=== release 1.1.2 === 2013-07-11 Sebastian Dröge * configure.ac: releasing 1.1.2 2013-07-10 17:16:14 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Only give sinks a new bus if they have no parent yet Otherwise we will remove the bus that would proxy messages to playsink and never set it again. If the sink is already in playsink, all failures are fatal anyway as it's either a sink that worked before or one that was set by the user. https://bugzilla.gnome.org/show_bug.cgi?id=701997 2013-07-10 13:22:04 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Store a/v/t sinks locally too, not just in playsink 2013-07-10 13:21:29 +0200 Sebastian Dröge * gst/playback/gstplaysink.c: playsink: ref_sink() any sinks that are set on playsink Otherwise the behaviour of the properties is inconsistent. 2013-07-10 13:20:34 +0200 Sebastian Dröge * tests/check/elements/playbin.c: playbin: Fix assumptions in the unit test Unused sinks are still set to READY now during autoplugging to check their caps. Also playsink owns a ref to the sinks too. 2013-07-10 13:00:21 +0200 Sebastian Dröge * gst/playback/gststreamsynchronizer.c: streamsynchronizer: Non-TIME segment streams are not waiting automatically This was leftover code from porting to 1.0 and fixes the playbin unit test. https://bugzilla.gnome.org/show_bug.cgi?id=701943 2013-07-09 23:04:49 +0200 Branko Subasic * win32/common/libgstrtp.def: win32: add missing rtp buffer methods 2013-07-09 14:55:57 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: * gst/playback/gstplaysink.c: playbin: Change sink ownership handling to be a bit more sane playbin will now only activate the sinks in a single place and will never change the states of any sinks that are owned by playsink. Also handle text-sinks the same way as audio/video sinks inside playbin. 2013-07-05 21:55:26 +0200 Piotr Drąg * po/POTFILES.in: po: update POTFILES.in https://bugzilla.gnome.org/show_bug.cgi?id=703684 2013-07-04 17:09:00 +0300 Sreerenj Balachandran * gst-libs/gst/video/colorbalance.c: colorbalance: Fix the typo in base_init(). 2013-07-04 12:54:59 -0400 Thibault Saunier * gst/adder/gstadder.c: adder: Do not send flush_start event with the stream lock taken FLUSH_START is not serialized, so the lock should not be taken when sending it. 2013-07-05 00:47:08 +0100 Marcin Lewandowski * gst-libs/gst/tag/id3v2frames.c: tag: ignore malformed ID3v2 TDAT frames Just skip them, don't cause criticals. https://bugzilla.gnome.org/show_bug.cgi?id=703283 2013-07-03 09:44:32 +0100 Tim-Philipp Müller * gst/audioresample/speex_resampler_int.c: audioresample: make explicit that neon is disabled and why https://bugzilla.gnome.org/show_bug.cgi?id=703477 2013-07-02 18:20:39 +0200 Carlos Rafael Giani * gst/audioresample/speex_resampler_int.c: audioresample: disable 16-bit integer NEON support it seems to be broken (produces no audio), plus the performance gain is small Signed-off-by: Carlos Rafael Giani 2013-07-02 14:25:28 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: If we had a previous autoplugged sink, try to reuse it https://bugzilla.gnome.org/show_bug.cgi?id=701997 2013-07-02 14:18:20 +0200 Sebastian Dröge * gst/playback/gstplaysink.c: playsink: If we switch sinks, make sure that the old sink is set to NULL 2013-07-02 14:02:57 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Don't change the state of sinks that we passed to playsink already 2013-07-02 14:01:52 +0200 Sebastian Dröge * gst/playback/gstplaysink.c: playsink: Consider new audio/video sinks when reconfiguring 2013-07-02 12:27:03 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Improve debug output regarding sink selection 2013-07-01 12:52:43 -0600 Brendan Long * gst/playback/gstplaybin2.c: playbin: Post an error message if a stream combiner doesn't return a request pad. 2013-07-01 13:45:25 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Only intersect to check if a sink can handle raw caps Doing a subset check requires fixed caps, which we might not have here. https://bugs.webkit.org/show_bug.cgi?id=116042 2013-07-01 10:39:02 +0100 Vincent Penquerc'h * gst-libs/gst/pbutils/descriptions.c: * gst-libs/gst/pbutils/missing-plugins.c: * gst-libs/gst/pbutils/pbutils-private.h: pbutils: allow describing unfixed caps if they share the same media type Caps description and missing plugin code does not really need caps to be fixed, and indeed they may not be if giving encodebin unfixed caps that correspond to an unknown encoder or muxer. So we relax the check, and allow unfixed caps if all the structures refer to the same media type. 2013-07-01 11:16:34 +0200 Sebastian Dröge * gst-libs/gst/video/gstvideodecoder.c: videodecoder: Send all pending events with type < CAPS before sending caps 2013-06-27 16:33:15 +0200 Mathieu Duponchelle * gst-libs/gst/video/gstvideoencoder.c: videoencoder: Send all pending events with type < CAPS before sending caps. https://bugzilla.gnome.org/show_bug.cgi?id=703196 2013-06-28 14:48:19 +0100 Vincent Penquerc'h * gst/typefind/gsttypefindfunctions.c: typefind: avoid too low mpeg/ts probability on small amount of data With the current test, we get into problems when we try to typefind a MPEG stream from a small amount of data, which can happen when we get data pushed from a HTTP source. We thus make a second test to give higher probability if all the potential headers were either pack or pes headers (ie, no potential header was unrecognized). This fixes an issue with a MPEG1/MP2 stream being properly discovered as video/mpeg from a file, but as audio/mpeg from souphttpsrc. https://bugzilla.gnome.org/show_bug.cgi?id=703256 2013-06-30 18:17:15 +0200 Sebastian Dröge * gst-libs/gst/video/gstvideodecoder.c: * gst-libs/gst/video/gstvideoencoder.c: video(enc|dec)oder: Don't return not-negotiated if flushing If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING instead from finish_frame(). https://bugzilla.gnome.org/show_bug.cgi?id=701763 2013-06-30 18:16:35 +0200 Sebastian Dröge * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: Don't return not-negotiated if flushing If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING instead from finish_frame(). https://bugzilla.gnome.org/show_bug.cgi?id=701763 2013-06-14 07:23:40 +0200 Edward Hervey * gst-libs/gst/pbutils/descriptions.c: * tests/check/libs/pbutils.c: pbutils: descriptions: Allow smart codec tag handling We already have internally the information on what type of stream (audio, video, container, subtitle, ...) a certain caps is. Instead of forcing callers to specify which CODEC_TAG category a certain caps is, use that information to make a smart choice. Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list (if tag is specified it will be used, if caps is invalid it will be rejected, ...). https://bugzilla.gnome.org/show_bug.cgi?id=702215 2013-06-19 09:25:48 +0200 Edward Hervey * gst-libs/gst/tag/gstxmptag.c: xmptag: Add a debug category Instead of using the default category 2013-06-27 12:23:27 +0200 Patricia Muscalu * gst/videotestsrc/gstvideotestsrc.c: videotestsrc: do not leak lines Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703177 2013-06-26 14:36:17 +0200 Ognyan Tonchev * gst-libs/gst/rtp/gstrtpbasepayload.c: rtpbasepayload: Do not leak the event when segment is delayed Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703119 2013-06-26 15:03:05 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: make read uncancelable when reading a message When we start to read a message, we need to continue reading until the end of the message or else we lose track and cause parse errors. Use a variable may_cancel to avoid cancelation after we read the first byte until we have the complete message. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703088 2013-06-21 20:41:15 +0200 Mathieu Duponchelle * gst-libs/gst/audio/gstaudiodecoder.c: audiodecoder: Don't return not-negotiated if flushing If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING. https://bugzilla.gnome.org/show_bug.cgi?id=701763 2013-06-23 12:07:41 +0200 Sebastian Dröge * ext/ogg/gstoggstream.c: ogg: The Daala headers are little endian, not big endian 2013-06-23 10:30:02 +0200 Sebastian Dröge * ext/ogg/gstoggmux.c: * ext/ogg/gstoggstream.c: ogg: Add Daala support 2013-06-21 19:04:43 +0200 Sebastian Dröge * gst-libs/gst/pbutils/descriptions.c: pbutils: Add VP9 description 2013-06-17 08:58:13 +0200 Edward Hervey * gst-libs/gst/video/gstvideodecoder.c: videodecoder: Fix drop frame handling at startup In the unlikely case that the decoder drops a frame before the first input frame is outputted, use the input segment (since it wasn't carried over to the output segment yet) https://bugzilla.gnome.org/show_bug.cgi?id=702502 2013-06-21 11:50:33 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: dispatch when initial buffer has data When we have data in the inital buffer, dispath the read function to read it even if the socket has no data to read. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702652 2013-06-20 17:28:46 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: manage writer child source better Only add the write child source when we have something to write or else we will dispatch forever without doing anything. 2013-06-19 13:21:45 +0200 Jonas Holmberg * gst-libs/gst/audio/gstaudioencoder.c: audioencoder: unref before memset Unref allocator and input_caps in encoder context before memsetting the context. 2013-06-19 09:22:50 +0200 Edward Hervey * gst-libs/gst/tag/gstxmptag.c: xmptag: More efficient GSList usage Instead of constantly appending (which gets more and more expensive), just prepend to the list (O(1)) and reverse the list before usage. https://bugzilla.gnome.org/show_bug.cgi?id=702545 2013-06-16 22:39:30 +0200 Branko Subasic * gst-libs/gst/rtp/gstrtpbuffer.c: * gst-libs/gst/rtp/gstrtpbuffer.h: * tests/check/libs/rtp.c: rtpbuffer: add gst_rtp_buffer_get_payload_bytes The function gst_rtp_buffer_get_payload can not be used in Python because it lacks necessary length parameter. This patch adds a new function, gst_rtp_buffer_get_payload_bytes, to use from Python bindings. The new function has the advisory "Rename to:" annotation so it can replace the gst_rtp_buffer_get_payload whan creating bindings. The function gst_rtp_buffer_get_extension_bytes is also added. It wraps gst_rtp_buffer_get_extension_data which doesn't work in Python due to incomplete annotation and because it returns the length as number of 32-bit words. https://bugzilla.gnome.org/show_bug.cgi?id=698562 2013-06-17 16:34:26 +0200 Ognyan Tonchev * gst-libs/gst/audio/gstaudiobasesrc.c: audiobasesrc: add 2 missing gst_buffer_unmap () calls There are 2 missing calls to gst_buffer_unmap () in the error handling in create (). Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467 2013-06-17 16:02:41 +0300 Sreerenj Balachandran * gst/playback/gstplaysink.c: playsink: Fix the block diagram of deinterlace bin. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702465 2013-06-13 11:08:20 -0600 Brendan Long * gst/playback/gstplaybin2.c: playbin: Emit {audio,text,video}-changed signals when pads are removed https://bugzilla.gnome.org/show_bug.cgi?id=702195 2013-06-11 15:22:50 +0200 Sebastian Dröge * gst/videoconvert/videoconvert.c: videoconvert: Fix leaking of the chroma resample helper objects 2013-06-10 14:43:35 +0300 Sreerenj Balachandran * tests/check/Makefile.am: * tests/check/elements/playbin-complex.c: tests: add more unit test for playbin Add unit test for autoplugging of video_decoder/video_sink combination based on capsfeatures. 2013-06-10 15:31:38 +0200 Sebastian Dröge * gst-libs/gst/rtsp/gstrtspconnection.c: rtspconnection: Make sure to set a sensible default port for the GSocketConnection Otherwise it will connect to port 0 if no port is given in the URI. https://bugzilla.gnome.org/show_bug.cgi?id=701798 2013-06-09 19:20:20 +0200 Sebastian Dröge * gst/adder/gstadder.c: adder: Reject segments that have a different rate than the output segment adder does no rate conversion. 2013-06-08 23:51:13 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: When activating a fixed sink, proxy error messages too If activating a fixed sink fails, everything will fail later anyway and we can just error out early. 2013-06-08 23:34:53 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Improve autoplugging of decoder/sink combinations by trying to activate the sink And if that fails don't bother autoplugging that sink. Also gives us more accurate sink caps. 2013-06-08 23:08:05 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Proxy the playbin context to the sinks 2013-06-08 23:04:43 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin: Proxy sink messages if we activate a sink in playbin already This makes sure the application gets any context related messages and can do whatever is required to a) get the sink a context or b) share the context with other elements in the pipeline. The proxying is necessary because the sink is not a child element of playbin, but instead will at a later point be a child of some bin inside playsink. https://bugzilla.gnome.org/show_bug.cgi?id=700967 2013-06-06 15:57:49 +0200 Sebastian Dröge * gst/playback/gstdecodebin2.c: decodebin: Let serialize queries before caps events through Otherwise we're going to deadlock forever because no autoplugging happens without having caps, but caps can never be send because we're blocking. Serialized queries before caps should never be sent unless really necessary. 2013-06-05 18:36:40 +0200 Sebastian Dröge * configure.ac: Back to development