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Re: [alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 16 Aug '08

16 Aug '08

Hello Stan, This is awesome I did find the problem it's listed in the /var/log/messages under ALSA. 1. When I bring up the "Pulse Volume Control" gui, the first tab "Playback" comes up empty, and grayed-out is "No Streams Available" 2. aplay does not produce sound, period! 3. Amrok's welcome sample does not produce sound out of the std tower speaker or the headphone jack. I didn't notice anything strange on the output of the alsa-info.sh file, but when I checked the var/log/messages, this is what I found: sudo cat /var/log/messages | grep ALSA Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x1 Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Surround Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Center Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave LFE Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Side Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped Aug 11 07:52:10 macprolinx kernel: ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped Let me know if I should send something else. // Salvador // Northwestern University ----- Original Message ---- From: stan <ghjeold_i_mwee(a)cox.net> To: S. Aguinaga <sa_jr(a)ieee.org> Cc: alsa-devel(a)alsa-project.org Sent: Friday, August 15, 2008 11:54:12 AM Subject: Re: [alsa-devel] latest Mac Pro (quad core) audio issues S. Aguinaga wrote: > Hello, > > I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. > > // Salvador > Fedora uses pulseaudio by default. How are you determining that there is no output? i.e. what program are you using, and how Try aplay -vv some.wav Try aplay -vv -Dhw:0,0 some.wav Try aplay -vv -Dplughw:0,0 some.wav Are there error messages or does it look like it is playing without sound? Are there messages in /var/log/messages regarding sound at boot or while trying to play. I don't use pulseaudio and don't know it very well, but there is more information here. http://www.pulseaudio.org Could you run a script which provides more information about your sound setup and post the link back here? The script is found at http://www.alsa-project.org/alsa-info.sh You can run the script so the results occur locally, by specifying the ---no-upload option. alsa-info.sh --no-upload You would then paste the output into your email instead of providing a link. This information will enable me to determine if your setup is OK, and look for other problems.

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[alsa-devel] [PATCH 1/2] wm8990: Fix routing of left DAC to speaker mixer
by Mark Brown 15 Aug '08

15 Aug '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8990.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) This ought to go into 2.6.27 - it's a fairly straightforward bug fix. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 3ecce51..9505a18 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, - {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"}, /* LONMIX */ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, -- 1.5.6.3

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[alsa-devel] WM8750 utilization
by Harald Radke 15 Aug '08

15 Aug '08

Hi there! Sorry, if this is the wrong list for this, just tell me if so (and maybe point out where to go instead *g*). I am trying to improve usage of the wm8750 codec on my Loox720 PDA. Its a PXA272 device. So far WM8750 driver works quite ok (via I2S), however there are some things that are kinda strange. The Loox720 has a headphone jack, a earpiece and a speaker. When driving LOUT1/ROUT1, sound goes via headphones, LOUT2/ROUT2 drives the earpiece. When setting OUT3 on ROUT1, the speaker plays fine. Strange: - when disabling either LOUT2 or ROUT2 (while the other is enabled), the speaker plays the sound (not sure if the earpiece still is active as speaker output is much louder, but I think so) - when disabling ROUT1/LOUT1 with LOUT2/ROUT2 active, the earpiece plays fine, however there is still some output on the headphone (really really silently) - Trying to utilize headphone switch via the WM8750 registers doesn't really work ): depending on the polarity bit either one of the OUTs is either not muted at all or activated at all. Any ideas or hints? Thanks a lot! Harald Radke

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[alsa-devel] latest Mac Pro (quad core) audio issues
by S. Aguinaga 15 Aug '08

15 Aug '08

Hello, I just installed fedora (9) on one of the latest Mac Pro, all is well except for the audio. I have no audio out of the main Tower speaker or via the headphones. Eventually I want to drive audio out of the optical ports. Any links to howto's or work-around would greatly appreciated. // Salvador

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[alsa-devel] SPDIF audio / non-audio bit
by Ben Stanley 15 Aug '08

15 Aug '08

Dear alsa-devel, I recently advised this list of some problems I experienced with the SPDIF audio / non-audio bit being set inappropriately while using the ca0106 driver [1]. Seeing as I got no replies, I am trying again. When playing AC3 encoded streams (Dolby Digital, DTS), the SPDIF non-audio bit is set. Otherwise, for PCM streams, the SPDIF is set to audio. Could someone explain to me what part of the software is responsible for correctly setting the SPDIF audio / non-audio bit? Is it * alsa-driver * alsa-lib * application ? It seems to me that currently some applications (MythTV, xine) set it correctly, but most other applications ignore it. So if it is set to non-audio (because for example MythTV crashed before re-setting it to audio), then all applications that are not aware of the setting have their audio broken. Could someone set me straight here so that I can try to produce a permanent fix for this? Thanks, Ben Stanley. [1] http://thread.gmane.org/gmane.linux.alsa.devel/54398/focus=55188

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.27-rc3-74-g027ee62
by Takashi Iwai 15 Aug '08

15 Aug '08

At Fri, 15 Aug 2008 12:39:19 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > @@ -362,10 +363,8 @@ static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream) > } > if (substream->pcm->device & 2) > runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID); > - if ((err = add_playback_constraints(runtime)) < 0) { > - kfree(dpcm); > + if ((err = add_playback_constraints(runtime)) < 0) > return err; > - } Run checkpatch.pl please... Takashi

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[alsa-devel] [Fwd: [Alsa-user] found bug, fixed bug, cannot tell anyone]
by stan 15 Aug '08

15 Aug '08

Forwarded as per request -------- Original Message -------- From: - Thu Aug 14 15:28:40 2008 Message-ID: <001d01c8fe5e$3e4eed40$0402a8c0@sj4> From: Wesley Johnson <johnson2412(a)usgo.net> To: <alsa-user(a)lists.sourceforge.net> Date: Thu, 14 Aug 2008 15:36:55 -0500 (CDT) MIME-Version: 1.0 How do you notify ALSA dev team about a bug that has been in 1.0.16 and 1.0.17. ??? Bugtracker will not let me login, nor email an account confirmation. Getting an ALSA home page account seems to have no purpose at all, it does not let you do anything. BUG: When you try to compile driver for ens1371 by itself the compile will fail. The driver for ens1371 has an include of the file for ens1370, but the makefule does not have this dependency so the ens1370 file is missing. The dev team is probably always compiling for all. Diff file that fixes this problem follows. *** pci/Makefile.orig Mon Jul 14 02:53:57 2008 --- pci/Makefile Thu Jul 17 21:25:42 2008 *************** *** 27,34 **** --- 27,35 ---- atiixp_modem.c: atiixp_modem.patch $(SND_TOPDIR)/alsa-kernel/pci/atiixp_modem.c bt87x.c: bt87x.patch $(SND_TOPDIR)/alsa-kernel/pci/bt87x.c cmipci.c: cmipci.patch $(SND_TOPDIR)/alsa-kernel/pci/cmipci.c ens1370.c: ens1370.patch $(SND_TOPDIR)/alsa-kernel/pci/ens1370.c + ens1371.c: ens1370.c fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c I have been trying for since July 17 to try to send this to ALSA dev team, and have been blocked at every attempt It is now Aug 13. If someone knows how to do this, please post a reply. Wesley D. Johnson --- Get FREE High Speed Internet from USFamily.Net! -- http://www.usfamily.net/mkt-freepromo.html --- ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

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Re: [alsa-devel] [Alsa-user] found bug, fixed bug, cannot tell anyone
by Rene Herman 15 Aug '08

15 Aug '08

On 14-08-08 22:36, Wesley Johnson wrote: > How do you notify ALSA dev team about a bug that has been in 1.0.16 > and 1.0.17. ??? Here will do, although the alsa-devel mailing list would've been better still. Added to CC. > Bugtracker will not let me login, nor email an account confirmation. > Getting an ALSA home page account seems to have no purpose at all, it > does not let you do anything. The homepage is a wiki so it surely does. > BUG: > When you try to compile driver for ens1371 by itself the compile will fail. > The driver for ens1371 has an include of the file for ens1370, but the > makefule does not have this dependency > so the ens1370 file is missing. The dev team is probably always compiling > for all. > > Diff file that fixes this problem follows. > > *** pci/Makefile.orig Mon Jul 14 02:53:57 2008 > --- pci/Makefile Thu Jul 17 21:25:42 2008 > *************** > *** 27,34 **** > --- 27,35 ---- > atiixp_modem.c: atiixp_modem.patch > $(SND_TOPDIR)/alsa-kernel/pci/atiixp_modem.c > bt87x.c: bt87x.patch $(SND_TOPDIR)/alsa-kernel/pci/bt87x.c > cmipci.c: cmipci.patch $(SND_TOPDIR)/alsa-kernel/pci/cmipci.c > ens1370.c: ens1370.patch $(SND_TOPDIR)/alsa-kernel/pci/ens1370.c > + ens1371.c: ens1370.c I expect this should be a straight copy of the ens1370.c line but never use the standalone alsa-driver package. Now that we're on alsa-devel, people who do will reply though. > fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c > intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c > maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c > via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c > > > I have been trying for since July 17 to try to send this to ALSA dev team, > and have been blocked at every attempt > It is now Aug 13. If someone knows how to do this, please post a reply. Well, not to unduly annoy you, and thank you very much for the report and the persistence, but basically all Linux development happens on mailinglists and the mailinglist information is right there on the homepage so, well... The alsa-devel mailinglist is moderated for non-subscribers; if you did email alsa-devel before and just weren't let through the moderation queue then please say so because in that case we definitely have a problem to fix. Rene.

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Re: [alsa-devel] sound-problems with ALC262
by Takashi Iwai 15 Aug '08

15 Aug '08

At Thu, 14 Aug 2008 07:09:41 +0200, Tobias Kaefer wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > > Takashi Iwai schrieb: > | At Wed, 13 Aug 2008 08:07:37 +0200, > | Tobias Kaefer wrote: > |> ... > |> "stan" told me to use this script > |> http://www.alsa-project.org/alsa-info.sh to create more information > |> about my system. Here is the pastebin link of the output of that script: > |> http://pastebin.ca/1169153 > | > | At a quick glance, I see no problem in the configuration. > | > | Does the problem exist even if you play via PCM "hw" device, such as > | > | % aplay -Dhw foo.wav > | > | ?? > | If your WAV file isn't with the supported format/rate, use plughw: > | > | % aplay -Dplughw foo.wav > > Yes. The problem still exists when using the PCM "hw" or "plughw" device. What happens when you load the driver with bdl_pos_adj=0 option? Takashi

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Re: [alsa-devel] [PATCH] hda: dynamically create capture mux controls
by Matthew Ranostay 15 Aug '08

15 Aug '08

Dynamically create capture mux volume controls when a output amp is detected. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index ff56432..eae6df1 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -794,7 +794,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), { } /* end */ }; @@ -908,12 +907,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - /* analog pc-beep replaced with digital beep support */ /* HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), @@ -931,11 +927,9 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -943,7 +937,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; @@ -953,12 +946,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -967,11 +957,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -982,15 +970,12 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2351,7 +2336,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) +static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, + int idx, const char *name, unsigned long val) { struct snd_kcontrol_new *knew; @@ -2371,6 +2357,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = stac92xx_control_templates[type]; + knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; @@ -2379,6 +2366,14 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char return 0; } + +/* add dynamic controls */ +static int stac92xx_add_control(struct sigmatel_spec *spec, int type, + const char *name, unsigned long val) +{ + return stac92xx_add_control_idx(spec, type, 0, name, val); +} + /* flag inputs as additional dynamic lineouts */ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) { @@ -2780,6 +2775,26 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, return 0; } +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + err = stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -3078,6 +3093,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2)

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