller driver exists in mainline. Is there some other series this depends on? > -Courtney This patch series has a dependency on the following patch series from linux-arm-msm: [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 -- Kenneth Westfield Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

1 0

Re: [alsa-devel] [PATCH 3/9] ASoC: ipq806x: add native LPAIF driver
by Kenneth Westfield 08 Dec '14

08 Dec '14

On Tue, November 25, 2014 1:44 pm, Mark Brown wrote: > On Wed, Nov 19, 2014 at 10:52:43AM -0800, Kenneth Westfield wrote: >> +static int lpaif_pcm_int_enable(uint8_t dma_ch) >> +{ >> + uint32_t intr_val; >> + uint32_t status_val; >> + unsigned long flags; >> + >> + if (dma_ch >= LPAIF_MAX_CHANNELS) { >> + pr_err("%s: invalid DMA channel given: %hhu\n", >> + __func__, dma_ch); > dev_err(). Now using it in most cases, except for two or three functions that would need to have a handle passed to it solely for the purpose of using dev_*. >> + while (intrsrc) { >> + dma_ch = lpaif_dai_find_dma_channel(intrsrc); >> + if (dma_ch != -1) { >> + if (lpaif_dai[dma_ch]->callback) { >> + >> + ret = lpaif_dai[dma_ch]->callback(intrsrc, >> + lpaif_dai[dma_ch]->private_data); >> + } >> + intrsrc &= ~LPAIF_PER_CH(dma_ch); >> + } else { >> + pr_err("%s: error getting channel\n", __func__); >> + break; >> + } >> + } >> + return ret; >> +} > This all looks like a very simple demux of a single register - don't we have a generic irqchip that can handle it? It looks like that's what you're trying to implement here, each DMA channel could be requesting the three interrupts it has separately rather than open coding an interrupt controller here. > I'm not actually immediately seeing one right now but it'd be simple to add (or regmap-irq could do it if we used a regmap, though it assumes threading and isn't a great fit). I will look into this. If one fits perfectly, great. If not, then I will add one. -- Kenneth Westfield Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

1 0

[alsa-devel] [PATCH] ASoC: pcm512x: Trigger auto-increment of register addresses on i2c
by Peter Rosin 08 Dec '14

08 Dec '14

From: Peter Rosin <peda(a)axentia.se> When the codec is connected using i2c, it will only auto-increment register addresses if msb (0x80) of the register address byte is set. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/codecs/pcm512x-i2c.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 4d62230bd378..d0547fa275fc 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -24,8 +24,13 @@ static int pcm512x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct regmap *regmap; + struct regmap_config config = pcm512x_regmap; - regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + /* msb needs to be set to enable auto-increment of addresses */ + config.read_flag_mask = 0x80; + config.write_flag_mask = 0x80; + + regmap = devm_regmap_init_i2c(i2c, &config); if (IS_ERR(regmap)) return PTR_ERR(regmap); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: request memory region in driver probe()
by Sylwester Nawrocki 08 Dec '14

08 Dec '14

The memory mapped registers region is common for both DAIs so request it in the I2S platform device driver probe for the platform device corresponding to the primary DAI, rather than in the ASoC DAI probe callback. While at it switch to devm_ioremap_resource(). This also drops the hard coded (0x100) register region size in the driver. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Tested on Odroid X2. sound/soc/samsung/i2s.c | 44 +++++++------------------------------------- 1 file changed, 7 insertions(+), 37 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 26d1a1a..fff429da 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -59,10 +59,8 @@ struct samsung_i2s_dai_data { struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* IOREMAP'd SFRs */ + /* Memory mapped SFR region */ void __iomem *addr; - /* Physical base address of SFRs */ - u32 base; /* Rate of RCLK source clock */ unsigned long rclk_srcrate; /* Frame Clock */ @@ -980,12 +978,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) goto probe_exit; } - i2s->addr = ioremap(i2s->base, 0x100); - if (i2s->addr == NULL) { - dev_err(&i2s->pdev->dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->clk = clk_get(&i2s->pdev->dev, "iis"); if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); @@ -1002,7 +994,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { - other->addr = i2s->addr; other->clk = i2s->clk; } @@ -1044,8 +1035,6 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) clk_disable_unprepare(i2s->clk); clk_put(i2s->clk); - - iounmap(i2s->addr); } i2s->clk = NULL; @@ -1164,7 +1153,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) u32 regs_base, quirks = 0, idma_addr = 0; struct device_node *np = pdev->dev.of_node; const struct samsung_i2s_dai_data *i2s_dai_data; - int ret = 0; /* Call during Seconday interface registration */ i2s_dai_data = samsung_i2s_get_driver_data(pdev); @@ -1229,16 +1217,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "Unable to get I2S SFR address\n"); - return -ENXIO; - } + pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(pri_dai->addr)) + return PTR_ERR(pri_dai->addr); - if (!request_mem_region(res->start, resource_size(res), - "samsung-i2s")) { - dev_err(&pdev->dev, "Unable to request SFR region\n"); - return -EBUSY; - } regs_base = res->start; pri_dai->dma_playback.dma_addr = regs_base + I2STXD; @@ -1247,7 +1229,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; - pri_dai->base = regs_base; pri_dai->quirks = quirks; pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs; @@ -1258,8 +1239,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.ch_name = "tx-sec"; @@ -1271,7 +1251,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_size = 4; - sec_dai->base = regs_base; + sec_dai->addr = pri_dai->addr; sec_dai->quirks = quirks; sec_dai->idma_playback.dma_addr = idma_addr; sec_dai->pri_dai = pri_dai; @@ -1280,8 +1260,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; + return -EINVAL; } devm_snd_soc_register_component(&pri_dai->pdev->dev, @@ -1293,17 +1272,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) samsung_asoc_dma_platform_register(&pdev->dev); return 0; -err: - if (res) - release_mem_region(regs_base, resource_size(res)); - - return ret; } static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; - struct resource *res; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; @@ -1313,9 +1286,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) other->sec_dai = NULL; } else { pm_runtime_disable(&pdev->dev); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res) - release_mem_region(res->start, resource_size(res)); } i2s->pri_dai = NULL; -- 1.7.9.5

1 1

[alsa-devel] Nodes, Widgets, ELD and Digital Surround 5.1 (HDMI) Output?
by bencoxdev＠gmail.com 08 Dec '14

08 Dec '14

Hi All, I have an Intel HDA compatible device that supports 5.1 digital audio output over HDMI. It reports two nodes and two widgets:- Root(0) | AFG(1) / \ / \ / \ Audio output(2) Pin Complex(3) This setup produces a “Digital Stereo (HDMI) Output” device at user level in the operating system. The codec reports its self to the operating system as follows (from /proc/asound/card0/codec#1) -------------------------------------------------------------------- AFG Function Id: 0x1 (unsol 1) No Modem Function Group found Default PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM 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 0x201: Stereo Digital Device: name="S3 HDMI 1", type="Audio", device=0 Converter: stream=0, channel=0 Digital: Digital category: 0x0 -------------------------------------------------------------------- Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000090: OUT HDMI Pin Default 0x185650f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Red DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 -------------------------------------------------------------------- the ELD from the receiving device is:- monitor_present 1 eld_valid 1 monitor_name EP-HDMI-RX connection_type HDMI eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0x1017 product_id 0x901 port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0x4f] FL/FR LFE FC RL/RR RLC/RRC sad_count 2 sad0_coding_type [0x2] AC-3 sad0_channels 6 sad0_rates [0xe0] 32000 44100 48000 sad0_max_bitrate 640000 sad1_coding_type [0x7] DTS sad1_channels 7 sad1_rates [0xc0] 44100 48000 sad1_max_bitrate 1536000 Does anyone know what changes would be required to this setup in order for the device to be recognised as “Digital Surround 5.1 (HDMI) Output” by alsa? TIA ---------------------------------------------------------------------- Regards Ben

4 11

[alsa-devel] [GIT PULL] ASoC updates for v3.19
by Mark Brown 08 Dec '14

08 Dec '14

The following changes since commit b2776bf7149bddd1f4161f14f79520f17fc1d71d: Linux 3.18 (2014-12-07 14:21:05 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19 for you to fetch changes up to 1810afd3e1ded09c53d4e966dddce3c7d484521f: Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next (2014-12-08 13:12:36 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.19 Lots and lots of changes this time around, the usual set of driver updates and a huge bulk of cleanups from Lars-Peter. Probably the most interesting thing for most users is the Intel driver updates which will (with some more machine integration work) enable support for newer x86 laptops. - Conversion of AC'97 drivers to use regmap, bringing us closer to the removal of the ASoC level I/O code. - Clean up a lot of old drivers that were open coding things that have subsequently been implemented in the core. - Some DAPM performance improvements. - Removal of the now seldom used CODEC mutex. - Lots of updates for the newer Intel SoC support, including support for the DSP and some Cherrytrail and Braswell machine drivers. - Support for Samsung boards using rt5631 as the CODEC. - Removal of the obsolete AFEB9260 machine driver. - Driver support for the TI TS3A227E headset driver used in some Chrombeooks. ---------------------------------------------------------------- My plan for the next cycle is to try to send one of these each time I send fixes, I meant to do that this time around during the second half of the cycle but ended up not sending fixes (or perhaps sending one lot of fixes at the same time as I'd applied a bunch of changes I wanted to cook in -next, can't remember exactly). Alban Bedel (1): ASoC: tegra: Read and use the GPIO flags of the headphone detect Alexandre Courbot (1): ASoC: jack: update calls to gpiod_get*() Anatol Pomozov (4): ASoC: rt5677: Add option to configure gpio as floating/pullup/pulldown ASoC: rt5677: Print more information if setting DAI clock failed ASoC: rt5677: fix build when kernel compiled without GPIOLIB support ASoC: rt5677: add build dependency to spi Andreas Ruprecht (1): ASoC: rockchip: i2s: Fix Kconfig for I2S device driver Andrew Jackson (1): ASoC: dwc: Allocate resources with devm_ioremap_resource Anil Kumar (1): ASoC: davinvi-mcasp: Balance pm_runtime_enable() on probe failure Axel Lin (2): ASoC: cs4265: Remove unused *dev field from struct cs4265_private ASoC: cs4271: Split SPI and I2C code into different modules Bard Liao (14): ASoC: rt286: reduce power consumption ASoC: rt5645: make bias level more reasonale ASoC: rt286: set combo jack by dmi ASoC: rt286: fix comment style ASoC: rt5645: Add JD function support ASoC: rt5645: Add ASRC support ASoC: rt5670: add rt5672 codec support ASoC: rt5645: remove unused rt5645_clk_sel_put ASoC: rt5645: add register setting for TDM ASoC: rt5645: two jacks for hp and mic ASoC: rt5670: make bias level more reasonable ASoC: rt5645: multiple JD mode support ASoC: rt5670: check if asrc is useable ASoC: rt5670: add DMIC ASRC support Ben Zhang (2): ASoC: rt5677: Include gpio driver header ASoC: rt5677: fix rt5677 spi driver build Charles Keepax (3): ASoC: arizona: Add support for 768kHz DMIC operation ASoC: wm_adsp: Use vmalloc to allocate firmware download buffer ASoC: wm_adsp: Move core_ena to be co-located with start bit Dan Carpenter (1): ASoC: sigmadsp: uninitialized variable in sigmadsp_activate_ctrl() Daniel Mack (1): ASoC: soc-compress: consolidate two identical branches Dmitry Eremin-Solenikov (3): ARM: pxa: spitz: register spitz-audio device ASoC: pxa: Convert spitz to use snd_soc_register_card() ASoC: pxa: prepare/unprepare clocks in pxa-ssp Dylan Reid (3): ASoC: max98090: Different comp tables for different pclks ASoC: add TI ts3a227e headset chip driver ASoC: rt5677: make volume TLV closer to reality Fabio Estevam (15): ASoC: fsl: imx-wm8962: Delete unneeded test before of_node_put ASoC: fsl: imx-sgtl5000: Delete unneeded test before of_node_put ASoC: fsl: imx-spdif: Delete unneeded test before of_node_put ASoC: fsl: eukrea-tlv320: Delete unneeded test before of_node_put ASoC: sgtl5000: Cleanup the comments ASoC: simple-card: Delete unneeded test before of_node_put ASoC: sgtl5000: Use the preferred form for passing a size of a struct ASoC: wm8962: Use the preferred form for passing a size of a struct ASoC: wm8731: Use the preferred form for passing a size of a struct ASoC: mxs: mxs-saif: Register the irq with the device name ASoC: fsl_ssi: Remove comment about SSI running only in slave mode ASoC: sgtl5000: Remove MCLK restriction ASoC: sgtl5000: Allow 8kHz playback in codec slave mode ASoC: mxs-sgtl5000: Remove MCLK restriction ASoC: fsl_ssi: Fix module unbound Fang, Yang A (1): ASoC: Intel: use correct firmware name Fengguang Wu (1): ASoC: Intel: mrfld: fix semicolon.cocci warnings Geert Uytterhoeven (2): ASoC: sh: fsi: Document SoC-specific bindings ASoC: rsnd: Document SoC-specific bindings JS Park (1): ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs Jarkko Nikula (3): ASoC: max98090: Fix digital microphone ASoC: max98090: Fix ill-defined sidetone route ASoC: max98090: Fix right sidetone connection Jean-Francois Moine (4): ASoC: simple-card: Remove useless casts ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: Remove 'const' from the device_node pointers Jean-Michel Hautbois (2): ASoC: sgtl5000: Add MicBias resistor support in DT ASoC: sgtl5000: Add MicBias voltage support Jiada Wang (1): ASoC: fsl_ssi: fix error path in probe Jianqun (2): ASoC: rockchip: i2s: add text after tristate for SND_SOC_ROCKCHIP_I2S ASoC: simple-card: add "invert" property for detect GPIOs Jie Yang (9): ASoC: Intel: Fix block is enabled multiple times issue ASoC: Intel: Work around to fix HW D3 potential crash issue ASoC: Intel: Add jack detection for Broadwell ASoC: Intel: Correct a macro for FW message ASoC: Intel: Fix the driver data not set issue ASoC: Intel: Fix stream volume set no effect issue on Broadwell ASoC: Intel: Remove useless loopback volume control for Broadwell ASoC: Intel: Correct the xmax volume ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell Jyri Sarha (3): ASoC: hdmi: Mark the maximum significant bits to HDMI codec ASoC: hdmi: HDMI codec doesn't benefit from pmdown delay ASoC: tlv320aic31xx: Fix off by one error in the loop stucture. Krishna Mohan Dani (5): ASoC: rt5631: Add device tree binding documentation ASoC: rt5631: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631/RT5631 codec driver ASoC: rt5631: Fixing compilation warning when DT is disabled ASoC: Samsung: Add arndale_rt5631 machine driver and binding Kuninori Morimoto (30): ASoC: rsnd: tidyup debug information when read/write ASoC: rsnd: tidyup RSND_DVC_VOLUME_NUM to RSND_DVC_CHANNELS ASoC: rsnd: add struct rsnd_dvc_cfg and control DVC settings ASoC: simple-card: add asoc_simple_card_parse_daifmt() ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: enable enumerated DVC valume settings ASoC: rsnd: tidyup debug message format and timing ASoC: rsnd: fallback to PIO mode if DMA mode was failed ASoC: rsnd: Add Volume Ramp support ASoC: rsnd: add .fallback callback ASoC: rsnd: add callback status check method ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily ASoC: rsnd: tidyup PIO/DMA mode settings method ASoC: rsnd: tidyup SSI interrupt enable/disable method ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() ASoC: rsnd: clear status register when HW start ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode ASoC: rsnd: show master clock rate when ADG probe ASoC: rsnd: move snd_kcontrol_new fucntions to core.c ASoC: rsnd: tidyup rsnd_io_to_runtime() macro ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() ASoC: rsnd: initialize SRC on rsnd_src_init() ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() ASoC: rsnd: share SSI starting method between PIO/DMA mode ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() ASoC: rsnd: add salvage support for under/over flow error on SSI ASoC: rsnd: rename SSI function name of PIO Lars-Peter Clausen (145): ASoC: ak4535: Cleanup manual bias level transitions ASoC: ak4535: Use table based setup for controls ASoC: ak4641: Cleanup manual bias level transitions ASoC: ak4642: Cleanup manual bias level transitions ASoC: ak4671: Cleanup manual bias level transitions ASoC: max98088: Cleanup manual bias level transitions ASoC: max98095: Cleanup manual bias level transitions ASoC: max9850: Cleanup manual bias level transitions ASoC: sta32x: Cleanup manual bias level transitions ASoC: sta350: Cleanup manual bias level transitions ASoC: sta529: Cleanup manual bias level transitions ASoC: stac9766: Cleanup manual bias level transitions ASoC: dapm: Reduce number of checked paths in dapm_widget_in_card_paths() ASoC: dapm: Remove always true path source/sink checks ASoC: dapm: Remove special DAI widget power check functions ASoC: dapm: Remove path 'walked' flag ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching ASoC: Use generic control handlers for S8 control ASoC: ad1980: Remove unused header ASoC: ad1980: Cleanup printk usage ASoC: ad1980: Use table based control setup ASoC: stac9766: Cleanup printk usage ASoC: stac9766: Use table based control setup ASoC: wm9705: Cleanup printk usage ASoC: wm9705: Use table based control setup ASoC: wm9712: Cleanup printk usage ASoC: wm9712: Use table based control setup ASoC: wm9713: Cleanup printk usage ASoC: wm9713: Use table based control setup ASoC: wm9713: Move driver state struct allocation to driver probe ASoC: wm9713: Use virtual control instead of virtual register ASoC: Remove snd_soc_platform_driver suspend/resume callbacks ASoC: Remove platform field from snd_soc_dai ASoC: mioa701_wm9713: Don't opencode CODEC register access ASoC: Remove card field from snd_soc_dai struct ASoC: Consolidate CPU and CODEC DAI probe ASoC: sn95031: Use table based control setup ASoC: tas2552: Use table based DAPM setup ASoC: wl1273: Use table based control setup ASoC: wm8737: Use table based DAPM and control setup ASoC: wm8961: Use table based DAPM and control setup ASoC: wm8995: Use table based DAPM and control setup ASoC: ad193x: Keep DAC output stage active in idle ASoC: ssm4567: Add DAC high-pass-filter control ASoC: ssm4567: Add support for setting the DAI format and TDM configuration ASoC: ssm4567: Add support for disabling the boost stage ASoC: wm9712/wm9713: Replace virtual registers with custom put/get callbacks ASoC: Forward calls to snd_soc_cache_sync() to regcache_sync() ASoC: Remove snd_soc_cache_sync() implementation ASoC: ab8500-codec: Move control lock to the driver level ASoC: max98095: Move mutex to the driver level ASoC: wm5102: Move ultrasonic response settings lock to the driver level ASoC: wm8731: Move the deemph lock to the driver level ASoC: wm8903: Move the deemph lock to the driver level ASoC: wm8958: Move DSP firmware lock to driver level ASoC: wm8962: Move DSP enable lock to the driver level ASoC: Remove CODEC mutex ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: adau1373: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: adau17x1: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device ASoC: mpc5200_dma: Don't overwrite ac97 device private_data ASoC: Properly handle AC'97 device lifetime management ASoC: Move AC'97 support to its own file ASoC: ac97: Use static ac97_bus ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() ASoC: ac97: Drop support for setting platform data via the CPU DAI ASoC: ac97: Drop delayed device registration ASoC: Drop ac97_control initialization from CODEC driver DAIs ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control ASoC: ac97: Push snd_ac97 pointer to the driver level ASoC: Add helper functions for deferred regmap setup ASoC: ad1980: Convert to regmap ASoC: ad1980: Remove ac97_read/ac97_write wrappers ASoC: sigmadsp: Refuse to load firmware files with a non-supported version ASoC: sigmadsp: Drop support support SIGMA_ACTION_DELAY ASoC: sigmadsp: Restructure in preparation for fw v2 support ASoC: sigmadsp: Add support for fw v2 ASoC: adau1701: Implement sigmadsp safeload ASoC: sirf-audio-codec: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98088: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98090: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98095: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: adau17x1: Mark DSP parameter memory as readable and precious ASoC: sigmadsp: Fix endianness conversion ASoC: cs42l51: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: cs42l73: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: uda134x: Remove is_powered_on_standby from platform data ASoC: uda134x: Cleanup manual bias level transitions ASoC: uda1380: Cleanup manual bias level transitions ASoC: alc5623: Cleanup bias level transitions ASoC: alc5632: Cleanup bias level transitions ASoC: rt5631: Cleanup bias level transitions ASoC: wm8350: Cleanup manual bias level transitions ASoC: wm8400: Cleanup manual bias level transitions ASoC: wm8510: Cleanup manual bias level transitions ASoC: wm8523: Cleanup manual bias level transitions ASoC: wm8580: Cleanup manual bias level transitions ASoC: wm8711: Cleanup manual bias level transitions ASoC: wm8728: Cleanup manual bias level transitions ASoC: wm8731: Cleanup manual bias level transitions ASoC: wm8737: Cleanup manual bias level transitions ASoC: wm8750: Cleanup manual bias level transitions ASoC: wm8776: Cleanup manual bias level transitions ASoC: wm8804: Cleanup manual bias level transitions ASoC: wm8900: Cleanup manual bias level transitions ASoC: wm8903: Cleanup manual bias level transitions ASoC: wm8940: Cleanup manual bias level transitions ASoC: wm8955: Cleanup manual bias level transitions ASoC: wm8960: Cleanup manual bias level transitions ASoC: wm8961: Cleanup manual bias level transitions ASoC: wm8983: Cleanup manual bias level transitions ASoC: wm8978: Cleanup manual bias level transitions ASoC: wm8974: Cleanup manual bias level transitions ASoC: wm8985: Cleanup manual bias level transitions ASoC: wm8988: Cleanup manual bias level transitions ASoC: wm8990: Cleanup manual bias level transitions ASoC: wm8991: Cleanup manual bias level transitions ASoC: wm8993: Cleanup manual bias level transitions ASoC: wm8994: Cleanup manual bias level transitions ASoC: wm8995: Cleanup manual bias level transitions ASoC: wm9081: Cleanup manual bias level transitions ASoC: wm9090: Cleanup manual bias level transitions ASoC: wm9712: Cleanup manual bias level transitions ASoC: wm9713: Cleanup manual bias level transitions ASoC: qi_lb60: Pass flags to gpiod_get() ASoC: Disable regmap helpers if regmap is disabled ASoC: cq93vc: Remove unused state struct ASoC: cq93vc: Cleanup manual bias level transitions ASoC: lm49453: Cleanup manual bias level transitions ASoC: sn95031: Cleanup bias level transitions ASoC: tlv320aic23: Cleanup manual bias level transitions ASoC: tlv320aix31xx: Cleanup manual bias level transitions ASoC: tlv320aic32x4: Cleanup manual bias level transitions ASoC: tlv320aic3x: Cleanup manual bias level transitions ASoC: tlv320dac33: Cleanup manual bias level transitions ASoC: twl4030: Cleanup manual bias level transitions ASoC: twl6040: Cleanup manual bias level transitions Liam Girdwood (11): ASoC: Intel: Add TDM support to HSW/BDW SSP port ASoC: Intel: Add 4 channel support to DSP. ASoC: Intel: Make HSW/BDW pointer debug verbose ASoC: Intel: Make ADSP memory block allocation more generic ASoC: Intel: Add generic support for DSP wake, sleep and stall ASoC: Intel: Add PM support to the HSW/BDW DSP core. ASoC: Intel: Add PM support to HSW/BDW IPC driver ASoC: Intel: Add PM support to HSW/BDW PCM driver ASoC: Intel: Add debug output when boot fails. ASoC: Intel: fix build with runtime PM disabled. ASoC: Intel: Fix build with CONFIG_SLEEP enabled. Mark Brown (46): Merge branch 'fix/sgtl5000' of git://git.kernel.org/.../broonie/sound into asoc-sgtl5000 Merge tag 'v3.18-rc2' into asoc-sgtl5000 ASoC: Intel: Add dependency on DesignWare DMA controller ASoC: core: Split ops out of soc-core.c Merge tag 'v3.18-rc3' into asoc-sgtl5000 Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'fix/max98090' of git://git.kernel.org/.../broonie/sound into asoc-max98090 Merge branches 'topic/wm9712' and 'topic/wm9713' of git://git.kernel.org/.../broonie/sound into asoc-wm971x Merge branch 'fix/davinci' of git://git.kernel.org/.../broonie/sound into asoc-davinci Merge branches 'topic/ad1980', 'topic/wm9705' and 'topic/wm971x' of git://git.kernel.org/.../broonie/sound into asoc-ac97 regmap: ac97: Add generic AC'97 callbacks Merge branch 'topic/ac97' of git://git.kernel.org/.../broonie/regmap into asoc-ac97 Merge branch 'topic/regmap' of git://git.kernel.org/.../broonie/sound into asoc-ac97 ASoC: samsung: Fix error handling for clock lookup ASoC: samsung: Fix non-DT use of I2S controller ASoC: wm5102: Initialize dac_comp_lock mutex Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/davinci', 'asoc/fix/max98090', 'asoc/fix/sigmadsp' and 'asoc/fix/tlv320aic31xx' into asoc-linus Merge remote-tracking branch 'asoc/topic/cache' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ad193x', 'asoc/topic/adau1373' and 'asoc/topic/adau17x1' into asoc-next Merge remote-tracking branches 'asoc/topic/adav80x', 'asoc/topic/adsp', 'asoc/topic/ak4535', 'asoc/topic/ak4641' and 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623', 'asoc/topic/alc5632', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/codec-mutex', 'asoc/topic/compress' and 'asoc/topic/cq93vc' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs4271', 'asoc/topic/cs42l51' and 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/doc', 'asoc/topic/dpcm', 'asoc/topic/dwc' and 'asoc/topic/fsi' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-card', 'asoc/topic/fsl-dt' and 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/intel', 'asoc/topic/jack', 'asoc/topic/jz4740' and 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branches 'asoc/topic/max98088', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max9850' and 'asoc/topic/mop500' into asoc-next Merge remote-tracking branches 'asoc/topic/multi-codec', 'asoc/topic/mxs-saif', 'asoc/topic/mxs-sgtl5000', 'asoc/topic/omap' and 'asoc/topic/pxa' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rockchip', 'asoc/topic/rt286' and 'asoc/topic/rt5631' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5670', 'asoc/topic/rt5677', 'asoc/topic/samsung' and 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/sigmadsp', 'asoc/topic/simple', 'asoc/topic/sirf' and 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm4567', 'asoc/topic/sta32x', 'asoc/topic/sta350', 'asoc/topic/sta529' and 'asoc/topic/stac9766' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tegra', 'asoc/topic/tfa9879', 'asoc/topic/tlv320aic23' and 'asoc/topic/tlv320aic31xx' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/ts3a227e' and 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branches 'asoc/topic/twl6040', 'asoc/topic/uda134x', 'asoc/topic/uda1380' and 'asoc/topic/wl1273' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8350', 'asoc/topic/wm8400', 'asoc/topic/wm8510', 'asoc/topic/wm8523' and 'asoc/topic/wm8580' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8711', 'asoc/topic/wm8728', 'asoc/topic/wm8731', 'asoc/topic/wm8737' and 'asoc/topic/wm8750' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8776', 'asoc/topic/wm8804', 'asoc/topic/wm8900', 'asoc/topic/wm8903' and 'asoc/topic/wm8940' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8955', 'asoc/topic/wm8960', 'asoc/topic/wm8961', 'asoc/topic/wm8962' and 'asoc/topic/wm8974' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8978', 'asoc/topic/wm8983', 'asoc/topic/wm8985', 'asoc/topic/wm8988' and 'asoc/topic/wm8990' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8991', 'asoc/topic/wm8993', 'asoc/topic/wm8994', 'asoc/topic/wm8995' and 'asoc/topic/wm9081' into asoc-next Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next Markus Elfring (3): ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_disable" ASoC: mop500: Deletion of unnecessary checks before the function call "of_node_put" Masanari Iida (1): ASoC: jack: Fix warning while make htmldocs caused by soc-jack.c Max Filippov (2): ASoC: tlv320aic23: make codecs selectable in Kconfig ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER Mengdong Lin (4): ASoC: rt5670 : Add ACPI match ID for Intel CHT/BSW platforms ASoC: Intel: add missing ACPI device table ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672 ASoC: Intel: add support for Cherrytrail and Braswell in SST driver Misael Lopez Cruz (3): ASoC: davinci-mcasp: Active slots depend on active serializers ASoC: tlv320aic3x: Add output driver pop reduction controls ASoC: davinci-mcasp: Add overrun/underrun event handling Mythri P K (1): ASoC: Intel: use lock when changing SST state. Nicolas Ferre (2): ASoC: snd-soc-afeb9260: delete driver as board has just been removed ASoC: Kconfig: remove not used SND_AT91_SOC_AFEB9260 option Nicolin Chen (11): ASoC: fsl_esai: Add indentation for binding doc to increase readability ASoC: fsl_spdif: Add indentation for binding doc to increase readability ASoC: fsl_sai: Add indentation for binding doc to increase readability ASoC: eukrea-tlv320: Add indentation for binding doc to increase readability ASoC: imx-audmux: Add indentation for binding doc to increase readability ASoC: imx-sgtl5000: Add indentation for binding doc to increase readability ASoC: imx-spdif: Add indentation for binding doc to increase readability ASoC: imx-wm8962: Add indentation for binding doc to increase readability ASoC: fsl-asoc-card: Don't bypass settings if cpu-dai is Master ASoC: fsl_esai: Use dynamic slot width as default ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai Oder Chiou (12): ASoC: rt5645: Add the workqueue of the jack detect function for the debouncing ASoC: rt5677: Support DSP function for VAD application ASoC: rt5677: add GPIO IRQ support ASoC: rt5677: Add TDM channel mapping function ASoC: rt5677: Minor coding style and typo fix ASoC: rt5677: Use specific r/w function for DSP mode ASoC: rt5677: Add TDM channel mux in DAC side of IF1 and IF2 ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function ASoC: rt5677: Align the reg_default table with tab character ASoC: rt5677: Follow the gpio naming rule to rename the irq function ASoC: rt5677: Set the slow charge of the vref in the end of the power sequences ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed Padmavathi Venna (3): ASoC: Samsung: Add quirk for internal DMA ASoC: samsung: add support for exynos7 I2S controller ASoC: samsung: ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk() Pavel Machek (1): ASoC: omap: enable sound support on n900 on devicetree-based boot Peter Rosin (3): ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. ASoC: tfa9879: New driver for NXP Semiconductors TFA9879 amplifier. ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing Peter Ujfalusi (10): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions ASoC: tlv320aic3x: Add TDM support ASoC: davinci-mcasp: Symmetric sample bits for IIS mode ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time ASoC: davinci-mcasp: Place constraint on number of channels ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual Qiao Zhou (2): ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE ASoC: soc-pcm: do not hw_free BE if it's still used Ramesh Babu (1): ASoC: core: Call mute for cpu dais as well Randy Dunlap (1): ASoC: cs42l51: depends on I2C Rasmus Villemoes (1): ASoC: dapm: Remove redundant cast Ricardo Neri (2): ASoC: Intel: Add new dependency for Broadwell machine ASoC: Intel: Add new dependency for Haswell machine Richard Fitzgerald (1): ASoC: arizona: Add DSP_B and LEFT_J mode support Subhransu S. Prusty (12): ASoC: Intel: mrfld: Replace pci_id with unique device id ASoC: Intel: mrfld - Define ipc_info structure ASoC: Intel: mrfld: Define sst_res_info for acpi ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream ASoC: mfld-compress: implement .power callback ASoC: Intel: mfld-pcm: Fix to Store device context in sst_data ASoC: Intel: move the driver wq init to a routine ASoC: Intel: move the lock and wq initialization to routine ASoC: Intel: move the driver context allocation to routine ASoC: Intel: modularize driver probe and remove ASoC: Intel: more probe modularization for sst ASoC: Intel: add BYTCR machine driver with RT5640 Sudip Mukherjee (2): ASoC: Intel: fix missing mutex ASoC: rt286: build warning of section mismatch Takashi Iwai (2): ASoC: eukrea-tlv320: Fix of_node_put() call with uninitialized object ASoC: core: Use snd_ctl_enum_info() Thomas Petazzoni (1): ASoC: cs42l51: make driver user-selectable Vinod Koul (27): ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mrfld: add DSP core controls ASoC: Intel: mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: mrfld: add the DSP mixers ASoC: intel: turn off COMPILE_TEST for medfield ASoC: Intel: mrfld - add the dsp sst driver ASoC: Intel: mrfld - Add DSP load and management ASoC: Intel: sst - add pcm ops handling ASoC: Intel: sst: Add IPC handling ASoC: Intel: sst: add stream operations ASoC: Intel: sst: Add some helper functions ASoC: Intel: sst: Add makefile and kconfig changes ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: Intel: sst: add runtime power management handling ASoC: Intel: sst: load firmware using async callback ASoC: Intel: sst - add compressed ops handling ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld - remove non static definition ASoC: Intel: mrfld- add ACPI module ASoC: Intel: cleanup runtime_pm initialization Zidan Wang (2): ASoC: wm8960: Add device tree support ASoC: wm8960: Move register initialisation to I2C driver probe() kbuild test robot (5): ASoC: rt5677: rt5677_irq_init() can be static ASoC: Intel: dw_pdata can be static ASoC: Intel: byt_rvp_platform_data can be static ASoC: Intel: chv_platform_data can be static ASoC: sigmadsp: fix simple_return.cocci warnings .../devicetree/bindings/sound/arndale.txt | 24 + .../bindings/sound/davinci-mcasp-audio.txt | 2 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 15 +- .../devicetree/bindings/sound/fsl,esai.txt | 44 +- .../devicetree/bindings/sound/fsl,spdif.txt | 37 +- .../devicetree/bindings/sound/fsl-sai.txt | 66 +- .../bindings/sound/imx-audio-sgtl5000.txt | 61 +- .../devicetree/bindings/sound/imx-audio-spdif.txt | 22 +- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 45 +- .../devicetree/bindings/sound/imx-audmux.txt | 22 +- .../devicetree/bindings/sound/max98090.txt | 2 + .../devicetree/bindings/sound/renesas,fsi.txt | 9 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 10 +- Documentation/devicetree/bindings/sound/rt5631.txt | 48 + Documentation/devicetree/bindings/sound/rt5677.txt | 17 + .../devicetree/bindings/sound/samsung-i2s.txt | 15 +- .../devicetree/bindings/sound/sgtl5000.txt | 13 + .../devicetree/bindings/sound/ts3a227e.txt | 26 + Documentation/devicetree/bindings/sound/wm8960.txt | 31 + MAINTAINERS | 6 + arch/arm/mach-pxa/spitz.c | 9 + arch/x86/include/asm/platform_sst_audio.h | 62 + drivers/base/regmap/Kconfig | 5 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-ac97.c | 114 ++ include/linux/mfd/arizona/core.h | 1 + include/linux/mfd/davinci_voicecodec.h | 7 - include/linux/platform_data/asoc-s3c.h | 1 + include/linux/regmap.h | 7 + include/sound/rcar_snd.h | 8 +- include/sound/rt5645.h | 4 + include/sound/rt5677.h | 10 + include/sound/soc-dai.h | 7 +- include/sound/soc-dapm.h | 9 +- include/sound/soc.h | 116 +- include/sound/uda134x.h | 12 - include/trace/events/asoc.h | 25 - sound/soc/Makefile | 6 +- sound/soc/atmel/Kconfig | 9 - sound/soc/atmel/Makefile | 1 - sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/atmel/snd-soc-afeb9260.c | 151 -- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 - sound/soc/cirrus/Kconfig | 3 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/Kconfig | 47 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 32 +- sound/soc/codecs/ac97.c | 18 +- sound/soc/codecs/ad193x.c | 14 +- sound/soc/codecs/ad1980.c | 212 +-- sound/soc/codecs/ad1980.h | 26 - sound/soc/codecs/adau1373.c | 6 +- sound/soc/codecs/adau1701.c | 86 +- sound/soc/codecs/adau1761.c | 25 +- sound/soc/codecs/adau1781.c | 33 +- sound/soc/codecs/adau17x1.c | 71 +- sound/soc/codecs/adau17x1.h | 10 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4535.c | 31 +- sound/soc/codecs/ak4641.c | 33 +- sound/soc/codecs/ak4642.c | 16 - sound/soc/codecs/ak4671.c | 13 - sound/soc/codecs/alc5623.c | 22 +- sound/soc/codecs/alc5632.c | 22 +- sound/soc/codecs/arizona.c | 34 +- sound/soc/codecs/cq93vc.c | 33 - sound/soc/codecs/cs4265.c | 2 - sound/soc/codecs/cs4271-i2c.c | 62 + sound/soc/codecs/cs4271-spi.c | 55 + sound/soc/codecs/cs4271.c | 155 +- sound/soc/codecs/cs4271.h | 11 + sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/hdmi.c | 2 + sound/soc/codecs/lm49453.c | 8 - sound/soc/codecs/max98088.c | 31 +- sound/soc/codecs/max98090.c | 201 ++- sound/soc/codecs/max98090.h | 8 + sound/soc/codecs/max98095.c | 23 +- sound/soc/codecs/max9850.c | 22 +- sound/soc/codecs/rt286.c | 231 ++- sound/soc/codecs/rt5631.c | 38 +- sound/soc/codecs/rt5645.c | 200 ++- sound/soc/codecs/rt5645.h | 19 +- sound/soc/codecs/rt5670.c | 136 +- sound/soc/codecs/rt5670.h | 6 + sound/soc/codecs/rt5677-spi.c | 130 ++ sound/soc/codecs/rt5677-spi.h | 21 + sound/soc/codecs/rt5677.c | 1198 ++++++++++++- sound/soc/codecs/rt5677.h | 162 +- sound/soc/codecs/sgtl5000.c | 111 +- sound/soc/codecs/sigmadsp-i2c.c | 81 +- sound/soc/codecs/sigmadsp-regmap.c | 46 +- sound/soc/codecs/sigmadsp.c | 711 +++++++- sound/soc/codecs/sigmadsp.h | 59 +- sound/soc/codecs/sirf-audio-codec.c | 6 +- sound/soc/codecs/sn95031.c | 15 +- sound/soc/codecs/ssm4567.c | 128 ++ sound/soc/codecs/sta32x.c | 21 +- sound/soc/codecs/sta350.c | 21 +- sound/soc/codecs/sta529.c | 35 +- sound/soc/codecs/stac9766.c | 60 +- sound/soc/codecs/tas2552.c | 10 +- sound/soc/codecs/tfa9879.c | 328 ++++ sound/soc/codecs/tfa9879.h | 202 +++ sound/soc/codecs/tlv320aic23.c | 21 +- sound/soc/codecs/tlv320aic31xx.c | 31 +- sound/soc/codecs/tlv320aic32x4.c | 24 +- sound/soc/codecs/tlv320aic3x.c | 228 ++- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/tlv320dac33.c | 2 - sound/soc/codecs/ts3a227e.c | 314 ++++ sound/soc/codecs/ts3a227e.h | 17 + sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 23 +- sound/soc/codecs/uda134x.c | 32 +- sound/soc/codecs/uda1380.c | 20 +- sound/soc/codecs/wl1273.c | 10 +- sound/soc/codecs/wm5102.c | 18 +- sound/soc/codecs/wm8350.c | 21 +- sound/soc/codecs/wm8400.c | 34 +- sound/soc/codecs/wm8510.c | 26 +- sound/soc/codecs/wm8523.c | 29 +- sound/soc/codecs/wm8580.c | 4 - sound/soc/codecs/wm8711.c | 27 +- sound/soc/codecs/wm8728.c | 34 +- sound/soc/codecs/wm8731.c | 37 +- sound/soc/codecs/wm8737.c | 49 +- sound/soc/codecs/wm8750.c | 25 +- sound/soc/codecs/wm8776.c | 31 +- sound/soc/codecs/wm8804.c | 3 - sound/soc/codecs/wm8900.c | 8 - sound/soc/codecs/wm8903.c | 43 +- sound/soc/codecs/wm8940.c | 22 +- sound/soc/codecs/wm8955.c | 33 +- sound/soc/codecs/wm8958-dsp2.c | 12 +- sound/soc/codecs/wm8960.c | 113 +- sound/soc/codecs/wm8961.c | 34 +- sound/soc/codecs/wm8962.c | 11 +- sound/soc/codecs/wm8974.c | 25 +- sound/soc/codecs/wm8978.c | 10 - sound/soc/codecs/wm8983.c | 27 +- sound/soc/codecs/wm8985.c | 28 +- sound/soc/codecs/wm8988.c | 27 +- sound/soc/codecs/wm8990.c | 24 +- sound/soc/codecs/wm8991.c | 32 +- sound/soc/codecs/wm8993.c | 12 - sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8994.h | 2 + sound/soc/codecs/wm8995.c | 17 +- sound/soc/codecs/wm9081.c | 7 - sound/soc/codecs/wm9090.c | 32 +- sound/soc/codecs/wm9705.c | 46 +- sound/soc/codecs/wm9712.c | 219 ++- sound/soc/codecs/wm9713.c | 230 +-- sound/soc/codecs/wm_adsp.c | 97 +- sound/soc/davinci/davinci-mcasp.c | 339 +++- sound/soc/davinci/davinci-mcasp.h | 17 + sound/soc/dwc/designware_i2s.c | 46 +- sound/soc/fsl/eukrea-tlv320.c | 5 +- sound/soc/fsl/fsl-asoc-card.c | 19 +- sound/soc/fsl/fsl_esai.c | 12 +- sound/soc/fsl/fsl_ssi.c | 17 +- sound/soc/fsl/imx-sgtl5000.c | 6 +- sound/soc/fsl/imx-spdif.c | 3 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/imx-wm8962.c | 6 +- sound/soc/fsl/mpc5200_dma.c | 3 - sound/soc/fsl/mpc5200_psc_ac97.c | 6 +- sound/soc/generic/simple-card.c | 160 +- sound/soc/intel/Kconfig | 44 +- sound/soc/intel/Makefile | 7 + sound/soc/intel/broadwell.c | 64 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 230 +++ sound/soc/intel/cht_bsw_rt5672.c | 285 ++++ sound/soc/intel/haswell.c | 14 +- sound/soc/intel/sst-atom-controls.c | 1208 ++++++++++++- sound/soc/intel/sst-atom-controls.h | 428 +++++ sound/soc/intel/sst-baytrail-dsp.c | 24 +- sound/soc/intel/sst-dsp-priv.h | 136 +- sound/soc/intel/sst-dsp.c | 31 + sound/soc/intel/sst-dsp.h | 28 +- sound/soc/intel/sst-firmware.c | 937 +++++++++-- sound/soc/intel/sst-haswell-dsp.c | 295 +++- sound/soc/intel/sst-haswell-ipc.c | 413 ++++- sound/soc/intel/sst-haswell-ipc.h | 25 +- sound/soc/intel/sst-haswell-pcm.c | 419 ++++- sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform-pcm.c | 154 +- sound/soc/intel/sst-mfld-platform.h | 5 + sound/soc/intel/sst/Makefile | 7 + sound/soc/intel/sst/sst.c | 437 +++++ sound/soc/intel/sst/sst.h | 546 ++++++ sound/soc/intel/sst/sst_acpi.c | 383 +++++ sound/soc/intel/sst/sst_drv_interface.c | 686 ++++++++ sound/soc/intel/sst/sst_ipc.c | 373 +++++ sound/soc/intel/sst/sst_loader.c | 456 +++++ sound/soc/intel/sst/sst_pci.c | 209 +++ sound/soc/intel/sst/sst_pvt.c | 449 +++++ sound/soc/intel/sst/sst_stream.c | 437 +++++ sound/soc/jz4740/qi_lb60.c | 11 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 7 - sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/omap/Kconfig | 8 +- sound/soc/omap/mcbsp.c | 3 +- sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/pxa-ssp.c | 16 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/pxa/spitz.c | 52 +- sound/soc/rockchip/Kconfig | 9 +- sound/soc/samsung/Kconfig | 8 +- sound/soc/samsung/Makefile | 2 + sound/soc/samsung/ac97.c | 4 +- sound/soc/samsung/arndale_rt5631.c | 150 ++ sound/soc/samsung/i2s-regs.h | 10 +- sound/soc/samsung/i2s.c | 244 ++- sound/soc/samsung/odroidx2_max98090.c | 4 +- sound/soc/sh/fsi.c | 9 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 236 ++- sound/soc/sh/rcar/dvc.c | 215 +-- sound/soc/sh/rcar/gen.c | 30 +- sound/soc/sh/rcar/rsnd.h | 92 +- sound/soc/sh/rcar/src.c | 101 +- sound/soc/sh/rcar/ssi.c | 233 ++- sound/soc/soc-ac97.c | 256 +++ sound/soc/soc-cache.c | 149 -- sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 1768 +++----------------- sound/soc/soc-dapm.c | 755 +++++---- sound/soc/soc-jack.c | 11 +- sound/soc/soc-ops.c | 952 +++++++++++ sound/soc/soc-pcm.c | 23 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 6 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/ux500/mop500.c | 8 +- 244 files changed, 17569 insertions(+), 5861 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt create mode 100644 drivers/base/regmap/regmap-ac97.c delete mode 100644 sound/soc/atmel/snd-soc-afeb9260.c delete mode 100644 sound/soc/codecs/ad1980.h create mode 100644 sound/soc/codecs/cs4271-i2c.c create mode 100644 sound/soc/codecs/cs4271-spi.c create mode 100644 sound/soc/codecs/cs4271.h create mode 100644 sound/soc/codecs/rt5677-spi.c create mode 100644 sound/soc/codecs/rt5677-spi.h create mode 100644 sound/soc/codecs/tfa9879.c create mode 100644 sound/soc/codecs/tfa9879.h create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/cht_bsw_rt5672.c create mode 100644 sound/soc/intel/sst/Makefile create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_drv_interface.c create mode 100644 sound/soc/intel/sst/sst_ipc.c create mode 100644 sound/soc/intel/sst/sst_loader.c create mode 100644 sound/soc/intel/sst/sst_pci.c create mode 100644 sound/soc/intel/sst/sst_pvt.c create mode 100644 sound/soc/intel/sst/sst_stream.c create mode 100644 sound/soc/samsung/arndale_rt5631.c create mode 100644 sound/soc/soc-ac97.c create mode 100644 sound/soc/soc-ops.c

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[alsa-devel] [PATCH] ASoC: atmel_ssc_dai/trivial: typo fix
by Nicolas Ferre 08 Dec '14

08 Dec '14

From: Matthieu Crapet <mcrapet(a)gmail.com> Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f399808a..775a2badd6a4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -267,7 +267,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, if (!ssc_p->dir_mask) { if (ssc_p->initialized) { /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); + pr_debug("atmel_ssc_dai: Stopping clock\n"); clk_disable(ssc_p->ssc->clk); free_irq(ssc_p->ssc->irq, ssc_p); -- 2.1.3

2 1

[alsa-devel] [PATCH v3] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 08 Dec '14

08 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 56 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 54 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ 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 031a1ae..ea5f0b7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,38 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1119,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: wm5102: Initialize dac_comp_lock mutex
by Mark Brown 08 Dec '14

08 Dec '14

Commit d74bcaaeb6682 (ASoC: wm5102: Move ultrasonic response settings lock to the driver level) created a driver local mutex for protecting the ultrasonic response settings but neglected to initialize that mutex, causing loud complaints from lockep and potential runtime failures. Fix this by initializing the mutex. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5102.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8c99b7cf47e0..f439ae052128 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1900,6 +1900,8 @@ static int wm5102_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5102); + mutex_init(&arizona->dac_comp_lock); + wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; -- 2.1.3

2 1

[alsa-devel] [PATCH 00/15 v4] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 08 Dec '14

08 Dec '14

This patchset is the rest of patches to enable ALSA Dice driver to handle more models and to improve OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084791.h… - move mutex operations to a condition in pcm.hw_params (dice/oxfw) - fix naming regression (oxfw) == Supplemental patch for userspace library This patchset adds Hwdep functionality for OXFW driver, thus userspace library needs a patch for entries of snd_hwdep_iface_t. Please apply below patch. [alsa-devel] [PATCH alsa-lib] hwdep: add OXFW driver support http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084824.h… Takashi Sakamoto (15): ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 10 +- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 194 ++++++++++ sound/firewire/dice/dice-pcm.c | 252 +++++++++---- sound/firewire/dice/dice-stream.c | 332 ++++++++++++---- sound/firewire/dice/dice-transaction.c | 5 - sound/firewire/dice/dice.c | 37 +- sound/firewire/dice/dice.h | 33 +- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-command.c | 153 ++++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++++++++ sound/firewire/oxfw/oxfw-pcm.c | 425 +++++++++++++++------ sound/firewire/oxfw/oxfw-proc.c | 113 ++++++ sound/firewire/oxfw/oxfw-stream.c | 667 +++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 181 +++++++-- sound/firewire/oxfw/oxfw.h | 104 ++++- 19 files changed, 2539 insertions(+), 377 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c -- 2.1.0

2 19

[alsa-devel] [PATCH] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Rickard Strandqvist 07 Dec '14

07 Dec '14

Removes some functions that are not used anywhere: hpi_subsys_get_version_ex() hpi_instream_wait_start() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/hpi.h | 4 ---- sound/pci/asihpi/hpifunc.c | 27 --------------------------- 2 files changed, 31 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..8338b6e 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1164,8 +1164,6 @@ u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, /* SubSystem */ /*************/ -u16 hpi_subsys_get_version_ex(u32 *pversion_ex); - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters); u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, @@ -1289,8 +1287,6 @@ u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_read_buf, u32 bytes_to_read); u16 hpi_instream_start(u32 h_instream); -u16 hpi_instream_wait_start(u32 h_instream); - u16 hpi_instream_stop(u32 h_instream); u16 hpi_instream_reset(u32 h_instream); diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..8fd02dd 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -93,18 +93,6 @@ static inline void hpi_send_recvV1(struct hpi_message_header *m, hpi_send_recv((struct hpi_message *)m, (struct hpi_response *)r); } -u16 hpi_subsys_get_version_ex(u32 *pversion_ex) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_VERSION); - hpi_send_recv(&hm, &hr); - *pversion_ex = hr.u.s.data; - return hr.error; -} - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) { struct hpi_message hm; @@ -950,21 +938,6 @@ u16 hpi_instream_start(u32 h_instream) return hr.error; } -u16 hpi_instream_wait_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_WAIT_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - u16 hpi_instream_stop(u32 h_instream) { struct hpi_message hm; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 06 Dec '14

06 Dec '14

sound/soc/xtensa/xtfpga-i2s.c:666:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- xtfpga-i2s.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -663,7 +663,6 @@ static struct platform_driver xtfpga_i2s .remove = xtfpga_i2s_remove, .driver = { .name = "xtfpga-i2s", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(xtfpga_i2s_of_match), .pm = &xtfpga_i2s_pm_ops, },

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Don't resubmit pending URBs at MIDI error recovery
by Takashi Iwai 06 Dec '14

06 Dec '14

In snd_usbmidi_error_timer(), the driver tries to resubmit MIDI input URBs to reactivate the MIDI stream, but this causes the error when some of URBs are still pending like: WARNING: CPU: 0 PID: 0 at ../drivers/usb/core/urb.c:339 usb_submit_urb+0x5f/0x70() URB ef705c40 submitted while active CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.16.6-2-desktop #1 Hardware name: FOXCONN TPS01/TPS01, BIOS 080015 03/23/2010 c0984bfa f4009ed4 c078deaf f4009ee4 c024c884 c09a135c f4009f00 00000000 c0984bfa 00000153 c061ac4f c061ac4f 00000009 00000001 ef705c40 e854d1c0 f4009eec c024c8d3 00000009 f4009ee4 c09a135c f4009f00 f4009f04 c061ac4f Call Trace: [<c0205df6>] try_stack_unwind+0x156/0x170 [<c020482a>] dump_trace+0x5a/0x1b0 [<c0205e56>] show_trace_log_lvl+0x46/0x50 [<c02049d1>] show_stack_log_lvl+0x51/0xe0 [<c0205eb7>] show_stack+0x27/0x50 [<c078deaf>] dump_stack+0x45/0x65 [<c024c884>] warn_slowpath_common+0x84/0xa0 [<c024c8d3>] warn_slowpath_fmt+0x33/0x40 [<c061ac4f>] usb_submit_urb+0x5f/0x70 [<f7974104>] snd_usbmidi_submit_urb+0x14/0x60 [snd_usbmidi_lib] [<f797483a>] snd_usbmidi_error_timer+0x6a/0xa0 [snd_usbmidi_lib] [<c02570c0>] call_timer_fn+0x30/0x130 [<c0257442>] run_timer_softirq+0x1c2/0x260 [<c0251493>] __do_softirq+0xc3/0x270 [<c0204732>] do_softirq_own_stack+0x22/0x30 [<c025186d>] irq_exit+0x8d/0xa0 [<c0795228>] smp_apic_timer_interrupt+0x38/0x50 [<c0794a3c>] apic_timer_interrupt+0x34/0x3c [<c0673d9e>] cpuidle_enter_state+0x3e/0xd0 [<c028bb8d>] cpu_idle_loop+0x29d/0x3e0 [<c028bd23>] cpu_startup_entry+0x53/0x60 [<c0bfac1e>] start_kernel+0x415/0x41a For avoiding these errors, check the pending URBs and skip resubmitting such ones. Reported-and-tested-by: Stefan Seyfried <stefan.seyfried(a)googlemail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index d3d49525a16b..5bfb695547f8 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -365,6 +365,8 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { + if (atomic_read(&in->urbs[j]->use_count)) + continue; in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } -- 2.2.0

2 2

[alsa-devel] [PATCH] ASoC: samsung: Fix error handling for clock lookup
by Mark Brown 05 Dec '14

05 Dec '14

Return the error code we got from clk_get() and check to make sure that clk_prepare_enable() worked. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 25919e090983..500b9ba5ebaa 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; + int ret; if (other && other->clk) { /* If this is probe on secondary */ samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, @@ -989,9 +990,14 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); iounmap(i2s->addr); - return -ENOENT; + return PTR_ERR(i2s->clk); + } + + ret = clk_prepare_enable(i2s->clk); + if (ret != 0) { + dev_err(&i2s->pdev->dev, "failed to enable clock: %d\n", ret); + return ret; } - clk_prepare_enable(i2s->clk); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- 2.1.3

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Clocks must be prepared before enabling them. Do this in one step. Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. This fixes the following warning: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1 at drivers/clk/clk.c:895 __clk_enable+0x24/0x9c() Modules linked in: CPU: 0 PID: 1 Comm: swapper Tainted: G W 3.18.0-rc7+ #245 [<c000dce8>] (unwind_backtrace) from [<c000bcf0>] (show_stack+0x10/0x14) [<c000bcf0>] (show_stack) from [<c001664c>] (warn_slowpath_common+0x60/0x80) [<c001664c>] (warn_slowpath_common) from [<c00166fc>] (warn_slowpath_null+0x18/0x20) [<c00166fc>] (warn_slowpath_null) from [<c02fd7ac>] (__clk_enable+0x24/0x9c) [<c02fd7ac>] (__clk_enable) from [<c02fdbb4>] (clk_enable+0x18/0x2c) [<c02fdbb4>] (clk_enable) from [<c0322688>] (atmel_ac97c_probe+0x154/0x694) [<c0322688>] (atmel_ac97c_probe) from [<c0235e08>] (platform_drv_probe+0x48/0x94) [<c0235e08>] (platform_drv_probe) from [<c02345f8>] (driver_probe_device+0x138/0x350) [<c02345f8>] (driver_probe_device) from [<c02348bc>] (__driver_attach+0x68/0x8c) [<c02348bc>] (__driver_attach) from [<c0232bd0>] (bus_for_each_dev+0x70/0x84) [<c0232bd0>] (bus_for_each_dev) from [<c0233cd8>] (bus_add_driver+0xfc/0x1f8) [<c0233cd8>] (bus_add_driver) from [<c0234f0c>] (driver_register+0x9c/0xe0) [<c0234f0c>] (driver_register) from [<c0008ac4>] (do_one_initcall+0x110/0x1c8) [<c0008ac4>] (do_one_initcall) from [<c053cd58>] (kernel_init_freeable+0xf8/0x1b8) [<c053cd58>] (kernel_init_freeable) from [<c03c0414>] (kernel_init+0x8/0xe4) [<c03c0414>] (kernel_init) from [<c00096d0>] (ret_from_fork+0x14/0x24) ---[ end trace cb88537fdc8fa201 ]--- atmel_ac97c fffa0000.sound: AC'97 0 does not respond - RESET atmel_ac97c fffa0000.sound: AC'97 0 access is not valid [0xffffffff], removing mixer. ------------[ cut here ]------------ Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 86f07e9..fa283d3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 2

[alsa-devel] [PATCH v7] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 05 Dec '14

05 Dec '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 154 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a4d7da..9b62457 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -533,6 +534,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..db60701 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -942,6 +942,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56..f4956e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1292,7 +1292,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1304,7 +1305,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..7e961ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -754,6 +754,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2721,6 +2751,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3206,6 +3239,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config = &config[w->params_select]; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3274,8 +3310,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3283,14 +3350,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3298,6 +3403,28 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3305,15 +3432,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

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[alsa-devel] usb-audio: I'd like to change two things in usb audio, but I'm not sure if it's right
by Johan Aires Rastén 05 Dec '14

05 Dec '14

So I'll try again, this time with the proper cc:s and a second question. I've come across two issues in the usb audio module that are causing problems with my sound card, but I'm thinking that changing them might break something for some other card. In sound/usb/mixer.c function check_input_term(...), for the case UAC_INPUT_TERMINAL and UAC_VERSION_2: The recursive call to get clock selectors messes up the naming of my terminals. At this stage the terminals are already properly named Mic and Line but the call overwrites them both with the unnamed clock selector "Unit 22". Are other UAC2 cards different, so that the clock selector contains the correct terminal name, or is there some other reasoning here that I'm not seeing? Could the call be removed, or perhaps only done if no terminal name has been found yet? In sound/usb/stream.c function snd_usb_add_audio_stream(...), near the end of the function: A newly created pcm is added with "list_add(&as->list, &chip->pcm_list);" which causes devices to be listed in reverse order, i.e. device 0 will be at the end of the list, and this leads to that capture streams end up in the wrong device under certain circumstances. The problem arises when the code under "look for an empty stream" earlier in the same function executes. It will attach the capture stream to the first compatible existing stream, but since they are reversed it will find the highest numbered device. In my case analog input will be grouped with digital out (device 1) instead of analog out (device 0). This is assuming that playback streams are defined before capture streams but it could happen in the other direction as long as there are several streams of one type. I also suspect that this is what causes the M-Audio Audiophile USB to have swapped inputs, for which there's a fix in USB-Audio.conf ( http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Aud… ). Thanks to debianuser on IRC for helping me find it. I would like to change this call to use list_add_tail for adding new items instead, but I'm worried that there are quirks or other fixes that expect the current behaviour. There are very few other references to pcm_list and none of them seem to care about ordering. Sorry for being bad at referencing code, I'm new at this. //Johan Rastén

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[alsa-devel] [PATCH 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b2a3225..ff2f743 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

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[alsa-devel] DMA buffer alignment
by Carlo Caione 05 Dec '14

05 Dec '14

Hi, I'm working with a SoC sound hw that requires that all the data written into the DMA memory is aligned a 64 byte boundary. Actually the alignment is required since the format of the data into the DMA memory is (for a 2 channel 16bit audio): 16 samples for the left channel followed by 16 samples for the right channel. If I try to write anything that is not aligned to the 64 bytes I got audio corruption (high pitched noise). Several user-space programs (like aplay) work well with my hw, since the buffer sent from the user-space is always (padded) of period-size bytes, that is aligned to my 64 bytes boundary. Others (like speaker-test) are not working because sometimes they snd_pcm_writei() a buffer that is not multiple of 64 bytes. In general every time I write into the DMA area a buffer that is not aligned, my hw pointer is not aligned anymore with the 64 bytes and the noise starts. IIUC it is not possible to force the user-space programs to send buffers with a fixed size so the problem should be solved in kernel-space (am I wrong on this point?). Any suggestion on how to tackle this problem? Thanks, -- Carlo Caione

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[alsa-devel] [PATCH v3] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 05 Dec '14

05 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Only use the resolution supported, ignoring the LSB of the volume register. This results in half the steps and 0.75dB per step, but reports accurate levels through TLV. Update the masks to reflect that these are registers have the LSB ignored. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 22 +++++++++++----------- sound/soc/codecs/rt5677.h | 24 ++++++++++++------------ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 133010d..81fe146 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -763,9 +763,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); @@ -817,13 +817,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), @@ -842,19 +842,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL, - RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, + RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), /* Sidetone Control */ diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index dbd9ffd..c0a625f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -306,10 +306,10 @@ #define RT5677_R_MUTE_SFT 7 #define RT5677_VOL_R_MUTE (0x1 << 6) #define RT5677_VOL_R_SFT 6 -#define RT5677_L_VOL_MASK (0x3f << 8) -#define RT5677_L_VOL_SFT 8 -#define RT5677_R_VOL_MASK (0x3f) -#define RT5677_R_VOL_SFT 0 +#define RT5677_L_VOL_MASK (0x7f << 9) +#define RT5677_L_VOL_SFT 9 +#define RT5677_R_VOL_MASK (0x7f << 1) +#define RT5677_R_VOL_SFT 1 /* LOUT1 Control (0x01) */ #define RT5677_LOUT1_L_MUTE (0x1 << 15) @@ -447,16 +447,16 @@ #define RT5677_SEL_DAC2_R_SRC_SFT 0 /* Stereo1 ADC Digital Volume Control (0x1c) */ -#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_STO1_ADC_L_VOL_SFT 8 -#define RT5677_STO1_ADC_R_VOL_MASK (0x7f) -#define RT5677_STO1_ADC_R_VOL_SFT 0 +#define RT5677_STO1_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_STO1_ADC_L_VOL_SFT 9 +#define RT5677_STO1_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_STO1_ADC_R_VOL_SFT 1 /* Mono ADC Digital Volume Control (0x1d) */ -#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_MONO_ADC_L_VOL_SFT 8 -#define RT5677_MONO_ADC_R_VOL_MASK (0x7f) -#define RT5677_MONO_ADC_R_VOL_SFT 0 +#define RT5677_MONO_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_MONO_ADC_L_VOL_SFT 9 +#define RT5677_MONO_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_MONO_ADC_R_VOL_SFT 1 /* Stereo 1/2 ADC Boost Gain Control (0x1e) */ #define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14) -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] 'No audio' bug on lenovo ultra dock
by Alexander Lochmann 05 Dec '14

05 Dec '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! The current linux kernel (3.17 as well as 3.18-rc7) does not support the audio output on a Lenovo Ultra Dock in conjunction with the X240. The issue is discussed in [1], which provides a patch as well. The patch is available at [2]. I'm asking you for integrating it into your linux tree. So it might be merged into the mainline kernel during the current or the next development phase. Thanks in advance! Regards, Alex [1] https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1385680 [2] https://launchpadlibrarian.net/188362011/lenovo-ultradock.patch - -- Technische Universität Dortmund Alexander Lochmann PGP key: 0xBC3EF6FD Otto-Hahn-Str. 16 phone: +49.231.7556141 D-44227 Dortmund fax: +49.231.7556116 http://ess.cs.tu-dortmund.de/Staff/al -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJUgXibAAoJEFk+7QW8Pvb9zSEP/0yezuJXBK9P8ejhfrmH7VxK PPNhsALQmgMa5vTPsepzBsbj5060yiyI8BTkt76h5d6QlLy3eKq8+AE4+hkGJNie sZtvbWPr8C0GP+8lnfRz6EFOdnWdXbEc4x1hSd7JVDgpNynzHvYZQgISVnxA1rFe NTqngYzFM8yBhUEKc0DYzvuCTw9J0VODWro9eD6tjuSI7/yyQmrLln7XihcLs3/3 VWcNYVBDzQOqjwOTJM7wPfprGgalr7vOBWo3WyshBnKcXmsnahswB+HSsiVWqtMd SZBoEfm811ocvtBFGfsuMkWJALtUfw3bBEPm2iV5jzKmK1CoCxCAuqTcUkiNEzCz m6f76LEtGzq10vpXsc9FmX1Mlbrt9mavE2a5cxoaxVEPJ2OVpwAU3ufgguwUkIrt L8E4RjBEHwttDbe4CPz1YSNz4j+OD4bxsRVjQGfpk8mVGTh0ljtno1chqxn0HZ5J +9Hto37DqDzywJnc4gYyhxTcAZ669qvdMow0avyPuEwGK///C9qgkGCgrm0W36Oq u/HU2lz9A6Vqxtzr8t4bVVEhiX+J2+HQAyf2OPx7o4iMyLAFoxvvWRZUJ3qvP1ZQ OlhXIHg22xTiy9WsVLiStSbT8xd6lV1UP/HotUwOjEoQLzv8pVh/5DH3fwTX5cIw jvRMnfYW+6Ai6Nn07Frj =Y9Kt -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH v2 0/2] fsl_ssi misc fix
by Jiada Wang 04 Dec '14

04 Dec '14

Hi, Changes in v2: - fix error path in probe - replace irq_of_parse_and_map with platform_get_irq Changes in v1: - free IRQ before irq_dispose_mapping Jiada Wang (2): ASoC: fsl_ssi: fix error path in probe ASoC: fsl_ssi: use platform_get_irq instead of irq_of_parse_and_map sound/soc/fsl/fsl_ssi.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix module unbound
by Fabio Estevam 04 Dec '14

04 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Trying to remove the snd-soc-fsl-ssi module leads to the following warning: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- The cause of problem and solution are well explained by Lars-Peter: "The driver creates the mapping by calling irq_of_parse_and_map(), so it also has to dispose the mapping. But the easy way out is to simply use platform_get_irq() instead of irq_of_parse_map(). In this case the mapping is not managed by the device but by the of core, so the device has not to dispose the mapping." Tested on a imx6q-sabresd board. Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 32a31d9..c528f16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1361,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(ssi_private->regs); } - ssi_private->irq = irq_of_parse_and_map(np, 0); + ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; @@ -1387,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi_private->soc->imx) { ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem); if (ret) - goto error_irqmap; + return ret; } ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, @@ -1458,10 +1458,6 @@ error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -error_irqmap: - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return ret; } @@ -1478,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: soc-pcm: do not hw_free BE if it's still used
by Qiao Zhou 04 Dec '14

04 Dec '14

Do not free BE hw if it's still used by other FE during dpcm runtime shutdown. Otherwise the BE runtime state will be STATE_HW_FREE and won't be updated to STATE_CLOSE when shutdown ends, because BE dai shutdown function won't close pcm when detecting BE is still under use. With STATE_HW_FREE, BE can't be triggered start again. This corner case can easily appear when one BE is used by two FE, without this patch "ASoC: dpcm: Fix race between FE/BE updates and trigger"(ea9d0d771fcd32cd56070819749477d511ec9117). One FE tries to shutdown but it's raced against xrun on another FE. It improves the be dai hw_free logic. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 57277dd..70e8088 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1664,6 +1664,10 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + /* do not free hw if this BE is used by other FE */ + if (be->dpcm[stream].users > 1) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing
by Peter Rosin 04 Dec '14

04 Dec '14

From: Peter Rosin <peda(a)axentia.se> If a snd_soc_card has any DAPM routes when it calls snd_soc_of_parse_audio_routing, those are clobbered without this change. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56ebc0f..377db486852e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4585,7 +4585,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes; + int num_routes, old_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -4603,7 +4603,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + old_routes = card->num_dapm_routes; + routes = devm_kzalloc(card->dev, + (old_routes + num_routes) * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -4611,9 +4613,11 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } + memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); + for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[i].sink); + 2 * i, &routes[old_routes + i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4621,7 +4625,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[i].source); + (2 * i) + 1, &routes[old_routes + i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4630,7 +4634,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; + card->num_dapm_routes += num_routes; card->dapm_routes = routes; return 0; -- 1.7.10.4

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Add ACPI device probing
by Ben Zhang 04 Dec '14

04 Dec '14

The rt5677 codec driver looks for ACPI device ID "RT5677CE", which is specified in coreboot. This patch allows platform data to be obtained via ACPI Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 52 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 50 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..384281d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <linux/firmware.h> #include <linux/gpio.h> #include <sound/core.h> @@ -4525,6 +4526,43 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) return 0; } +#ifdef CONFIG_ACPI + +static unsigned long long rt5677_parse_acpi_entry(struct device *dev, + acpi_string name) +{ + acpi_handle handle = ACPI_HANDLE(dev); + unsigned long long val; + acpi_status status; + + status = acpi_evaluate_integer(handle, name, NULL, &val); + if (ACPI_FAILURE(status)) { + dev_err(dev, "Failed to parse ACPI entry %s, default to 0: %d\n", + name, status); + return 0; + } + return val; +} + +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ + rt5677->pdata.dmic2_clk_pin = (enum rt5677_dmic2_clk) + rt5677_parse_acpi_entry(dev, "DCLK"); + rt5677->pdata.in1_diff = (bool)rt5677_parse_acpi_entry(dev, "IN1"); + rt5677->pdata.in2_diff = (bool)rt5677_parse_acpi_entry(dev, "IN2"); + rt5677->pdata.lout1_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT1"); + rt5677->pdata.lout2_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT2"); + rt5677->pdata.lout3_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT3"); + rt5677->pdata.jd1_gpio = rt5677_parse_acpi_entry(dev, "JD1"); + rt5677->pdata.jd2_gpio = rt5677_parse_acpi_entry(dev, "JD2"); + rt5677->pdata.jd3_gpio = rt5677_parse_acpi_entry(dev, "JD3"); +} +#else +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ +} +#endif + static struct regmap_irq rt5677_irqs[] = { [RT5677_IRQ_JD1] = { .reg_offset = 0, @@ -4604,6 +4642,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + rt5677->pow_ldo2 = -EINVAL; if (i2c->dev.of_node) { ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); if (ret) { @@ -4611,8 +4650,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, ret); return ret; } - } else { - rt5677->pow_ldo2 = -EINVAL; + } else if (ACPI_HANDLE(&i2c->dev)) { + rt5677_parse_acpi(rt5677, &i2c->dev); } if (gpio_is_valid(rt5677->pow_ldo2)) { @@ -4708,10 +4747,19 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_id[] = { + { "RT5677CE", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_id); +#endif + static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5677_acpi_id), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, -- 2.1.0.rc2.206.gedb03e5

6 35

[alsa-devel] [PATCH v2] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 04 Dec '14

04 Dec '14

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 676 +++++++++++++++++++++ 7 files changed, 707 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index acb4523..d7bfeec 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10204,6 +10204,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..9ca12e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -56,6 +56,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..99d7222 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -33,3 +33,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..bd66be2 --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,676 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

1 0

[alsa-devel] [RFC 0/2] get rid of remaining users of attach_adapter
by Wolfram Sang 04 Dec '14

04 Dec '14

The attach_adapter mechanism of the I2C framework is deprecated for years. There are two users left, drivers for old Macintosh computers. I recently got the idea of replacing this mechanism with a custom one with the help of deferred probing. Because I don't have the hardware, I worked on a seperate branch where I modified the windtunnel driver to be runnable on my Renesas Lager board (ARM). I first verified that the attach_adapter method was used, then the driver rightfully failed on detecting the i2c clients. Using my custom mechanism, the same happens: all busses get scanned, then the clients cannot be detected. Since I didn't change the actual detection code, I assume that it should be working on those Macintosh devices as well. The keywest driver is only compile-tested. That being said, I'd be more than happy, if we could find someone willing to test these patches. If they could be applied, we can _finally_ get rid of this legacy mechanism and clean up the i2c core. The branch I used for ARM compilation is here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/attach_adapter_removal_with_arm_compile-experimental Thanks, Wolfram Wolfram Sang (2): macintosh: therm_windtunnel: drop using attach adapter sound: ppc: keywest: drop using attach adapter drivers/macintosh/therm_windtunnel.c | 25 +++++++++++++++++++++++-- sound/ppc/keywest.c | 21 ++++++++++++++++++--- 2 files changed, 41 insertions(+), 5 deletions(-) -- 2.1.3

2 4

[alsa-devel] [PATCH v2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Dec '14

04 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. If we use PLL to generate sysclk, wm8960_set_dai_pll() will enable PLL. But PLL doesn't power down after using codec. So power down PLL in suspend(). Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 59 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ 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 031a1ae..2c7a97a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,41 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + /* Power down PLL to save power */ + regmap_update_bits(wm8960->regmap, WM8960_POWER2, 0x1, 0); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1122,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 2

[alsa-devel] [PATCH 4/5] ASoC: dwc: Add devicetree support for Designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

Convert to driver to use either platform_data or device-tree for configuration of the device. When using device-tree, the I2S block's configuration is read from the relevant registers: this reduces the amount of information required in the device tree. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 238 ++++++++++++++++---- 3 files changed, 222 insertions(+), 49 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt new file mode 100644 index 0000000..cdee591 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/designware-i2s.txt @@ -0,0 +1,32 @@ +DesignWare I2S controller + +Required properties: + - compatible : Must be "snps,designware-i2s" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the APB interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "apb_plck" for the clock to the APB interface, "i2sclk" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects one or two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x0 0x7ff90000 0x0 0x1000>; + clocks = <&scpi_i2sclk 0>, <&soc_refclk100mhz>; + clock-names = "i2sclk", "apb_pclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index e334900..d50e085 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,6 +1,7 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" depends on CLKDEV_LOOKUP + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index c497ada..083779d 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> /* common register for all channel */ #define IER 0x000 @@ -54,19 +55,46 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +/* + * Component parameter register fields - define the I2S block's + * configuration. + */ +#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25) +#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22) +#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19) +#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16) +#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9) +#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7) +#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6) +#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5) +#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4) +#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2) +#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0) + +#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10) +#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7) +#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3) +#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0) + #define MAX_CHANNEL_NUM 8 #define MIN_CHANNEL_NUM 2 +union snd_dma_data { + struct i2s_dma_data pd; + struct snd_dmaengine_dai_dma_data dt; +}; + struct dw_i2s_dev { void __iomem *i2s_base; struct clk *clk; int active; unsigned int capability; struct device *dev; + bool using_pd; /* data related to DMA transfers b/w i2s and DMAC */ - struct i2s_dma_data play_dma_data; - struct i2s_dma_data capture_dma_data; + union snd_dma_data play_dma_data; + union snd_dma_data capture_dma_data; struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; @@ -153,7 +181,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - struct i2s_dma_data *dma_data = NULL; + union snd_dma_data *dma_data = NULL; if (!(dev->capability & DWC_I2S_RECORD) && (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) @@ -227,8 +255,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - /* Iterate over set of channels - independently controlled. - */ + /* Iterate over set of channels - independently controlled. */ do { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), @@ -251,13 +278,18 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, config->sample_rate = params_rate(params); - if (!dev->i2s_clk_cfg) - return -EINVAL; + if (dev->using_pd) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk; - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; + /* TODO: Validate sample rate against permissible set */ + bitclk = config->sample_rate * config->data_width * 2; + clk_set_rate(dev->clk, bitclk); } return 0; @@ -331,6 +363,31 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) #define dw_i2s_resume NULL #endif +/* Maximum resolution of a channel - not uniformly spaced */ +static const u32 fifo_width[] = { + 12, 16, 20, 24, 32, 0, 0, 0 +}; + +/* Width of (DMA) bus */ +static const u32 bus_widths[] = { + DMA_SLAVE_BUSWIDTH_1_BYTE, + DMA_SLAVE_BUSWIDTH_2_BYTES, + DMA_SLAVE_BUSWIDTH_4_BYTES, + DMA_SLAVE_BUSWIDTH_UNDEFINED +}; + +/* PCM format to support channel resolution */ +static const u32 formats[] = { + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S32_LE, + 0, + 0, + 0 +}; + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -340,11 +397,6 @@ static int dw_i2s_probe(struct platform_device *pdev) unsigned int cap; struct snd_soc_dai_driver *dw_i2s_dai; - if (!pdata) { - dev_err(&pdev->dev, "Invalid platform data\n"); - return -EINVAL; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); @@ -373,47 +425,114 @@ static int dw_i2s_probe(struct platform_device *pdev) return PTR_ERR(dev->i2s_base); } - cap = pdata->cap; - dev->capability = cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (pdata) { + dev->using_pd = true; + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock device\n"); + return -ENODEV; + } - /* Set DMA slaves info */ + /* Set DMA slaves info */ + + dev->play_dma_data.pd.data = pdata->play_dma_data; + dev->capture_dma_data.pd.data = pdata->capture_dma_data; + dev->play_dma_data.pd.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.pd.addr = res->start + I2S_RXDMA; + dev->play_dma_data.pd.max_burst = 16; + dev->capture_dma_data.pd.max_burst = 16; + dev->play_dma_data.pd.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.pd.addr_width = + DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.pd.filter = pdata->filter; + dev->capture_dma_data.pd.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " designware: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } - dev->play_dma_data.data = pdata->play_dma_data; - dev->capture_dma_data.data = pdata->capture_dma_data; - dev->play_dma_data.addr = res->start + I2S_TXDMA; - dev->capture_dma_data.addr = res->start + I2S_RXDMA; - dev->play_dma_data.max_burst = 16; - dev->capture_dma_data.max_burst = 16; - dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->play_dma_data.filter = pdata->filter; - dev->capture_dma_data.filter = pdata->filter; + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "designware: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + } else { + /* + * Read component parameter registers to extract + * the I2S block's configuration. + */ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); + u32 bus_width = bus_widths[COMP1_APB_DATA_WIDTH(comp1)]; + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); + u32 max_size; + + dev->using_pd = false; + + dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + /* + * Code presumes all channels are configured with the same + * word size. + */ + if (COMP1_TX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "playback capable\n"); + + dev->capability |= DWC_I2S_PLAY; + max_size = COMP1_TX_WORDSIZE_0(comp1); + dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; + dev->play_dma_data.dt.addr_width = bus_width; + dev->play_dma_data.dt.chan_name = "TX"; + dev->play_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size]) >> 8; + dev->play_dma_data.dt.maxburst = 16; + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = + 1 << (COMP1_TX_CHANNELS(comp1) + 1); + dw_i2s_dai->playback.formats = formats[max_size]; + dw_i2s_dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + } + if (COMP1_RX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "record capable\n"); + + dev->capability |= DWC_I2S_RECORD; + max_size = COMP2_RX_WORDSIZE_0(comp2); + dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; + dev->capture_dma_data.dt.addr_width = bus_width; + dev->capture_dma_data.dt.chan_name = "RX"; + dev->capture_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size] >> 8); + dev->capture_dma_data.dt.maxburst = 16; + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = + 1 << (COMP1_RX_CHANNELS(comp1) + 1); + dw_i2s_dai->capture.formats = formats[max_size]; + dw_i2s_dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + } + } - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); + ret = clk_prepare(dev->clk); + if (ret < 0) + goto err_clk_put; ret = clk_enable(dev->clk); if (ret < 0) goto err_clk_put; - if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " designware: play supported\n"); - dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->playback.channels_max = pdata->channel; - dw_i2s_dai->playback.formats = pdata->snd_fmts; - dw_i2s_dai->playback.rates = pdata->snd_rates; - } - - if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "designware: record supported\n"); - dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->capture.channels_max = pdata->channel; - dw_i2s_dai->capture.formats = pdata->snd_fmts; - dw_i2s_dai->capture.rates = pdata->snd_rates; - } - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, @@ -423,6 +542,15 @@ static int dw_i2s_probe(struct platform_device *pdev) goto err_clk_disable; } + if (!dev->using_pd) { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, + "Could not register PCM: %d\n", ret); + return ret; + } + } + return 0; err_clk_disable: @@ -443,12 +571,24 @@ static int dw_i2s_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dw_i2s_of_match[] = { + { .compatible = "snps,designware-i2s", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, dw_i2s_of_match); +#endif + static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = dw_i2s_of_match, +#endif }, }; -- 1.7.1

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[alsa-devel] [PATCH 2/5] ASoC: dwc: Don't allow negative use counts
by Andrew Jackson 04 Dec '14

04 Dec '14

Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 08f0229..f8946bd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -280,7 +280,8 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev->active--; + if (dev->active > 0) + dev->active--; i2s_stop(dev, substream); break; default: -- 1.7.1

3 6

[alsa-devel] [PATCH 1/5] ASoC: dwc: Allocate resources with devm_ioremap_resource
by Andrew Jackson 04 Dec '14

04 Dec '14

Prepare for the introduction of device-tree support by re-ordering some of the allocations and using devm_iomap_resource to simplify IO mapping. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 46 ++++++++++++++++----------------------- 1 files changed, 19 insertions(+), 27 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e961388..08f0229 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -338,31 +338,34 @@ static int dw_i2s_probe(struct platform_device *pdev) return -EINVAL; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "no i2s resource defined\n"); - return -ENODEV; - } - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(&pdev->dev, "i2s region already claimed\n"); - return -EBUSY; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - dev->i2s_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!dev->i2s_base) { - dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); return -ENOMEM; } + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dev->i2s_base)) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return PTR_ERR(dev->i2s_base); + } + cap = pdata->cap; dev->capability = cap; dev->i2s_clk_cfg = pdata->i2s_clk_cfg; @@ -388,13 +391,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_clk_put; - dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); - if (!dw_i2s_dai) { - dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); - ret = -ENOMEM; - goto err_clk_disable; - } - if (cap & DWC_I2S_PLAY) { dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; @@ -411,10 +407,6 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->capture.rates = pdata->snd_rates; } - dw_i2s_dai->ops = &dw_i2s_dai_ops; - dw_i2s_dai->suspend = dw_i2s_suspend; - dw_i2s_dai->resume = dw_i2s_resume; - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, -- 1.7.1

3 5

[alsa-devel] [PATCH 3/5] ASoC: dwc: Iterate over all channels
by Andrew Jackson 04 Dec '14

04 Dec '14

On the Designware core, the channels are independent and not combined in higher registers. So as more channels are added, more registers need to be updated. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 32 +++++++++++++++++++------------- 1 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index f8946bd..c497ada 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -227,19 +227,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); - } else { - i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); - } + /* Iterate over set of channels - independently controlled. + */ + do { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + } while (ch_reg-- > 0); i2s_write_reg(dev->i2s_base, CCR, ccr); -- 1.7.1

3 4

[alsa-devel] [PATCH 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Andrew Jackson (5): ASoC: dwc: Allocate resources with devm_ioremap_resource ASoC: dwc: Don't allow negative use counts ASoC: dwc: Iterate over all channels ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap .../devicetree/bindings/sound/designware-i2s.txt | 32 ++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 313 ++++++++++++++------ 3 files changed, 260 insertions(+), 86 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

2 2

[alsa-devel] [PATCH 0/2] rockchip: i2s: patches to add optional clock output from
by Jianqun Xu 04 Dec '14

04 Dec '14

The clock output from rockchip SoCs' CRU to chip outside is from the same source as hclk which is for i2s controller. We make it optional because some audio codec outside not need the master clock from SoC but from other clock source. Jianqun Xu (2): ASoC: rockchip-i2s: dt: add an optional property "i2s_clk_out" ASoC: rockchip: i2s: add support for grabbing output clock to codec Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 8 ++++++-- sound/soc/rockchip/rockchip_i2s.c | 11 +++++++++++ 2 files changed, 17 insertions(+), 2 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 03 Dec '14

03 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Switch to use TLV_DB_MINMAX as this allows the minimum to be off by only 0.005dB. This is an improvement from the ~500dB is was misreported by before. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a4c71f9 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,9 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); -- 2.1.2.330.g565301e

3 6

[alsa-devel] [PATCH 0/2] ALSA: ctxfi: Delete an unnecessary check before kfree()
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 20:35:41 +0100 Another update suggestion was taken into account after a patch was applied from static source code analysis. Markus Elfring (2): Deletion of an unnecessary check before the function call "kfree" One function call less in get_daio_rsc() after error detection sound/pci/ctxfi/ctdaio.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) -- 2.1.3

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