FEC may only be used when PLC is enabled on the audio decoder,
as it relies on empty buffers to generate audio from the next
buffer. Hooking to the gap events doesn't work as the audio
decoder does not like more buffers output than it sends.
The length of data to generate using FEC from the next packet
is determined by rounding the gap duration to nearest. This
ensures that duration imprecision does not cause quantization
to 2.5 milliseconds less than available. Doing so causes the
Opus API to fail decoding. Such duration imprecision is common
in live cases.
The buffer to consider when determining the length of audio
to be decoded is the previous buffer when using FEC, and the
new buffer otherwise. In the FEC case, this means we determine
the amount of audio from the previous buffer, whether it was
missing or not (and get the data either from this buffer, or
the current one if the previous one was missing).
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
Add missing ':' to tile_ws and tile_hs fields documentation to avoid
bad render of these two fields, mark reserved bytes as private to hide
field and avoid gtkdoc warning and add parameters description to
documented macro to avoid gtkdoc warnings.
https://bugzilla.gnome.org/show_bug.cgi?id=761132
Reduce resolution, which shouldn't make any difference
to what's tested here. Makes test finish in less than
half the time it took before (8s vs. 21s).
analyze_new_pad() can return a new decode chain, which might have a new
GstDecodePad in the end. We should use those two for expose_pad() and not the
original ones that were passed to analyze_new_pad().
This fails when having a demuxer element that has raw pads immediately or
if a decoder with raw caps is after an adaptive demuxer.
https://bugzilla.gnome.org/show_bug.cgi?id=760949
In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
RGB formats and warn when the GstVideoInfo colorimetry is wrong.
In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
for RGB formats and warn about inconsistent caps.
See https://bugzilla.gnome.org/show_bug.cgi?id=759624
For RGB formats, the matrix in the colorimetry (conversion from YUV to
RGB) is irrelevant and we should ignore it and assume the identity
transform for everything we do.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624
value of 32768L << 16 and 1L << 31 is 2147483648
but it exceeds the positive range of int which is 2147483647
resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
https://bugzilla.gnome.org/show_bug.cgi?id=760769
It's quite unexpected behaviour that various subclass settings are just
reset before set_format(). Unfortunately changing this now has the risk
of breaking existing code but we should reconsider this for 2.0.
[..] when resetting group start time. In GES, we are usually connected
to the streamsynchronizer on one audio and one video pad.
When seeking the timeline, both nlecompositions often output their flush_start
before any of them has output its flush_stop.
The current code, when receiving the first flush stop was using the
running time of the start of the second composition, which could
be pretty much anything, and means nothing at that point.
This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
both when setting flushing and when checking it.
https://bugzilla.gnome.org/show_bug.cgi?id=750013
When blocking input pads, we also need to properly set the appropriate
pending flag.
Without this, when switching stream types after initial configuration
(like going from Audio+Video to Audio+Video+Sub) playsink would never
wait for *all* input streams to be blocked (it would just wait for the
new input pad (text in this case) to be blocked).
Since the reconfiguration might introduce unlinking/relinking of elements,
we need to ensure that *ALL* input streams are blocked.
Failure to do so would result in having some input streams pushing data
to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
(returning GST_FLOW_NOT_LINKED).
A later optimization could involve only blocking the input pads that
might be involved in reconfiguration. But better be safe than sorry for
now :)
Elements usually require that all fields on their caps are present
on the fixed caps they receive. Using intersection won't verify it,
resort to using is_subset() checks.
https://bugzilla.gnome.org/show_bug.cgi?id=760477
When the input and output formats are the same and in a possible
intermediate format, avoid unpack and pack.
Never do passthrough channel mixing.
Only do dithering and noise shaping in S32 format
Add support for float and int16 mixing
Remove in-place processing, this simplifies things as we won't be using it.
Don't do clipping for float audio formats
Process as many samples as we can from the input and return the number
of processed samples from the chain. This simplifies some code.
Fix the IN_WRITABLE handling, don't overwrite the flags.
Those accept caps are actually checking if downstream supports
some particular caps to check if it need to negotiate a different
format. Checking only the next element with accept-caps is not enough
to guarantee that it is supported.
Using a caps query makes it obtain the supported caps for downstream
as a whole instead of only the next element.
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
