December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [BUG] Race in spdif initialisation / ASoC oopses on module removal
by Russell King - ARM Linux 16 Dec '13

16 Dec '13

So, I have the following as modules, and Ubuntu being Ubuntu, it loads them in an apparantly random order on each boot. The order in which it loaded it this time was (most recent first): snd_soc_fsl_spdif imx_sdma imx_pcm_dma snd_soc_imx_spdif snd_soc_spdif_tx which produced these messages: imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-spdif sound-spdif.17: ASoC: CPU DAI (null) not registered imx-spdif sound-spdif.17: snd_soc_register_card failed: -517 platform sound-spdif.17: Driver imx-spdif requests probe deferral imx-sdma 20ec000.sdma: initialized imx-sdma 20ec000.sdma: loaded firmware 1.1 Unloading and reloading snd_soc_imx_spdif then caused this: fsl-spdif-dai 2004000.spdif: Missing dma channel for stream: 0 fsl-spdif-dai 2004000.spdif: ASoC: pcm constructor failed: -22 imx-spdif sound-spdif.17: ASoC: can't create pcm S/PDIF PCM Playback :-22 imx-spdif sound-spdif.17: ASoC: failed to instantiate card -22 imx-spdif sound-spdif.17: snd_soc_register_card failed: -22 imx-spdif: probe of sound-spdif.17 failed with error -22 fsl-spdif-dai 2004000.spdif: Missing dma channel for stream: 0 fsl-spdif-dai 2004000.spdif: ASoC: pcm constructor failed: -22 imx-spdif sound-spdif.17: ASoC: can't create pcm S/PDIF PCM Playback :-22 imx-spdif sound-spdif.17: ASoC: failed to instantiate card -22 imx-spdif sound-spdif.17: snd_soc_register_card failed: -22 and unloading all of them (except for imx_sdma and snd_soc_spdif_tx) and reloading them in this order (most recent first): snd_soc_imx_spdif snd_soc_fsl_spdif imx_pcm_dma then produced: imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok Clearly something isn't right here. Unloading snd_soc_imx_spdif to then try and re-provoke the problem then caused this: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90() sysfs: can not remove 'dapm_widget', no directory Modules linked in: snd_soc_imx_spdif(-) snd_soc_fsl_spdif imx_pcm_dma fuse bnep rfcomm bluetooth imx_sdma imx_thermal imx2_wdt snd_soc_spdif_tx hid_cypress [last unloaded: imx_pcm_dma] CPU: 0 PID: 1301 Comm: rmmod Not tainted 3.13.0-rc4+ #387 Backtrace: [<c0012640>] (dump_backtrace) from [<c00127dc>] (show_stack+0x18/0x1c) r6:00000393 r5:c016bfa4 r4:00000000 r3:00000000 [<c00127c4>] (show_stack) from [<c0683a18>] (dump_stack+0x70/0x90) [<c06839a8>] (dump_stack) from [<c00244e0>] (warn_slowpath_common+0x74/0x94) r4:dab89d30 r3:00000000 [<c002446c>] (warn_slowpath_common) from [<c00245a4>] (warn_slowpath_fmt+0x38/0x40) r8:c689ce00 r7:00000000 r6:00000000 r5:c077c04c r4:dba2a318 [<c0024570>] (warn_slowpath_fmt) from [<c016bfa4>] (sysfs_hash_and_remove+0x84/0x90) r3:c077c04c r2:c06b3c38 [<c016bf20>] (sysfs_hash_and_remove) from [<c016a00c>] (sysfs_remove_file_ns+0x18/0x1c) r7:da916900 r6:00000000 r5:db2c900c r4:dba2a318 [<c0169ff4>] (sysfs_remove_file_ns) from [<c03687d4>] (device_remove_file+0x20/0x24) [<c03687b4>] (device_remove_file) from [<c04f532c>] (snd_soc_dapm_free+0x1c/0x22c) [<c04f5310>] (snd_soc_dapm_free) from [<c04f0c98>] (soc_remove_codec+0x34/0x98) r10:dabae59c r9:00000000 r8:c689ce00 r7:da916900 r6:00000000 r5:db2c900c r4:dba2a200 r3:00000000 [<c04f0c64>] (soc_remove_codec) from [<c04f3188>] (soc_remove_dai_links.clone.28+0x3c0/0x3ec) r4:00000000 r3:00000000 [<c04f2dc8>] (soc_remove_dai_links.clone.28) from [<c04f3248>] (snd_soc_unregister_card+0x94/0xc8) r10:d8435000 r9:db924204 r8:d8435000 r7:d079d480 r6:00000001 r5:00000608 r4:dabae4b8 [<c04f31b4>] (snd_soc_unregister_card) from [<c04ffa70>] (devm_card_release+0x14/0x18) r6:00000003 r5:db924010 r4:dab89e60 r3:c04ffa5c [<c04ffa5c>] (devm_card_release) from [<c036ec74>] (release_nodes+0x190/0x1f8) [<c036eae4>] (release_nodes) from [<c036ed94>] (devres_release_all+0x38/0x54) r10:00000000 r9:dab88000 r8:c000eb44 r7:bef87600 r6:db924044 r5:bf131014 r4:db924010 [<c036ed5c>] (devres_release_all) from [<c036b7a8>] (__device_release_driver+0x80/0xd4) r4:db924010 r3:bf12f000 [<c036b728>] (__device_release_driver) from [<c036bf28>] (driver_detach+0xbc/0xc0) r5:bf131014 r4:db924010 [<c036be6c>] (driver_detach) from [<c036b490>] (bus_remove_driver+0x54/0x98) r6:dab89f3c r5:bf131058 r4:bf131014 r3:db0c8000 [<c036b43c>] (bus_remove_driver) from [<c036c588>] (driver_unregister+0x30/0x4c) r4:bf131014 r3:c68a7a80 [<c036c558>] (driver_unregister) from [<c036d21c>] (platform_driver_unregister+0x14/0x18) r4:00000000 r3:bf12f2d0 [<c036d208>] (platform_driver_unregister) from [<bf12f2e4>] (imx_spdif_driver_exit+0x14/0xd30 [snd_soc_imx_spdif]) [<bf12f2d0>] (imx_spdif_driver_exit [snd_soc_imx_spdif]) from [<c008d764>] (SyS_delete_module+0x140/0x190) [<c008d624>] (SyS_delete_module) from [<c000e980>] (ret_fast_syscall+0x0/0x48) r7:00000081 r6:000120a8 r5:bef87600 r4:00000880 ---[ end trace 09423e64ab60df46 ]--- Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = dab9c000 [00000008] *pgd=2062f831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] SMP ARM Modules linked in: snd_soc_imx_spdif(-) snd_soc_fsl_spdif imx_pcm_dma fuse bnep rfcomm bluetooth imx_sdma imx_thermal imx2_wdt snd_soc_spdif_tx hid_cypress [last unloaded: imx_pcm_dma] CPU: 0 PID: 1301 Comm: rmmod Tainted: G W 3.13.0-rc4+ #387 task: db0c8000 ti: dab88000 task.ti: dab88000 PC is at soc_remove_codec+0x78/0x98 LR is at trace_hardirqs_on+0x14/0x18 pc : [<c04f0cdc>] lr : [<c0066514>] psr: 600f0013 sp : dab89dc8 ip : 00000020 fp : dab89ddc r10: dabae59c r9 : 00000000 r8 : c689ce00 r7 : da916900 r6 : 00000000 r5 : db2c900c r4 : dba2a200 r3 : 00000000 r2 : 00100100 r1 : dabac810 r0 : dabae5f8 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 2ab9c059 DAC: 00000015 Process rmmod (pid: 1301, stack limit = 0xdab88240) Stack: (0xdab89dc8 to 0xdab8a000) 9dc0: 00000000 00000000 dab89e24 dab89de0 c04f3188 c04f0c70 9de0: d84355bc dabae5a0 00200200 00100100 00200200 00100100 d079d480 dabae4b8 9e00: 00000608 00000001 d079d480 d8435000 db924204 d8435000 dab89e44 dab89e28 9e20: c04f3248 c04f2dd4 c04ffa5c dab89e60 db924010 00000003 dab89e54 dab89e48 9e40: c04ffa70 c04f31c0 dab89e94 dab89e58 c036ec74 c04ffa68 dab89e94 a00f0013 9e60: dabae400 d079d480 00000002 db924010 bf131014 db924044 bef87600 c000eb44 9e80: dab88000 00000000 dab89eac dab89e98 c036ed94 c036eaf0 bf12f000 db924010 9ea0: dab89ec4 dab89eb0 c036b7a8 c036ed68 db924010 bf131014 dab89ee4 dab89ec8 9ec0: c036bf28 c036b734 db0c8000 bf131014 bf131058 dab89f3c dab89efc dab89ee8 9ee0: c036b490 c036be78 c68a7a80 bf131014 dab89f14 dab89f00 c036c588 c036b448 9f00: bf12f2d0 00000000 dab89f24 dab89f18 c036d21c c036c564 dab89f34 dab89f28 9f20: bf12f2e4 c036d214 dab89fa4 dab89f38 c008d764 bf12f2dc 00000000 5f646e73 9f40: 5f636f73 5f786d69 69647073 bef80066 dab89f74 dab89f60 c005f0d4 c0069020 9f60: 0077a008 00000000 00000880 bef87600 000120a8 00000081 bf131058 00000880 9f80: dab89f84 00000000 00000880 bef87600 000120a8 00000081 00000000 dab89fa8 9fa0: c000e980 c008d630 00000880 bef87600 bef87600 00000880 00009778 bef875f4 9fc0: 00000880 bef87600 000120a8 00000081 00000001 000120bc 00000001 00000000 9fe0: b6f03840 bef875fc 00008f75 b6f0384c 800f0010 bef87600 00000000 00000000 Backtrace: [<c04f0c64>] (soc_remove_codec) from [<c04f3188>] (soc_remove_dai_links.clone.28+0x3c0/0x3ec) r4:00000000 r3:00000000 [<c04f2dc8>] (soc_remove_dai_links.clone.28) from [<c04f3248>] (snd_soc_unregister_card+0x94/0xc8) r10:d8435000 r9:db924204 r8:d8435000 r7:d079d480 r6:00000001 r5:00000608 r4:dabae4b8 [<c04f31b4>] (snd_soc_unregister_card) from [<c04ffa70>] (devm_card_release+0x14/0x18) r6:00000003 r5:db924010 r4:dab89e60 r3:c04ffa5c [<c04ffa5c>] (devm_card_release) from [<c036ec74>] (release_nodes+0x190/0x1f8) [<c036eae4>] (release_nodes) from [<c036ed94>] (devres_release_all+0x38/0x54) r10:00000000 r9:dab88000 r8:c000eb44 r7:bef87600 r6:db924044 r5:bf131014 r4:db924010 [<c036ed5c>] (devres_release_all) from [<c036b7a8>] (__device_release_driver+0x80/0xd4) r4:db924010 r3:bf12f000 [<c036b728>] (__device_release_driver) from [<c036bf28>] (driver_detach+0xbc/0xc0) r5:bf131014 r4:db924010 [<c036be6c>] (driver_detach) from [<c036b490>] (bus_remove_driver+0x54/0x98) r6:dab89f3c r5:bf131058 r4:bf131014 r3:db0c8000 [<c036b43c>] (bus_remove_driver) from [<c036c588>] (driver_unregister+0x30/0x4c) r4:bf131014 r3:c68a7a80 [<c036c558>] (driver_unregister) from [<c036d21c>] (platform_driver_unregister+0x14/0x18) r4:00000000 r3:bf12f2d0 [<c036d208>] (platform_driver_unregister) from [<bf12f2e4>] (imx_spdif_driver_exit+0x14/0xd30 [snd_soc_imx_spdif]) [<bf12f2d0>] (imx_spdif_driver_exit [snd_soc_imx_spdif]) from [<c008d764>] (SyS_delete_module+0x140/0x190) [<c008d624>] (SyS_delete_module) from [<c000e980>] (ret_fast_syscall+0x0/0x48) r7:00000081 r6:000120a8 r5:bef87600 r4:00000880 Code: e594100c e5842068 e584306c e5913080 (e5930008) ---[ end trace 09423e64ab60df47 ]---

2 1

[alsa-devel] Where to find recent fix for HD-Audio runtime PM?
by Lin, Mengdong 16 Dec '13

16 Dec '13

Hi Takashi, If convenient, would you please point out your previous patches to make HD-Audio runtime PM stable? I guess 'ALSA: hda - Refactor pm notification in hda_codec.c' is only one of them. We observed audio failure (dead communication) after S3 if runtime PM is enabled on Haswell with Kernel v3.9. We hope to back port your previous fix and further check this issue. Many thanks! Mengdong

2 2

[alsa-devel] [PATCH v2] ASoC: SGTL5000: Fix kernel failed while trying to get optional VDDD supply.
by Xiubo Li 16 Dec '13

16 Dec '13

The SGTL5000 requires 2 external power supplies: VDDA and VDDIO. An optional third external power supply VDDD may be provided externally to achieve lower power.If an external supply is not used for VDDD, the SGTL5000 driver will register it's own regulator device, and then provides the VDDD supply consumer, and now there will be two regulator devices exist, local regulator for VDDD and platform regulator for VDDIO, VDDA. ******************* * *---------|3.3V VDDIO * * * SGTL5000 codec *---x VDDD * * * *---------|3.3V VDDA ******************* If an external supply is not used for VDDD, in the DT or architecture-specific file, only "VDDA-supply" and "VDDIO-supply" properties will be presented. This caused the following kernel failed while trying to get the external VDDD supply before trying to register it's own regulator device. sgtl5000 0-000a: Failed to get supply 'VDDD': -19 Here use regulator_get_optional() trying to look at the fact that whether the external VDDD supply is used or not. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..90d6d1e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1298,6 +1298,21 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) return 0; } +static int sgtl5000_external_vddd_used(struct snd_soc_codec *codec) +{ + struct regulator *consumer; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + consumer = regulator_get_optional(codec->dev, + sgtl5000->supplies[VDDD].supply); + if (IS_ERR(consumer)) + return 0; + + regulator_put(consumer); + + return 1; +} + static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { int reg; @@ -1310,11 +1325,15 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + if (sgtl5000_external_vddd_used(codec)) { + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - if (!ret) + if (ret) + return ret; + external_vddd = 1; - else { + } else { ret = sgtl5000_replace_vddd_with_ldo(codec); if (ret) return ret; -- 1.8.4

4 4

[alsa-devel] [PATCH] ALSA: hda - limit mic boost on Asus UX31[A, E]
by Oleksij Rempel 15 Dec '13

15 Dec '13

This both devices need limit for internal dmic. Signed-off-by: Oleksij Rempel <linux(a)rempel-privat.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4ad9d1..b617aa4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3798,6 +3798,7 @@ enum { ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -4075,6 +4076,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269VB_FIXUP_DMIC, + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4189,8 +4196,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), - SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_LIMIT_INT_DMIC_BOOST), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 1.8.3.2

2 7

[alsa-devel] [RFC][PATCH 00/13] bebob: a new driver for BridgeCo BeBoB based device
by Takashi Sakamoto 15 Dec '13

15 Dec '13

In previous series of patch, I showed enhancement of current firewire-lib. So I should show the way to use new functions in each driver side. This driver is under development but functionalities for streaming seems to be fixed. I'm happy to receive your comments. This is a new driver for BridgeCo BeBoB based device. FFADO (successor of FreeBoB project) currently give driver and mixer control in user space. My aim of this series of patch is just to give a support for kernel streaming. In theory, this driver just support kernel streaming and related functionality. But for my debugging, this module still includes some functionality which should be in user space. Additionally, there is a heavily customized version of BeBoB. For some reasons, this driver give a few functionality to control such device. Takashi Sakamoto (13): bebob: Add skelton for BridgeCo BeBoB based devices bebob: Add command and stream management bebob: Add proc interface for debug bebob: Add functionality to change sampling rate and control interface bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface to lock/unlock streaming bebob: Prepare for device specific operations bebob: add support for M-Audio Firewire series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for Terratec PHASE series bebob: Add support for Yamaha GO series bebob: Add support for Presonus FIREBOX/FIREPOD include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 29 ++ sound/firewire/Makefile | 1 + sound/firewire/bebob/Makefile | 6 + sound/firewire/bebob/bebob.c | 450 ++++++++++++++++ sound/firewire/bebob/bebob.h | 255 +++++++++ sound/firewire/bebob/bebob_command.c | 423 +++++++++++++++ sound/firewire/bebob/bebob_control.c | 331 ++++++++++++ sound/firewire/bebob/bebob_focusrite.c | 390 ++++++++++++++ sound/firewire/bebob/bebob_hwdep.c | 207 ++++++++ sound/firewire/bebob/bebob_maudio.c | 910 +++++++++++++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 182 +++++++ sound/firewire/bebob/bebob_pcm.c | 420 +++++++++++++++ sound/firewire/bebob/bebob_presonus.c | 75 +++ sound/firewire/bebob/bebob_proc.c | 164 ++++++ sound/firewire/bebob/bebob_stream.c | 724 ++++++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 143 ++++++ sound/firewire/bebob/bebob_yamaha.c | 176 +++++++ 19 files changed, 4889 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_control.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_presonus.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c -- 1.8.3.2

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[alsa-devel] [PATCH] ASOC:DAPM: extend dapm kcontrol to support runtime route update
by Nenghua Cao 15 Dec '13

15 Dec '13

From: nhcao <nhcao(a)marvell.com> DPCM can dynamically alter the FE to BE PCM links at runtime based on mixer setting updates. DAPM has provided common get/put function for mixer/mux. But these function doesn't call soc_dpcm_runtime_update() func to update pcm links. This patch defines a DPCM DPCM kcontrol. For this control, the common get/put function will update FE to BE links dynamically. This patch has no impact to the current code. Change-Id: I45e291f467e4fa17e21aa8923c2b712d52067aca Signed-off-by: nhcao <nhcao(a)marvell.com> --- include/sound/control.h | 4 ++++ include/sound/soc-dapm.h | 32 ++++++++++++++++++++++++++++++++ sound/core/control.c | 1 + sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++---- 4 files changed, 61 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 5358892..3788284 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -44,6 +44,8 @@ struct snd_kcontrol_new { unsigned int index; /* index of item */ unsigned int access; /* access rights */ unsigned int count; /* count of same elements */ + /* this kcontrol impact fe<->be relationship at runtime */ + unsigned char dpcm_checked; snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; snd_kcontrol_put_t *put; @@ -63,6 +65,8 @@ struct snd_kcontrol { struct list_head list; /* list of controls */ struct snd_ctl_elem_id id; unsigned int count; /* count of same elements */ + /* this kcontrol impact fe<->be relationship at runtime */ + unsigned char dpcm_checked; snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; snd_kcontrol_put_t *put; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 56ebdfc..8962884 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -336,6 +336,38 @@ struct device; .put = snd_soc_dapm_put_pin_switch, \ .private_value = (unsigned long)xname } +/* dapm & dpcm kcontrol types */ +#define SOC_DAPM_DPCM_SINGLE(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DAPM_DPCM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_DAPM_DPCM_ENUM(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = snd_soc_dapm_put_enum_double, \ + .private_value = (unsigned long)&xenum } +#define SOC_DAPM_DPCM_ENUM_VIRT(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_virt, \ + .put = snd_soc_dapm_put_enum_virt, \ + .private_value = (unsigned long)&xenum } +#define SOC_DAPM_DPCM_VALUE_ENUM(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .dpcm_checked = 1, .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_value_enum_double, \ + .put = snd_soc_dapm_put_value_enum_double, \ + .private_value = (unsigned long)&xenum } + /* dapm stream operations */ #define SND_SOC_DAPM_STREAM_NOP 0x0 #define SND_SOC_DAPM_STREAM_START 0x1 diff --git a/sound/core/control.c b/sound/core/control.c index d8aa206..34df2c4 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -254,6 +254,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); + kctl.dpcm_checked = ncontrol->dpcm_checked; kctl.info = ncontrol->info; kctl.get = ncontrol->get; kctl.put = ncontrol->put; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcade13..b568183 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2868,6 +2868,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val; int connect, change; struct snd_soc_dapm_update update; + int ret = 0; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, @@ -2901,12 +2902,16 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, card->update = &update; } - soc_dapm_mixer_update_power(card, kcontrol, connect); + ret = soc_dapm_mixer_update_power(card, kcontrol, connect); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2955,6 +2960,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2978,12 +2984,16 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -3019,6 +3029,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; + int ret = 0; if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; @@ -3028,9 +3039,13 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, value = ucontrol->value.enumerated.item[0]; change = dapm_kcontrol_set_value(kcontrol, value); if (change) - soc_dapm_mux_update_power(card, kcontrol, value, e); + ret = soc_dapm_mux_update_power(card, kcontrol, value, e); mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -3097,6 +3112,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -3120,12 +3136,16 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; - soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); + + if ((ret > 0) && (kcontrol->dpcm_checked == 1)) + soc_dpcm_runtime_update(card); + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); -- 1.7.0.4

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[alsa-devel] Add support for Focusrite Saffire 6 USB
by Эдуард Гильмутдинов 14 Dec '13

14 Dec '13

Hello! I have external audio card Focusrite Saffire 6 USB. This device works with the Linux kernel if I add the following corrections: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..158beb4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..3ea9627 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -660,10 +660,11 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) return err; } -/* +/* * Novation Twitch DJ controller + * Focusrite(Novation) Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ Help me to add this corrections in the Linux kernel mainline. Thank you!

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Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa: Disable timer-scheduling for PCMs with the BATCH flag
by Raymond Yau 14 Dec '13

14 Dec '13

2013/12/6 Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> > On Sat, 2013-11-30 at 18:07 +0100, Lars-Peter Clausen wrote: > > PCM Devices which have the BATCH flag set update the PCM pointer only > with > > period size granularity. Using timer based scheduling does not have any > > advantage in this mode. For one devices which have that flag set usually > update > > the position pointer in software after getting the period interrupt. So > > disabling the period interrupt is not possible for this kind of devices. > > Furthermore writing to or reading from the buffer slice for the current > period > > is not possible since the position inside the buffer is not known. On > the other > > hand the tsched algorithm seems to get easily confused for this kind of > > hardware, which results in garbled audio output. This typically means > that timer > > based scheduling needs to be manually disabled on systems with such > devices. > > Auto disabling tsched in this case allows these systems to run with the > default > > configuration. > > > > Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > --- > > src/modules/alsa/alsa-util.c | 4 ++++ > > 1 file changed, 4 insertions(+) > > > > diff --git a/src/modules/alsa/alsa-util.c b/src/modules/alsa/alsa-util.c > > index 75f5858..4b24e47 100644 > > --- a/src/modules/alsa/alsa-util.c > > +++ b/src/modules/alsa/alsa-util.c > > @@ -245,6 +245,10 @@ int pa_alsa_set_hw_params( > > if (!pa_alsa_pcm_is_hw(pcm_handle)) > > _use_tsched = false; > > > > + /* The PCM pointer is only updated with period granularity */ > > + if (snd_pcm_hw_params_is_batch(hwparams)) > > + _use_tsched = false; > > + > > #if (SND_LIB_VERSION >= ((1<<16)|(0<<8)|24)) /* API additions in 1.0.24 > */ > > if (_use_tsched) { > > > > Thanks! I applied the patch. > > -- > Are you patch sufficient ? In alsa-lib , snd_pcm_sw_params_set_avail_min always return zero int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val) { assert(pcm && params); /* Fix avail_min if it's below period size. The period_size * defines the minimal wake-up timing accuracy, so it doesn't * make sense to set below that. */ if (val < pcm->period_size) val = pcm->period_size; params->avail_min = val; return 0; } http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… /* We need at last one frame in the used part of the buffer */ - avail_min = (snd_pcm_uframes_t) u->hwbuf_unused_frames + 1; + avail_min = (snd_pcm_uframes_t) u->hwbuf_unused / u->frame_size + 1; do pulseaudio need to keep at least one periods instead of more than 1 frames for those driver which update hwptr in periods ?

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Correct S24_3LE support
by Peter Ujfalusi 13 Dec '13

13 Dec '13

Correct the hw_params callback to configure the codec correctly in case of S24_3LE format. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb4b386..31b6ae2334bc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S20_3LE: data |= (0x01 << 4); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: data |= (0x02 << 4); break; case SNDRV_PCM_FORMAT_S32_LE: -- 1.8.5.1

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[alsa-devel] ASoC updates for v3.13
by Mark Brown 13 Dec '13

13 Dec '13

The following changes since commit 374b105797c3d4f29c685f3be535c35f5689b30e: Linux 3.13-rc3 (2013-12-06 09:34:04 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc3 for you to fetch changes up to e20ab019e28dcf09c2727aa69e2a073ed66718b3: Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/tegra' and 'asoc/fix/wm8962' into asoc-linus (2013-12-12 00:30:04 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few driver and error handling fixes plus a fix to ensure that we mute streams when we should. The Atmel trigger addition is a fix to ensure that we do the correct sequence of interactions with the hardware. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dma' into asoc-linus Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/tegra' and 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (3): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver Stephen Warren (2): ASoC: tegra: fix uninitialized variables in set_fmt ASoC: don't leak on error in snd_dmaengine_pcm_register sound/soc/atmel/atmel_ssc_dai.c | 30 ++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/wm8962.c | 13 ++++++++++++ sound/soc/fsl/imx-wm8962.c | 2 -- sound/soc/soc-generic-dmaengine-pcm.c | 38 +++++++++++++++++++++++++---------- sound/soc/soc-pcm.c | 5 +++-- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 ++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 9 files changed, 84 insertions(+), 28 deletions(-)

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