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authorLinus Torvalds <torvalds@linux-foundation.org>2023-04-27 10:58:37 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2023-04-27 10:58:37 -0700
commit1c15ca4e4efaddb78f83eed31eeee34c522c3ae2 (patch)
treea528054028d13fb3361ec72663c7fce7b619564b /Documentation/sound/cards/sb-live-mixer.rst
parent34b62f186db9614e55d021f8c58d22fc44c57911 (diff)
parentbaa6584a24494fbbd2862270d39e61b86987cc91 (diff)
Merge tag 'sound-6.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "At this time, it's an interesting mixture of changes for both old and new stuff. Majority of changes are about ASoC (lots of systematic changes for converting remove callbacks to void, and cleanups), while we got the fixes and the enhancements of very old PCI cards, too. Here are some highlights: ALSA/ASoC Core: - Continued effort of more ASoC core cleanups - Minor improvements for XRUN handling in indirect PCM helpers - Code refactoring of PCM core code ASoC: - Continued feature and simplification work on SOF, including addition of a no-DSP mode for bringup, HDA MLink and extensions to the IPC4 protocol - Hibernation support for CS35L45 - More DT binding conversions - Support for Cirrus Logic CS35L56, Freescale QMC, Maxim MAX98363, nVidia systems with MAX9809x and RT5631, Realtek RT712, Renesas R-Car Gen4, Rockchip RK3588 and TI TAS5733 ALSA: - Lots of works for legacy emu10k1 and ymfpci PCI drivers - PCM kselftest fixes and enhancements" * tag 'sound-6.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (586 commits) ALSA: emu10k1: use high-level I/O in set_filterQ() ALSA: emu10k1: use high-level I/O functions also during init ALSA: emu10k1: fix error handling in snd_audigy_i2c_volume_put() ALSA: emu10k1: don't stop DSP in _snd_emu10k1_{,audigy_}init_efx() ALSA: emu10k1: fix SNDRV_EMU10K1_IOCTL_SINGLE_STEP ALSA: emu10k1: skip Sound Blaster-specific hacks for E-MU cards ALSA: emu10k1: fixup DSP defines ALSA: emu10k1: pull in some register definitions from kX-project ALSA: emu10k1: remove some bogus defines ALSA: emu10k1: eliminate some unused defines ALSA: emu10k1: fix lineup of EMU_HANA_* defines ALSA: emu10k1: comment updates ALSA: emu10k1: fix snd_emu1010_fpga_read() input masking for rev2 cards ALSA: emu10k1: remove unused emu->pcm_playback_efx_substream field ALSA: emu10k1: remove unused `resume` parameter from snd_emu10k1_init() ALSA: emu10k1: minor optimizations ALSA: emu10k1: remove remaining cruft from snd_emu10k1_emu1010_init() ALSA: emu10k1: remove apparently pointless EMU_HANA_OPTION_CARDS reads ALSA: emu10k1: remove apparently pointless FPGA reads ALSA: emu10k1: stop doing weird things with HCFG in snd_emu10k1_emu1010_init() ...
Diffstat (limited to 'Documentation/sound/cards/sb-live-mixer.rst')
-rw-r--r--Documentation/sound/cards/sb-live-mixer.rst17
1 files changed, 10 insertions, 7 deletions
diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst
index 357fcd619d39..819886634400 100644
--- a/Documentation/sound/cards/sb-live-mixer.rst
+++ b/Documentation/sound/cards/sb-live-mixer.rst
@@ -15,7 +15,7 @@ The ALSA driver programs this portion of chip by default code
IEC958 (S/PDIF) raw PCM
=======================
-This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+This PCM device (it's the 3rd PCM device (index 2!) and first subdevice
(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
little endian streams without any modifications to the digital output
(coaxial or optical). The universal interface allows the creation of up
@@ -33,9 +33,9 @@ Digital mixer controls
These controls are built using the DSP instructions. They offer extended
functionality. Only the default built-in code in the ALSA driver is described
here. Note that the controls work as attenuators: the maximum value is the
-neutral position leaving the signal unchanged. Note that if the same destination
-is mentioned in multiple controls, the signal is accumulated and can be wrapped
-(set to maximal or minimal value without checking of overflow).
+neutral position leaving the signal unchanged. Note that if the same destination
+is mentioned in multiple controls, the signal is accumulated and can be clipped
+(set to maximal or minimal value without checking for overflow).
Explanation of used abbreviations:
@@ -46,11 +46,11 @@ ADC
analog to digital converter
I2S
one-way three wire serial bus for digital sound by Philips Semiconductors
- (this standard is used for connecting standalone DAC and ADC converters)
+ (this standard is used for connecting standalone D/A and A/D converters)
LFE
- low frequency effects (subwoofer signal)
+ low frequency effects (used as subwoofer signal)
AC97
- a chip containing an analog mixer, DAC and ADC converters
+ a chip containing an analog mixer, D/A and A/D converters
IEC958
S/PDIF
FX-bus
@@ -313,6 +313,9 @@ MANUALS/PATENTS
ftp://opensource.creative.com/pub/doc
-------------------------------------
+Note that the site is defunct, but the documents are available
+from various other locations.
+
LM4545.pdf
AC97 Codec
m2049.pdf