January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] No internal speakers on laptop, suspend-to-ram and resume makes them work
by Wolfy 11 Jan '15

11 Jan '15

I have a Panasonic Toughbook CF-52 with intel hda and AD1833. No sound from internal speakers, but sound from headphones jack works correctly. Based on a forum post I found regarding the same issue in Win7, I tried suspending to ram and resuming, and sound works afterwards, until the next reboot. debianuser (very very helpful!) on irc said I should post this here, as different settings after resume shouldn't be, suggesting a bug. Plus, sound not working for a pretty large laptop brand suggests a fix/quirk of some kind is needed, even if the real bug is in the laptop. Some notes / things tried: Headphones output seems to work correctly on boot. No alsamixer sliders/muting/switches result in sound from speakers. Googling shows this affects a pretty wide variety of panasonic laptops, not just mine or my model. Googling shows this happens in win7 (but not earlier windows versions) unless a "Hotkey Appendix Driver" from panasonic is some random combination (depending on post) of installed, uninstalled and reinstalled. The keyboard does have a few audio hotkeys, but they seem to be software only, and are simply grabbed by xfce's volumed, where they seem to do exactly what they should. But apparently the panasonic windows one does something else with audio from within the hotkey driver? Perhaps it mutes it by toggling some non-obvious bit? Or despite being named "hotkey", it also handles the headphones jack automute? Googling found some reports saying that booting into windows (with the proper extra driver installed) then rebooting into linux will make sound work "for a while", even over reboots. I do not have windows to test these posts. This suggests it's something being saved somewhere that survives poweroff. Googling didn't find anyone with this problem who managed to make sound work correctly without windows. (not good!) It always works (internal speakers come on) after a suspend-to-ram and resume. It always stops working after a restart, suspend-to-disk or any other kind of reboot. Reloading all snd* modules makes no change. If it's not working, it stays not working, if it's working, it stays working. Suspending and resuming fixes it even if all the snd* modules are unloaded before suspending and loaded after resuming. When working (after suspend and resume), the labels shown in alsamixer are wrong - for example, Headphones now controls the internal speakers, while the Speakers slider does exactly nothing. Loading the module with generic, nofixup, ref models did nothing. I installed hda-jack-retask to see if I could make it work that way, but I was unable to make hda-jack-retask do anything at all - even setting every pin to not connected results in absolutely no change in operation. One forum thread on this problem, with some hdajackretask screenshots, is: http://www.toughbooktalk.com/viewtopic.php?f=39&t=1909 I've attached three alsa-info.sh files. The "broken" one is with sound not working. I then suspended and resumed, then generated the "working" one. The third is a diff between them, to save people time trying to spot changes. Thanks, --Randy

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[alsa-devel] [PATCHv2 1/2] ALSA: fix emu8000 DRAM sizing for AWE64 Value
by David Flater 11 Jan '15

11 Jan '15

Applicable to any kernel since 2013: The special case added in commit 1338fc97d07a did not handle the possibility that the address space on an AWE64 Value would wrap around at 512 KiB. That is what it does, so the memory is still not detected on those cards. Fix that with a logic clean-up that eliminates the need for a special case. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). CT4380 is commonly said to come with 1 MiB of DRAM, but Creative's AWE Control app agreed that mine has only 512 KiB. It has the same memory chip as the CT4520. The affected function first appeared in alsa-driver-0.3.0 and was merged in linux-2.5.5. Its somewhat different ancestor was in sound/oss/awe_wave.c. sound/isa/sb/emu8000.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff..3dcf80e 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -378,13 +378,12 @@ init_arrays(struct snd_emu8000 *emu) static void size_dram(struct snd_emu8000 *emu) { - int i, size, detected_size; + int i, size; if (emu->dram_checked) return; size = 0; - detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -392,11 +391,12 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_WRITE(emu, UNIQUE_ID1); snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */ + snd_emu8000_write_wait(emu); + + goto size_dram_ID1_check; /* If that fails, we have no RAM. */ while (size < EMU8000_MAX_DRAM) { - size += 512 * 1024; /* increment 512kbytes */ - /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -424,23 +424,20 @@ size_dram(struct snd_emu8000 *emu) * wraps back to the beginning; so check to see if the * initial value has been overwritten. */ + + size_dram_ID1_check: EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) - break; /* we must have wrapped around */ + break; snd_emu8000_read_wait(emu); - /* Otherwise, it's valid memory. */ - detected_size = size + 512 * 1024; - } - - /* Distinguish 512 KiB from 0. */ - if (detected_size == 0) { - snd_emu8000_read_wait(emu); - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) - detected_size = 512 * 1024; + /* + * Otherwise, it's valid memory. If a write succeeds at + * the beginning of a 512 KiB page we assume that the whole + * page is there. + */ + size += 512 * 1024; /* increment 512 KiB */ } /* wait until FULL bit in SMAxW register is false */ @@ -455,9 +452,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, detected_size/1024); + emu->port1, size/1024); - emu->mem_size = detected_size; + emu->mem_size = size; emu->dram_checked = 1; } -- 1.8.4

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[alsa-devel] [PATCH - alsa-lib 1/1] include/global.h: missing struct on FreeBSD
by rumko＠rumko.net 11 Jan '15

11 Jan '15

From: Rumko <rumko(a)rumko.net> On FreeBSD, struct timespec is not accessible through <time.h>, but through <sys/tiespec.h>. Add specific include for that header on that platform only. Signed-off-by: Rumko <rumko(a)rumko.net> diff --git a/include/global.h b/include/global.h index 16a26dc..70be1f9 100644 --- a/include/global.h +++ b/include/global.h @@ -31,6 +31,10 @@ /* for timeval and timespec */ #include <time.h> +#if defined(__FreeBSD__) +#include <sys/timespec.h> +#endif + #ifdef __cplusplus extern "C" { #endif -- 2.2.1

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[alsa-devel] [PATCH] ALSA: ice1712: wm8766: Remove some unused functions
by Rickard Strandqvist 11 Jan '15

11 Jan '15

Removes some functions that are not used anywhere: snd_wm8766_set_power() snd_wm8766_set_master_mode() 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/ice1712/wm8766.c | 16 ---------------- sound/pci/ice1712/wm8766.h | 2 -- 2 files changed, 18 deletions(-) diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 21b373b..f7ac8d5 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -183,22 +183,6 @@ void snd_wm8766_set_if(struct snd_wm8766 *wm, u16 dac) snd_wm8766_write(wm, WM8766_REG_IFCTRL, val | dac); } -void snd_wm8766_set_master_mode(struct snd_wm8766 *wm, u16 mode) -{ - u16 val = wm->regs[WM8766_REG_DACCTRL3] & ~WM8766_DAC3_MSTR_MASK; - - mode &= WM8766_DAC3_MSTR_MASK; - snd_wm8766_write(wm, WM8766_REG_DACCTRL3, val | mode); -} - -void snd_wm8766_set_power(struct snd_wm8766 *wm, u16 power) -{ - u16 val = wm->regs[WM8766_REG_DACCTRL3] & ~WM8766_DAC3_POWER_MASK; - - power &= WM8766_DAC3_POWER_MASK; - snd_wm8766_write(wm, WM8766_REG_DACCTRL3, val | power); -} - void snd_wm8766_volume_restore(struct snd_wm8766 *wm) { u16 val = wm->regs[WM8766_REG_DACR1]; diff --git a/sound/pci/ice1712/wm8766.h b/sound/pci/ice1712/wm8766.h index c119f84..18c8d9d 100644 --- a/sound/pci/ice1712/wm8766.h +++ b/sound/pci/ice1712/wm8766.h @@ -155,8 +155,6 @@ struct snd_wm8766 { void snd_wm8766_init(struct snd_wm8766 *wm); void snd_wm8766_resume(struct snd_wm8766 *wm); void snd_wm8766_set_if(struct snd_wm8766 *wm, u16 dac); -void snd_wm8766_set_master_mode(struct snd_wm8766 *wm, u16 mode); -void snd_wm8766_set_power(struct snd_wm8766 *wm, u16 power); void snd_wm8766_volume_restore(struct snd_wm8766 *wm); int snd_wm8766_build_controls(struct snd_wm8766 *wm); -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: lola: lola_mixer: Remove some unused functions
by Rickard Strandqvist 11 Jan '15

11 Jan '15

Removes some functions that are not used anywhere: lola_restore_mixer() lola_save_mixer() 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/lola/lola.h | 2 -- sound/pci/lola/lola_mixer.c | 43 ------------------------------------------- 2 files changed, 45 deletions(-) diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index f0b1000..9125916 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -513,8 +513,6 @@ int lola_init_mixer_widget(struct lola *chip, int nid); void lola_free_mixer(struct lola *chip); int lola_create_mixer(struct lola *chip); int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute); -void lola_save_mixer(struct lola *chip); -void lola_restore_mixer(struct lola *chip); int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update); /* proc */ diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 782f4d8..47ac93a 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -348,49 +348,6 @@ int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute) return lola_codec_flush(chip); } -void lola_save_mixer(struct lola *chip) -{ - /* mute analog output */ - if (chip->mixer.array_saved) { - /* store contents of mixer array */ - memcpy_fromio(chip->mixer.array_saved, chip->mixer.array, - sizeof(*chip->mixer.array)); - } - lola_setup_all_analog_gains(chip, PLAY, true); /* output mute */ -} - -void lola_restore_mixer(struct lola *chip) -{ - int i; - - /*lola_reset_setups(chip);*/ - if (chip->mixer.array_saved) { - /* restore contents of mixer array */ - memcpy_toio(chip->mixer.array, chip->mixer.array_saved, - sizeof(*chip->mixer.array)); - /* inform micro-controller about all restored values - * and ignore return values - */ - for (i = 0; i < chip->mixer.src_phys_ins; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_SOURCE_GAIN, - i, 0); - for (i = 0; i < chip->mixer.src_stream_outs; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_SOURCE_GAIN, - chip->mixer.src_stream_out_ofs + i, 0); - for (i = 0; i < chip->mixer.dest_stream_ins; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_DESTINATION_GAIN, - i, 0); - for (i = 0; i < chip->mixer.dest_phys_outs; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_DESTINATION_GAIN, - chip->mixer.dest_phys_out_ofs + i, 0); - lola_codec_flush(chip); - } -} - /* */ -- 1.7.10.4

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[alsa-devel] [PATCH 0/4] ASoC: Make set_tdm_slot() semantics consistent
by Lars-Peter Clausen 11 Jan '15

11 Jan '15

This is something we shortly discussed during the ALSA mini conference in Duesseldorf last year. We want to do more useful stuff with the snd_soc_set_tdm_slot() API and build generic infrastructure on top of it. But there are currently two different incompatible semantics of how the tx_mask and rx_mask parameters of snd_soc_dai_set_tdm_slot() are used. If we want to be able to use the API in a generic way we need to fix this inconsistency. Both interpretations agree that the slot masks should be represented as bitmasks which bit 0 representing the first slot, bit 1 the second slot and so on. But where they differ is in the meaning of value of the bit. Some implementations assume that if a bit is set the channel is enabled and if the bit is cleared the channel is disabled, others have inverse semantics, they assume that if a bit is cleared the channel is enabled and if the bit is set the channel is disabled. The former is used by most drivers and later is only used by some the Freescale DAI drivers and the MC13783 CODEC driver. This appears to be a result of Freescale DAI drivers directly passing the masks through to the registers which expect the inverted semantics. And the MC13783 appears to have picked up the same meaning since it is used on a board which has a Freescale DAI on the CPU side of the DAI link. This series updates the MC13783 and Freescale DAI drivers to use the same semantics as other drivers and also updates the documentation of the snd_soc_dai_set_tdm_slot() function to be more specific on the intended semantics. - Lars Lars-Peter Clausen (4): ASoC: mc13783: Update set_tdm_slot() semantics ASoC: fsl: Update set_tdm_slot() semantics ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() ASoC: Update snd_soc_dai_set_tdm_slot() documentation sound/soc/codecs/mc13783.c | 10 +++++----- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_esai.c | 3 +++ sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/fsl_utils.c | 27 --------------------------- sound/soc/fsl/fsl_utils.h | 3 --- sound/soc/fsl/imx-mc13783.c | 5 ++--- sound/soc/fsl/imx-ssi.c | 5 ++--- sound/soc/fsl/wm1133-ev1.c | 4 ++-- sound/soc/soc-core.c | 20 ++++++++++++++++---- 10 files changed, 33 insertions(+), 50 deletions(-) -- 1.8.0

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[alsa-devel] [PATCHv2 2/2] ALSA: log emu8000 DRAM size at level INFO
by David Flater 11 Jan '15

11 Jan '15

Applicable after PATCHv2 1/2: Detected sound font memory goes unreported unless the kernel was built with ALSA debugging enabled. Elevate that to a printk at level INFO. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). sound/isa/sb/emu8000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 3dcf80e..7429a51 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -451,7 +451,7 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_CLOSE); snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); - snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", + snd_printk(KERN_INFO "EMU8000 [0x%lx]: %d KiB on-board DRAM detected\n", emu->port1, size/1024); emu->mem_size = size; -- 1.8.4

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[alsa-devel] [PATCH] fm801: remove FSF address
by Andy Shevchenko 11 Jan '15

11 Jan '15

The FSF address is subject to change. So, we remove it from the header. While here, remove a line with my old and non-functional email from the header. Moreover, nowadays we have a Git VCS that contains a history of changes. Signed-off-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- sound/pci/fm801.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 52555bda..1fdd92b6 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -2,8 +2,6 @@ * The driver for the ForteMedia FM801 based soundcards * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * - * Support FM only card by Andy Shevchenko <andy(a)smile.org.ua> - * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -14,10 +12,6 @@ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ #include <linux/delay.h> -- 1.8.3.101.g727a46b

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[alsa-devel] [PATCH] ALSA: hda - fixup input_free_device called after input_unregister_device
by David Henningsson 11 Jan '15

11 Jan '15

Input_unregister_device will internally free the device, so input_free_device should not be called. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 09d2131..a50e15e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3548,7 +3548,6 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, break; case HDA_FIXUP_ACT_FREE: input_unregister_device(spec->kb_dev); - input_free_device(spec->kb_dev); spec->kb_dev = NULL; } #endif -- 1.9.1

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Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Raymond Yau 11 Jan '15

11 Jan '15

> > > >If Combo audio connector does not support conventional microphone, this mean > >the driver should not create mic jack control for node 0x1a > > > >http://support.lenovo.com/us/en/documents/pd029567 > > > >Connect headphones or a headset that has a 3.5-mm (0.14-inch), 4-pole plug > >to the combo audio connector to listen to the sound from the computer. > > I do not have any 4-pole (3-rings) equipment to test. > A conventional 3-pole (2-ring) microphone certainly does not do its job. > > Then again, I ditched unpowered microphones (most external ones are). > The internal mic next to the internal webcam is powered and provides > better gain. > > The Mic control in alsamixer should stay. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… The debug version of driver message indicate whether the codec can differentiate headphone , headset (ctia or omtp) or not http://voices.canonical.com/david.henningsson/2014/03/07/headset-jacks-on-n… http://bazaar.launchpad.net/~unity-settings-daemon-team/unity-settings-daem… Headphone Mic Jack - indicates headphone and mic-in mode share the same jack, i e, not two separate jacks. Hardware cannot distinguish between a headphone and a mic. Headset Mic Phantom Jack - indicates headset jack where hardware can not distinguish between headphones and headsets Headset Mic Jack - indicates headset jack where hardware can distinguish between headphones and headsets. There is no use popping up a dialog in this case, unless we already need to do this for the mic-in mode. control.14 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } What is the usage of this jack detect control if conventional mic is not supported ?

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