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authorLinus Torvalds <torvalds@linux-foundation.org>2022-05-25 16:55:16 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2022-05-25 16:55:16 -0700
commitd7227785e384d4422b3ca189aa5bf19f462337cc (patch)
treefbd117e5e3f31a8561f6b6d159413bc744f460e5 /include/sound/soc-component.h
parent2518f226c60d8e04d18ba4295500a5b0b8ac7659 (diff)
parent60571929d06b028800f27b51a7c81de1144944cf (diff)
Merge tag 'sound-5.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "Not much dramatic changes at this time, but we've received quite a lot of changes for ASoC, while there are still a few fixes and quirks for usual HD- and USB-auido. Here are some highlights. ASoC: - Overhaul of endianness specification for data formats, avoiding needless restrictions due to CODECs - Initial stages of Intel AVS driver merge - Introduction of v4 IPC mechanism for SOF - TDM mode support for AK4613 - Support for Analog Devices ADAU1361, Cirrus Logic CS35L45, Maxim MAX98396, MediaTek MT8186, NXP i.MX8 micfil and SAI interfaces, nVidia Tegra186 ASRC, and Texas Instruments TAS2764 and TAS2780 Others: - A few regression fixes after the USB-audio endpoint management refactoring - More enhancements for Cirrus HD-audio codec support (still ongoing) - Addition of generic serial MIDI driver" * tag 'sound-5.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (504 commits) ALSA: hda/realtek - Add new type for ALC245 ALSA: usb-audio: Configure sync endpoints before data ALSA: ctxfi: fix typo in comment ALSA: cs5535audio: fix typo in comment ALSA: ctxfi: Add SB046x PCI ID ALSA: usb-audio: Add missing ep_idx in fixed EP quirks ALSA: usb-audio: Workaround for clock setup on TEAC devices ALSA: lola: Bounds check loop iterator against streams array size ASoC: max98090: Move check for invalid values before casting in max98090_put_enab_tlv() ASoC: rt1308-sdw: add the default value of register 0xc320 ASoC: rt9120: Use pm_runtime and regcache to optimize 'pwdnn' logic ASoC: rt9120: Fix 3byte read, valule offset typo ASoC: amd: acp: Set Speaker enable/disable pin through rt1019 codec driver. ASoC: amd: acp: Set Speaker enable/disable pin through rt1019 codec driver ASoC: wm2000: fix missing clk_disable_unprepare() on error in wm2000_anc_transition() ASoC: codecs: lpass: Fix passing zero to 'PTR_ERR' ASoC: SOF: sof-client-ipc-flood-test: use pm_runtime_resume_and_get() ASoC: SOF: mediatek: remove duplicate include in mt8195.c ASoC: SOF: mediatek: Add mt8195 debug dump ASoC: SOF: mediatek: Add mediatek common debug dump ...
Diffstat (limited to 'include/sound/soc-component.h')
-rw-r--r--include/sound/soc-component.h9
1 files changed, 9 insertions, 0 deletions
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 766dc6f009c0..5a764c3099d3 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -169,6 +169,15 @@ struct snd_soc_component_driver {
unsigned int idle_bias_on:1;
unsigned int suspend_bias_off:1;
unsigned int use_pmdown_time:1; /* care pmdown_time at stop */
+ /*
+ * Indicates that the component does not care about the endianness of
+ * PCM audio data and the core will ensure that both LE and BE variants
+ * of each used format are present. Typically this is because the
+ * component sits behind a bus that abstracts away the endian of the
+ * original data, ie. one for which the transmission endian is defined
+ * (I2S/SLIMbus/SoundWire), or the concept of endian doesn't exist (PDM,
+ * analogue).
+ */
unsigned int endianness:1;
unsigned int non_legacy_dai_naming:1;