The external time should be moved only as much as needed
to get back to the ideal center point, so that the clock
is still allowed to drift both directions after the correction.
This reduces excessive back and forth corrections that were
caused by the assumption of a linear drift.
https://bugzilla.gnome.org/show_bug.cgi?id=788006
Otherwise subclasses might accidentially use the old audioinfo/caps.
None of the subclasses currently uses the audioinfo/caps, but future
subclasses might.
https://bugzilla.gnome.org/show_bug.cgi?id=795827
In the situation described in
https://bugzilla.gnome.org/show_bug.cgi?id=795397,
downstream_caps consists of two structures, the first with
the preferred rate, if at all possible (44100), the second
containing the full range of allowed rates, as audioresample
correctly tries to negotiate passthrough caps.
As audioaggregator cannot perform rate conversion, it wants
to return a fixated rate in its getcaps implementation,
however it previously directly used the first structure in
the caps allowed downstream, without taking the filter into
consideration, to determine the rate to fixate to.
With this, we first intersect our downstream caps with the
filter, in order not to fixate to an unsupported rate.
We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
When outputting more than two channels, a channel-mask has to be
specified in the output caps.
We follow the same heuristic as other cases, when downstream
does not specify a channel-mask, we use that of the first
configured pad, and if there was none we generate a fallback
mask.
https://bugzilla.gnome.org/show_bug.cgi?id=794257
Don't reuse the offset variables will contain a sample offset for an
intermediate time value. Instead add a segment_pos variable of type
GstClockTime for this. Use The clock-time macros to check if we got
a valid time.
When an empty mix matrix is passed, audio-channel-mixer
will now generate a (potentially truncated) identity matrix,
this replicates the behaviour of audiomixmatrix in first-channels
mode.
https://bugzilla.gnome.org/show_bug.cgi?id=788833
Acording to the logic this cannot happen (we already check this before). So
add a assert like we do above and remove the check. This make it clearer that
we check for the offset range.
Also remove a dead assignment since we reassign this a few lines below.
This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.
https://bugzilla.gnome.org/show_bug.cgi?id=787560
+ Refactor previous constructor to call on that new constructor
+ Reimplement is_passthrough to strictly check whether the matrix
is an identity matrix, comparing channel-masks was incorrect:
the mixer may be remixing from a list of positions to the same
list of positions, but ordered differently, and reciprocally,
the mixer may be remixing from a list of positions to another
list of positions identically ordered
+ Remove unused tmp field, must have been a refactoring leftover
https://bugzilla.gnome.org/show_bug.cgi?id=785471
Only adjusting the base_ts might lead to a negative ts and as such integer
overflow into a huge timestamp which then propagates into the granulepos
and so on. Instead, resync to incoming buffer timestamp using both base_ts
and sample count rather than only base_ts.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=785948
This is now needed as GstClock does not do that internally anymore,
because that broke bindings.
And mark the function correctly as (transfer full), which it already was
before.
https://bugzilla.gnome.org/show_bug.cgi?id=743062
Optimize LE<->BE conversion by adding a dedicated fast path instead of
using the generic converter. Implement transform_ip function in order to do the
endian swap in place.
This saves buffer allocation for the intermediate format, can be done in place
and also performs the conversion in one step instead of unpack-convert-pack.
For all bit widths the naive algorithm is implemented, which provides the best
performance when compiled with -O3. ORC was considered but eventually removed
as it requires a dedicated function for in-place conversion (due to the
"restrict" parameters).
A more complex algorithm for the 24-bit conversion with unrolled loop and
32-bit processing is implemented in the #if 0 section. It performs better if
compiled with -O2. With -O3 however the naive algorithm performs better.
https://bugzilla.gnome.org/show_bug.cgi?id=773073
It is not needed to store a pointer to every single chain element to free it.
Instead walk the channel list backwards and free the chain elements one by one.
Rename GstAudioConverter->chain_pack to chain_end.
https://bugzilla.gnome.org/show_bug.cgi?id=773073
Refuse to answer BYTES queries ourselves. The only
time they make sense is on raw elementary streams,
in which case upstream would already have answered.
They especially don't make sense for encoders to answer
based on upstream values - although perhaps later
we could make it do TIME->BYTES conversion on the source
pad based on bitrate.
https://bugzilla.gnome.org/show_bug.cgi?id=757631
gst_audio_buffer_reorder_channels() was always mapping the buffer read-write
regardless whether any reordering was needed. If the from and to channel order
is identical return immediately without remapping the buffer.
Add a small helper function gst_audio_channel_positions_equal() which is used
in both gst_audio_reorder_channels() and gst_audio_buffer_reorder_channels().
https://bugzilla.gnome.org/show_bug.cgi?id=773833
All the GstAudioClock method declarations required object of GstClock type
as a first argument, but in fact, required GstAudioClock object (runtime
check in function body). Instead of checking type in run-time, we can
change functions declaration, to accept only GstAudioClock methods. Then,
runtime check is not necessary anymore, since always GstAudioClock object
is passed to a function.
https://bugzilla.gnome.org/show_bug.cgi?id=756628
Seen on the Jenkins CI:
FAILED: subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o
ccache cc '-Isubprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta' '-fdiagnostics-color=always' '-I../subprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/.' '-I../subprojects/gst-plugins-base/.' '-Isubprojects/gst-plugins-base/gst-libs' '-I../subprojects/gst-plugins-base/gst-libs' '-Isubprojects/gstreamer/libs' '-I../subprojects/gstreamer/libs' '-Isubprojects/gstreamer/.' '-I../subprojects/gstreamer/.' '-pipe' '-Wall' '-Winvalid-pch' '-DHAVE_CONFIG_H' '-msse4.1' '-fPIC' '-O0' '-g' '-fPIC' '-I/usr/include/glib-2.0' '-I/usr/lib/glib-2.0/include' '-pthread' '-Isubprojects/gstreamer/gst' '-MMD' '-MQ' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' '-MF' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o.d' -o 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' -c ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c
In file included from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h:24:0,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-private.h:23,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-macros.h:25,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.h:23,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c:24:
../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
#include <gst/audio/audio-enumtypes.h>
^
compilation terminated.
This makes sure that we only build files that need explicit SIMD support
with the relevant CFLAGS. This allows the rest of the code to be built
without, and specific SSE* code is only called after runtime checks for
CPU features.
https://bugzilla.gnome.org/show_bug.cgi?id=729276
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
Elements inherited from GstAudioDecoder, supporting PLC and introducing
delay produce invalid timestamps. Good example is opusdec with in-band FEC
enabled. After receiving GAP event it delays the audio concealment until
the next buffer arrives. The next buffer will have DISCONT flag set which
will make GstAudioDecoder to reset it's internal state, thus forgetting
the timestamp of GAP event. As a result the concealed audio will have the
timestamp of the next buffer (with DISCONT flag) but not the timestamp
from the event.
As said in its doc GST_AUDIO_CHANNEL_POSITION_NONE is meant to be used
for "position-less channels, e.g. from a sound card that records 1024
channels; mutually exclusive with any other channel position".
But at the moment using such positions would raise a
'g_return_if_reached' warning as gst_audio_get_channel_reorder_map()
would reject it.
Fix this by preventing any attempt to reorder in such case as that's not
what we want anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=763799
We currently don't log much about channel positions making debugging
harder as it should be. This is the first step in my attempt to improve
this.
https://bugzilla.gnome.org/show_bug.cgi?id=763985
There is a small window of time where the audio ringbuffer thread
can access the parent thread variable, before it's initialized
by the parent thread. The patch replaces this variable use by
g_thread_self().
https://bugzilla.gnome.org/show_bug.cgi?id=764865
Since the allocation query caps contains memory size and the pad's caps
contains the display size, an audio encoder or decoder might need to allocate
a different buffer size than the size negotiated in the caps.
This patch splits this logic distinction for audiodecoder and audioencoder.
Thus the user, if needs a different allocation caps, should set it through
gst_audio_{encoder,decoder}_set_allocation_cap() before calling the negotiate()
vmethod. Otherwise the allocation_caps will be the same as the caps in the
src pad.
https://bugzilla.gnome.org/show_bug.cgi?id=764421
