- 14 Sep, 2016 14 commits
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The basic idea is this: 1. For *larger* rtx-rtt, weigh a new measurement as before 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less 3. For very large measurements, consider them "outliers" and count them a lot less The idea being that reducing the rtx-rtt is much more harmful then increasing it, since we don't want to be underestimating the rtt of the network, and when using this number to estimate the latency you need for you jitterbuffer, you would rather want it to be a bit larger then a bit smaller, potentially losing rtx-packets. The "outlier-detector" is there to prevent a single skewed measurement to affect the outcome too much. On wireless networks, these are surprisingly common. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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Assuming equidistant packet spacing when that's not true leads to more loss than necessary in the case of reordering and jitter. Typically this is true for video where one frame often consists of multiple packets with the same rtp timestamp. In this case it's better to assume that the missing packets have the same timestamp as the last received packet, so that the scheduled lost timer does not time out too early causing the packets to be considered lost even though they may arrive in time. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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There is no need to schedule another EXPECTED timer if we're already past the retry period. Under normal operation this won't happen, but if there are more timers than the jitterbuffer is able to process in real-time, scheduling more timers will just make the situation worse. Instead, consider this packet as lost and move on. This scenario can occur with high loss rate, low rtt and high configured latency. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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This patch fixes an issue with the estimated gap duration when there is a gap immediately after a lost timer has been processed. Previously there was a discrepancy beteen the gap in seqnum and gap in dts which would cause wrong calculated duration. The issue would only be seen with retranmission enabled since when it's disabled lost timers are only created when a packet is received and the actual gap length and last dts is known. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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The default -1 gives the old behavior. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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Stats should also be collected for unsuccessful packets. rtx-rtt is very important for determining the necessary configured latency on the jitterbuffer. It's especially important to be able to increase the latency when retransmitted packets arrive too late and are considered lost. This patch includes these late packets in the calculation of the various rtx stats, making them more correct and useful. Also in the case where the original packet arrives after a NACK is sent, the received RTX packet should update the stats since it provides useful information about RTT. The RTT is only updated if and only if all requested retranmissions are received. That way the RTT is guaranteed to make sense. If not we don't know which request the packet is a response to and the RTT may be bogus. A consequence of this patch is that RTT is not updated for a request when one of the RTX packets for that seqnum is lost, but that since measured RTT will be more accurate. The implementation store the RTX information from the timed out timers and use this when the retransmitted packet arrives. For performance these timers are stored separately from the "normal" timers in order to not impact performance (see attached performance test). https://bugzilla.gnome.org/show_bug.cgi?id=769768
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Add num-pushed and num-lost. Expose num-late, num-duplicates and avg-jitter. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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When disabled we can save some iterations over timers. There is probably an argument for rtx-delay-reorder to exist, but for normal operations, handling jitter (reordering) is something a jitterbuffer should do, and this variable feels like functionality that is not "in-sync" with what the jitterbuffer is trying to achieve. Example: You have 50ms jitter on your network, and are receiving audio packets with 10ms durations. An audio packet should not be considered late until its rtx-timeout has expired (and hence a rtx-event is sent), but with rtx-delay-reorder, events will be sent pretty much all the time due to the jitter on the network. Point being: The jitterbuffer should adapt its size to the measured network jitter, and then rtx-delay-reorder needs to adapt as well, or simply get out of the way and let the other (better) rtx-mechanisms do their job. Also change find_timer to only use seqnum as an argument, since there will only ever be one timer per seqnum at any given time. In the one case where the type matters, the caller simply checks the type. https://bugzilla.gnome.org/show_bug.cgi?id=769768
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Olivier Crête authoredCID #1372887
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Olivier Crête authored
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Sebastian Dröge authored
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- 13 Sep, 2016 1 commit
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Thomas Scheuermann authored
If jackd changes the buffer size or sample rate, jackaudiosink hangs and can't be stopped. This also happens if jack is configured as slave and a gstreamer pipeline is started on the slave machine while the jack master isn't running yet. If the the jack master is started it changes the buffer size / sample rate and jackaudiosink can't be stopped. This fix calls jack_shutdown_cb when jack_sample_rate_cb or jack_buffer_size_cb is called. https://bugzilla.gnome.org/show_bug.cgi?id=771272
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- 12 Sep, 2016 2 commits
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Sebastian Dröge authored
And actually calculate the field duration instead of a frame duration so that we can properly timestamp output frames in fields=all mode. This is probably still broken for reverse playback in telecine mode.
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- 10 Sep, 2016 2 commits
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Jan Schmidt authored
From b18d820 to f980fd9
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Xabier Rodriguez Calvar authored
Cause later it will try to use the crypto info array to get an index and attach on of the positions as buffer's crypto info. https://bugzilla.gnome.org/show_bug.cgi?id=770951
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- 09 Sep, 2016 1 commit
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Jan Schmidt authored
From f49c55e to b18d820
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- 07 Sep, 2016 2 commits
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Nicolas Dufresne authoredThose where missing the the dev release tarballs for 1.9.2 which prevented building from tarball on OSX platform
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- 06 Sep, 2016 8 commits
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Vincent Penquerc'h authored
Keep track of the last parsed channels/rate fields so they can be used even if the element was not yet configured. https://bugzilla.gnome.org/show_bug.cgi?id=769278
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Olivier Crête authored
This may cause a few packets to be processed by the parser, but it's better than never pushing out buffers from a slightly broken stream where no marker bits are set.
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Sebastian Dröge authored
This is only supported right now if after a demuxer that supports reverse playback, e.g. with DV container inside AVI container.
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- 05 Sep, 2016 1 commit
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Thibault Saunier authored
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- 01 Sep, 2016 5 commits
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Mathieu Duponchelle authored
+ And provide a "youtube" preset, which based on https://support.google.com/youtube/answer/1722171 sets faststart to True. https://bugzilla.gnome.org/show_bug.cgi?id=751559
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
We use gdk_cairo_create() which is deprecated since 3.22.
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- 30 Aug, 2016 1 commit
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Jan Schmidt authored
QuickTime.h is no longer available on OS X 10.12 (Sierra), and both the header and the framework seem unnecessary for compilation - at least as of 10.11 (El Capitan). https://bugzilla.gnome.org/show_bug.cgi?id=770526
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- 26 Aug, 2016 3 commits
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Josep Torra authored
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Josep Torra authored
Fixes build in OSX when running 'make check' in gst-uninstalled.
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