| Commit message (Collapse) | Author | Age | Files | Lines |
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opus decoder can convert from different number of channels, no
need to check, just let it negotiate and create a new decoder if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=755059
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Avoids useless check of downstream caps when handling an
accept-caps query
Elements: dtsdec, faad, gsmdec, mpg123audiodec, opusdec,
sbcdec, adpcmdec, sirendec
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Previously, PLC frames always had a length of 120ms, which caused audio
quality degradation and synchronization errors. Fix this by calculating an
appropriate length for the PLC frame.
The length must be a multiple of 2.5ms. Calculate a multiple of 2.5ms that
is nearest to the current PLC length. Any leftover PLC length that didn't
make it into this frame is accumulated for the next PLC frame.
https://bugzilla.gnome.org/show_bug.cgi?id=725167
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structure has this field
Just set the rate/channels directly if the caps don't have this field.
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Use the helper function available in the base class instead.
https://bugzilla.gnome.org/show_bug.cgi?id=748585
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This way we let opusdec do the resampling if needed and don't carry
around buffers with a too high sample rate if not required.
While Opus always uses 48kHz internally, this information from the
header specifies which frequencies are safe to drop.
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The max latency parameter is "the maximum time an element
synchronizing to the clock is allowed to wait for receiving all
data for the current running time" (docs/design/part-latency.txt).
https://bugzilla.gnome.org/show_bug.cgi?id=744338
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This now follows the design docs everywhere, especially the maximum latency
handling.
https://bugzilla.gnome.org/show_bug.cgi?id=744106
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Set the channels and rate back to their default values in _stop because they
are used to renegotiate when needed.
See https://bugzilla.gnome.org/show_bug.cgi?id=692950
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https://bugzilla.gnome.org/show_bug.cgi?id=687520
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When the decoder received a NULL buffer, it tried to
unmap a not mapped buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=686829
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where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
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opus + jpegformat plugin builds fail when gstreamer is configured with
--disable-gst-debug as they are checking the GST_DISABLE_DEBUG symbol
instead of GST_DISABLE_GST_DEBUG.
Signed-off-by: Peter Korsgaard <jacmet@sunsite.dk>
https://bugzilla.gnome.org/show_bug.cgi?id=683850
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because as slomo noted, in fact pretty much all the code in there is mine.
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It's at byte offset 16, not 14.
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The base audio decoder sends zero size packets, not NULL buffers,
to signal dropped packets.
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... so as to avoid leaking caps or manipulating NULL caps.
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Conflicts:
docs/libs/Makefile.am
ext/kate/gstkatetiger.c
ext/opus/gstopusdec.c
ext/xvid/gstxvidenc.c
gst-libs/gst/basecamerabinsrc/Makefile.am
gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
gst-libs/gst/video/gstbasevideocodec.c
gst-libs/gst/video/gstbasevideocodec.h
gst-libs/gst/video/gstbasevideodecoder.c
gst-libs/gst/video/gstbasevideoencoder.c
gst/asfmux/gstasfmux.c
gst/audiovisualizers/gstwavescope.c
gst/camerabin2/gstcamerabin2.c
gst/debugutils/gstcompare.c
gst/frei0r/gstfrei0rmixer.c
gst/mpegpsmux/mpegpsmux.c
gst/mpegtsmux/mpegtsmux.c
gst/mxf/mxfmux.c
gst/videomeasure/gstvideomeasure_ssim.c
gst/videoparsers/gsth264parse.c
gst/videoparsers/gstmpeg4videoparse.c
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There are two of them, unintuitively enough; the one passed
to the encoder should not be the one that gets written to the
file. The former maps the input to an ordering which puts
paired channels first, while the latter moves the channels
to Vorbis order. So add code to calculate both, and we now
have properly paired channels where appropriate.
https://bugzilla.gnome.org/show_bug.cgi?id=665078
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https://bugzilla.gnome.org/show_bug.cgi?id=665078
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https://bugzilla.gnome.org/show_bug.cgi?id=665078
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https://bugzilla.gnome.org/show_bug.cgi?id=665078
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https://bugzilla.gnome.org/show_bug.cgi?id=665078
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https://bugzilla.gnome.org/show_bug.cgi?id=662664
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Conflicts:
ext/opus/gstopusdec.c
ext/opus/gstopusparse.c
gst-libs/gst/video/gstbasevideodecoder.c
gst-libs/gst/video/gstbasevideodecoder.h
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https://bugzilla.gnome.org/show_bug.cgi?id=664815
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Conflicts:
ext/faac/gstfaac.c
ext/opus/gstopusdec.c
ext/opus/gstopusenc.c
gst/audiovisualizers/gstspacescope.c
gst/colorspace/colorspace.c
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It's very similar to the basic API, and is a superset ot it,
which will allow encoding and decoding more than 2 channels.
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It would ideally be better to leave this to a rgvolume element,
but we don't control the pipeline. So do it by default, and allow
disabling it via a property, so the correct volume should always
be output.
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This allows reconstruction of lost packets if FEC info is included
in the next packet, at the cost of extra latency. Since we do not
know if the stream has FEC (and this can change at runtime), we
always incur the latency, even if we never lose any frame, or see
any FEC information. Off by default.
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Conflicts:
ext/opus/gstopusdec.c
ext/opus/gstopusenc.c
ext/opus/gstopusparse.c
gst/audiovisualizers/gstwavescope.c
gst/filter/Makefile.am
gst/filter/gstfilter.c
gst/filter/gstiir.c
gst/playondemand/gstplayondemand.c
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