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In the C plug-ins, mark the constructor and destructor functions as
such. While there, comment out a new target to run the tests; they are
broken, but not because of this modification. This allows us to use
cc(1) to link the plug-ins, thus working around a bug in the cwrappers
for ld(1).
Bump PKGREVISION, since this generates a different binary now that SSP
and FORTIFY are enabled.
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This helps pass the RELRO check for this package.
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1.39:
* Tests:
* Require `hypothesis <https://hypothesis.readthedocs.io>`__
* Run pycodestyle/pyflakes tests by default. Skip with ``--no-quality`` or
``-m no quality`` when using pytest directly.
* Python 3.3 is no longer supported
* MP3: Improved bitrate accuracy for files with XING header
* ASF: Fix case where some tags resulted in broken ASFUnicodeAttribute
instances
* Add support for filesystems which don't support opening files read/write
(gvfs over fuse for example)
* mid3v2: Add support for USLT
* Minor improvements
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This notably fixes building with RELRO enabled.
Bump PKGREVISION, since this generates a different binary now that SSP and
FORTIFY are enabled.
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This notably fixes building with RELRO enabled.
Bump PKGREVISION, since this generates a different binary now that SSP and
FORTIFY are enabled.
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1.25.7
------
- mpg123:
-- Do not play with cursor and inverse video for progress bar
when TERM=dumb.
-- Fix parsing of host port for numerical IPv6 addresses (just did
not work before, only for textual host names).
- libmpg123:
-- Proper fix for the xrpnt overflow problems by correctly
initialising certain tables for MPEG 2.x layer III. The checks that
catch the resulting overflow are still in place, but likely superfluous
now. Note that this means certain valid files would have been misdecoded
before, if anyone actually produced them. Thanks to Robert Hegemann for
the fix!
-- Silently handle granules with part2_3_length == 0, but
scalefac_compress != 0 (ignore the latter).
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See https://github.com/westes/flex/issues/241
The problem is fixed in master branch and the workaround could be removed
in the next version update.
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* -Werror does not work properly since id3lib makes warnings
Changes:
v3.6.1 2017-10-04 Johnny A. Solbu <johnny@solbu.net>
* Patch from Adrian Reber: fix compiler warning
and enable -Werror and -Wall by default
* Translation update: Norwegian
v3.6.0 2017-09-14 Johnny A. Solbu <johnny@solbu.net>
* Patch from Mike Gilbode: Select DiscDB entry from multiple results
* Translation updates.
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Many bug fixes and security fixes.
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- Revert "Move gdbus-codegen users to py-glib2-tools by including
glib2/buildtools.mk" 1f764df
- while here change to TOOL_DEPENDS
- switch from py-glib2-tools to glib2-tools
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Version 0.8.9.0:
OOB Write and Read fixes + a number of divide by zero fixes.
(ABC, PAT, AMF, MDL, PSM, XM, IT, MMCMP, MID)
There were some patches 2010-2016 which were recorded here.
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Changes not documented.
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When it is found, the mpris2 plugin is built, which lead to "random"
PLIST problems. Depend on it to always build it.
Bump PKGREVISION.
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Changelog:
* Fix a crash in filter modules related to flat volumes and volume
sharing
* Fix a crash when the bluetooth adapter reports weird MTU size
* Disable bluetooth MTU autodetection by default
* Add mixer handling back for hardware that doesn't have any alsa-lib
configuration
* Prioritize USB devices over built-in sound cards (11.0 was supposed
to have this feature, but the implementation turned out to be
incomplete)
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openal-soft-1.18.2:
Fixed resetting the FPU rounding mode after certain function calls on
Windows.
Fixed use of SSE intrinsics when building with Clang on Windows.
Fixed a crash with the JACK backend when using JACK1.
Fixed use of pthread_setnane_np on NetBSD.
Fixed building on FreeBSD with an older freebsd-lib.
OSS now links with libossaudio if found at build time (for NetBSD).
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libopenmpt 0.3.1 (2017-09-28)
Changelog since libopenmpt 0.2.6774-beta20.
libopenmpt 0.3.1
[Bug] Windows: libopenmpt resource did not compile for release versions.
libopenmpt 0.3.0 (2017-09-27, not released)
[New] New error handling functionality in the C API, which in particular allows distinguishing potentially transient out-of-memory errors from parse errors during module loading.
[New] New API openmpt::module::get_selected_subsong() (C++) and openmpt_module_get_selected_subsong() (C).
[New] Faster file header probing API openmpt::probe_file_header() and openmpt::probe_file_header_get_recommended_size (C++), and openmpt_probe_file_header(), openmpt_probe_file_header_without_filesize(), openmpt_probe_file_header_from_stream() and openmpt_probe_file_header_get_recommended_size() (C).
[New] New API openmpt::could_open_probability() (C++) and openmpt_could_open_probability() (C). This fixes a spelling error in the old 0.2 API.
[New] openmpt123: openmpt123 can now open M3U, M3U8, M3UEXT, M3U8EXT and PLSv2 playlists via the --playlist option.
[New] openmpt123: openmpt123 now supports very fast file header probing via the --probe option.
[New] Libopenmpt now supports building for Windows 10 Universal (Windows Store 8.2) APIs with MSVC, and also for the older Windows Runtime APIs with MinGW-w64.
[New] New API header libopenmpt_ext.h which implements the libopenmpt extension APIs also for the C interface.
[New] The Reverb effect (S99 in S3M/IT/MPTM, and X99 in XM) is now implemented in libopenmpt.
[New] For Amiga modules, a new resampler based on the Amiga’s sound characteristics has been added. It can be activated by passing the render.resampler.emulate_amiga ctl with a value of 1. Non-Amiga modules are not affected by this, and setting the ctl overrides the resampler choice specified by OPENMPT_MODULE_RENDER_INTERPOLATIONFILTER_LENGTH or openmpt::module::RENDER_INTERPOLATIONFILTER_LENGTH. Support for the MOD command E0x (Set LED Filter) is also available when the Amiga resampler is enabled.
[Change] libopenmpt versioning changed and follows the more conventional major.minor.patch as well as the recommendations of the SemVer scheme now. In addition to the SemVer requirements, pre-1.0.0 versions will also honor API and ABI stability in libopenmpt (i.e. libopenmpt ignores SemVer Clause 4).
[Change] The output directories of the MSVC build system were changed to bin/vs2015-shared/x86-64-win7/ (and similar) layout which allows building in the same tree with different compiler versions without overwriting other outputs.
[Change] The emscripten build now exports libopenmpt as ‘libopenmpt’ instead of the default ‘Module’.
[Change] Android: The build system changed. The various Android.mk files have been merged into a single one which can be controlled using command line options.
[Change] The Makefile build system now passes std=c++11 to the compiler by default. Older compilers may still work if you pass STDCXX=c++0x to the make invocation.
[Change] The Makefile option ANCIENT=1 is gone.
[Change] The optional dependencies on libltdl or libdl are gone. They are no longer needed for any functionality.
[Regression] Compiling client code using the C++ API now requires a compiler running in C++11 mode.
[Regression] Support for GCC 4.1, 4.2, 4.3, 4.4, 4.5, 4.6, 4.7 has been removed.
[Regression] Support for Clang 3.0, 3.1, 3.2, 3.3 has been removed.
[Regression] Support for Emscripten versions older than 1.31.0 has been removed.
[Regression] Support for Android NDK versions older than 11 has been removed.
[Regression] Visual Studio 2008, 2010, 2012, 2013 support has been removed.
[Regression] Dynamic run-time loading of libmpg123 is no longer supported. Libmpg123 must be linked at link-time now.
[Regression] xmp-openmpt: xmp-openmpt now requires XMPlay 3.8 or later and compiling xmp-openmpt requires an appropriate XMPlay SDK with XMPIN_FACE >= 4.
[Regression] Support for libmpg123 older than 1.13.0 has been removed.
[Regression] Un4seen unmo3 support has been removed.
[Bug] C++ API: openmpt::exception did not define copy and move constructors or copy and move assignment operators in libopenmpt 0.2. The compiler-generated ones were not adequate though. libopenmpt 0.3 adds the appropriate special member functions. This adds the respective symbol names to the exported ABI, which, depending on the compiler, might or might not have been there in libopenmpt 0.2. The possibly resulting possible ODR violation only affects cases that did crash in the libopenmpt 0.2 API anyway due to memory double-free, and does not cause any further problems in practice for all known platforms and compilers.
[Bug] The C API could crash instead of failing gracefully in out-of-memory situations.
[Bug] The test suite could fail on MacOSX or FreeBSD in non-fatal ways when no locale was active.
[Bug] libopenmpt_stream_callbacks_fd.h and libopenmpt_stream_callbacks_file.h were missing in Windows development packages.
[Bug] libopenmpt on Windows did not properly guard against current working directory DLL injection attacks.
[Bug] localtime() was used to determine the version of Schism Tracker used to save IT and S3M files. This function is not guaranteed to be thread-safe by the standard and is now no longer used.
[Bug] Possible crashes with malformed IT, ITP, AMS, MDL, MED, MPTM, PSM and Startrekker files.
[Bug] Possible hangs with malformed DBM, MPTM and PSM files.
[Bug] Possible hangs with malformed files containing cyclic plugin routings.
[Bug] Excessive loading times with malformed ITP / truncated AMS files.
[Bug] Plugins did not work correctly when changing the sample rate between two render calls.
[Bug] Possible NULL-pointer dereference read during obscure out-of-memory situations while handling exceptions in the C API.
[Bug] libmodplug: libmodplug.pc was wrong.
[Bug] Cross-compiling libopenmpt with autotools for Windows now properly sets -municode and -mconsole as well as all required Windows system libraries.
[Bug] foo_openmpt: Interpolation filter and volume ramping settings were confused in previous versions. This version resets both to the defaults.
[Bug] libmodplug: The CSoundFile::Read function in the emulated libmodplug C++ API returned the wrong value, causing qmmp (and possibly other software) to crash.
Support for SoundTracker Pro II (STP) and Digital Tracker (DTM) modules.
Increased accuracy of the sample position and sample rate to drift less when playing very long samples.
Various playback improvements for IT and XM files.
Channel frequency could wrap around after some excessive portamento / down in some formats since libopenmpt 0.2-beta17.
Playback improvements for S3M files made with Impulse Tracker and Schism Tracker.
ParamEq plugin emulation didn’t do anything at full gain (+15dB).
All standard DMO effects are now also emulated on non-Windows and non-MSVC systems.
Added libopenmpt_stream_callbacks_buffer.h which adds openmpt_stream_callbacks support for in-memory buffers, possibly even only using a truncated prefix view into a bigger file which is useful for probing.
Avoid enabling some ProTracker-specific quirks for MOD files most likely created with ScreamTracker 3.
Tremolo effect only had half the intended strength in MOD files.
Pattern loops ending on the last row a pattern were not executed correctly in S3M files.
Work-around for reading MIDI macros and plugin settings in some malformed IT files written by old UNMO3 versions.
Improve tracker detection in IT format.
Playback fixes for 8-channel MED files
Do not set note volume to 0 on out-of-range offset in XM files.
Better import of some slide commands in SFX files.
Sample 15 in “Crew Generation” by Necros requires short loops at the beginning of the sample to not be ignored. Since we need to ignore them in some (non-ProTracker) modules, we heuristically disable the old loop sanitization behaviour based on the module channel count.
Both normal and percentage offset in PLM files were handled as percentage offset.
MT2 files with instruments that had both sample and plugin assignments were not read correctly.
Some valid FAR files were rejected erroneously.
Support for VBlank timing flag and comment field in PT36 files.
Improved accuracy of vibrato command in DIGI / DBM files.
STM: Add support for “WUZAMOD!” magic bytes and allow some slightly malformed STM files to load which were previously rejected.
Detect whether “hidden” patterns in the order list of SoundTracker modules should be taken into account or not.
Tighten heuristics for rejecting invalid 669, M15, MOD and ICE files and loosen them in other places to allow some valid MOD files to load.
Improvements to seeking: Channel panning was not always updated from instruments / samples when seeking, and out-of-range global volume was not applied correctly in some formats.
seek.sync_samples=1 did not apply PTM reverse offset effect and the volume slide part of combined volume slide + vibrato commands.
If the order list was longer than 256 items and there was a pattern break effect without a position jump on the last pattern of the sequence, it did not jump to the correct restart order.
Makefile has now explicit support for FreeBSD with no special option or configuration required.
openmpt123: Improved section layout in man page.
libmodplug: Added all missing C++ API symbols that are accessible via the public libmodplug header file.
Autotools build system now has options --disable-openmpt123, --disable-tests and --disable-examples which may be desireable when cross-compiling.
Windows binary packages now ship with libmpg123 included.
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Fixes build without Jack.
From PR 52575 by John D. Baker.
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arts was removed from pkgsrc
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arts was removed from pkgsrc
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arts was removed from pkgsrc
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arts was removed from pkgsrc
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arts was removed from pkgsrc.
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hsaudiotag is a pure Python library that lets you read metadata
(bitrate, sample rate, duration and tags) from mp3, mp4, wma, ogg,
flac and aiff files. It can only read tags, not write to them, but
unlike more complete libraries (like Mutagen), it is BSD licensed,
making it suitable for most projects. It is also backed by a nifty
test suite.
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Announced in https://mail-index.netbsd.org/pkgsrc-users/2017/09/10/msg025556.html
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Create the needed makefile fragment for Darwin.
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Release Name: 1.14
====================
Tag.pm: Quiet warnings from 5.22.
Start implementing handlers: instead of FRAM(langs)[opts], one can use, e.g., func_name(ID3v1,Cue)[arg1][arg2]
# ID3v2::_frame_select_by_descr is missing //s
New configuration variable ampersand_joiner (default '; ').
New method _auto_field_from() (abstracted from _auto_field()).
Change logic of calling ->can() in _auto_field_from().
_parse_rex_microinterpolate() would not update $ecount on seeing %%.
When parsing with %=c etc: with %==c the match fails if there is no comment.
(Checked the same way as for %{c:}.)
Support some of %-escapes not being matched (e.g, due to alternatives in a REx).
(0-length matches were ignored anyway [when join()ing].)
parse_rex(), parse_rex_match() may return an extra result (if %{handler}s are present).
Recognize ID3v2 frame names as [A-Z]{3}[A-Z\d] (was \w{4}).
(Detection frame/vs/handler happens via PACKAGES; so if lang codes inf/cue appear, we may be in trouble.)
Use the same code in parse(_rex)?_prepare. (Now parse() allows the same %-constructs as parse_rex().)
ID3v2.pm:
New method have_one_of_frames().
New methods *_have() (for simplest fields: title, comment, track, artist, album, genre, year).
ImageExifTool.pm:
Comprehensive docs.
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openal-soft-1.18.1:
Fixed an issue where resuming a source might not restart playing it.
Fixed PulseAudio playback when the configured stream length is much less
than the requested length.
Fixed MMDevAPI capture with sample rates not matching the backing device.
Fixed int32 output for the Wave Writer.
Fixed enumeration of OSS devices that are missing device files.
Added correct retrieval of the executable's path on FreeBSD.
Added a config option to specify the dithering depth.
Added a 5.1 decoder preset that excludes front-center output.
openal-soft-1.18.0:
Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
Implemented 3D processing for some effects. Currently implemented for
Reverb, Compressor, Equalizer, and Ring Modulator.
Implemented 2-channel UHJ output encoding. This needs to be enabled with a
config option to be used.
Implemented dual-band processing for high-quality ambisonic decoding.
Implemented distance-compensation for surround sound output.
Implemented near-field emulation and compensation with ambisonic rendering.
Currently only applies when using the high-quality ambisonic decoder or
ambisonic output, with appropriate config options.
Implemented an output limiter to reduce the amount of distortion from
clipping.
Implemented dithering for 8-bit and 16-bit output.
Implemented a config option to select a preferred HRTF.
Implemented a run-time check for NEON extensions using /proc/cpuinfo.
Implemented experimental capture support for the OpenSL backend.
Fixed building on compilers with NEON support but don't default to having
NEON enabled.
Fixed support for JACK on Windows.
Fixed starting a source while alcSuspendContext is in effect.
Fixed detection of headsets as headphones, with MMDevAPI.
Added support for AmbDec config files, for custom ambisonic decoder
configurations. Version 3 files only.
Added backend-specific options to alsoft-config.
Added first-, second-, and third-order ambisonic output formats. Currently
only works with backends that don't rely on channel labels, like JACK,
ALSA, and OSS.
Added a build option to embed the default HRTFs into the lib.
Added AmbDec presets to enable high-quality ambisonic decoding.
Added an AmbDec preset for 3D7.1 speaker setups.
Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
the provided ambdec presets.
Added the ability for MMDevAPI to open devices given a Device ID or GUID
string.
Added an option to the example apps to open a specific device.
Increased the maximum auxiliary send limit to 16 (up from 4). Requires
requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
attribute.
Increased the default auxiliary effect slot count to 64 (up from 4).
Reduced the default period count to 3 (down from 4).
Slightly improved automatic naming for enumerated HRTFs.
Improved B-Format decoding with HRTF output.
Improved internal property handling for better batching behavior.
Improved performance of certain filter uses.
Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
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GStreamer is a library that allows the construction of graphs of
media-handling components, ranging from simple Ogg/Vorbis playback to
complex audio (mixing) and video (non-linear editing) processing.
Applications can take advantage of advances in codec and filter technology
transparently. Developers can add new codecs and filters by writing a
simple plugin with a clean, generic interface.
GStreamer is released under the LGPL.
This package is part of the 'ugly' plugins for GStreamer. It provides the
mpg123 plugin, which allows MP3 decoding.
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Highlights
new msdk plugin for Intel's Media SDK for hardware-accelerated
video encoding and decoding on Intel graphics hardware on Windows
or Linux.
x264enc can now use multiple x264 library versions compiled for
different bit depths at runtime, to transparently provide support
for multiple bit depths.
videoscale and videoconvert now support multi-threaded scaling
and conversion, which is particularly useful with higher
resolution video.
h264parse will now automatically insert AU delimiters if needed
when outputting byte-stream format, which improves standard
compliance and is needed in particular for HLS playback on
iOS/macOS.
rtpbin has acquired bundle support for incoming streams
Major new features and changes Noteworthy new API
The video library gained support for a number of new video
formats:
GBR_12LE, GBR_12BE, GBRA_12LE, GBRA_12BE (planar 4:4:4
RGB/RGBA, 12 bits per channel) GBRA_10LE, GBRA_10BE (planar
4:4:4:4 RGBA, 10 bits per channel) GBRA (planar 4:4:4:4
ARGB, 8 bits per channel) I420_12BE, I420_12LE (planar 4:2:0
YUV, 12 bits per channel) I422_12BE,I422_12LE (planar 4:2:2
YUV, 12 bits per channel) Y444_12BE, Y444_12LE (planar 4:4:4
YUV, 12 bits per channel) VYUY (another packed 4:2:2 YUV
format)
The high-level GstPlayer API was extended with functions for
taking video snapshots and enabling accurate seeking. It can
optionally also use the still-experimental playbin3 element
now.
New Elements
msdk: new plugin for Intel's Media SDK for hardware-accelerated
video encoding and decoding on Intel graphics hardware on Windows
or Linux. This includes an H.264 encoder/decoder (msdkh264dec,
msdkh264enc), an H.265 encoder/decoder (msdkh265dec, msdkh265enc),
an MJPEG encoder/encoder (msdkmjpegdec, msdkmjpegenc), an MPEG-2
video encoder (msdkmpeg2enc) and a VP8 encoder (msdkvp8enc).
iqa is a new Image Quality Assessment plugin based on DSSIM,
similar to the old (unported) videomeasure element.
The faceoverlay element, which allows you to overlay SVG graphics
over a detected face in a video stream, has been ported from
0.10.
our ffmpeg wrapper plugin now exposes/maps the ffmpeg Opus audio
decoder (avdec_opus) as well as the GoPro CineForm HD / CFHD
decoder (avdec_cfhd), and also a parser/writer for the IVF
format (avdemux_ivf and avmux_ivf).
audiobuffersplit is a new element that splits raw audio buffers
into equal-sized buffers
audiomixmatrix is a new element that mixes N:M audio channels
according to a configured mix matrix.
The timecodewait element got renamed to avwait and can operate
in different modes now.
The opencv video processing plugin has gained a new dewarp
element that dewarps fisheye images.
ttml is a new plugin for parsing and rendering subtitles in
Timed Text Markup Language (TTML) format. For the time being
these elements will not be autoplugged during media playback
however, unless the GST_TTML_AUTOPLUG=1 environment variable
is set. Only the EBU-TT-D profile is supported at this point.
New element features and additions
x264enc can now use multiple x264 library versions compiled for
different bit depths at runtime, to transparently provide support
for multiple bit depths. A new configure parameter
--with-x264-libraries has been added to specify additional paths
to look for additional x264 libraries to load. Background is
that the libx264 library is always compile for one specific bit
depth and the x264enc element would simply support the depth
supported by the underlying library. Now we can support multiple
depths.
x264enc also picks up the interlacing mode automatically from
the input caps now and passed interlacing/TFF information
correctly to the library.
videoscale and videoconvert now support multi-threaded scaling
and conversion, which is particularly useful with higher
resolution video. This has to be enabled explicitly via the
"n-threads" property.
videorate's new "rate" property lets you set a speed factor on
the output stream
splitmuxsink's buffer collection and scheduling was rewritten
to make processing and splitting deterministic; before it was
possible for a buffer to end up in a different file chunk in
different runs. splitmuxsink also gained a new "format-location-full"
signal that works just like the existing "format-location"
signal only that it is also passed the primary stream's first
buffer as argument, so that it is possible to construct the
file name based on metadata such as the buffer timestamp or any
GstMeta attached to the buffer. The new "max-size-timecode"
property allows for timecode-based splitting. splitmuxsink will
now also automatically start a new file if the input caps change
in an incompatible way.
fakesink has a new "drop-out-of-segment" property to not drop
out-of-segment buffers, which is useful for debugging purposes.
identity gained a "ts-offset" property.
both fakesink and identity now also print what kind of metas
are attached to buffers when printing buffer details via the
"last-message" property used by gst-launch-1.0 -v.
multiqueue: made "min-interleave-time" a configurable property.
video nerds will be thrilled to know that videotestsrc's snow
is now deterministic. videotestsrc also gained some new properties
to make the ball pattern based on system time, and invert colours
each second ("animation-mode", "motion", and "flip" properties).
oggdemux reverse playback should work again now. You're welcome.
playbin3 and urisourcebin now have buffering enabled by default,
and buffering message aggregation was fixed.
tcpclientsrc now has a "timeout" property
appsink has gained support for buffer lists. For backwards
compatibility reasons users need to enable this explicitly with
gst_app_sink_set_buffer_list_support(), however. Once activated,
a pulled GstSample can contain either a buffer list or a single
buffer.
splitmuxsrc reverse playback was fixed and handling of sparse
streams, such as subtitle tracks or metadata tracks, was improved.
matroskamux has acquired support for muxing G722 audio; it also
marks all buffers as keyframes now when streaming only audio,
so that tcpserversink will behave properly with audio-only
streams.
qtmux gained support for ProRes 4444 XQ, HEVC/H.265 and CineForm
(GoPro) formats, and generally writes more video stream-related
metadata into the track headers. It is also allows configuration
of the maximum interleave size in bytes and time now. For
fragmented mp4 we always write the tfdt atom now as required
by the DASH spec.
qtdemux supports FLAC, xvid, mp2, S16L and CineForm (GoPro)
tracks now, and generally tries harder to extract more video-related
information from track headers, such as colorimetry or interlacing
details. It also received a couple of fixes for the scenario
where upstream operates in TIME format and feeds chunks to
qtdemux (e.g. DASH or MSE).
audioecho has two new properties to apply a delay only to certain
channels to create a surround effect, rather than an echo on
all channels. This is useful when upmixing from stereo, for
example. The "surround-delay" property enables this, and the
"surround-mask" property controls which channels are considered
surround sound channels in this case.
webrtcdsp gained various new properties for gain control and
also exposes voice activity detection now, in which case it
will post "voice-activity" messages on the bus whenever the
voice detection status changes.
The decklink capture elements for Blackmagic Decklink cards
have seen a number of improvements:
decklinkvideosrc will post a warning message on "no signal"
and an info message when the signal lock has been (re)acquired.
There is also a new read-only "signal" property that can
be used to query the signal lock status. The GAP flag will
be set on buffers that are captured without a signal lock.
The new drop-no-signal-frames will make decklinkvideosrc
drop all buffers that have been captured without an input
signal. The "skip-first-time" property will make the source
drop the first few buffers, which is handy since some devices
will at first output buffers with the wrong resolution
before they manage to figure out the right input format and
decide on the actual output caps.
decklinkaudiosrc supports more than just 2 audio channels
now.
The capture sources no longer use the "hardware" timestamps
which turn out to be useless and instead just use the
pipeline clock directly.
srtpdec now also has a readonly "stats" property, just like
srtpenc.
rtpbin gained RTP bundle support, as used by e.g. WebRTC. The
first rtpsession will have a rtpssrcdemux element inside splitting
the streams based on their SSRC and potentially dispatch to a
different rtpsession. Because retransmission SSRCs need to be
merged with the corresponding media stream the ::on-bundled-ssrc
signal is emitted on rtpbin so that the application can find
out to which session the SSRC belongs.
rtprtxqueue gained two new properties exposing retransmission
statistics ("requests" and "fulfilled-requests")
kmssink will now use the preferred mode for the monitor and
render to the base plane if nothing else has set a mode yet.
This can also be done forcibly in any case via the new
"force-modesetting" property. Furthermore, kmssink now allows
only the supported connector resolutions as input caps in order
to avoid scaling or positioning of the input stream, as kmssink
can't know whether scaling or positioning would be more appropriate
for the use case at hand.
waylandsink can now take DMAbuf buffers as input in the presence
of a compatible Wayland compositor. This enables zero-copy
transfer from a decoder or source that outputs DMAbuf. It will
also set surface opacity hint to allow better rendering
optimization in the compositor.
udpsrc can be bound to more than one interface when joining a
multicast group, this is done by giving a comma separate list
of interfaces such as multicast-iface="eth0,eth1".
Plugin moves
dataurisrc moved from gst-plugins-bad to core
The rawparse plugin containing the rawaudioparse and rawvideoparse
elements moved from gst-plugins-bad to gst-plugins-base. These
elements supersede the old videoparse and audioparse elements.
They work the same, with just some minor API changes. The old
legacy elements still exist in gst-plugins-bad, but may be
removed at some point in the future.
timecodestamper is an element that attaches time codes to video
buffers in form of GstVideoTimeCodeMetas. It had a "clock-source"
property which has now been removed because it was fairly useless
in practice. It gained some new properties however: the
"first-timecode" property can be used to set the inital timecode;
alternatively "first-timecode-to-now" can be set, and then the
current system time at the time the first buffer arrives is
used as base time for the time codes.
Plugin removals
The mad mp1/mp2/mp3 decoder plugin was removed from gst-plugins-ugly,
as libmad is GPL licensed, has been unmaintained for a very
long time, and there are better alternatives available. Use the
mpg123audiodec element from the mpg123 plugin in gst-plugins-ugly
instead, or avdec_mp3 from the gst-libav module which wraps the
ffmpeg library. We expect that we will be able to move mp3
decoding to gst-plugins-good in the next cycle seeing that most
patents around mp3 have expired recently or are about to expire.
The mimic plugin was removed from gst-plugins-bad. It contained
a decoder and encoder for a video codec used by MSN messenger
many many years ago (in a galaxy far far away). The underlying
library is unmaintained and no one really needs to use this
codec any more. Recorded videos can still be played back with
the MIMIC decoder in gst-libav.
Miscellaneous API additions
Request pad name templates passed to gst_element_request_pad()
may now contain multiple specifiers, such as e.g. src_%u_%u.
gst_buffer_iterate_meta_filtered() is a variant of
gst_buffer_iterate_meta() that only returns metas of the requested
type and skips all other metas.
gst_pad_task_get_state() gets the current state of a task in a
thread-safe way.
gst_uri_get_media_fragment_table() provides the media fragments
of an URI as a table of key=value pairs.
gst_print(), gst_println(), gst_printerr(), and gst_printerrln()
can be used to print to stdout or stderr. These functions are
similar to g_print() and g_printerr() but they also support all
the additional format specifiers provided by the GStreamer
logging system, such as e.g. GST_PTR_FORMAT.
a GstParamSpecArray has been added, for elements who want to
have array type properties, such as the audiomixmatrix element
for example. There are also two new functions to set and get
properties of this type from bindings:
gst_util_set_object_array() gst_util_get_object_array()
various helper functions have been added to make it easier to
set or get GstStructure fields containing caps-style array or
list fields from language bindings (which usually support
GValueArray but don't know about the GStreamer specific fundamental
types):
gst_structure_get_array() gst_structure_set_array()
gst_structure_get_list() gst_structure_set_list()
a new 'dynamic type' registry factory type was added to register
dynamically loadable GType types. This is useful for automatically
loading enum/flags types that are used in caps, such as for
example the GstVideoMultiviewFlagsSet type used in multiview
video caps.
there is a new GstProxyControlBinding for use with GstController.
This allows proxying the control interface from one property
on one GstObject to another property (of the same type) in
another GstObject. So e.g. in parent-child relationship, one
may need to call gst_object_sync_values() on the child and have
a binding (set elsewhere) on the parent update the value. This
is used in glvideomixer and glsinkbin for example, where
sync_values() on the child pad or element will call sync_values()
on the exposed bin pad or element.
Note that this doesn't solve GObject property forwarding, that
must be taken care of by the implementation manually or using
GBinding.
gst_base_parse_drain() has been made public for subclasses to
use.
`gst_base_sink_set_drop_out_of_segment()' can be used by
subclasses to prevent GstBaseSink from dropping buffers that
fall outside of the segment.
gst_calculate_linear_regression() is a new utility function to
calculate a linear regression.
gst_debug_get_stack_trace is an easy way to retrieve a stack
trace, which can be useful in tracer plugins.
allocators: the dmabuf allocator is now sub-classable, and there
is a new GST_CAPS_FEATURE_MEMORY_DMABUF define.
video decoder subclasses can use the newly-added function
gst_video_decoder_allocate_output_frame_with_params() to pass
a GstBufferPoolAcquireParams to the buffer pool for each buffer
allocation.
the video time code API has gained a dedicated GstVideoTimeCodeInterval
type plus related API, including functions to add intervals to
timecodes.
There is a new libgstbadallocators-1.0 library in gst-plugins-bad,
which may go away again in future releases once the
GstPhysMemoryAllocator interface API has been validated by more
users and was moved to libgstallocators-1.0 from gst-plugins-base.
GstPlayer
New API has been added to:
get the number of audio/video/subtitle streams:
gst_player_media_info_get_number_of_streams()
gst_player_media_info_get_number_of_video_streams()
gst_player_media_info_get_number_of_audio_streams()
gst_player_media_info_get_number_of_subtitle_streams()
enable accurate seeking: gst_player_config_set_seek_accurate()
and gst_player_config_get_seek_accurate()
get a snapshot image of the video in RGBx, BGRx, JPEG, PNG or
native format: gst_player_get_video_snapshot()
selecting use of a specific video sink element
(gst_player_video_overlay_video_renderer_new_with_sink())
If the environment variable GST_PLAYER_USE_PLAYBIN3 is set,
GstPlayer will use the still-experimental playbin3 element and
the GstStreams API for playback.
Miscellaneous changes
video caps for interlaced video may contain an optional
"field-order" field now in the case of interlaced-mode=interleaved
to signal that the field order is always the same throughout
the stream. This is useful to signal to muxers such as mp4mux.
The new field is parsed from/to GstVideoInfo of course.
video decoder and video encoder base classes try harder to proxy
interlacing, colorimetry and chroma-site related fields in caps
properly.
The buffer stored in the PROTECTION events is now left unchanged.
This is a change of behaviour since 1.8, especially for the
mssdemux element which used to decode the base64 parsed data
wrapped in the protection events emitted by the demuxer.
PROTECTION events can now be injected into the pipeline from
the application; source elements deriving from GstBaseSrc will
forward those downstream now.
The DASH demuxer is now correctly parsing the MSPR-2.0
ContentProtection nodes and emits Protection events accordingly.
Applications relying on those events might need to decode the
base64 data stored in the event buffer before using it.
The registry can now also be disabled by setting the environment
variable GST_REGISTRY_DISABLE=yes, with similar effect as the
GST_DISABLE_REGISTRY compile time switch.
Seeking performance with gstreamer-vaapi based decoders was
improved. It would recreate the decoder and surfaces on every
seek which can be quite slow.
more robust handling of input caps changes in videoaggregator-based
elements such as compositor.
Lots of adaptive streaming-related fixes across the board (DASH,
MSS, HLS). Also:
mssdemux, the Microsoft Smooth Streaming demuxer, has seen
various fixes for live streams, duration reporting and
seeking.
The DASH manifest parser now extracts MS PlayReady
ContentProtection objects from manifests and sends them
downstream as PROTECTION events. It also supports multiple
Period elements in external xml now.
gst-libav was updated to ffmpeg 3.3 but should still work with
any 3.x version.
GstEncodingProfile has been generally enhanced so it can, for
example, be used to get possible profiles for a given file
extension. It is now possible to define profiles based on element
factory names or using a path to a .gep file containing a
serialized profile.
audioconvert can now do endianness conversion in-place. All
other conversions still require a copy, but e.g. sign conversion
and a few others could also be implemented in-place now.
The new, experimental playbin3 and urisourcebin elements got
many bugfixes and improvements and should generally be closer
to a full replacement of the old elements.
interleave now supports > 64 channels.
OpenCV elements, grabcut and retinex has been ported to use
GstOpencvVideoFilter base class, increasing code reuse and
fixing buffer map/unmap issues. Redundant copie of images has
been removed in edgedetect, cvlaplace and cvsobel. This comes
with various cleanup and Meson support.
OpenGL integration
As usual the GStreamer OpenGL integration library has seen
numerous fixes and performance improvements all over the place,
and is hopefully ready now to become API stable and be moved
to gst-plugins-base during the 1.14 release cycle.
The GStreamer OpenGL integration layer has also gained support
for the Vivante EGL FB windowing system, which improves performance
on platforms such as Freescale iMX.6 for those who are stuck
with the proprietary driver. The qmlglsink element also supports
this now if Qt is used with eglfs or wayland backend, and it
works in conjunction with gstreamer-imx of course.
various qmlglsrc improvements
Tracing framework and debugging improvements
New tracing hooks have been added to track GstMiniObject and
GstObject ref/unref operations.
The memory leaks tracer can optionally use this to retrieve
stack traces if enabled with e.g.
GST_TRACERS=leaks(filters="GstEvent,GstMessage",stack-traces-flags=full)
The GST_DEBUG_FILE environment variable, which can be used to
write the debug log output to a file instead of printing it to
stderr, can now contain a name pattern, which is useful for
automated testing and continuous integration systems. The
following format specifiers are supported:
%p: will be replaced with the PID %r: will be replaced with
a random number, which is useful for instance when running
two processes with the same PID but in different containers.
Tools
gst-inspect-1.0 can now list elements by type with the new
--types command-line option, e.g. gst-inspect-1.0 --types=Audio/Encoder
will show a list of audio encoders.
gst-launch-1.0 and gst_parse_launch() have gained a new operator
(:) that allows linking all pads between two elements. This is
useful in cases where the exact number of pads or type of pads
is not known beforehand, such as in the uridecodebin : encodebin
scenario, for example. In this case, multiple links will be
created if the encodebin has multiple profiles compatible with
the output of uridecodebin.
gst-device-monitor-1.0 now shows a gst-launch-1.0 snippet for
each device that shows how to make use of it in a gst-launch-1.0
pipeline string.
GStreamer RTSP server
The RTSP server now also supports Digest authentication in
addition to Basic authentication.
The GstRTSPClient class has gained a pre-*-request signal and
virtual method for each client request type, emitted in the
beginning of each rtsp request. These signals or virtual methods
let the application validate the requests, configure the
media/stream in a certain way and also generate error status
codes in case of an error or a bad request.
GStreamer VAAPI
GstVaapiDisplay now inherits from GstObject, thus the VA display
logging messages are better and tracing the context sharing is
more readable.
When uploading raw images into a VA surfaces now VADeriveImages
are tried fist, improving the upload performance, if it is
possible.
The decoders and the post-processor now can push dmabuf-based
buffers to downstream under certain conditions. For example:
GST_GL_PLATFORM=egl gst-play-1.0 video-sample.mkv
--videosink=glimagesink
Refactored the wrapping of VA surface into gstreamer memory,
adding lock when mapping and unmapping, and many other fixes.
Now vaapidecodebin loads vaapipostproc dynamically. It is
possible to avoid it usage with the environment variable
GST_VAAPI_DISABLE_VPP=1.
Regarding encoders: they have primary rank again, since they
can discover, in run-time, the color formats they can use for
upstream raw buffers and caps renegotiation is now possible.
Also the encoders push encoding info downstream via tags.
About specific encoders: added constant bit-rate encoding mode
for VP8 and H265 encoder handles P010_10LE color format.
Regarding decoders, flush operation has been improved, now the
internal VA encoder is not recreated at each flush. Also there
are several improvements in the handling of H264 and H265
streams.
VAAPI plugins try to create their on GstGL context (when
available) if they cannot find it in the pipeline, to figure
out what type of VA Display they should create.
Regarding vaapisink for X11, if the backend reports that it is
unable to render correctly the current color format, an internal
VA post-processor, is instantiated (if available) and converts
the color format.
GStreamer Editing Services and NLE
Enhanced auto transition behaviour
Fix some races in nlecomposition
Allow building with msvc
Added a UNIX manpage for ges-launch
API changes:
Added ges_deinit (allowing the leak tracer to work properly)
Added ges_layer_get_clips_in_interval Finally hide internal
symbols that should never have been exposed
GStreamer validate
Port gst-validate-launcher to python 3
gst-validate-launcher now checks if blacklisted bugs have been
fixed on bugzilla and errors out if it is the case
Allow building with msvc
Add ability for the launcher to run GStreamer unit tests
Added a way to activate the leaks tracer on our tests and fix
leaks
Make the http server multithreaded
New testsuite for running various test scenarios on the DASH-IF
test vectors
GStreamer Python Bindings
Overrides has been added for IntRange, Int64Range, DoubleRange,
FractionRange, Array and List. This finally enables Python
programmers to fully read and write GstCaps objects.
Build and Dependencies
Meson build files are now disted in tarballs, for jhbuild and
so distro packagers can start using it. Note that the Meson-based
build system is not 100% feature-equivalent with the autotools-based
one yet.
Some plugin filenames have been changed to match the plugin
names: for example the file name of the encoding plugin in
gst-plugins-base containing the encodebin element was
libgstencodebin.so and has been changed to libgstencoding.so.
This affects only a handful of plugins across modules.
Developers who install GStreamer from source and just do make
install after updating the source code, without doing make
uninstall first, will have to manually remove the old installed
plugin files from the installation prefix, or they will get
'Cannot register existing type' critical warnings.
Most of the docbook-based documentation (FAQ, Application
Development Manual, Plugin Writer's Guide, design documents)
has been converted to markdown and moved into a new gst-docs
module. The gtk-doc library API references and the plugins
documentation are still built as part of the source modules
though.
GStreamer core now optionally uses libunwind and libdw to
generate backtraces. This is useful for tracer plugins used
during debugging and development.
There is a new libgstbadallocators-1.0 library in gst-plugins-bad
(which may go away again in future releases once the
GstPhysMemoryAllocator interface API has been validated by more
users).
gst-omx and gstreamer-vaapi modules can now also be built using
the Meson build system.
The qtkitvideosrc element for macOS was removed. The API is
deprecated since 10.9 and it wasn't shipped in the binaries
since a few releases.
Platform-specific improvements Android
androidmedia: add support for VP9 video decoding/encoding and
Opus audio decoding (where supported)
OS/X and iOS
avfvideosrc, which represents an iPhone camera or, on a Mac, a
screencapture session, so far allowed you to select an input
device by device index only. New API adds the ability to select
the position (front or back facing) and device-type (wide angle,
telephoto, etc.). Furthermore, you can now also specify the
orientation (portrait, landscape, etc.) of the videostream.
Bugs fixed in 1.12
More than 635 bugs have been fixed during the development of 1.12.
This list does not include issues that have been cherry-picked into
the stable 1.10 branch and fixed there as well, all fixes that ended
up in the 1.10 branch are also included in 1.12.
This list also does not include issues that have been fixed without
a bug report in bugzilla, so the actual number of fixes is much
higher. Stable 1.12 branch
After the 1.12.0 release there will be several 1.12.x bug-fix
releases which will contain bug fixes which have been deemed suitable
for a stable branch, but no new features or intrusive changes will
be added to a bug-fix release usually. The 1.12.x bug-fix releases
will be made from the git 1.12 branch, which is a stable branch.
1.12.0
1.12.0 was released on 4th May 2017.
1.12.1
The first 1.12 bug-fix release (1.12.1) was released on 20 June
2017. This release only contains bugfixes and it should be safe to
update from 1.12.x. Major bugfixes in 1.12.1
Various fixes for crashes, assertions, deadlocks and memory
leaks Fix for regression when seeking to the end of ASF files
Fix for regression in (raw)videoparse that caused it to omit
video metadata Fix for regression in discoverer that made it
show more streams than actually available Numerous bugfixes to
the adaptive demuxer base class and the DASH demuxer Various
playbin3/urisourcebin related bugfixes Vivante DirectVIV (imx6)
texture uploader works with single-plane (e.g. RGB) video formats
now Intel Media SDK encoder now outputs valid PTS and keyframe
flags OpenJPEG2000 plugin can be loaded again on MacOS and
correctly displays 8 bit RGB images now Fixes to DirectSound
source/sink for high CPU usage and wrong latency/buffer size
calculations gst-libav was updated to ffmpeg n3.3.2 ... and
many, many more!
1.12.2
The second 1.12 bug-fix release (1.12.2) was released on 14 July
2017. This release only contains bugfixes and it should be safe to
update from 1.12.x. Major bugfixes in 1.12.2
Various fixes for crashes, assertions, deadlocks and memory
leaks Regression fix for playback of live HLS streams Regression
fix for crash when playing back a tunneled RTSP stream Regression
fix for playback of RLE animations in MOV containers Regression
fix for RTP GSM payloading producing corrupted output Major
bugfixes to the MXF demuxer, mostly related to seeking and fixes
to the frame reordering handling in the MXF muxer and demuxer
Fix for playback of mono streams on MacOS More fixes for index
handling of ASF containers Various fixes to adaptivedemux, DASH
and HLS demuxers Fix deadlock in gstreamer-editing-services
during class initialization ... and many, many more!
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1.25.5
------
- Avoid another buffer read overflow in the ID3 parser on 32 bit platforms
(bug 254).
1.25.4
------
- Better configure checks for i?86-apple-darwin (bug 253).
- libmpg123:
-- Prevent harmless call to memcpy(NULL, NULL, 0).
-- More early checking of ID3v2 encoding values to avoid bogus text being
stored.
1.25.3
------
- libmpg123:
-- Better checks for xrpnt overflow in III_dequantize_sample() before each
use, avoiding false positives and catching cases that were rendered
harmless by alignment-enlarged buffers.
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Changelog:
PulseAudio 11.0 release notes
Changes at a glance
Support for newer AirPlay hardware
USB and bluetooth devices preferred over internal sound cards
The default sink and source configuration is remembered better
Bluetooth HSP headset role implemented
Bluetooth HFP audio gateway role implemented (requires oFono)
Bluetooth HSP audio gateway and HFP hands-free unit roles can be enabled simultaneously
Upmixing can now be disabled without bad side effects
Avoid having unavailable sinks or sources as the default
Option to avoid resampling more often
Option to automatically switch bluetooth profile to HSP more often
Better latency regulation in module-loopback
Changed module argument names in module-ladspa-sink and module-virtual-surround-sink
Fixed input device handling on Windows
Improved bluetooth MTU configuration (warning! this causes some hardware to not work any more, see the details below for how to fix it)
GNU Hurd support
Applications can request LADSPA or virtual surround filtering for their streams
Support for 32-bit applications on 64-bit systems in padsp
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