4 files changed, 6 insertions, 3 deletions

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 809d72b8eff1..a46ddb85e83a 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -244,6 +244,7 @@ STAC9227/9228/9229/927x
   5stack-no-fp  D965 5stack without front panel
   dell-3stack   Dell Dimension E520
   dell-bios     Fixes with Dell BIOS setup
+  dell-bios-amic Fixes with Dell BIOS setup including analog mic
   volknob       Fixes with volume-knob widget 0x24
   auto          BIOS setup (default)
 
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c3c912d023cc..42a0a39b77e6 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -454,6 +454,8 @@ The generic parser supports the following hints:
 - need_dac_fix (bool): limits the DACs depending on the channel count
 - primary_hp (bool): probe headphone jacks as the primary outputs;
   default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+  line-in/surround, mic/clfe jacks)
 - multi_cap_vol (bool): provide multiple capture volumes
 - inv_dmic_split (bool): provide split internal mic volume/switch for
   phase-inverted digital mics
diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44
index 0e41576fa13e..67b2ea1cc31d 100644
--- a/Documentation/sound/alsa/README.maya44
+++ b/Documentation/sound/alsa/README.maya44
@@ -120,7 +120,7 @@ Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on
     Make sure this is not turned on while any other source is connected to input 1/2.
     It might damage the source and/or the maya44 card.
 
-Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+Mic/Line input: if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
 
 Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
 Bypass 1: same for channel 3+4.
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
index 0bcc55155911..fd74ff26376e 100644
--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@@ -73,7 +73,7 @@ The main requirements are:
 
 Design
 
-The new API shares a number of concepts with with the PCM API for flow
+The new API shares a number of concepts with the PCM API for flow
 control. Start, pause, resume, drain and stop commands have the same
 semantics no matter what the content is.
 
@@ -130,7 +130,7 @@ the settings should remain the exception.
 The timestamp becomes a multiple field structure. It lists the number
 of bytes transferred, the number of samples processed and the number
 of samples rendered/grabbed. All these values can be used to determine
-the avarage bitrate, figure out if the ring buffer needs to be
+the average bitrate, figure out if the ring buffer needs to be
 refilled or the delay due to decoding/encoding/io on the DSP.
 
 Note that the list of codecs/profiles/modes was derived from the