The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request
periodically as a ping. The code in gst_rtspsrc_handle_request forms an OK
response and sends, but doesn't call gst_rtsp_message_unset to free the memory
after sending the response. This results in a constant slow memory leak.
Fixes#624770
That is, in files that have no index (Cue), perform seek by scanning for
nearest cluster with timecode before requested position. Scanning is done
as a combination of interpolation and sequential scan.
Fixes#617368.
Timestamp rounding issues could lead to going backwards 2 keyframe periods
(rather than only 1). While this is not necessarily a problem, it might
potentially place additional (buffering) load on downstream and could be
avoided (because We Can).
Fixes#623629.
There seems to be a bug in libmp4v2 that generates a MPEG4BitRateBox as
(bufferSizeDB, avgBitrate, maxBitrate) instead of (bufferSizeDB,
maxBitrate, avgBitrate), according to the spec. I used the mp4file
output while writing this code, so the order is wrong. This patches
fixes that.
https://bugzilla.gnome.org/show_bug.cgi?id=623654
PluginInfo is quite a sizeable struct, let's not allocate it on the
stack, especially not if we're copying it over into another dynamically
allocated copy anyway.
Fixes#570761.
If we restart the Stream in the case of doing a transition from
PAUSED_TO_READY and back with READY_TO_PAUSED aso. the duration of the video
will get calculated even if we have a avi header with that information.
Signed-off-by: Michael Grzeschik <m.grzeschik@pengutronix.de>
On windows builds, sets source address for bind to INADDR_ANY, while
maintaining the original multicast group address for subsequent join.
Fixes#595978
Skip everything before the @ sign in the url location. VLC uses that as the
remote address to connect to (but we ignore it for now). This makes our udp urls
compatible with the ones used by VLC.
Fixes#597695
Keep a global connection for aggregate control but also keep stream connections
for non-aggregate control.
Add some helper methods to connect/close/flush the connections.
Subsequent entry time calculations use blockalign value to determine
number of frames per chunk, and blockalign == 1 is then most unlikely to result
in reasonable values (which also aligns with "spec").
So matroska's Block structure has no keyframe flag, only the SimpleBlock has it.
To detect keyframes in Blocks, it is just the BlockGroup container that needs
to have a ReferenceBlock attached if it is a delta frame in video.
Use new dts audio typefinder from -base to check if the PCM data
contains a dts stream. This way we recognise more varieties more
reliably and also detect the dts stream if there isn't a frame
sync right at the start of the data.
Fixes#413942.
Start cluster at every keyframe or when we would overflow the previous
cluster's relative timestamp field. This would avoid as much as possible
starting clusters at non-keyframes.
Don't send them upstream because for upstream a BYTES seek
might make sense but is completely wrong because upstream
can't seek to a byte position of the audio or video stream.
Also don't build the index in push mode for non-TIME seeks,
things will go wrong here otherwise.
This allows us to skip delta units earlier and is a bit clearer in my
opinion. It also makes only video buffers ever be delta units, not
just for SimpleBlock as before.
When the keyframe bit of SimpleBlock Flags wasn't set, the buffer was being
marked with GST_BUFFER_FLAG_DELTA_UNIT, causing all buffers to be skipped
after a seek. This may be a problem with the Sorenson Squish encoder, but
arguably the keyframe bit should only be applied to video.
Fixes bug #620358.
Even though we don't use delivery-method in our payloader, older versions of
the theora payloader in gstreamer required it. As such we need to keep this
around in the caps for backwards-compatibility.
This reverts part of 49463a37cbFixes#618940
When we calculate the frame duration, we need to use the amount of
frames in the _previous_ packet, not the current packet. The frame duration is
needed to correctly de-interleave interleaved streams. This fixes the case where
there are a variable number of frames in a packet.
Fixes#620494
This commit basically puts _get_caps() in sync with accept_caps().
If we don't have a master pad OR the master pad caps aren't negotiated
then we just return the downstream allowed caps.
If we have a master pad with negotiated caps, we return those caps
with a free range of width/height/framerate
When using RTP_JITTER_BUFFER_MODE_BUFFER, make sure that the ringbuffer doesn't
get stuck buffering forever when there isn't enough data left to fill the
buffer.
In demuxer and muxer use the gst_util_uint64 scaling functions rather than
standard integer division. Add warnings (to be changed to debug) for debugging
the timestamp and duration.
