Log message:
Whitespace fixes, needed for gmake-3.82.

Log message:
put back URL to upstream bug report
noticed by wiz

Log message:
update to 0.0.21
changes: bugfixes

Log message:
Adjust line number for safe side.
It's too far and warnings in do-patch.

Log message:
Bump PKGREVISION for libpng shlib name change.
Also add some patches to remove use of deprecated symbols and fix other
problems when looking for or compiling against libpng-1.4.x.

Log message:
Fix build with libnice-0.0.11 and depend on it.

Bump PKGREVISION.

Fixes PR 43241 by Muhammad Hallaj Subery.

Log message:
Bump revision for PYTHON_VERSION_DEFAULT change.

Log message:
Bump PKGREVISION for gupnp/gssdp API changes.

Log message:
Update to 0.0.17:

    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

Log message:
Update to 0.0.16:

    Version 0.0.16

    rtpspecialsource: Remove want_source() method

    get_codec() function does the same thing

    rtpdtmfsoundsource: Implement get_codec method

    rtpdtmfeventsource: Implement get_codec method

    rtpspecialsource: Add new get_codec method

    rtp: Check if the codec changed when removing special sources

    rtp: Allow checking if a codec is valid for sending even if it has no way to \ 
build a codecbin

    rtpcodecnego: Fix doc string

    rtpspecialsource: Move static function closer to its use place

    rtpspecialsource: Fix over-80 line

    rtpsession: Check/update secondary sources even if the primary one doesn't change

    tests: Tests changing the dtmf PT mid-call

    tests: Make sure dtmf events are really received

    test: Test changing the dtmf_id

    tests: dtmf method is not always auto

    rtpsession: Only emit send-codec-changed message after the special codecs \ 
have been changed

    rtpsession: Don't leak iterator on linking failure

    rtpsession: Cleanup send codecbin on failure

    rtpsession: Print error on session dispose problems

    rtpdtmfsoundsource: Correctly check the presence of elements

    rawudp: Use %d for ints, not %s

    configure: quiet automake portability bs

    msnstream: Make send sink async=false for now

    msnstream: Don't keep lock into set_remote_candidates

    tests: Test invalid property name in fs_element_added_notifier_from_keyfile

    element-added-notifier: Don't crash on invalid property

    rtpconference: Don't assert on non-existing sdes parts

    rtpspecialsource: Dispose is not always called twice, cleanup in finalize

    rtpsession: Remove useless ref

    Version 0.0.15.1

    Version 0.0.15

    Require gst-p-bad 0.10.14 for mimic

    tests: Unlock src before setting it to playing

    tests: Refrain from using the thread unsafe version of failure in the nice test

    rtpsession: Keep ref on stream while associating substreams to it

    rtpsubstream: Remove another double-unlock in error case

    rtpsession: Don't double-unlock

    rtpsession: Fix leaking caps on signals after dispose

    rtpsession: Fix potential leak if already disposed

    rtpsubstrea: Remove unused variable

    elementaddednotifier: Use g_connect_signal_object

    Otherwise each element had a ref on the notifier and relied on the not thread
    safe weak references.

    rawudp: Emit local candidates if there are no local interfaces suitable for UPnP

    rawudp: Add some UPnP debug messages

    glib-gen: Use single = instead of == for portability

    msnconnection: Check return values from recv()

    msnsession: Conference must always set before get_property

    msnsession: Only try to lock conference if it has been set

    rtpsession: Initialise variable to NULL

    Makes coverity happy

    msnconnection: Remove unused variables

    rtpstream: Correct documentation

    rtpsession: Unref transmitter src/sink in dispose

    Unref element from g_object_get(), fixes leak

    elementaddednotifier: Unref element in iterator loop

    Fixes leak

    elementadded: Use gst_value_deserialize to read properties

    Use the existing function instead of having our own less-capable \ 
re-implementation

    Version 0.0.14.1