January 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] ASoC: pxa2xx-i2s
by Mark Brown 09 May '09

09 May '09

On Wed, Jan 21, 2009 at 10:46:17AM -0600, Brian G Rhodes wrote: > Mark Brown wrote: > >What is the clock master in your system? > The master is the i2s controller sysclk. The dac2x is in slave mode. > Though the i2s controller isn't directly supplying the bit clock to the > codec in this case. We're using the PLL on the dac2x. Just to confirm, the codec is the clock master on the I2S bus itself?

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[alsa-devel] [PATCH 0/2] ASoC updates
by Mark Brown 07 May '09

07 May '09

The following changes since commit 32c8dabc97d436582298ebd0e33af041c69f5a4b: Mark Brown (1): ASoC: Remove obsolete declaration of struct snd_soc_clock_info are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: tlv320aic3x: control additions and cleanups Grazvydas Ignotas (1): ASoC: Add support for OMAP3 Pandora sound/soc/codecs/tlv320aic3x.c | 18 ++- sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 311 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 334 insertions(+), 5 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c

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[alsa-devel] Bug 2552 typo in ice1724.c
by Alan Horstmann 31 Mar '09

31 Mar '09

Hi, I noticed this trivial typo report on bugtrack, -no 2552 and thought it may as well be fixed, and the bug closed. Patch inline & attached. Not sure if it needs a sign-off? Alan ice1724.c-name-typo-fix.diff Fix trivial name string typo as reported in bug 2552 Signed-off-by: Alan Horstmann <gineera(a)aspect135.co.uk> --- alsa-kernel/pci/ice1712/ice1724-orig.c 2008-06-27 09:11:58.000000000 +0100 +++ alsa-kernel/pci/ice1712/ice1724.c 2009-01-05 11:11:33.000000000 +0000 @@ -1239,7 +1239,7 @@ if (ice->force_pdma4 || ice->force_rdma1) name = "ICE1724 Secondary"; else - name = "IEC1724 IEC958"; + name = "ICE1724 IEC958"; err = snd_pcm_new(ice->card, name, device, play, capt, &pcm); if (err < 0) return err;

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Re: [alsa-devel] Bugs on aspire one A150
by Maxim Levitsky 16 Mar '09

16 Mar '09

Alan Jenkins wrote: > Maxim Levitsky wrote: >> I have just bought an Aspire one A150, XP version, >> as it was the only available here, and installed ubuntu on it. >> >> Bugs I discovered so far: >> >> >> ** 1 - embedded controler works in polling mode, due to this: >> >> [ 0.708571] ACPI: EC: non-query interrupt received, switching to interrupt mode >> [ 1.224043] ACPI: EC: missing confirmations, switch off interrupt mode. >> >> >> Maybe this is the reason for the fact that gnome power manager freezes when I unplug >> the AC, and freezes often when I try to see battery status. >> >> >> (Note: same is seen on my acer aspire 5720) > > That sounds like a known issue. It has been resolved by "ACPI: EC: revert msleep patch". Happily Len submitted it for mainline this week. You will also find it if you try the acpi-test git tree. We're all hoping 2.6.28 will be much improved in terms of reliable operation of different ECs :). > Yes, this almost fixes all issues with that. Almost because, it looks like the EC changes screen brightness on his own when battery is plugged/unplugged, but so does the gnome-power-manager, and thus it still hangs as before on battery removal (but doesn't hang otherwise) I disabled that behaver in gnome-power-manager and now no more hangs. I see that this fix fixes the polled mode. Any chance to make irq mode work? >> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by default in ubuntu... >> wireless more or less works, but kernel log is full of backtraces. > > Well, that doesn't tell us much. Did they still happen after upgrading to 2.6.28-rc3? Can we see them? Fixed completely, looking through the git log I have seen a commit that fixes exactly same backtrace I had. No more errors in kernel log. > >> Was able to connect to my WPA2 access point. >> Sometimes wireless fails completely, especially after suspend to ram. >> Advanced features like monitor/injection work, but when I changed the card's >> mac address it stopped working. >> I also noticed that if I then start airodump, then wireless works with new mac. This still not fixed, I ask about this again in separate mail. >> >> >> >> ** 3 - internal mic doesn't work. >> tried model=acer, model=auto. >> Overall it seems that alsa misprograms O/B realteck >> codec. >> I talk about this later. Surprisingly, this is almost fixed. I can record from both internal mic and external mic (Quality is not so good, but is the same as in windows) Almost, is that to increase capture level, I have to set "one" channel of it. Eg: if I set 100% left and 100% right then capture level is very low. But when I set 100% left and 0% right or vice versa it works fine. Also the "mic boost" doesn't change capture volume for internal mic. also the "mic boost" if set to high value works like analog loopback (I hear what I say in mic in headphones, same happens on my main notebook acer 5720) >> Have same issue on my acer 5720 >> >> >> >> ** 4 - wireless led doesn't work. >> ath5k devs, can you fix this? >> >> >> ** 5 - coretemp doesn't show cpu temperature, >> I have seen somewhere that atom support same thermal diode as core2 >> and only patch to detect it is needed. >> Please include such path in 2.6.28 if exists. >> >> >> ** 6 - both card readers are missing from lspci, is this normal? > > A similar bug has been reported as a regression: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11828> > > so one assumes that it worked on the machines with linux pre-installed. Hopefully without requiring any hacks. > > It seems that for now a workaround may be to pass the option debug_quirks=1 to the sdhci module... > > <http://marc.info/?l=linux-kernel&m=122509648027303&w=2> > > ...or that it may help if you insert an SD card before booting. > > Apparently the reporter also investigated pcie hotplug. Probably the BIOS doesn't provide the normal support. You can try "modprobe pciehp pciehp_force=1", maybe it helps the kernel discover the devices. It worked for something else on my EeePC. But then it will reportedly disappear the ethernet controller. > > However, at the moment pciehp can cause delays of 10s of seconds during resume. I am aware of this, but don't yet have a SD card to test. > >> ** 7 - opengl is broken, I tried running neverball in fullscreen, but it shows wrong/corrupted >> textures (compiz is running, and working well thought) Without compiz neveball works fine (slow, but this is very slow hardware) >> >> Now I installed 2.6.28-rc3 (actually latest -git) >> >> >> Sadly nothing improved. >> I only found few regressions: >> >> >> >> * Wireless now drops connection from my WPA2 access point after few minutes. >> I changed temporarily its wireless protection to WEP and open mode, and both work >> fine, maybe this is related to periodic key exchange WPA does? Now fixed, big thanks. >> >> * System doesn't reboot/shutdown, just hangs. > > Could be this: > > <http://bugzilla.kernel.org/show_bug.cgi?id=11942> > > As a fix, 8fd145917fb62368a9b80db59562c20576238f5a was reverted from the acpi tree. This has also been submitted to mainline. Also fixed big thanks. Also noticed another bug: If I suspend/resume with compiz running, on resume I see wallpaper and mouse cursor, everything hangs for minute or two, and sometimes forever. 2.6.27 worked fine. Also hibernate doesn't work on my main notebook, but it is probably fixed, I update kernel to really latest -git and tell you. Big thanks for bugfixes, you saved me a lot of work and bisecting. Biggest solvable problem now is sound support now I think, datasheets are there, and I take a look at them. Best regards, Maxim Levitsky

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[alsa-devel] [PATCH] ASoC: Complete Beagleboard support
by Steve Sakoman 11 Mar '09

11 Mar '09

Commit dc06102a0c8b5aa0dd7f9a40ce241e793c252a87 in the asoc tree did not include the necessary Kconfig and Makefile changes. This patch completes the support for Beagleboard Signed-off-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/Kconfig | 10 ++++++++++ sound/soc/omap/Makefile | 2 ++ 2 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4f7f040..ccd8973 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA select SND_SOC_TWL4030 help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_OMAP_SOC_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Beagleboard. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 76fedd9..0c9e4ac 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.5.6.3

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[alsa-devel] SoC scenarii API
by Robert Jarzmik 08 Mar '09

08 Mar '09

Hi Liam and Mark, As mioa701 submission was stopped due to the need of a generic scenarii API, this a first attempt to design such an API in order to unblock mioa701 submission. This is oversimplified, no convenience macros are provided, just external data structures and namespace polution. I'm expecting a lot of comments from both of you. Cheers. -- Robert enum pin_change { SCEN_PIN_OFF = 0, /* Disable the pin */ SCEN_PIN_ON, /* Enable the pin */ SCEN_PIN_LEAVE /* Leave the pin in previous state */ }; struct setup_mixmux { char *mixname; /* Codec Mixer or Muxer name */ int val; /* Codec Mixer or Muxer value to enforce */ }; /** * struct soc_scenario - describes one sound usecase * @name: Scenario name, value as will be seen in alsa "SoC Mode" alsa control * @pin_setup: Pin configuration to perform on scenario activation * Table of all pins, as they should be configured. Each elements is a * pin_change value, describing how to handle a specific pin. * Table size must be the same as in snd_soc_scenario_init(). * @mixer_cleanup: Mixers/muxes to set up in phase (b) * Table of all mixers/muxes to modify, NULL terminated. * @mixer_setup: Mixers/muxers to set up in phase (c) * Table of all mixers/muxes to modify, NULL terminated * @lvol_master: Left volume aliased to "SoC Volume" * @rvol_master: Right volume aliased to "SoC Volume" * @lvol_mute: Left volume mute aliased to "SoC Volume" mute control * @rvol_mute: Right volume mute aliased to "SoC Volume" mute control * * This structure describes what a scenario change implies. The behaviour is to : * a) enable several pins, disable others, leave others in their state * (understand here snd_soc_dapm_enable_pin) * b) set up some mixers and muxers to prepare the change, and clean up. * This should be normally used to restore a "default state" if need be, before * activating the mixers/muxers into their final state. * Note: table in NULL, or always terminated by NULL pointer * Note: this cleanup may/should be shared amongst all scenarii * c) set up some mixers and muxers to the final state * This should be normally used to restore a "default state" if need be, before * activating the mixers/muxers into their final state. * Note: table in NULL, or always terminated by NULL pointer */ struct soc_scenario { const char *name; /* scenario name */ const unsigned char pin_setup[]; /* pin_change for pins */ const struct setup_mixmux mixes_cleanup[]; /* mix cleanup */ const struct setup_mixmux mixes_setup[]; /* mix setup */ const char *lvol_master; /* left volume master */ const char *rvol_master; /* right volume master */ const char *lvol_mute; /* left volume mute */ const char *rvol_mute; /* right volume mute */ }; /** * snd_soc_scenario_init - initialize soc scenarii * @codec: codec associated to the pins/mixers/muxes/volumes/mutes * @scenario: table of scenarii * @nb_scenarii: number of scenarii * @pin_names: table of pin names * @nb_pins: number of pins handled by scenario * * Initialise the soc scenarii engine. The first scenario (0) will be used. * By default, the user could leave this scenario as non intrusive (not a single * pin changed, no mixers/muxes changes, and volume master inactive). */ int snd_soc_scenario_init(struct snd_soc_codec *codec, struct soc_scenario *scenario, int nb_scenarii, char *pin_names[], int nb_pins); /** * snd_soc_scenario_init - initialize soc scenarii * @codec: codec used for the init */ void snd_soc_scenario_release(struct snd_soc_codec *code);

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Re: [alsa-devel] [Fwd: HP DV4 1155 SE - No Sound through speakers...]
by Jean-Pierre André 24 Feb '09

24 Feb '09

Hi, > Hi all, I have an HP DV4 1155-SE, I've been unable to get sound working > through the speakers, although it does work through the headhpones. I ran > alsa-.upgrade which downloaded compiled, and installed 1.0.19 - still no > luck. > I have the same IDT codec 92HD71B7X on another HP DV4 model, and I have tracked the problem to be the kernel configuring IRQ 22 through the APIC, but interrupts triggerring IRQ 20 (on kernel 2.6.27 and alsa 1.0.19). It would be interesting if you could confirm this by analyzing /proc/interrupts. The workarounds I know of are : - adding irqpoll to the boot options, - or adding noapic to the boot options, - using mplayer to play music. There are with option "snd-hda-intel model=hp-m4" (this option is for HP DV5/DV7, but HP DV4 is probably different) I have not enough knowledge to tell where the problem lies : kernel, apic, alsa. Regards Jean-Pierre > I ran alsa-info, output here: > http://www.alsa-project.org/db/?f=844b7ee6468067460e17b7fc694e49052072a26f > > I filed a bug, here: > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4374 > > Any ideas? >

4 32

[alsa-devel] PCM delay compensation
by Takashi Iwai 23 Feb '09

23 Feb '09

Hi, the patch below (to the latest sound git tree) adds the extra delay count for USB-audio driver. This change will appear as the return value of snd_pcm_delay(). Could you check whether it's appropriate behavior you've wanted? thanks, Takashi diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 40c5a6f..dcbdc60 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -269,6 +269,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ + snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e61e125..df7c3fa 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -597,14 +597,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { status->avail = snd_pcm_playback_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || - runtime->status->state == SNDRV_PCM_STATE_DRAINING) + runtime->status->state == SNDRV_PCM_STATE_DRAINING) { status->delay = runtime->buffer_size - status->avail; - else + status->delay += runtime->delay; + } else status->delay = 0; } else { status->avail = snd_pcm_capture_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - status->delay = status->avail; + status->delay = status->avail + runtime->delay; else status->delay = 0; } @@ -2423,6 +2424,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, n = snd_pcm_playback_hw_avail(runtime); else n = snd_pcm_capture_avail(runtime); + n += runtime->delay; break; case SNDRV_PCM_STATE_XRUN: err = -EPIPE; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6e70ba4..7d5a103 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -629,6 +629,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += offs; if (subs->hwptr_done >= runtime->buffer_size) subs->hwptr_done -= runtime->buffer_size; + runtime->delay += offs; spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = offs * stride; if (period_elapsed) @@ -638,12 +639,20 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, /* * process after playback data complete - * - nothing to do + * - decrease the delay count again */ static int retire_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { + unsigned long flags; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); return 0; } @@ -1542,6 +1551,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->phase = 0; + runtime->delay = 0; /* clear urbs (to be sure) */ deactivate_urbs(subs, 0, 1);

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[alsa-devel] set_params:1073: not enough memory
by maarten van es 20 Feb '09

20 Feb '09

Hello all, I am trying to get alsa to work using buildroot to crosscompile to i686-linux-uclibc. I am using the latest git version of alsa-lib and a recent git version of alsa-util. I am stuck with the same error as in the message included below when using aplay. So far I have been able to narrow this down to the function set_params in aplay.c, lines 1070, 1071: chunk_bytes = chunk_size * bits_per_frame / 8; audiobuf = realloc(audiobuf, chunk_bytes); chunk_size is 0 here and obviously the realloc fails. Any ideas on what's going on here? For the record, "aplay -L" lists the correct devices, "aplay -D null test.wav" works ok and the card works correctly using alsa version 1.0.16 on standard suse 11.0. Best regards Maarten van Es Hello all, I'm trying to get alsa 1.0.18 working on a ARM PXA processor and I'm stuck with the following error message: aplay /Sinus.raw -fcd Playing raw data '/Sinus.raw' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1073: not enough memory alsa kernel was compiled with alsa 1.0.16, alsa.conf, located in /usr/share/alsa is the basic: pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } also aplay -l and alsaconf store show the message: ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 However the card shows up /proc/asound/cards and alsamixer is able to access it. Any ideas? Best regards Michael

2 2

[alsa-devel] [PATCH] ALSA: hda - Consider additional capture source/selector in ALC889
by Herton Ronaldo Krzesinski 13 Feb '09

13 Feb '09

Currently code for capture source support in ALC889 only considers capture mixers. This change adds aditional support for ADC+selector present in ALC889, taking into account also the presence of an additional DMIC connection item in the selector. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 105 ++++++++++++++++++++++++++++++----------- 1 files changed, 77 insertions(+), 28 deletions(-) Note: I did this patch without hardware to test on, just based on alsa-info.txt from one machine with ALC889 plus the ALC889 datasheet available from Realtek website, so this is just build tested. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 863ab95..d81cb5e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -238,6 +238,13 @@ enum { ALC883_MODEL_LAST, }; +/* styles of capture selection */ +enum { + CAPT_MUX = 0, /* only mux based */ + CAPT_MIX, /* only mixer based */ + CAPT_1MUX_MIX, /* first mux and other mixers */ +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -276,7 +283,7 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ + int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -294,7 +301,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ @@ -396,7 +403,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->is_mix_capture) { + if (spec->capture_style && + !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -4130,7 +4138,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -4279,7 +4287,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -5487,7 +5495,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -5647,7 +5655,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -7087,7 +7095,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->is_mix_capture = 1; /* matrix-style capture */ + spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -7155,10 +7163,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids + static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -8977,6 +8989,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err = alc880_parse_auto_config(codec); + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; if (err < 0) return err; @@ -8990,6 +9004,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* setup input_mux for ALC889 */ + if (codec->vendor_id == 0x10ec0889) { + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + for (i = 1; i < 3; i++) + memcpy(&spec->private_imux[i], + &spec->private_imux[0], + sizeof(spec->private_imux[0])); + imux->items[imux->num_items].label = "Int DMic"; + imux->items[imux->num_items].index = 0x0b; + imux->num_items++; + spec->num_mux_defs = 3; + spec->input_mux = spec->private_imux; + } + } + return 1; /* config found */ } @@ -9053,14 +9087,36 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC888 Analog"; spec->stream_name_digital = "ALC888 Digital"; } + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; spec->stream_name_digital = "ALC889 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); + spec->adc_nids = alc889_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc889_capsrc_nids; + spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style + capture */ break; default: spec->stream_name_analog = "ALC883 Analog"; spec->stream_name_digital = "ALC883 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -9071,13 +9127,6 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - spec->is_mix_capture = 1; /* matrix-style capture */ if (!spec->cap_mixer) set_capture_mixer(spec); @@ -10512,7 +10561,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -10881,7 +10930,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -11539,7 +11588,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -11657,7 +11706,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -12511,7 +12560,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, */ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; imux->items[imux->num_items].label = "Int Mic"; imux->items[imux->num_items].index = 0x05; imux->num_items++; @@ -12567,7 +12616,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; /* set default input source */ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, @@ -13483,7 +13532,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -13620,7 +13669,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); @@ -14724,7 +14773,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -14803,7 +14852,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; set_capture_mixer(spec); @@ -16397,7 +16446,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -16528,7 +16577,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) @@ -16613,7 +16662,7 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); -- 1.6.1

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