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) {

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[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

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