Changes since v1.21:
v1.22 - 27.04.2020
- Fixed crash when loading XMs with pattern lengths above 256. They are now
safely truncated, and the user gets a warning message about the truncation.
- Allow loading overflown BPM/speed values from quirky XMs, up to BPM 999
and speed 99 ("jk_error_txt.xm" is an example for quirky BPM/speed values).
- Tiny code cleanup
ver 0.21.23 (2020/04/23)
* protocol
- add tag fallback for AlbumSort
* storage
- curl: fix corrupt "href" values in the presence of XML entities
- curl: unescape "href" values
* input
- nfs: fix crash bug
- nfs: fix freeze bug on reconnect
* decoder
- gme: adapt to API change in the upcoming version 0.7.0
* output
- alsa: implement channel mapping for 5.0 and 7.0
* player
- drain outputs at end of song in "single" mode
* Windows
- fix case insensitive search
Changes since v1.20:
v1.21 - 22.04.2020
- macOS/Linux: The initial working directory would not get properly set when
opening Disk Op.
- Bugfix: The custom directory paths in the config could contain unwanted junk
at the end if the loaded config file was saved from real FT2.
- Bugfix: If you were skilled enough to beat the very last Nibbles level, your
potential entry in the highscore table would show garbled level graphics.
- Show a warning if either of the Buttons/Desktop palette colors are so dark
that you might have issues seeing the walls in Nibbles.
- "Space Pigs" palette replaced with a new (low contrast) "Dark mode" palette
- Some more small miscellaneous fixes not worth of a mention
Release 2020.04.05
- Support writing audio to named pipe (audio_pipe)
- Add resampling (sample_rate)
- Fix station list not showing more than 95 stations
Pragha is a Lightweight Music Player, based on GTK and SQLite.
It is completely written in C, constructed to be fast, light, and
simultaneously tries to be complete without obstructing the daily work.
opus-tools 0.2
Sep 18, 2018
In this release the Opus decoder opusdec has been converted to use the
opusfile library, and the Opus encoder opusenc has been converted to
use the libopusenc library. These libraries make it easy to robustly
read and write Ogg Opus audio files, and enable some new features.
opusdec enhancements include:
* Read directly from http or https sources
* New option --force-stereo will force stereo output
* Improved support for chained input files with differing sample rate
or channel count
* A summary is displayed for METADATA_BLOCK_PICTURE tags rather than
displaying the base64-encoded data
opusenc enhancements include:
* Delayed decision support allows the encoder to look ahead up to two
seconds in order to improve encoding decisions
* The options --music and --speech can be used to tune low bitrate
audio for music or speech, overriding automatic detection
* The option --no-phase-inv disables the use of phase inversion for
intensity stereo, which can be useful for streams that are likely
to be downmixed to mono after decoding
* New --tracknumber shortcut for setting tracknumber metadata
Additionally:
* The opusinfo utility can display the demixing matrix from Ogg Opus
files using ambisonics channel mapping family 3
* The experimental opusrtp tool supports new options to specify RTP
payload type, Ogg Opus output file, original sample rate, and
number of channels, and supports improved transmit timing,
arbitrary network devices, and IPv6
* Numerous bug fixes are also included
Version 1.5.0 -- April 15, 2020
===============================
- Added support for fpcalc -raw -signed, which helps with easier PostgreSQL integration
- Added support for using libavresample instead of libswresample
- Fixed possible crash in chromaprint_decode_fingerprint
- Fixed unit tests on big endian CPUs
Bug fixes since 4.0.1:
* Fixed a regression where the Enter key did not work in the playlist
* Fixed the seek bar changing width as the time counter increments
* Fixed the title bar getting stuck showing "Buffering"
* Added a workaround to ensure that the toolbar is not hidden on startup
* Fixed the Composer field not being saved when updating ID3v2 tags
Changes since 46.1:
DIN Is Noise 46.2 @ Chennai, India
* optimised GUI controls a bit
* improved mirroring curve about horizontal, vertical axes
+ new horizontal, vertical local axes
+ new seed curves in fractaliser plugin
! fixed bug in identifying and setting audio devices !
! introduced in DIN 46 :( !
Compile error log: (Similar message found on aarch64)
ld: mute.o: relocation R_X86_64_32 against `.rodata.str1.1' can not be used when making a shared object; recompile with -fPIC
(please advice me if I'm doing wrong, thanks)
5.3.0:
fixed: OSS-Fuzz issues 19925, 19928, 20060, 20448
(no CVEs or vulnerabilities)
fixed: trailing garbage characters on imported ID3v2 TXXX tags
fixed: various minor undefined behavior and memory access issues
fixed: sanitize tag extraction names for length and path inclusion
improved: reformat wvunpack "help" and split into long + short versions
added: regression testing to Travis CI for OSS-Fuzz crashers
While here, adjust some hacks around differences between NetBSD ossaudio
and OSSv4:
It turns out the problem was Audacity attempting to initialize with
an absurdly high sample rate. When a sample rate that's outside the
range of the kernel's maximum of 192000 is passed to SNDCTL_DSP_SPEED,
-1 was returned (until my commits today).
This is a violation of the OSSv4 spec in NetBSD. The spec states that if
an unsupported sample rate is requested, a supported one should be picked
and returned.
Changes since v1.19:
v1.20 - 13.04.2020
- x86_64/amd64 users now get 32.32 fixed-point precision in the audio channel
mixer, an extra 192kHz mixing frequency mode (may only work with a larger
buffer size), and double the amount of phases in the cubic spline table. This
will only make an improved outcome in very few cases (like extremely high/low
channel rates), so don't expect to hear any difference. Non-x86_64 users now
lose 96kHz mode as it was a bit unsuitable with 16.16 fixed-point precision.
- The "Volume" function in Smp. Ed. now has more precision and takes less time
to complete. Not that it was slow to begin with...
- Increased the minimum .WAV rendering frequency from 8kHz to 44.1kHz, because
the mixer delta calculations could overflow on such low rates.
- Some more small miscellaneous changes not worth of a mention
- Code cleanup
- Use AUDIO_INITINFO rather than AUDIO_GETINFO for initialization
This is recommended by Solaris and NetBSD documentation.
- Don't open the device with O_NONBLOCK then remove the flag.
AFAIK this is a workaround for old implementations that wouldn't
allow the device to be opened in blocking mode if it was already
in use. Either way, it hasn't been necessary for a long time.
- Support S8 and S32 audio formats.
