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

[alsa-devel] [PATCH] ASoC: Intel: fix ifnullfree.cocci warnings
by kbuild test robot 13 Apr '15

13 Apr '15

sound/soc/intel/common/sst-ipc.c:287:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values. NULL check before some freeing functions is not needed. Based on checkpatch warning "kfree(NULL) is safe this check is probably not required" and kfreeaddr.cocci by Julia Lawall. Generated by: scripts/coccinelle/free/ifnullfree.cocci CC: Jin Yao <yao.jin(a)linux.intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sst-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -283,8 +283,7 @@ void sst_ipc_fini(struct sst_generic_ipc if (ipc->tx_thread) kthread_stop(ipc->tx_thread); - if (ipc->msg) - kfree(ipc->msg); + kfree(ipc->msg); } EXPORT_SYMBOL_GPL(sst_ipc_fini);

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Re: [alsa-devel] [Intel-gfx] [PATCH 15/19] drm/i915: HSW cdclk support
by Ville Syrjälä 13 Apr '15

13 Apr '15

On Mon, Apr 13, 2015 at 12:43:14PM +0300, Mika Kahola wrote: > On Fri, 2015-04-10 at 16:10 +0200, Takashi Iwai wrote: > > At Fri, 10 Apr 2015 16:27:39 +0300, > > Mika Kahola wrote: > > > > > > On Thu, 2015-04-09 at 17:17 +0200, Takashi Iwai wrote: > > > > At Thu, 9 Apr 2015 15:51:27 +0200, > > > > Daniel Vetter wrote: > > > > > > > > > > On Thu, Apr 09, 2015 at 04:41:26PM +0300, Mika Kahola wrote: > > > > > > On Thu, 2015-04-09 at 11:32 +0200, Daniel Vetter wrote: > > > > > > > On Thu, Apr 09, 2015 at 10:24:24AM +0300, Mika Kahola wrote: > > > > > > > > I did some testing on audio part with HDMI-HDMI and DP-HDMI cables > > > > > > > > connected to my Haswell box. Before applying the patch I tested as a > > > > > > > > reference with the latest -nightly (04-08-2015), 4.0-rc7. Unfortunately, > > > > > > > > I failed to get any audio over HDMI cable. For a reference I tested with > > > > > > > > the very same setup the vanillla kernel from Linus tree 4.0-rc7 and with > > > > > > > > that kernel the audio worked ok. Then I did some GIT bisecting and it > > > > > > > > turned out that the first commit that I failed to get audio working was > > > > > > > > aa2fee4286e43b4784982b17669b02cc99c1ae55. > > > > > > > > > > > > > I rerun the bisecting and this time the result was > > > > > > > > > > > > commit 0a599838737a2527c35e4d94f794aefe59df1781 > > > > > > Merge: 2d846c7 a59d719 > > > > > > Author: Takashi Iwai <tiwai(a)suse.de> > > > > > > Date: Wed Apr 8 11:29:56 2015 +0200 > > > > > > > > > > > > Merge branch 'for-linus' into for-next > > > > > > > > > > > > Back merge HD-audio quirks to for-next branch, so that we can apply > > > > > > a couple of more quirks. > > > > > > > > > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > > > > > Adding Takashi and intel audio folks. > > > > > > > > The bisecting looks odd. The commit you pointed is a back-merge from > > > > 4.0-rc to next branch, so this merge itself shouldn't bring so many > > > > stuff -- at least about the sound part. > > > > > > > > The diff in sound/* is found below. As you can see, the only change > > > > relevant with HDMI is the chunk in sound/pci/hda/hda_intel.c for > > > > HD-audio controller, but it's specific to Skylake, thus this must be > > > > irrelevant with your hardware. > > > > > > > > Please double-check. > > > > > > I double-checked the bisecting and I ended up with the same result. You're > > > right, it does seems odd why audio on my Haswell box stopped working. > > > > > > Any ideas how to proceed from here? > > > > Try to check whether the reverting the diff of sound/* manually fixes > > the problem. (i.e. apply diff -R for the diff I showed in the > > previous mail.) > > > > If it doesn't fix but still reverting the whole commit does fix, it > > means that something else got broken by the merge. > > > > > > Takashi > > > > I reverted the diff but unfortunately there were no difference. > > For curiosity, I tested the audio with the latest -nightly and I got > the audio working again. I wonder what has changed but it seems that > the problem is now solved. BTW we now have a WARN triggering on BSW due to hda trying to query the current cdclk frequency. https://bugs.freedesktop.org/show_bug.cgi?id=90012 I'm not sure if hda really needs to know the cdclk freq on BSW or not. But in either case I think we should just kill the HAS_DDI check. Once these patches land the code will anyway work on all platforms. Mika, can you add that to your list of things to do on top of these patches? > > -Mika- > > > > > > > > > Cheers, > > > Mika > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > === > > > > % git diff 0a599838737a2527c35e4d94f794aefe59df1781^..0a599838737a2527c35e4d94f794aefe59df1781 sound > > > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > > > index 9bcc5457a83e..e1c210515581 100644 > > > > --- a/sound/pci/hda/hda_intel.c > > > > +++ b/sound/pci/hda/hda_intel.c > > > > @@ -1967,7 +1967,7 @@ static const struct pci_device_id azx_ids[] = { > > > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > > > /* Sunrise Point */ > > > > { PCI_DEVICE(0x8086, 0xa170), > > > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > > > /* Sunrise Point-LP */ > > > > { PCI_DEVICE(0x8086, 0x9d70), > > > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > > > index e0c06f9a0e80..7f46d063af57 100644 > > > > --- a/sound/pci/hda/patch_realtek.c > > > > +++ b/sound/pci/hda/patch_realtek.c > > > > @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) > > > > { > > > > /* We currently only handle front, HP */ > > > > static hda_nid_t pins[] = { > > > > - 0x0f, 0x10, 0x14, 0x15, 0 > > > > + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 > > > > }; > > > > hda_nid_t *p; > > > > for (p = pins; *p; p++) > > > > @@ -2870,6 +2870,8 @@ static void alc283_init(struct hda_codec *codec) > > > > > > > > if (!hp_pin) > > > > return; > > > > + > > > > + msleep(30); > > > > hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); > > > > > > > > /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ > > > > @@ -3564,6 +3566,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3619,6 +3622,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_write_coef_idx(codec, 0x45, 0xc489); > > > > snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > > > > alc_process_coef_fw(codec, coef0255); > > > > @@ -3688,6 +3692,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3742,6 +3747,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3796,6 +3802,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3841,6 +3848,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > msleep(300); > > > > val = alc_read_coef_idx(codec, 0x46); > > > > @@ -4993,6 +5001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > > > > SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), > > > > SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), > > > > SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > > > + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), > > > > SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > > > SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), > > > > SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), > > > > @@ -5173,6 +5182,16 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > > > > {0x17, 0x40000000}, > > > > {0x1d, 0x40700001}, > > > > {0x21, 0x02211050}), > > > > + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, > > > > + {0x12, 0x90a60140}, > > > > + {0x13, 0x40000000}, > > > > + {0x14, 0x90170110}, > > > > + {0x19, 0x411111f0}, > > > > + {0x1a, 0x411111f0}, > > > > + {0x1b, 0x411111f0}, > > > > + {0x1d, 0x40700001}, > > > > + {0x1e, 0x411111f0}, > > > > + {0x21, 0x02211020}), > > > > SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, > > > > {0x12, 0x90a60130}, > > > > {0x13, 0x40000000}, > > > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > > > > index 353532b8aee4..8f63e8c8b46c 100644 > > > > --- a/sound/usb/quirks.c > > > > +++ b/sound/usb/quirks.c > > > > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > > > > > > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > > > > { > > > > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > > > > - return chip->usb_id == USB_ID(0x045E, 0x076D); > > > > + /* devices which do not support reading the sample rate. */ > > > > + switch (chip->usb_id) { > > > > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > > > > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > > > > + return true; > > > > + } > > > > + return false; > > > > } > > > > > > > > /* Marantz/Denon USB DACs need a vendor cmd to switch > > > > > > > > > > > > > _______________________________________________ > Intel-gfx mailing list > Intel-gfx(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/intel-gfx -- Ville Syrjälä Intel OTC

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[alsa-devel] [PATCH] ctl: fix to handle several elements added by one operation for userspace element
by Takashi Sakamoto 13 Apr '15

13 Apr '15

An element instance can have several elements with the same feature. Some userspace applications can add such an element instance by add operation with the number of elements. Then, the element instance gets a memory object to keep states of these elements. But the element instance has just one memory object for the elements. This causes the same result to each read/write operations to the different elements. This commit fixes this bug by allocating enough memory objects to the element instance for each of elements. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index ccb1ca2..5b2c352 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1029,7 +1029,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct user_element { struct snd_ctl_elem_info info; struct snd_card *card; - void *elem_data; /* element data */ + char *elem_data; /* element data */ unsigned long elem_data_size; /* size of element data in bytes */ void *tlv_data; /* TLV data */ unsigned long tlv_data_size; /* TLV data size */ @@ -1078,9 +1078,12 @@ static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct user_element *ue = kcontrol->private_data; + unsigned int size = ue->elem_data_size; + char *src = ue->elem_data + + snd_ctl_get_ioff(kcontrol, &ucontrol->id) * size; mutex_lock(&ue->card->user_ctl_lock); - memcpy(&ucontrol->value, ue->elem_data, ue->elem_data_size); + memcpy(&ucontrol->value, src, size); mutex_unlock(&ue->card->user_ctl_lock); return 0; } @@ -1090,11 +1093,14 @@ static int snd_ctl_elem_user_put(struct snd_kcontrol *kcontrol, { int change; struct user_element *ue = kcontrol->private_data; + unsigned int size = ue->elem_data_size; + char *dst = ue->elem_data + + snd_ctl_get_ioff(kcontrol, &ucontrol->id) * size; mutex_lock(&ue->card->user_ctl_lock); - change = memcmp(&ucontrol->value, ue->elem_data, ue->elem_data_size) != 0; + change = memcmp(&ucontrol->value, dst, size) != 0; if (change) - memcpy(ue->elem_data, &ucontrol->value, ue->elem_data_size); + memcpy(dst, &ucontrol->value, size); mutex_unlock(&ue->card->user_ctl_lock); return change; } @@ -1278,7 +1284,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (err < 0) return err; memcpy(&kctl->id, &info->id, sizeof(kctl->id)); - kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, + kctl->private_data = kzalloc(sizeof(struct user_element) + private_size * count, GFP_KERNEL); if (kctl->private_data == NULL) { kfree(kctl); -- 2.1.0

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[alsa-devel] [PATCH] ASoC: Intel: fix array_size.cocci warnings
by kbuild test robot 12 Apr '15

12 Apr '15

sound/soc/intel/haswell/sst-haswell-ipc.c:646:28-29: WARNING: Use ARRAY_SIZE Use ARRAY_SIZE instead of dividing sizeof array with sizeof an element Semantic patch information: This makes an effort to find cases where ARRAY_SIZE can be used such as where there is a division of sizeof the array by the sizeof its first element or by any indexed element or the element type. It replaces the division of the two sizeofs by ARRAY_SIZE. Generated by: scripts/coccinelle/misc/array_size.cocci CC: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sst-haswell-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -643,7 +643,7 @@ static void hsw_fw_ready(struct sst_hsw /* log the FW version info got from the mailbox here. */ memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); pinfo = &fw_info[0]; - for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + for (i = 0; i < ARRAY_SIZE(tmp); i++) tmp[i] = strsep(&pinfo, " "); dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " "version: %s.%s, build %s, source commit id: %s\n",

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Re: [alsa-devel] Issues and/or possible bugs in alsa
by Yomi Ogunwumi 12 Apr '15

12 Apr '15

I don't think I've disabled anything. • Yomi On Apr 12, 2015 12:31 PM, "Raymond Yau" <superquad.vortex2(a)gmail.com> wrote: > > > > I'm not sure if it worked with pervious kernel versions. There was a > time I rolled back earlier kernel versions — The result was the same, it > didn't work. > > > > It might have worked with Windows 8, but I had installed Arch soon after > obtaining the laptop. > > >Second. When I plug in these headphones, I can't use the internal > microphone on this laptop, it mutes. (AFAICT) These headphones do not have > a microphone built in, they're just an ordinary pair of headphones. > However, I was given a workaround in #alsa on freenode. In order to use the > laptop's internal microphone with my headphones plugged in, I have to do : > sudo hda-verb /dev/snd/hwC1D0 0x22 SET_CONNECT_SEL 6 — it works, but it > isn't really ideal. > > 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In > Amp-In caps: N/A > Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 > 0x00] [0x00 0x00] [0x00 0x00] > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12* > > Do you mean disable jack detection cannot work around this jack detection > fault ? > > Using hda-emu, the driver seem select internal mic when jack detection is > disabled on startup > > >> > >> How about disable jack detect using hint jack_detect=0 ? > >> > >> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… > >> > >> you have to select mic jack using capture source and manually mute the > speaket >

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[alsa-devel] [PATCH 1/2] ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

Some callers of n810_ext_control() pass the card DAPM context and some pass the CODEC DAPM context. Given that some of the widgets that are accessed in the function are in the card's context, always passing it is the obvious choice. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/n810.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 5d7f9ce..617eae3 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -98,12 +98,11 @@ static int n810_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - n810_ext_control(&codec->dapm); + n810_ext_control(&rtd->card->dapm); return clk_prepare_enable(sys_clkout2); } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: davinci-evm: Use card DAPM context to access widgets
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/davinci/davinci-evm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 8c2b9be..731fb0d 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -117,7 +117,6 @@ static const struct snd_soc_dapm_route audio_map[] = { static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_codec *codec = rtd->codec; struct device_node *np = card->dev->of_node; int ret; @@ -136,9 +135,9 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) } /* not connected */ - snd_soc_dapm_nc_pin(&codec->dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(&codec->dapm, "HPLCOM"); - snd_soc_dapm_nc_pin(&codec->dapm, "HPRCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); return 0; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: mop500_ab8500: Use card DAPM context to access widgets
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/ux500/mop500_ab8500.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index aa65370..b81a7a4c 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -362,7 +362,7 @@ struct snd_soc_ops mop500_ab8500_ops[] = { int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; struct device *dev = rtd->card->dev; struct mop500_ab8500_drvdata *drvdata; int ret; @@ -407,23 +407,23 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + ret = snd_soc_dapm_disable_pin(dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 6"); return ret; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: wm1133-ev1: Use card DAPM context to access widgets
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/wm1133-ev1.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 0653aa8..b454972 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -202,7 +202,6 @@ static struct snd_soc_jack_pin mic_jack_pins[] = { static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, @@ -216,7 +215,7 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias"); return 0; } -- 1.8.0

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[alsa-devel] [PATCH 1/2] ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The SND_AT91_SOC_SAM9G20_WM8731 and SND_AT91_SOC_SAM9X5_WM8731 machine driver symbols select SND_SOC_WM8731 which depends on SND_SOC_I2C_AND_SPI. So the machine driver symbols need to depend on SND_SOC_I2C_AND_SPI as well, otherwise we might end up with a invalid configuration, which will sooner or later upset the randconfig auto-builders. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1579e99..2f185e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,7 +25,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 -- 1.8.0

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