The problem is that the filesrc and souphttpsrc are behaving
differently regarding the calculation of the segment boundaries. The
filesrc is using a non-inclusive boundaries, while the souphttpsrc
uses inclusive. Currently the hlsdemux calculates the boundaries as
inclusive, so for this reason there is no problem with the souphttpsrc,
but there is an issue in the filesrc.
The GstSegment is non-inclusive, so the proposed solution is to use
non-inclusive boundaries in the hlsdemux in order to be consistent.
Make the change in the hlsdemux, will break the souphttpsrc, which
will expect inclusive boundaries, but the hlsdemux will offer
non-inclusive. This change makes sure that the non-inclusive
boundaries are converted to inclusive.
https://bugzilla.gnome.org/show_bug.cgi?id=748316
These allow a failed request to be retried after the given number of seconds
instead of failing the pipeline. Take account of the Retry-After header if
present. Add retries parameter that controls the number of times an HTTP
request will be retried before failing.
https://bugzilla.gnome.org/show_bug.cgi?id=756318
If the QtDemuxStream are re-used they may already have caps which used
to be leaked.
Reproduced using the
validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=756561
Negotiation to audio/x-raw,format=S8 was not possible because S8 does
not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
https://bugzilla.gnome.org/show_bug.cgi?id=756387
They now use the new GstAudioVisualizer base class
from gst-plugins-base/gst-libs/gst/pbutils
Also fixed undefined reference to gst_audio_visualizer_get_type
Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
https://bugzilla.gnome.org/show_bug.cgi?id=742875
If soup message is not created then the same should not be passed
on, which is resulting in segfault. Hence throwing a warning message
and returning
https://bugzilla.gnome.org/show_bug.cgi?id=755326
Add statitics from each rtp source to the rtp session property.
'source-stats' is a GValueArray where each element is a GstStructure of
stats for one rtp source.
The availability of new stats is signaled via g_object_notify.
https://bugzilla.gnome.org/show_bug.cgi?id=752669
Buffer is added to the internal cache, and pushed only when accumulated
buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
is not freed. Freeing the cache when the state changes from PAUSED to READY.
https://bugzilla.gnome.org/show_bug.cgi?id=754212
By not doing this, the muxer is not effectively a rtpmuxer, rather a
funnel, since it should be a single stream that exists the muxer.
If not specified, take the first ssrc seen on a sinkpad, allowing upstream
to decide ssrc in "passthrough" with only one sinkpad.
Also, let downstream ssrc overrule internal configured one
We hence has the following order for determining the ssrc used by
rtpmux:
0. Suggestion from GstRTPCollision event
1. Downstream caps
2. ssrc-Property
3. (First) upstream caps containing ssrc
4. Randomly generated
https://bugzilla.gnome.org/show_bug.cgi?id=752694
