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author | Linus Torvalds <torvalds@linux-foundation.org> | 2016-01-17 12:05:31 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2016-01-17 12:05:31 -0800 |
commit | a016af2e70bfca23f2f5de7d8708157b86ea374d (patch) | |
tree | bfe3c0c6ea9d52d4ec6ea021b0626a53c83e7d9f /sound/soc/codecs/ak4613.c | |
parent | e535d74bc50df2357d3253f8f3ca48c66d0d892a (diff) | |
parent | c3b1681375dc6e71d89a3ae00cc3ce9e775a8917 (diff) | |
download | blackbird-op-linux-a016af2e70bfca23f2f5de7d8708157b86ea374d.tar.gz blackbird-op-linux-a016af2e70bfca23f2f5de7d8708157b86ea374d.zip |
Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've had quite busy weeks in this cycle. Looking at ALSA core, the
significant changes are a few fixes wrt timer and sequencer ioctls
that have been revealed by fuzzer recently. Other than that, ASoC
core got a few updates about DAI link handling, but these are rather
straightforward refactoring.
In drivers scene, ASoC received quite lots of new drivers in addition
to bunch of updates for still ongoing Intel Skylake support and
topology API. HD-audio gained a new HDMI/DP hotplug notification via
component. FireWire got a pile of code refactoring/updates with
SCS.1x driver integration.
More highlights are shown below.
[ NOTE: this contains also many commits for DRM. This is due to the
pull of drm stable branch into sound tree, as the base of i915 audio
component work for HD-audio. The highlights below don't contain
these DRM changes, as these are supposed to be pulled via drm tree
in anyway sooner or later. ]
Core:
- Handful fixes to harden ALSA timer and sequencer ioctls against
races reported by syzkaller fuzzer
- Irq description string can be unique to each card; only for
HD-audio for now
ASoC:
- Conversion of the array of DAI links to a list for supporting
dynamically adding and removing DAI links
- Topology API enhancements to make everything more component based
and being able to specify PCM links via topology
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production; we really need to get to the
point where that can be done
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come
- Lots of new features and cleanups for the Renesas drivers
- ANC support for WM5110
- New drivers: Imagination Technologies IPs, Atmel class D speaker,
Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
- Rename PCM1792a driver to be generic pcm179x
HD-Audio:
- Use audio component for i915 HDMI/DP hotplug handling
- On-demand binding with i915 driver
- bdl_pos_adj parameter adjustment for Baytrail controllers
- Enable power_save_node for CX20722; this shouldn't lead to
regression, hopefully
- Kabylake HDMI/DP codec support
- Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
machines
- A few code refactoring
FireWire:
- Lots of code cleanup and refactoring
- Integrate the support of SCS.1x devices into snd-oxfw driver;
snd-scs1x driver is obsoleted
USB-audio:
- Fix possible NULL dereference at disconnection
- A regression fix for Native Instruments devices
Misc:
- A few code cleanups of fm801 driver"
* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
ALSA: timer: Code cleanup
ALSA: timer: Harden slave timer list handling
ALSA: hda - Add fixup for Dell Latitidue E6540
ALSA: timer: Fix race among timer ioctls
ALSA: hda - add codec support for Kabylake display audio codec
ALSA: timer: Fix double unlink of active_list
ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
ALSA: hda - fix the headset mic detection problem for a Dell laptop
ALSA: hda - Fix white noise on Dell Latitude E5550
ALSA: hda_intel: add card number to irq description
ALSA: seq: Fix race at timer setup and close
ALSA: seq: Fix missing NULL check at remove_events ioctl
ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
ASoC: AMD: Add missing include file
ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
ALSA: usb: Add native DSD support for Oppo HA-1
ASoC: Make aux_dev more like a generic component
ASoC: bcm2835: cleanup includes by ordering them alphabetically
ASoC: AMD: Manage ACP 2.x SRAM banks power
...
Diffstat (limited to 'sound/soc/codecs/ak4613.c')
-rw-r--r-- | sound/soc/codecs/ak4613.c | 118 |
1 files changed, 83 insertions, 35 deletions
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 07a266460ec3..647f69de6baa 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -70,18 +70,11 @@ #define FMT_MASK (0xf8) /* CTRL2 */ +#define DFS_MASK (3 << 2) #define DFS_NORMAL_SPEED (0 << 2) #define DFS_DOUBLE_SPEED (1 << 2) #define DFS_QUAD_SPEED (2 << 2) -struct ak4613_priv { - struct mutex lock; - - unsigned int fmt; - u8 fmt_ctrl; - int cnt; -}; - struct ak4613_formats { unsigned int width; unsigned int fmt; @@ -92,6 +85,16 @@ struct ak4613_interface { struct ak4613_formats playback; }; +struct ak4613_priv { + struct mutex lock; + const struct ak4613_interface *iface; + + unsigned int fmt; + u8 oc; + u8 ic; + int cnt; +}; + /* * Playback Volume * @@ -126,7 +129,7 @@ static const struct reg_default ak4613_reg[] = { { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, }; -#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE_TO_VAL(fmts) ((fmts - ak4613_iface) << 3) #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ @@ -240,7 +243,7 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, priv->cnt = 0; } if (!priv->cnt) - priv->fmt_ctrl = NO_FMT; + priv->iface = NULL; mutex_unlock(&priv->lock); } @@ -265,13 +268,35 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface, + int is_play, + unsigned int fmt, unsigned int width) +{ + const struct ak4613_formats *fmts; + + fmts = (is_play) ? &iface->playback : &iface->capture; + + if (fmts->fmt != fmt) + return false; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + return false; + } else { + if (fmts->width < width) + return false; + } + + return true; +} + static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); - const struct ak4613_formats *fmts; + const struct ak4613_interface *iface; struct device *dev = codec->dev; unsigned int width = params_width(params); unsigned int fmt = priv->fmt; @@ -305,33 +330,27 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, * It doesn't support TDM at this point */ fmt_ctrl = NO_FMT; - for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { - fmts = (is_play) ? &ak4613_iface[i].playback : - &ak4613_iface[i].capture; - - if (fmts->fmt != fmt) - continue; + ret = -EINVAL; + iface = NULL; - if (fmt == SND_SOC_DAIFMT_RIGHT_J) { - if (fmts->width != width) - continue; - } else { - if (fmts->width < width) + mutex_lock(&priv->lock); + if (priv->iface) { + if (ak4613_dai_fmt_matching(priv->iface, is_play, fmt, width)) + iface = priv->iface; + } else { + for (i = ARRAY_SIZE(ak4613_iface); i >= 0; i--) { + if (!ak4613_dai_fmt_matching(ak4613_iface + i, + is_play, + fmt, width)) continue; + iface = ak4613_iface + i; + break; } - - fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); - break; } - ret = -EINVAL; - if (fmt_ctrl == NO_FMT) - goto hw_params_end; - - mutex_lock(&priv->lock); - if ((priv->fmt_ctrl == NO_FMT) || - (priv->fmt_ctrl == fmt_ctrl)) { - priv->fmt_ctrl = fmt_ctrl; + if ((priv->iface == NULL) || + (priv->iface == iface)) { + priv->iface = iface; priv->cnt++; ret = 0; } @@ -340,8 +359,13 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto hw_params_end; + fmt_ctrl = AUDIO_IFACE_TO_VAL(iface); + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); - snd_soc_write(codec, CTRL2, ctrl2); + snd_soc_update_bits(codec, CTRL2, DFS_MASK, ctrl2); + + snd_soc_write(codec, ICTRL, priv->ic); + snd_soc_write(codec, OCTRL, priv->oc); hw_params_end: if (ret < 0) @@ -431,6 +455,28 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), }; +static void ak4613_parse_of(struct ak4613_priv *priv, + struct device *dev) +{ + struct device_node *np = dev->of_node; + char prop[32]; + int i; + + /* Input 1 - 2 */ + for (i = 0; i < 2; i++) { + snprintf(prop, sizeof(prop), "asahi-kasei,in%d-single-end", i + 1); + if (!of_get_property(np, prop, NULL)) + priv->ic |= 1 << i; + } + + /* Output 1 - 6 */ + for (i = 0; i < 6; i++) { + snprintf(prop, sizeof(prop), "asahi-kasei,out%d-single-end", i + 1); + if (!of_get_property(np, prop, NULL)) + priv->oc |= 1 << i; + } +} + static int ak4613_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -458,7 +504,9 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, if (!priv) return -ENOMEM; - priv->fmt_ctrl = NO_FMT; + ak4613_parse_of(priv, dev); + + priv->iface = NULL; priv->cnt = 0; mutex_init(&priv->lock); |