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author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-10-10 22:13:25 -0400 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-10-10 22:13:25 -0400 |
commit | a2ce35273c2f1aa0dcddd8822681d64ee5f31852 (patch) | |
tree | fac2b4d526b6ca6657ea7c1b0a25efe76a3a0b5e /sound/soc/soc-dapm.c | |
parent | bf65dea87e87c53ba4f97c6432761498bc977efd (diff) | |
parent | fd1a2a90d08b0052fa52bd36cebd0592c9e537c2 (diff) | |
download | talos-op-linux-a2ce35273c2f1aa0dcddd8822681d64ee5f31852.tar.gz talos-op-linux-a2ce35273c2f1aa0dcddd8822681d64ee5f31852.zip |
Merge tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This time it's a relatively calm update batch, but the amount isn't
too small in the end. Here we go over some highlights:
ALSA core:
- One major change is the support of nonatomic PCM operations. This
allows the trigger and other callbacks to call schedule(), which
would be useful for mailbox type communications. Already some
drivers (Digigram ones) have been converted to use together with
threaded irqs as an example.
- Improvement / fixes of DSD PCM format support
HD-audio:
- Large volume of rewrites are found in Realtek codec driver for
converting Dell and HP quirks to generic forms.
- Inverted dmic code cleanup from David.
- Realtek COEF access has been optimized.
- Now HD-audio jack infrastructure allows multiple callbacks, which
fixes / simplifies the jack-dependent power controls on STAC/IDT
and VIA codecs.
- Many additional device-specific fixups as usual
- A few deadcode cleanups, CA0132 code cleanup, etc.
ASoC:
- More componentization work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to
need a lot quirks over time due to the lack of any firmware
description of the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32,
Everest Semiconductor ES8328 and Freescale cards using the ASRC in
newer i.MX processors.
- A few simple-card fixes, mostly cleanups but also a fix for
interaction between GPIO 0 and simple-card.
Misc:
- Virtuoso / Oxygen updates by Clemens
- USB-audio: Yamaha MOTIF XF MIDI port name fixes
- Conversion of kernel messages to standard dev_*() in ctxfi driver"
* tag 'sound-3.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (251 commits)
ASoC: mc13783: Ensure we only try to dereference valid of_nodes
ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl
ALSA: hda - Add dock port support to Thinkpad L440 (71aa:501e)
ALSA: Allow pass NULL dev for snd_pci_quirk_lookup()
ASoC: imx-es8328: Fix of_node_put() call with uninitialized object
ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb()
ASoC: simple-card: Initialize headphone and mic GPIO numbers
ASoC: imx-es8328: Fix missing return code in imx_es8328_probe()
ALSA: hda - Add dock support for Thinkpad T440 (17aa:2212)
ALSA: usb: caiaq: check for cdev->n_streams > 1
ASoC: 88pm860x-codec: Fix possibly missing string termination
ASoC: core: fix use after free in snd_soc_remove_platform()
ASoC: soc-dapm: fix use after free
ALSA: hda - Make the inv dmic handling for Realtek use generic parser
ALSA: hda - Add Inverted Internal mic for Samsung Ativ book 9 (NP900X3G)
ALSA: hda - Add inverted internal mic for Asus Aspire 4830T
ASoC: Intel: byt-rt5640: fix coccinelle warnings
ASoC: fsl_esai doc: Add "fsl,vf610-esai" as compatible string
ASoC: da732x: Remove unnecessary KERN_ERR in pr_err()
ASoC: simple-card: Fix detect gpio documentation.
...
Diffstat (limited to 'sound/soc/soc-dapm.c')
-rw-r--r-- | sound/soc/soc-dapm.c | 51 |
1 files changed, 36 insertions, 15 deletions
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 177bd8639ef9..c61cb9cedbcd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -326,12 +326,13 @@ static struct list_head *dapm_kcontrol_get_path_list( list_for_each_entry(path, dapm_kcontrol_get_path_list(kcontrol), \ list_kcontrol) -static unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) +unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); return data->value; } +EXPORT_SYMBOL_GPL(dapm_kcontrol_get_value); static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, unsigned int value) @@ -591,9 +592,9 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, int shared; struct snd_kcontrol *kcontrol; bool wname_in_long_name, kcname_in_long_name; - char *long_name; + char *long_name = NULL; const char *name; - int ret; + int ret = 0; prefix = soc_dapm_prefix(dapm); if (prefix) @@ -652,15 +653,17 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); - kfree(long_name); - if (!kcontrol) - return -ENOMEM; + if (!kcontrol) { + ret = -ENOMEM; + goto exit_free; + } + kcontrol->private_free = dapm_kcontrol_free; ret = dapm_kcontrol_data_alloc(w, kcontrol); if (ret) { snd_ctl_free_one(kcontrol); - return ret; + goto exit_free; } ret = snd_ctl_add(card, kcontrol); @@ -668,17 +671,18 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, dev_err(dapm->dev, "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", w->name, name, ret); - return ret; + goto exit_free; } } ret = dapm_kcontrol_add_widget(kcontrol, w); - if (ret) - return ret; + if (ret == 0) + w->kcontrols[kci] = kcontrol; - w->kcontrols[kci] = kcontrol; +exit_free: + kfree(long_name); - return 0; + return ret; } /* create new dapm mixer control */ @@ -1683,6 +1687,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, } } +static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) +{ + if (dapm->idle_bias_off) + return true; + + switch (snd_power_get_state(dapm->card->snd_card)) { + case SNDRV_CTL_POWER_D3hot: + case SNDRV_CTL_POWER_D3cold: + return dapm->suspend_bias_off; + default: + break; + } + + return false; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1706,7 +1726,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->idle_bias_off) + if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1772,7 +1792,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - if (!d->idle_bias_off) + if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); @@ -3109,7 +3129,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); |