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

[alsa-devel] alsa-tools 1.0.28 regression with sblive/emu10k1: init_live shell script broken.
by Roc Vallès 27 Jun '14

27 Jun '14

I've recently upgraded to alsa-tools 1.0.28. There's apparently a regression: live_init, the script that is bundled for sound blaster live soundcard initialization, has been broken. Without this, the soundcard can only be used at a very basic level (just 2 speakers, no equalization or other effects, most inputs unavailable). Reporting here as the bugtracker is down. Correct run (1.0.17): # sh /usr/bin/init_live # Broken run (1.0.28): # /usr/bin/init_live /usr/bin/init_live: line 49: log: command not found /usr/bin/init_live: line 51: local: can only be used in a function /usr/bin/init_live: line 55: [: 0: unary operator expected /usr/bin/init_live: line 60: runlo10k1: command not found /usr/bin/init_live: line 62: log: command not found rename: not enough arguments Usage: rename [options] expression replacement file... Options: -v, --verbose explain what is being done -s, --symlink act on symlink target -h, --help display this help and exit -V, --version output version information and exit For more details see rename(1). /usr/bin/init_live: line 79: log: command not found /usr/bin/init_live: line 81: runlo10k1: command not found /usr/bin/init_live: line 84: connect: command not found /usr/bin/init_live: line 90: log: command not found /usr/bin/init_live: line 92: runlo10k1: command not found /usr/bin/init_live: line 95: copy: command not found /usr/bin/init_live: line 97: local: can only be used in a function /usr/bin/init_live: line 98: local: can only be used in a function /usr/bin/init_live: line 100: log: command not found /usr/bin/init_live: line 103: runlo10k1: command not found /usr/bin/init_live: line 105: connect: command not found /usr/bin/init_live: line 108: volume: command not found /usr/bin/init_live: line 110: local: can only be used in a function /usr/bin/init_live: line 111: local: can only be used in a function /usr/bin/init_live: line 112: local: can only be used in a function /usr/bin/init_live: line 113: local: can only be used in a function /usr/bin/init_live: line 115: log: command not found /usr/bin/init_live: line 118: runlo10k1: command not found /usr/bin/init_live: line 120: connect: command not found /usr/bin/init_live: line 123: master_volume: command not found /usr/bin/init_live: line 125: local: can only be used in a function /usr/bin/init_live: line 126: local: can only be used in a function /usr/bin/init_live: line 127: local: can only be used in a function /usr/bin/init_live: line 128: local: can only be used in a function /usr/bin/init_live: line 130: log: command not found /usr/bin/init_live: line 133: runlo10k1: command not found /usr/bin/init_live: line 135: connect: command not found /usr/bin/init_live: line 138: switch: command not found /usr/bin/init_live: line 140: local: can only be used in a function /usr/bin/init_live: line 141: local: can only be used in a function /usr/bin/init_live: line 142: local: can only be used in a function /usr/bin/init_live: line 144: log: command not found /usr/bin/init_live: line 148: runlo10k1: command not found /usr/bin/init_live: line 150: connect: command not found /usr/bin/init_live: line 153: switch_2x2: command not found /usr/bin/init_live: line 155: local: can only be used in a function /usr/bin/init_live: line 156: local: can only be used in a function /usr/bin/init_live: line 157: local: can only be used in a function /usr/bin/init_live: line 158: local: can only be used in a function /usr/bin/init_live: line 160: log: command not found /usr/bin/init_live: line 163: runlo10k1: command not found /usr/bin/init_live: line 165: connect: command not found /usr/bin/init_live: line 166: connect: command not found /usr/bin/init_live: line 169: switch_5x2: command not found /usr/bin/init_live: line 171: local: can only be used in a function /usr/bin/init_live: line 172: local: can only be used in a function /usr/bin/init_live: line 173: local: can only be used in a function /usr/bin/init_live: line 174: local: can only be used in a function /usr/bin/init_live: line 176: log: command not found /usr/bin/init_live: line 179: runlo10k1: command not found /usr/bin/init_live: line 181: connect: command not found /usr/bin/init_live: line 182: connect: command not found /usr/bin/init_live: line 185: master_switch: command not found /usr/bin/init_live: line 187: local: can only be used in a function /usr/bin/init_live: line 188: local: can only be used in a function /usr/bin/init_live: line 189: local: can only be used in a function /usr/bin/init_live: line 190: local: can only be used in a function /usr/bin/init_live: line 192: log: command not found /usr/bin/init_live: line 195: runlo10k1: command not found /usr/bin/init_live: line 197: connect: command not found /usr/bin/init_live: line 200: playback: command not found /usr/bin/init_live: line 205: log: command not found /usr/bin/init_live: line 208: switch: command not found /usr/bin/init_live: line 209: volume: command not found /usr/bin/init_live: line 211: add_output: command not found /usr/bin/init_live: line 214: playback_noswitch: command not found /usr/bin/init_live: line 219: log: command not found /usr/bin/init_live: line 222: volume: command not found /usr/bin/init_live: line 224: add_output: command not found /usr/bin/init_live: line 227: capture: command not found /usr/bin/init_live: line 232: log: command not found /usr/bin/init_live: line 235: switch: command not found /usr/bin/init_live: line 236: volume: command not found /usr/bin/init_live: line 238: connect: command not found /usr/bin/init_live: line 241: capture_noswitch: command not found /usr/bin/init_live: line 246: log: command not found /usr/bin/init_live: line 249: volume: command not found /usr/bin/init_live: line 251: connect: command not found /usr/bin/init_live: line 254: master: command not found /usr/bin/init_live: line 256: local: can only be used in a function /usr/bin/init_live: line 257: local: can only be used in a function /usr/bin/init_live: line 258: local: can only be used in a function /usr/bin/init_live: line 260: log: command not found /usr/bin/init_live: line 263: master_volume: command not found /usr/bin/init_live: line 264: master_switch: command not found /usr/bin/init_live: line 267: dump_patches: command not found

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[alsa-devel] [RFC] Add DummyDevice property to UCM
by Felipe Tonello 27 Jun '14

27 Jun '14

Hi, I would like to see what you guys think about adding a DummyDevice official property to a UCM device. This is very useful for users to configure properly when, let's say, the PCM is a dummy device. It is a common practice on embedded systems to have dummy PCMs. BR Felipe

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[alsa-devel] [PATCH] ucm: Document PlaybackPCMIsDummy and CapturePCMIsDummy values
by Tanu Kaskinen 27 Jun '14

27 Jun '14

At least PulseAudio needs special handling for dummy devices. To allow that to happen automatically, the UCM configuration should contain the information about which PCMs are dummy. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- include/use-case.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/include/use-case.h b/include/use-case.h index 4e13249..f30168f 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -258,7 +258,17 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * Recommended names for values: * TQ - Tone Quality * PlaybackPCM - full PCM playback device name + * PlaybackPCMIsDummy - Valid values: "yes" and "no". If set to "yes", the + * PCM named by the PlaybackPCM value is a dummy device, + * meaning that opening it enables an audio path in the + * hardware, but writing to the PCM device has no + * effect. * CapturePCM - full PCM capture device name + * CapturePCMIsDummy - Valid values: "yes" and "no". If set to "yes", the + * PCM named by the CapturePCM value is a dummy device, + * meaning that opening it enables an audio path in the + * hardware, but reading from the PCM device has no + * effect. * PlaybackRate - playback device sample rate * PlaybackChannels - playback device channel count * PlaybackCTL - playback control device name -- 1.9.3

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[alsa-devel] [PATCH 01/16] ASoC: cs4270: Remove redundant OOM message
by Sachin Kamat 27 Jun '14

27 Jun '14

Let memory subsystem handle the error logging. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/cs4270.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 9947a9583679..e6d4ff9fd992 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -664,10 +664,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), GFP_KERNEL); - if (!cs4270) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs4270) return -ENOMEM; - } /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: samsung: Remove unused variable from idma.c
by Sachin Kamat 27 Jun '14

27 Jun '14

‘iiscon’ is not used in the function. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/samsung/idma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 8cc5770abb39..db6cefa18017 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -261,10 +261,9 @@ static int idma_mmap(struct snd_pcm_substream *substream, static irqreturn_t iis_irq(int irqno, void *dev_id) { struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; - u32 iiscon, iisahb, val, addr; + u32 iisahb, val, addr; iisahb = readl(idma.regs + I2SAHB); - iiscon = readl(idma.regs + I2SCON); val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; -- 1.7.9.5

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Re: [alsa-devel] No sound on Thinkpad T440s in Docking Station
by Jim Minter 27 Jun '14

27 Jun '14

Hello, I'm getting the same issue reported by Joschi in May, but with a docked T540p. I can confirm that doing the following fixes the issue for me: ==== 8< ==== # cat <<EOF >/lib/firmware/t540p.fw [codec] 0x10ec0292 0x17aa2210 0 [pincfg] 0x16 0x21211010 0x19 0x21a11010 EOF # cat <<EOF >/etc/modprobe.d/hda-intel.conf options snd-hda-intel patch=t540p.fw,t540p.fw,t540p.fw EOF Then rmmod/modprobe the snd-hda-intel kernel module, or just reboot. ==== 8< ==== I can see http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=1c37… which I presume resolves this problem for T440s and T440p users. Please could an additional line, I'm guessing as follows, be added to sound/pci/hda/patch_realtek.c to include T540p users in this happy group? + SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), Thanks! Jim

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[alsa-devel] [PATCH] ALSA: hda - Add another headset pin quirk for some Dell machines
by David Henningsson 27 Jun '14

27 Jun '14

Another quirk to make the headset mic work on some new Dell machines. Cc: Hui Wang <hui.wang(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1297581 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 60bf5ee..4ebbe03 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5199,6 +5199,18 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x411111f0}, {0x1d, 0x40700001}, {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x13, 0x90a60140}, + {0x14, 0x90170110}, + {0x15, 0x0221401f}, + {0x16, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x411111f0}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), {} }; -- 1.9.1

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[alsa-devel] [PATCH 0/3 v3] arizona: Improvements to codec DVFS control
by Charles Keepax 27 Jun '14

27 Jun '14

Hi, Respin of Richard's DVFS series, from me since he is on holiday. I have based these on Mark's ASoC for-next branch as there a few changes in there to the same parts of code. But let me know if you would rather have a series based on Lee's tree instead. This set of changes cleans up the way the drivers handle the control of internal subsystem clocks for codecs with DVFS. This previously relied on the clock control happening as a side-effect of changing the regulator voltage, rather than explicitly, which assumed that you were using the built-in regulator. Also DVFS control was being done from the ADSP driver despite that driver supposedly being codec- agnostic. There are other cases on some codecs where DVFS control is needed so this sequence of patches cleans up the way DVFS is implemented and adds the extra case of DVFS boost for higher sample rates on WM8997 and WM5102 Changes since v2: - Correct some small issues (wrong printf specifier, clarity on PRE_PMD/POST_PMD event) - Added warning for calling arizona_dvfs_down with flags that are not enabled - Squashed patches 2 and 4, so the SUBSYS bit is not set from two seperate places Changes since v1: - Changed mask variable to flags to be more consistent with its usage - Fixed minor tabbing issue - Adding missing SND_SOC_DAPM_PRE_PMD event flag in WM_ADSP2_E Richard Fitzgerald (3): mfd: arizona: Export function to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control drivers/mfd/arizona-core.c | 87 ++++++++++++++++++++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 5 -- include/linux/mfd/arizona/core.h | 12 +++++ sound/soc/codecs/arizona.c | 22 +++++++++- sound/soc/codecs/wm5102.c | 47 +++++++++++++++++++- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 73 +------------------------------- sound/soc/codecs/wm_adsp.h | 13 +++--- 8 files changed, 174 insertions(+), 87 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH V2] ASoC: mcbsp: remove unnecessary null test before kfree
by Fabian Frederick 27 Jun '14

27 Jun '14

Fix checkpatch warning: WARNING: kfree(NULL) is safe this check is probably not required Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- V2: replace subject "sound/soc/omap/mcbsp.c" by "ASoC: mcbsp:" (suggested by Peter Ujfalusi) sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 1.8.4.5

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[alsa-devel] infinity wait in snd_pcm_wait_nocheck
by 김은희 27 Jun '14

27 Jun '14

My system plays and captures short streams using by dmix device after boot. But that system sometimes go infinity wait state when stream plays. I use alsa-lib-1.0.24.1. but v1.0.27 and 1.0.28 also occur same err. So I have below questions. - What event that poll (in snd_pcm_wait_nocheck) wait? I know snd_pcm_wait function is for wait PCM ready. But I want to know details. - How can I avoid that problem? I changed timeout(in snd_pcm_wait) from -1 to 100. But system go infinity loop. And I attach status and params when this problem occurred. - status state: RUNNING owner_pid : 1213 trigger_time: 5.285075511 tstamp : 5.333190511 delay : -832 avail : 17216 avail_max : 17216 ----- hw_ptr : 832 appl_ptr : 0 - hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 1024 buffer_size: 16384 - sw_params tstamp_mode: ENABLE period_step: 1 avail_min: 1024 start_threshold: 1 stop_threshold: 1073741824 silence_threshold: 0 silence_size: 1073741824 boundary: 1073741824 Please reply about it. Thanks. ----- Eunhee Kim

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