GTK-Doc specifies that, by default, the caller owns returned objects, so that the caller should free it when it is done. However, in the case of this function, the returned GstAudioInfo is owned by the decoder, so this default choice is incorrect. This creates double free problems when using GStreamer Rust bindings, because they are generated using the information contained in the docs.
Fix this by correctly specifying that the caller does not own the returned object.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1032>
Currently max-errors gets set during init to default or via property.
However, if a decoder element calls gst_audio_decoder_reset with 'full'
argument set to TRUE, it would result in all the fields of context being
zeroed with memset. This effectively results in max-errors getting a
value of 0 overriding the default or user requested value set during
init.
This would result in calls to GST_AUDIO_DECODER_ERROR which track error
counts and allow max-errors, to be ineffective.
To fix this move max-errors out of GstAudioDecoderContext, as changes to
context should not affect this. The error_count is anyways also in
GstAudioDecoderPrivate and not in context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/946>
For audio we copy metas that have no tags at all, or that only have the
"audio" and/or "audio-channels" tag. Audio codecs don't change the
audio aspect of the stream and in almost all cases don't change the
number of channels. They might however change the sample rate (e.g.
Opus). Subclasses that change the number of channels will have to
override ::transform_meta() accordingly.
For video we copy metas that have no tags at all, or that only have the
"video" and/or "video-size" and/or "video-orientation" tag. Video codecs
don't change the "video" aspect of the stream and in almost all cases
don't change the resolution or orientation. Subclasses that rescale or
change the orientation will have to override ::transform_meta()
accordingly.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/576#note_610581
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/801>
When receiving an instant-rate-change event, store the updated
seek flags and replace the flags in any input segments with them
to allow for instant switching between trickmodes and not.
Matroskademux will send gap event when lag of video and audio is over 3 seconds.
audiodecoder needs to handle gap event and set default output caps.
Only audio info is set, while output caps is ignored. This cause the assertion failed.
Need to fill output caps in gst_audio_decoder_negotiate_default_caps() with
negotiated caps to avoid critical info printed when check it later.
This allows us to output audio samples without discarding
any input frames, which is useful for some formats/codecs
(e.g. the MonkeysAudio decoder implementation in ffmpeg
which will might return e.g. 16 output buffers for an
input buffer for certain files).
In the past decoder implementations just concatenated
the returned audio buffers until a full frame had been
decoded, but that's no longer possible to do efficiently
when the decoder returns audio samples in non-interleaved
layout.
Allowing subframes to be output before the entire input
frame is decoded can also be useful to decrease startup
latency/delay.
https://gitlab.freedesktop.org/gstreamer/gst-libav/issues/49
This patch adds API in the audio decoder base class for setting the arbitrary
caps on the source pad. Previously only caps converted from audio info were
possible. This is particularly useful when subclass wants to set caps features
for audio decoder producing metadata.
This is useful if the output buffers are planar and have extra padding
on each plane, in which case size/bpf does not represent the number of
valid samples.
https://bugzilla.gnome.org/show_bug.cgi?id=705977
Aggregation will break the layout, as it concatenates buffers,
and fixing it here would be much more inefficient than configuring
the actual decoder implementation to output larger buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=705977
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
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.
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
gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
In that case, use gst_pad_peer_query_caps() with template caps as filter
to have negotiated output caps properly before forwarding GAP event.
https://bugzilla.gnome.org/show_bug.cgi?id=761218
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.
API: gst_audio_channel_get_fallback_mask()
Before we just merged everything in pretty much random ways
ad-hoc instead of keeping state properly. In 0.10 that was
how it worked, but in 1.x the tag events sent should always
reflect the latest state and replace any previous tags.
So save the upstream (stream) tags, and save the tags set
by the decoder subclass with merge mode, and then update
the merged tags whenever either of those two changes.
This slightly changes the behaviour of gst_audio_decoder_merge_tags()
in case it is called multiple times, since now any call replaces
the previously-set tags. However, it leads to much more predictable
outcomes, and also we are not aware of any subclass which sets this
multiple times and expects all the tags set to be merged.
If more complex tag merging scenarios are required, we'll have
to add a new vfunc for that or the subclass has to intercept
the upstream tags itself and send merged tags itself.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer
This reverts commit 8a64592481.
Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary
API: gst_audio_decoder_set_use_default_pad_acceptcaps
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.
API: gst_audio_decoder_sink_query_default
https://bugzilla.gnome.org/show_bug.cgi?id=753623
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
