Detailed changes v1.2.7.2 v1.2.8: Difference between revisions
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Latest revision as of 06:39, 25 October 2022
Detailed changelog between 1.2.7 and 1.2.8 releases
Changelog between 1.2.7.2 and 1.2.8 releases
alsa-utils
Core
- Release v1.2.8
- configure: automake - use subdir-objects option
- configure: use AC_DISABLE_STATIC (for topology modules)
ALSA Control (alsactl)
- alsactl: Print driver name in info
alsatplg (topology)
- configure: use AC_DISABLE_STATIC (for topology modules)
amixer
- amixer: fix the help for 'events' command
aplay/arecord
- aplay: remove wrongly merged UCM code
The code origin comes from me. Unfortunately, it was merged
by mistake to another commit. Remove it for now.
- aplay: fix the capture file length regression
The commit 4b959a4 ("aplay: Fix for arecord recording ghost data")
intruduced a regression (file may contain more samples than
expected). Correct this.
Detailed changelog between 1.2.5 and 1.2.8 releases
Changelog between 1.2.7.2 and 1.2.8 releases
tinycompress
Core
- Release v1.2.8
- README: mention official mirror
Since alsa-project is 'the' official mirror, we should still keep these links
- README: remove old alsa-project link
git.alsa-project is not main repo now, so remove that in favour of the github link
Utilities
- cplay: add parentheses around comparison in operand of ‘&’
We get a warning:
cplay.c: In function ‘find_adts_header’:
cplay.c:259:41: warning: suggest parentheses around comparison in operand of ‘&’ [-Wparentheses]
259 | if ((buf[0] != 0xff) || (buf[1] & 0xf0 != 0xf0))
Resolve by adding a parentheses around "&" case
- cplay: remove set but not used warnings
cplay.c: In function ‘find_adif_header’:
cplay.c:367:13: warning: variable ‘object_type’ set but not used [-Wunused-but-set-variable]
367 | int object_type;
| ^~~~~~~~~~~
cplay.c:366:13: warning: variable ‘bitrate’ set but not used [-Wunused-but-set-variable]
366 | int bitrate;
| ^~~~~~~
cplay.c:91:27: warning: ‘DEFAULT_CODEC_ID’ defined but not used [-Wunused-const-variable=]
91 | static const unsigned int DEFAULT_CODEC_ID = SND_AUDIOCODEC_PCM;
| ^~~~~~~~~~~~~~~~
While at it, cleanup the code a bit as well removing now redundant
braces around if-else
So remove these unused variables
- utils: cplay: Add support for ID3v2 tag skip
This patch will allow playing MP3 files containing ID3v2
tag at the beginning of the file.
This is done simply by parsing the ID3v2 header, extracting
the header size and seeking to that position (+10 bytes
from the header size which is not included in the 'header
size' field of the ID3v2 header)
- utils: cplay: Add support for pause/resume
We want to be able to pause and resume the stream
just like the ALSA utilitary aplay does.
In order to do so, we first need to mark the read operations
on stdin as nonblocking and then enable the noncanonical mode
for the terminal. What this does is it makes the read operations
nonblocking and it makes the input available immediately.
After doing so, we can check if we receive a SPACE or ENTER
character from user and do pause/resume depending on current
stream state (and by this I mean if the stream is currently
paused and we receive a SPACE or ENTER then we resume it
and vice-versa)
- utils: cplay: Reset file cursor after MP3 header parse
We need to reset the file cursor to the beginning of the file.
Initially, the program would simply get stuck polling the
compress fd. This was probably because of the fact that the
codec would hang because of the fact that it was expecting
to receive the MP3 data along with its associated MP3 header.
This was not the case for the first (header, data) pair because,
after parsing the first header, the file cursor would point
at the beginning of the data region.
By resetting the file cursor to the beginning of the file,
the codec will receive all the (header, data) pairs it
actually expects.
Suggested-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Detailed changelog between 1.2.7.2 and 1.2.8 releases
Changelog between 1.2.7.2 and 1.2.8 releases
alsa-lib
Core
- Release v1.2.8
- add FreeBSD build support (except test/)
Note: NetBSD and OpenBSD has both <endian.h> and <sys/endian.h>
- add NetBSD/OpenBSD build support (except test/)
Control API
- control: namehint - do not support 'card' devices
The card: devices do not exist. Do not return -ENOENT,
but -EINVAL in this case.
- control: eld - skip trailing spaces
PCM API
- pcm: rate: fix the crash for the partial period copy
The size argument in snd_pcm_rate_commit_area can be smaller
than cont in drain. Return the original code. It should
not make things worse. Perhaps, we can fill the rest of
the period with silence in this case.
- add DragonFlyBSD build support (except test/)
due to undefined reference to 'environ' error, ucm_exec.c uses workaround.
- pcm: hw_params - copy periods and buffer_time to the local variable
It is expected to return the correct numbers for get_periods and
get_buffer_time requests. Copy the appropriate fields in the
hw_params call.
- pcm: fix the dshare delay reporting
Pulseaudio uses full ring buffer when the incorrect delay is reported.
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/198
- pcm: direct plugins: make three more symbols private to library
Those symbols should not be exported:
snd_pcm_direct_poll_descriptors
snd_pcm_direct_check_xrun
snd_pcm_direct_slave_recover
Use Case Manager API
- ucm: do not handle multiple Syntax field updates
It is useful to include a toplevel configuration file from another
toplevel configuration file. Ignore the further Syntax updates
(assuming the that the parent knows what to do).
Also, parse the Syntax field in own function.
- add DragonFlyBSD build support (except test/)
due to undefined reference to 'environ' error, ucm_exec.c uses workaround.
- add FreeBSD build support (except test/)
Note: NetBSD and OpenBSD has both <endian.h> and <sys/endian.h>
- doc: ucm - more volume notes
- doc: ucm - add sequence graphs
- ucm: add NULL check for card_name in open
- ucm: fix few memory-leaks in the error paths
Async helpers
- add NetBSD/OpenBSD build support (except test/)
Compatibility routines
- add NetBSD/OpenBSD build support (except test/)
Configuration
- Remove non existent SPDIF output on ThinkPad USB-C Dock Gen2
- add DragonFlyBSD build support (except test/)
due to undefined reference to 'environ' error, ucm_exec.c uses workaround.
- add NetBSD/OpenBSD build support (except test/)
Documentation
- doc: ucm - more volume notes
- doc: ucm - add sequence graphs
Kernel Headers
- add FreeBSD build support (except test/)
Note: NetBSD and OpenBSD has both <endian.h> and <sys/endian.h>
- add NetBSD/OpenBSD build support (except test/)
alsa-ucm-conf
Configuration
- USB-Audio: Add support for Arturia Minifuse 1
Recode Minifuse 2 config to support Minifuse 1, too.
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/220
- Rockchip/max98090: add HDMI device
- USB-Audio: ALC4080: Add MSI MPG X670E Carbon Wifi (USB0db0:d6e7)
- USB-Audio: ALC4080 - add back SPDIF switch
It seems that 'IEC958 Playback Switch' is valid for the hardware
with the provided USB mapping in the driver. For others, it appears
that this switch is 'PCM PLayback Switch' with index 2.
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/227
- USB-Audio: ALC4080 - detect Speaker / Front Headphone controls
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/227
- ucm2: simplify acp3xalc5682m98.conf
Use SectionUseCase block only one time.
BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/146
- ucm2: Support acp3xalc5682m98 on Chromebook "zork"
- ucm2: mt8195-sof: Fix wrong JackControl for headphone
The value set for the headphone's JackControl doesn't match any jack
control available. Change it to the right one.
Tested-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
- USB-Audio: UR24C - add Steinberg UR24C (USB0499:174d)
- USB-Audio: alc4080 - add MSI MEG X570S Ace Max (ID 0db0:a47c)
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/186
- USB-Audio: alc4080 - add MSI MPG Z690 CARBON WIFI DDR5 (ID 0db0:005a)
- USB-Audio: Fix Motu M2/M4 regex expression (wrong field)
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/214
- sdm845: add LENOVO Yoga C630 support
LENOVO YOGA C630 has MultiMedia1 dai setup as primary Playback so add required
HiFi configuration as well.
- ucm2: codecs: wcd934x: Add enable disable sequences
- ucm2: USB-Audio: Add correct support for Rane SL-1
- ucm2: Add support for MT8195 Cherry Tomato Chromebook with SOF
Add support for the Acer Chromebook Spin 513 CP513-2H, powered
by MediaTek Kompanio 1380 (MT8195) with a HiFi 4 Audio DSP running
SoundOpenFirmware.
This machine uses MT6359 PMIC, with RT1019 and RT5682 as headphones
and speaker codecs/amps.
This configuration has been successfully tested on Debian 11
with PipeWire 0.3.56 + wireplumber.
- USB-Audio: Motu M4 - add new USB ID 07fd:0008
- USB-Audio: ALC4080 - add MSI X570S EDGE MAX WIFI USB ID 0db0:151f
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/172
- USB-Audio: ALC4080 - add Gigabyte Z590 Aorus Pro AX (USB 0414:a00e)
- ucm2: update DEBUG.md (download latest)
- USB-Audio: ALC4080 - correct S/PDIF PCM device for USB 0db0:1feb
USB 0db0:1feb - MSI Edge Wifi Z690
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/210
- USB-Audio: ALC4080 - fix S/PDIF and Mic2 PCM values
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/210
- USB-Audio: ALC4080 - fix PCM,1 mixer control value settings
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/210
- ucm2: add DEBUG.md
- USB-Audio: ALC4080 - add 0db0:b202 MSI MAG Z690 Tomahawk Wifi
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/189
- ucm2: sof-glkda7219max: add initial support
Speaker and headphone work.
Headset mic works.
Automatic detection of headset/headphone work with additional kernel
changes to remap jack pins, queued for review at
https://github.com/thesofproject/linux/pull/3681
DMIC doesn't seem to work, but this could be because of the 4ch
capture. We need the ability to extract 2ch as done by CRAS with
CaptureChannelMap "0 0 -1 -1 -1 -1 -1 -1 -1 -1 -1"
BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/150
- ucm2: HDA - add support for Internal Mic
- USB-Audio: Add Behringer UMC202HD configuration
- USB-Audio: Adding Focusrite Scarlett 2i4 gen2
- USB-Audio: ALC4080 - add ASUS ROG Strix B660-F Gaming WiFi USB ID
Fix for ALC4080 (ASUS ROG Strix B660-F Gaming WiFi)
- USB-Audio: Add profile for MSI MEG Z690I Unify
Very similar to c710aa0eb2fe184072b312a8afa5c4e301e93a65, but
this board only has 3 devices, Mic, Line-Out and Line-In. Thus
we are using the same conf. Mainly enabling the Mic by default,
but we also need to disable the unused devices from the MSI
MPG X570S Max WiFi.
- USB-Audio: Realtek ALC4080 cleanups
Based on alsa-info.sh output, this is a complete cleanup for the ALC4080
codecs. The simple mixer controls names are fixed based on the amixer
output contents. Because I cannot test those changes directly with hardware,
please, create an issue or pull request to fix the possible regressions.
PCM playback devices: 0,1,2,3
PCM capture devices: 0,1,2
Simple playback mixer controls: 'PCM',0 8ch
'PCM',1 2ch
'PCM',2 mono
Simple capture mixer controls: 'Line',0 2ch
'Mic',0 2ch
'Analog In',0 2ch
Jacks: 'Speaker - Output Jack'
'Headphone - Output Jack'
'Mic - Input Jack'
'Line - Input Jack'
'Analog In - Input Jack'
Alsa-info: https://launchpadlibrarian.net/573795872/AlsaInfo.txt
MAG X570S TORPEDO MAX (MS-7D54) USB0db0:a073
Alsa-info: https://github.com/alsa-project/alsa-ucm-conf/files/9078972/alsa-info.txt
MAG X570S TOMAHAWK MAX WIFI (MS-7D54) USB0db0:a073
Alsa-info: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/uploads/8fb28cc3c7f98c642d0d1f0e8521dc49/alsa-info.txt
MPG Z690 EDGE WIFI DDR4 (MS-7D31) USB0db0:1feb
Alsa-info: http://alsa-project.org/db/?f=131d4da535c4ec4545c74616f83f393f1c9152dd
ROG STRIX Z590-F GAMING WIFI USB0b05:1996
- USB-Audio: alc4080 - add multichannel variants
- USB-Audio: Aorus-Master-Main-Audio - fix the secondary card lookup
There was missing bracket.
- Behringer UCM204HD - use S32_LE format for dshare/dsnoop
BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/192
- rt715-sdca: use sensible capture gain value
The value of 124 used for FU02 is way too much, with saturation even in low-volume cases. The 0dB value (47) is a much better initialization value.
BugLink: https://github.com/thesofproject/linux/issues/3766
Resolves: https://github.com/alsa-project/alsa-ucm-conf/pull/193
- USB-Audio: add MOTU M2 config
MOTU M2 and M4 has similiar USB ID. Use the long card name to
identify the M4 variant.
Description
- Release v1.2.8
- README: fix tar compress arguments in README
- README: add the latest config download info