+=== release 1.5.1 ===
+
+2015-06-07 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.5.1
+
+2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: No flush during Teardown.
+ When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
+ backlog is empty it can happen that just a part of a message will be
+ sent and rest is in backlog queue. If then flush during teardown
+ just a part of message will be sent.This can lead to client miss
+ teardown response since it expect to get the last part of message.
+ The flushing during teardown was introduced to fix a deadlock that now
+ is fixed more generally in handle_request by temporary setting backlog
+ size to unlimited.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
+
+2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner and the
+ current version of the common submodule.
+
+2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Use single-include rtsp header to make sure we get all definitions
+
+2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Mark some more functions static
+
+2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only unblock the media in suspend() when actually changing the state
+ Otherwise we're going to lose a few packets for live streams during DESCRIBE.
+
+2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-video-rtx.c:
+ examples: Use AVPF profile for the RTX example
+
+2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Only add RTX to the SDP when using a feedback profile
+
+2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: get valid clock-rate from last-sample
+ clock-rate in last-sample's caps is integer, not unsigned.
+ To get this value properly, variable needs to be type-casted to int.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747614
+
+2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ autogen.sh: only run autopoint if gettext requested in configure.ac
+ Not just because there happens to be a po directory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ Revert "configure.ac: uncomment gettext version setup"
+ This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
+ We don't need a gettext setup here and there's no po
+ directory either, so no reason why autopoint would be
+ run in the first place.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
+
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-sdp.c:
+ * examples/test-video-rtx.c:
+ * examples/test-video.c:
+ * tests/test-cleanup.c:
+ * tests/test-reuse.c:
+ Fix timeout function signatures across tests and examples
+
+2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * configure.ac:
+ configure.ac: uncomment gettext version setup
+ Fixes autogen.sh. It would run autopoint, which would complain
+ that it could not find the gettext version in configure.ac.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-video-rtx.c:
+ test-video-rtx: set exact payload type to PCMA payloader
+ Setting wrong payload type causes failure to do retransmission through audio stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: fix to get valid each stream data for request-aux-sender signal
+ Because of duplicated g_signal_connect for request-aux-sender signal,
+ wrong stream pointer is passed to the signal handler.
+ Instead of passing each stream, pass stream array and get the relevant stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * acinclude.m4:
+ * autogen.sh:
+ Update autogen.sh to latest version from common
+ Fixes build after aclocal_check etc. helpers have been removed.
+
+2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Limit the queues to 1 buffer
+ We only need them to be able to pre-roll, queueing up more data here
+ is only going to harm latency and memory usage.
+
+2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Update comment and ASCII art to the latest code
+ We have a queue in front of the udpsink too to prevent the pipeline from
+ locking up.
+
+2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: Properly return first rtptime
+ Instead we where returning first GstBuffer timestamp. This would result
+ in clock skew and unwanted behaviour in RTSP playback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746479
+
+2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't leave buffer mapped
+ If the seq is NULL, the RTP buffer was left mapped. We should always
+ unmap the buffer.
+
+2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * README:
+ Fix typo in README
+
+2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * tests/check/gst/client.c:
+ Fix double semicolons
+
+2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
+ This gives more accurate values than asking the payloader. There might be
+ queueing happening between the payloader and the sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't seek for PLAY if the position will not change
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't include payload type in the caps for framesize
+ When the sdp media attribute framesize are converted to caps
+ the <payload> should not be included.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
+ Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
+
+2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: add payload type to the sdp framesize attribute
+ The sdp framesize attribute is desribed in RFC6064. It is specified
+ for payloading of H263 and has the following form
+ a=framesize:<payload type> <width>-<height>. The <width>-<height> part
+ should be added to the caps in a payloader and the <payload type> should
+ be added by the rtsp-server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
+
+2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: fix tainted variable
+ Insignificant but this keeps Coverity happy.
+ CID #1268404
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ examples: Add a simple example of network synch for live streams.
+ An example server and client that works for synchronising live streams
+ only - as it can't support pause/play.
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: Add functions to set/get the media gtype
+ Allow specifying the GType of a GstRtspMedia subclass to create
+ as a simpler way to get the factory to create a custom
+ GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
+
+2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix double unlock in _get_buffer_size()
+ Fixes an abort when calling gst_rtsp_media_get_buffer_size()
+ because of double g_mutex_unlock () usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745434
+
+2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-session: Use monotonic time for RTSP session timeout
+ Changed RTSP session timeout handling to monotonic time
+ and deprecating the API for current system time.
+ This fixes timeouts when the system time changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743346
+
+2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-client: Only error out in PLAY if seeking actually failed
+ If the media was just not seekable, we continue from whatever position we are
+ and let the client decide if that is what is wanted or not.
+ Only if the actual seek failed, we can't really recover and should error out.
+
+2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Add necessary queues between tee and multiudpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=744379
+
+2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: If seeking fails, don't wait forever for the media to preroll again
+ Instead error out properly the same way as if the SEEKING query already
+ failed.
+
+2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: minor code formatting fix
+
+2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix logic for collect_streams
+ Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
+ all streams it knows if it got any, and can check if the transport mode is OK.
+ CID #1268400
+
+2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't set the transport mode based on what elements we find
+ Just print a warning if the one that was set before disagrees with what
+ elements we found. It must already be set to something before as this
+ function is called after we received the SDP from ANNOUNCE in RECORD mode,
+ and we would reject ANNOUNCE if the RECORD flag was not set.
+
+2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: rename shadowed variable
+ We have two different 'sink' variables here,
+ rename one of them for clarity.
+
+2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix awkward if clause
+
+2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: improve uri argument handling and accept file names
+ Print an error if the argument passed is not a URI and can't
+ be converted into one, or no arguments have been provided.
+
+2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: don't remove mount point after 10 seconds
+ It's very irritating when trying to test stuff repeatedly
+ and serves no real purpose other than showing that it can
+ be done.
+
+2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add new test-record to .gitignore
+
+2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media: Use flags to distinguish between PLAY and RECORD media
+
+2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Set latency for playback-style example to 2s instead of 200ms
+
+2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: add some unit tests for ANNOUNCE and RECORD
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix a couple of leaks in handle_announce
+
+2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Expose latency setting for setting the rtpbin latency
+
+2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
+
+2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
+
+2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ Add initial support for RECORD
+ We currently only support media that is RECORD or PLAY only, not both at once.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: RTCP and RTP transport cache cookies seperated
+ RTCP packets were not sent because the same tr_cache_cookie was used for
+ both RTP and RTCP. So only one of the tr_cache lists were populated
+ depending on which one was sent first. If the tr_cache list is not
+ populated then no packets can be sent. Most often this happened to be
+ RTCP. Now seperate RTCP and RTP transport cache cookies are added which
+ resulted in both the tr_cache_lists to be populated regardless of which
+ one was sent first.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
+
+2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix false compiler warning
+ rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
+
+2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: log interleaved data received
+
+2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
+
+2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
+
+2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use a random session ID in the SDP
+ RFC4566 Section 5.2 says that it should make the username, session id,
+ nettype, addrtype and unicast address tuple globally unique. Always using
+ 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
+ Instead let's create a 64 bit random number, which at least brings us
+ closer to the goal of global uniqueness.
+ https://tools.ietf.org/html/rfc4566#section-5.2
+
+2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Don't call gst_init() and gst_get_option_group()
+ The latter calls the former at the appropriate time.
+
+2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Drop trailing \0 of RTSP DATA messages
+ We add a trailing \0 in GstRTSPConnection to make parsing of
+ string message bodies easier (e.g. the SDP from DESCRIBE) but
+ for actual data this means we have to drop it or otherwise
+ create invalid data.
+
+2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
+ Fixes crash when two threads access handle_new_sample() at the same
+ time, one for RTP, one for RTCP.
+ Otherwise, when iterating over the transports cache, it might be modified by
+ another thread at the same time if the transports cookie has changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742954
+
+2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set format=TIME on our app sources for TCP
+
+2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
+ This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
+ RFC 2326 states that session IDs may consist of alphanumeric as well as
+ the safe characters $-_.+ -- N.B. the percent character is not allowed.
+ Previously the session ID was URI-escaped, this meant that any character
+ which was not alphanumeric or any of the characters +-._~ would be
+ percent encoded. While the RFC (surprisingly) mentions that linear white
+ space in session IDs should be URI-escaped, it does not say anything
+ about other characters. Moreover no white space is allowed in the
+ session ID. Finally the percent character which is the result of
+ URI-escaping is not allowed in a session ID.
+ So there is no reason to do any URI-escaping, and now it is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742869
+
+2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Fix 'make check' from top-level directory
+
+2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Add command-line parsing and take a 'port' argument
+ This allows users to run multiple servers on different ports for testing.
+ Only done for examples that actually take arguments and hence are capable of
+ outputting different streams for each instance on each port.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742115
+
+2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Add a send_message default signal handler
+ This allows subclasses to easily hook into the response sending
+ mechanism without doing everything from a signal, which seems
+ awkward from subclasses.
+
+2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * Makefile.am:
+ * configure.ac:
+ configure: add --disable-examples switch
+ https://bugzilla.gnome.org/show_bug.cgi?id=741678
+
+2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-video-rtx.c:
+ examples: add a retransmisison example implementing RFC4588
+ Currently only SSRC-multiplexed rtx streams are supported
+
+2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix some minor memory leaks
+
+2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Some minor cleanup
+
+2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix compiler warnings
+ rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+ rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+
+2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ media: implement ssrc-multiplexed retransmission support
+ based off RFC 4588 and the server-rtpaux example in -good
+
+2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: Ref transports in hash table.
+ Also ref streams for transports.
+ This solves a crash when reciving a rtcp after teardown but before
+ client finalize.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
+
+2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor cleanup of cached media
+
+2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: Remove FIXME
+ The session leak is now fixed, lets remove those FIXME comments.
+
+2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test to setup two sessions on one connection
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test setup with tcp transport
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Configure transport after creating session media
+ The default implementation of configure_client_transport() in
+ rtsp-client uses the session media when it chooses channels for
+ interleaved traffic.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ client: Stop caching media in client when doing setup
+ If the media has been managed by a session media, it should not be
+ cached in the client any longer. The GstRTSPSessionMedia object is now
+ responsible for unpreparing the GstRTSPMedia object using
+ gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
+ session media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: unref srtp decoder when leaving bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=739481
+
+2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: mikey memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739383
+
+2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Parallelise 'make check-valgrind'
+
+2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 36388a1 to a8c8939
+
+2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: deactivate media when shutting down from paused
+ This was only done when going directly from playing.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
+
+2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ rtsp-client: add stream transport to context
+ We add the stream transport to the context so we can get the configured
+ client stream transport in the setup request signal.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
+
+2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock even not all transports have been removed
+ We don't want to keep the lock even we return FALSE because not all the
+ transports have been removed. This could lead into a deadlock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737797
+
+2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set session media to NULL without the lock
+ We need to set session medias to NULL without the client lock otherwise
+ we can end up in a deadlock if another thread is waiting for the lock
+ and media unprepare is also waiting for that thread to end.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737690
+
+2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Set state to UNPREPARING in all cases
+
+2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set state to unpreparing when unprepare is initiated
+ https://bugzilla.gnome.org/show_bug.cgi?id=737675
+
+2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Remove backlog limit while processings requests
+ If the backlog limit is kept two cases of deadlocks may be
+ encountered when streaming over TCP. Without the backlog
+ limit this deadlocks can not happen, at the expence of
+ memory usage.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
+
+2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not free main context before rtsp watch
+ https://bugzilla.gnome.org/show_bug.cgi?id=737110
+
+2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Extend unit test timeout to accomodate for valgrind
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ rtsp-*: Treat sending packets to clients as keepalive
+ As long as gst-rtsp-server can successfully send RTP/RTCP data to
+ clients then the client must be reading. This change makes the server
+ timeout the connection if the client stops reading.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Allow backlog to grow while expiring session
+ Allow the send backlog in the RTSP watch to grow to unlimited size while
+ attempting to bring the media pipeline to NULL due to a session
+ expiring. Without this change the appsink element cannot change state
+ because it is blocked while rendering data in the new_sample callback.
+ This callback will block until it has successfully put the data into the
+ send backlog. There is a chance that the send backlog is full at this
+ point which means that the callback may block for a long time, possibly
+ forever. Therefore the media pipeline may also be prevented from
+ changing state for a long time.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Make old compilers happy
+ rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
+ Just in case that guint8 doesn't fit in a pointer. Just in case ...
+
+2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: raise the backlog limits before pausing
+ We need to raise the backlog limits before pausing the pipeline or else
+ the appsink might be blocking in the render method in wait_backlog() and
+ we would deadlock waiting for paused.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make define for the WATCH_BACKLOG
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: simplify session transport handling
+ link/unlink of the transport in a session was done to keep track of all
+ TCP transports and to send RTP/RTCP data to the streams. We can simplify
+ that by putting all the TCP transports in a hashtable indexed with the
+ channel number.
+ We also don't need to link/unlink the transports when we pause/resume
+ the streams. The same effect is already achieved when we pause/play the
+ media. Indeed, when we pause the media, the transport is removed from
+ the media and the callbacks will not be called anymore.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736041
+
+2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: make method to handle received data
+ Make a method to handle the data received on a channel. It sends the
+ data to the stream of the transport on the RTP or RTCP pads based on
+ the channel number.
+
+2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-mp4.c:
+ test: add example of dumping RTCP reports
+
+2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Make sure that sequence numbers are monotonic after pause
+ The sequence number is not monotonic for RTP packets after pause. The
+ reason is basepayloader generates a randon sequence number when the
+ pipeline goes from ready to pause. With this fix generation of sequence
+ number will be monotonic when going from pause to play request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736017
+
+2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Protect saved clients watch with a mutex
+ Fixes a crash when close() is called while merging clients
+ in handle_tunnel(). In that case close() would destroy the
+ watch while it is still being used in handle_tunnel().
+ https://bugzilla.gnome.org/show_bug.cgi?id=735570
+
+2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Remove the multicast group udp sources when removing from the bin
+
+2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Query position and stop time only on the RTP parts of the pipeline
+ The RTCP parts, in specific the RTCP udpsinks, are not flushed when
+ seeking and will always continue counting the time. This leads to
+ the NPT after a backwards seek to be something completely different
+ to the actual seek position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732644
+
+2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-appsrc.c:
+ examples: fix another reference leak
+ gst_rtsp_media_get_element() returns a new ref.
+
+2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * examples/test-appsrc.c:
+ examples: unref element after usage
+ gst_bin_get_by_name_recurse_up() returns an element
+ reference that must be unreffed after usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734546
+
+2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ signals: Fix copy-pasto in target-state signal offset
+
+2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Listen on the multicast group for RTP/RTCP packets
+ When a UDP multicast transport is used it is expected that the server listens
+ for RTP and RTCP packets on the multicast group with the corresponding port.
+ Without this we will never get RTCP packets from clients in multicast mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732238
+
+2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.4.0 ===
-2014-07-19 Sebastian Dröge <slomo@coaxion.net>
+2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.4.0
+ * gst-rtsp-server.doap:
+ Release 1.4.0
2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
-This is GStreamer RTSP Server 1.4.0
+This is GStreamer RTSP Server 1.5.1
-Changes since 1.2:
-
-New API:
- • GstMessageType has GST_MESSAGE_EXTENDED added. All types before
- that can be used together as a flags type as before, but from
- that message onwards the types are just counted incrementally.
- This was necessary to be able to add more message types.
- In 2.0 GstMessageType will just become an enum and not a flags
- type anymore.
- • GstDeviceMonitor for device probing, e.g. to list all available
- audio or video capture devices. This is the replacement for
- GstPropertyProbe from 0.10.
- • Events accumulate the running-time offset now when travelling
- through pads, as set by the gst_pad_set_offset() function. This
- allows to compensate for this in the QOS event for example.
- • GstBuffer has a new flag "tag-memory" that is set automatically
- when memory is added or removed to a buffer. This allows buffer
- pools to detect if they can recycle a buffer or need to reset
- it first.
- • GstToc has new API to mark GstTocEntries as loops.
- • A not-authorized resource error has been defined to notify
- applications that accessing the resource has failed because
- of missing authorization and to distinguish this case from others.
- This change is actually already in 1.2.4.
- • GstPad has a new flag "accept-intersect", that will let the default
- ACCEPT_CAPS query handler do an intersection instead of subset check.
- This is interesting for parser elements that can handle incomplete
- caps.
- • GstCollectPads has support for flushing and a default handler for
- SEEK events now.
- • New GstFlowAggregator helper object that simplifies handling of
- flow returns in elements with multiple source pads. Additionally
- GstPad now always stores the last flow return and provides an
- API to retrieve it.
- • GstSegment has new API to offset the running time by a specific
- value and this is used in GstPad to allow positive and negative
- offsets in gst_pad_set_offset() in all situations.
- • Support for h265/HEVC and VP8 has been added to the codec utils and codec
- parsers library, and was integrated into various elements.
- • API for adjusting the TLS validation of RTSP connection has been added.
- • The RTSP and SDP library has MIKEY (RFC 3830) support now, and
- there is API to distinguish between the different RTSP profiles.
- • API to access RTP time information and statistics.
- • Support for auxiliary streams was added to rtpbin.
- • Support for tiled, raw video formats has been added.
- • GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
- events and merge custom tags into them consistently.
- • GstBufferPool has support for flushing now.
- • playbin/playsink has support for application provided audio and video
- filters.
- • GstDiscoverer has new and simplified API to get details about missing
- plugins and information to pass to the plugin installer.
- • The GL library was merged from gst-plugins-gl to gst-plugins-bad,
- providing a generic infrastructure for handling GL inside GStreamer
- pipelines and a plugin with some elements using these, especially
- a video sink. Supported platforms currently are Android, Cocoa (OS X),
- DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
- Wayland and EGL platforms.
- This replaces eglglessink and also is supposed to replace osxvideosink.
- • New GstAggregator base class in gst-plugins-bad. This is supposed to
- replace GstCollectPads in the future and fix long-known shortcomings
- in its API. Together with the base class some elements are provided
- already, like a videomixer (compositor).
-
-
-Major changes:
- • New plugins and elements:
- ∘ v4l2videodec element for accessing hardware codecs on
- platforms that make them accessible via V4L2, e.g.
- Samsung Exynos. This comes together with major refactoring
- of the existing V4L2 elements and the corresponding
- infrastructure.
- The v4l2videodec element replaces the mfcdec element.
- ∘ New downloadbuffer element that replaces the download
- buffering feature of queue2. Compared to queue2's code
- it is much simpler and only for this single use case.
- A noteworthy new feature is that it's downloading gaps
- in the already downloaded stream parts when nothing else
- is to be downloaded.
- This is now used by playbin when download buffering is
- enabled.
- ∘ rtpstreampay and rtpstreamdepay elements for transmitting
- RTP packets over a stream API (e.g. TCP) according to
- RFC 4571.
- ∘ rtprtx elements for standard compliant implementation of
- retransmissions, integrated into the rtpmanager plugin.
- ∘ audiomixer element that mixes multiple audio streams together
- into a single one while keeping synchronization. This is
- planned to become the replacement of the adder element.
- ∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
- ∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
- ∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
- ∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
- ∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
- are available on OS X and iOS now.
-
- • Other changes:
- ∘ gst-libav now uses libav 10.2, and gained support for H265/HEVC.
- ∘ Support for hardware codecs and special memory types has been
- improved with bugfixes and feature additions in various plugins
- and base classes.
- ∘ Various bugfixes and improvements to buffering in queue2 and
- multiqueue elements.
- ∘ dvbsrc supports more delivery mechanisms and other features
- now, including DVB S2 and T2 support.
- ∘ The MPEGTS library has support for many more descriptors.
- ∘ Major improvements to tsdemux and tsparse, especially time and
- seeking related.
- ∘ souphttpsrc now has support for keep-alive connections,
- compression, configurable number of retries and configuration
- for SSL certificate validation.
- ∘ hlsdemux has undergone major refactoring and works more
- reliable now and supports more HLS features like trick modes.
- Also fragments are pushed downstream while they're downloaded
- now instead of waiting for each fragment to finish.
- ∘ dashdemux and mssdemux are now also pushing fragments downstream
- while they're downloaded instead of waiting for each fragment to
- finish.
- ∘ videoflip can automatically flip based on the orientation tag.
- ∘ openjpeg supports the OpenJPEG2 API.
- ∘ waylandsink was refactored and should be more useful now. It also
- includes a small library which most likely is going to be removed
- in the future and will result in extensions to the GstVideoOverlay
- interface.
- ∘ gst-rtsp-server supports SRTP and MIKEY now.
- ∘ gst-libav encoders are now negotiating any profile/level settings
- with downstream via caps.
- ∘ Lots of fixes for coverity warnings all over the place.
- ∘ Negotiation related performance improvements.
- ∘ 800+ fixed bug reports, and many other bug fixes and other
- improvements everywhere that had no bug report.
-
-Things to look out for:
- • The eglglessink element was removed and replaced by the glimagesink
- element.
- • The mfcdec element was removed and replaced by v4l2videodec.
- • osxvideosink is only available in OS X 10.6 or newer.
- • On Android the namespace of the automatically generated Java class
- for initialization of GStreamer has changed from com.gstreamer to
- org.freedesktop.gstreamer to prevent namespace pollution.
- • On iOS you have to update your gst_ios_init.h and gst_ios_init.m in
- your projects from the one included in the binaries if you used the
- GnuTLS GIO module before. The loading mechanism has slightly changed.
projects / platform / upstream / gstreamer.git / commitdiff