Version 0.0.17
tests: Add test for telephone-event events parameter nego
rtpspecificnego: Add handling of telephone-event event ranges
tests: Skip tests if no local candidates are produced
rtcpfilter: Reduce the packet size when reducing the packet
tests: Skip libnice tests if it finds no local candidates
rtpdtmfsoundsource: Respect the ptime/maxptime too
tests: Add test ptime/maxptime passing
rtpsession: Set the ptime/maxptime on the send codec bin caps
rtpcodecnego: Negotiate the ptime/maxptime
rtpconference: Add function to make gst caps while keeping the ptime
rtpcodecnego: Add function to copy the list of codecs with the send-side ptime
tests; Add test for fscodec ptime/maxptime handling
codec: Add ptime
codec: Add maxptime
tests: Take rtpsession lock during message emissions
This ensures that it is not held across message emissions.
tests: Add debug-blocks
rtpsubstream: Keep ref on substream while callbacks are invoked
rtpsubstream: Put codec/codecbin inside loop
rtpsubstream: Use rw-lock to make sure the substream really stops
rtp: Move locking into callback
rtpsubstream: Don't hold session lock too much while setting new codecbin
rtpsubstream: Move modification locking to blocked function
Also allow only one thread to be in substream blocked function at once.
rtp: Move substream blocking logic into substream
rtp: Don't include marshaller headers in headers
rtp: Depend on the correct var for marshaller list generation
rtcpfilter: Add gst-p-base paths to Makefile.am
Patch contributed by Armijn Hemel <armijn@loohuis-consulting.nl>
rawudp: Remove upnp-request-timeout, it was a terrible idea
Substitute deprecated Glib symbol: g_mapped_file_free
Use g_mapped_file_unref if Glib >= 2.22 is available
http://bugs.freedesktop.org/show_bug.cgi?id=21422
rtpsession: Only add stream to list if its creation worked
README: Require gst-p-bad 0.10.17 for dtmfsrc
dtmfsrc can do do more than 8000 Hz, that has only been fixed in
gst-plugins-bad 0.10.17
rtpdtmfsound: Try hardwired PCMx only if the clock-rate is 8000
rtp: Lookup codec with config is always for sending, so make it explicit
Also, the dtmf sound will always get a valid codec now.
rtpconference: Make message about gst_bin_add failure more accurate
rtpdtmfsoundsource: Ignore codecs that don't have a blueprint
tests: Test dtmf as sound
tests: Make recv-pipeline per test
rtpdtmfsoundsource: Use main codec if PCMA/U are not available
rtpspecialsource: Make local class_get_codec function static
rtp: Regroup CodecBlueprint related functions in one place
rtpspecialsource: Rename negotiated_codecs to negotiated_codec_associations
This way, the list contents can be guessed
rtpsession: Don't need to set queue-delay anymore
rtpsession: Split codecbin generation from factory from profile
tests: Make it build against GUPnP 0.13
msnsession: Check if dispose has already been called
fstransmitter: uint can't be < 0
rawudp: Bring upnp discovery timeout down to 2 seconds
tests: Verify that it is not possible to disable all codecs
Add a reserve-pt to guarantee that it is not possible to disable all codecs
rtpcodecnego: Verify if there are any valid local codecs left after applying preferences
rtpsession: Make error message less cryptic
Version 0.0.16.1
97aa8b28ba