From 5f98203bd753c32666c8fa7a2fde6d186c2a4247 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Sebastian=20Dr=C3=B6ge?= Date: Thu, 24 Dec 2015 13:59:15 +0100 Subject: [PATCH] Release 1.7.1 --- ChangeLog | 1705 ++++++++++++++++- NEWS | 64 +- RELEASE | 106 +- configure.ac | 6 +- docs/plugins/inspect/plugin-adder.xml | 4 +- docs/plugins/inspect/plugin-alsa.xml | 4 +- docs/plugins/inspect/plugin-app.xml | 4 +- docs/plugins/inspect/plugin-audioconvert.xml | 4 +- docs/plugins/inspect/plugin-audiorate.xml | 4 +- docs/plugins/inspect/plugin-audioresample.xml | 4 +- docs/plugins/inspect/plugin-audiotestsrc.xml | 4 +- docs/plugins/inspect/plugin-cdparanoia.xml | 4 +- docs/plugins/inspect/plugin-encoding.xml | 4 +- docs/plugins/inspect/plugin-gio.xml | 4 +- docs/plugins/inspect/plugin-libvisual.xml | 4 +- docs/plugins/inspect/plugin-ogg.xml | 4 +- docs/plugins/inspect/plugin-pango.xml | 4 +- docs/plugins/inspect/plugin-playback.xml | 4 +- docs/plugins/inspect/plugin-subparse.xml | 4 +- docs/plugins/inspect/plugin-tcp.xml | 4 +- docs/plugins/inspect/plugin-theora.xml | 4 +- .../inspect/plugin-typefindfunctions.xml | 4 +- docs/plugins/inspect/plugin-videoconvert.xml | 4 +- docs/plugins/inspect/plugin-videorate.xml | 4 +- docs/plugins/inspect/plugin-videoscale.xml | 4 +- docs/plugins/inspect/plugin-videotestsrc.xml | 4 +- docs/plugins/inspect/plugin-volume.xml | 4 +- docs/plugins/inspect/plugin-vorbis.xml | 4 +- docs/plugins/inspect/plugin-ximagesink.xml | 4 +- docs/plugins/inspect/plugin-xvimagesink.xml | 4 +- gst-plugins-base.doap | 30 + win32/common/_stdint.h | 4 +- win32/common/audio-enumtypes.c | 113 +- win32/common/audio-enumtypes.h | 16 + win32/common/config.h | 14 +- win32/common/pbutils-enumtypes.c | 44 + win32/common/pbutils-enumtypes.h | 4 + 37 files changed, 2064 insertions(+), 146 deletions(-) diff --git a/ChangeLog b/ChangeLog index ddf310395..2a5e3d949 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,1710 @@ +=== release 1.7.1 === + +2015-12-24 Sebastian Dröge + + * configure.ac: + releasing 1.7.1 + +2015-12-24 12:22:04 +0100 Sebastian Dröge + + * po/nl.po: + * po/sv.po: + * po/zh_CN.po: + po: Update translations + +2015-12-11 15:38:00 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-profile.c: + encodebin: Implement an encoding profile serialization format + https://bugzilla.gnome.org/show_bug.cgi?id=759356 + +2015-12-21 00:43:49 +0100 Koop Mast + + * configure.ac: + configure: Make -Bsymbolic check work with clang. + Update the -Bsymbolic check with the version glib has. This version + works with clang. + https://bugzilla.gnome.org/show_bug.cgi?id=759713 + +2015-12-03 11:53:05 +0900 Kazunori Kobayashi + + * gst-libs/gst/app/gstappsrc.c: + appsrc: Clear is_eos flag when receiving the flush-stop event + The EOS event can be propagated to the downstream elements when + is_eos flag remains set even after leaving the flushing state. + This fix allows this element to normally restart the streaming + after receiving the flush event by clearing the is_eos flag. + https://bugzilla.gnome.org/show_bug.cgi?id=759110 + +2015-12-16 18:11:05 -0300 Thiago Santos + + * tests/examples/playback/playback-test.c: + examples: playback-test: remove unused variables + audiosink and videosink string variables are unused + +2015-11-30 10:28:55 +1100 Matthew Waters + + * gst/playback/gstplaybin2.c: + playbin: only add the template caps when the result is empty + Unconditionally adding the template caps when proxying the caps query will play + havoc with decoders that attempt to choose an output format based on some caps + features. Creating a sink that does not include those caps features and a + decoder/parser/etc that preferentially chooses some specific caps feature when + available, will always return the decoder/parser/etc template caps and choose a + feature that downstream will be unable to support. + Fix by limiting the addition of the template caps to when the result is actually + empty. + https://bugzilla.gnome.org/show_bug.cgi?id=758212 + +2015-12-17 13:39:01 +0100 Sebastian Dröge + + * configure.ac: + configure: Don't use AG_GST_CHECK_FEATURE for checking for gio-unix-2.0 + It's meant to be used for external plugins that can then all be disabled via + --disable-external. gio-unix-2.0 however is just an optional dependency for + the TCP unit test. + Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part + there needs to be an AM_CONDITIONAL for the feature with FALSE. + +2015-12-16 17:07:54 +0100 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + Revert "decodebin2: fix deadlock on chain shutdown" + This reverts commit 77dc09c3a9a5e5e371e189f39b5557db440a8dc9. + It can cause the FLUSH_START/STOP events to go to the sink elements, which + then causes state changes and various other problems. We shouldn't really + flush downstream here, the idea is to do *draining*. + Apart from that the testcase for the original bug here works without this + commit now. + +2015-12-16 11:12:00 +0000 Luis de Bethencourt + + * gst/tcp/gstmultifdsink.c: + multifdsink: fix typo in GST_WARNING_OBJECT + This should make easier to parse the debug logs. + s/fnctl/fcntl + +2014-04-10 15:36:15 +0100 Vincent Penquerc'h + + * gst/videorate/gstvideorate.c: + videorate: remove dead code + Since the loops increasing count from 0 are always run at least + once (if count < 1), count will always be at least one when + compared to the drop/dup conditions. + Coverity 1139674 + +2015-12-16 10:45:48 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-converter.c: + * gst-libs/gst/audio/audio-converter.h: + * win32/common/libgstaudio.def: + audio-converter: rework the main processing loop + Rework the main processing loop. We now create an audio processing + chain from small core functions. This is very similar to how the + video-converter core works and allows us to statically calculate an + optimal allocation strategy for all possible combinations of operations. + Make sure we support non-interleaved data everywhere. + Add functions to calculate in and out frames and latency. + +2015-12-16 10:44:16 +0100 Wim Taymans + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: clear convert object + +2015-12-16 09:35:38 +0100 Sebastian Dröge + + * docs/plugins/gst-plugins-base-plugins.args: + * docs/plugins/gst-plugins-base-plugins.hierarchy: + * docs/plugins/gst-plugins-base-plugins.signals: + * docs/plugins/inspect/plugin-adder.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-audioconvert.xml: + * docs/plugins/inspect/plugin-audiorate.xml: + * docs/plugins/inspect/plugin-audioresample.xml: + * docs/plugins/inspect/plugin-audiotestsrc.xml: + * docs/plugins/inspect/plugin-cdparanoia.xml: + * docs/plugins/inspect/plugin-encoding.xml: + * docs/plugins/inspect/plugin-gio.xml: + * docs/plugins/inspect/plugin-libvisual.xml: + * docs/plugins/inspect/plugin-ogg.xml: + * docs/plugins/inspect/plugin-pango.xml: + * docs/plugins/inspect/plugin-playback.xml: + * docs/plugins/inspect/plugin-subparse.xml: + * docs/plugins/inspect/plugin-tcp.xml: + * docs/plugins/inspect/plugin-theora.xml: + * docs/plugins/inspect/plugin-typefindfunctions.xml: + * docs/plugins/inspect/plugin-videoconvert.xml: + * docs/plugins/inspect/plugin-videorate.xml: + * docs/plugins/inspect/plugin-videoscale.xml: + * docs/plugins/inspect/plugin-videotestsrc.xml: + * docs/plugins/inspect/plugin-volume.xml: + * docs/plugins/inspect/plugin-vorbis.xml: + * docs/plugins/inspect/plugin-ximagesink.xml: + * docs/plugins/inspect/plugin-xvimagesink.xml: + docs: update to git + +2015-12-14 13:59:02 -0500 Nicolas Dufresne + + * ext/alsa/gstalsasrc.c: + Revert "alsasrc: Disable HW timestamp" + This reverts commit 3642e9a3913a35c00f379034780c27298d09929c. + +2015-11-10 12:54:23 -0500 Xavier Claessens + + * gst-libs/gst/allocators/gstfdmemory.h: + * gst-libs/gst/app/gstappsink.h: + * gst-libs/gst/app/gstappsrc.h: + * gst-libs/gst/audio/audio-info.h: + * gst-libs/gst/audio/gstaudiobasesink.h: + * gst-libs/gst/audio/gstaudiobasesrc.h: + * gst-libs/gst/audio/gstaudiocdsrc.h: + * gst-libs/gst/audio/gstaudioclock.h: + * gst-libs/gst/audio/gstaudiodecoder.h: + * gst-libs/gst/audio/gstaudioencoder.h: + * gst-libs/gst/audio/gstaudiofilter.h: + * gst-libs/gst/audio/gstaudioringbuffer.h: + * gst-libs/gst/audio/gstaudiosink.h: + * gst-libs/gst/audio/gstaudiosrc.h: + * gst-libs/gst/pbutils/encoding-profile.h: + * gst-libs/gst/pbutils/encoding-target.h: + * gst-libs/gst/pbutils/gstdiscoverer.h: + * gst-libs/gst/pbutils/install-plugins.h: + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.h: + * gst-libs/gst/rtp/gstrtpbasedepayload.h: + * gst-libs/gst/rtp/gstrtpbasepayload.h: + * gst-libs/gst/rtsp/gstrtspurl.h: + * gst-libs/gst/sdp/gstmikey.h: + * gst-libs/gst/sdp/gstsdpmessage.h: + * gst-libs/gst/tag/gsttagdemux.h: + * gst-libs/gst/tag/gsttagmux.h: + * gst-libs/gst/video/colorbalancechannel.h: + * gst-libs/gst/video/gstvideodecoder.h: + * gst-libs/gst/video/gstvideoencoder.h: + * gst-libs/gst/video/gstvideofilter.h: + * gst-libs/gst/video/gstvideopool.h: + * gst-libs/gst/video/gstvideosink.h: + * gst-libs/gst/video/gstvideoutils.h: + * gst-libs/gst/video/video-info.h: + * gst-libs/gst/video/video-overlay-composition.h: + base: Add g_autoptr() support to all types + https://bugzilla.gnome.org/show_bug.cgi?id=754464 + +2015-09-24 18:26:51 -0400 Nicolas Dufresne + + * ext/alsa/gstalsasrc.c: + alsasrc: Disable HW timestamp + This is a workaround for broken pulse module. + +2015-12-14 19:03:33 +0100 Sebastian Dröge + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes + +2015-12-14 10:57:19 -0500 Evan Callaway + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Use relative URI for non-proxy tunneled requests + Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we + are using a proxy server. Also, send Host header for compatability with + HTTP/1.1 and some HTTP/1.0 servers. + https://bugzilla.gnome.org/show_bug.cgi?id=758922 + +2015-12-14 09:10:16 -0500 Evan Callaway + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + * win32/common/libgstrtsp.def: + rtspconnection: Support authentication during tunneling setup + gst_rtsp_connection_connect_with_response accepts a response pointer + which it fills with the response from setup_tunneling if the + connection is configured to be tunneled. The motivation for this is to + allow the caller to inspect the response header to determine if + additional authentication is required so that the connection can be + retried with the appropriate authentication headers. + The function prototype of gst_rtsp_connection_connect has been + preserved for compatability with existing code and wraps + gst_rtsp_connection_connect_with_response. + https://bugzilla.gnome.org/show_bug.cgi?id=749596 + +2015-12-14 13:11:21 +0100 Sebastian Dröge + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields + CID 1139615 + +2015-12-10 17:46:26 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-channel-mix.c: + * gst-libs/gst/audio/audio-channel-mix.h: + * gst-libs/gst/audio/audio-converter.c: + * gst-libs/gst/audio/audio-quantize.c: + * gst-libs/gst/audio/audio-quantize.h: + * gst/audioconvert/gstaudioconvert.c: + audio: adapt API for non-interleaved formats + Allow an array of sample blocks to be passed to the channel mix and + quantizer functions to support non-interleaved formats. + +2015-12-10 16:26:40 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-converter.c: + * gst-libs/gst/audio/audio-converter.h: + audio-converter: improve API for non-interleaved formats + Make it possible to pass an array of sample blocks when dealing with + non-interleaved formats. + +2015-12-12 17:49:28 +0100 Luis de Bethencourt + + * gst-libs/gst/riff/riff-media.c: + riff: add FourCC aliases + Support media using the aliases defined in http://www.fourcc.org/ that are + exact duplicates of already known codes. + +2015-12-12 17:04:21 +0100 Luis de Bethencourt + + * gst-libs/gst/riff/riff-media.c: + riff: use defined FourCC + Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h, + like gst_riff_create_audio_caps() does. + +2015-12-11 14:42:09 +0000 Julien Isorce + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: add some debug around pool negotiation + It lets us know easily which pool is activated or + inactivated during the negotiation. + https://bugzilla.gnome.org/show_bug.cgi?id=720597 + +2015-12-11 21:42:00 +0800 Song Bing + + * gst-libs/gst/video/convertframe.c: + video/convertframe: Add crop meta support via videocrop + https://bugzilla.gnome.org/show_bug.cgi?id=759329 + +2015-12-11 11:01:53 +0000 Tim-Philipp Müller + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepay: when setting discont flag make sure rtpbuffer is current + Depayloaders will look at rtpbuffer->buffer for the discont flag. + When we set the discont flag on a buffer in the rtp base depayloader + and we have to make the buffer writable, make sure the rtpbuffer + actually contains the newly-flagged buffer, not the original input + buffer. This was introduced with the addition of the process_rtp_packet + vfunc, but would only trigger if the input buffer wasn't flagged + already and was not writable already. + +2015-12-11 00:18:30 +0000 Tim-Philipp Müller + + * tests/check/libs/rtpbasedepayload.c: + tests: rtpbasedepayload: add test for seqnum gap discont setting + The problem was triggered only when the input buffers were not + writable, so add extra ref to test this code path. + +2015-12-11 10:25:00 +0000 Tim-Philipp Müller + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepay: fix possible refcounting issue when detecting a discont + When we detect a discont and the input buffer isn't already flagged + as discont, handle_buffer() does a gst_buffer_make_writable() on the + input buffer in order to set the flag. This assumed it had ownership + of the input buffer though, which it didn't. This would still work + fine in most scenarios, but could lead to crashes or mini object + unref criticals in some cases when a discont is detected, e.g. when + using pcapparse in front of a depayloader. This problem was + introduced in bc14cdf529e. + +2015-12-10 12:18:04 +0100 Wim Taymans + + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + multisocketsink: add GstNetworkMessage event + Add a property and logic to send a GstNetworkMessage event containing + the message that was received from a client. This can be used to + implement simply bidirectional communication. + +2015-12-10 12:14:37 +0100 Wim Taymans + + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + multisocketsink: add dispatched event + Add a property and logic to send a GstNetworkMessageDispatched + event upstream to notify that a buffer has been sent. This can be used + to keep track of what client received what buffers. + +2015-12-04 11:17:37 +0100 Wim Taymans + + * gst/tcp/gstsocketsrc.c: + * gst/tcp/gstsocketsrc.h: + socketsrc: handle GstNetworkMessage events + Add a property to handle GstNetworkMessage events. These events contain + a buffer that is sent on the socket to allow for simple bidirectional + communication. + +2015-12-09 17:16:26 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-converter.c: + * gst-libs/gst/audio/audio-converter.h: + * gst/audioconvert/gstaudioconvert.c: + audio-convert: improve converter API + Improve the converter API to allow for an max input and output number of + samples and return the number of consumed/produced samples. + +2015-12-08 11:15:34 +0100 Philippe Normand + + * gst-libs/gst/app/gstappsrc.c: + appsrc: duration query support based on the size property + https://bugzilla.gnome.org/show_bug.cgi?id=759126 + +2015-12-07 09:08:05 -0500 Nicolas Dufresne + + * autogen.sh: + * common: + Automatic update of common submodule + From b319909 to 86e4663 + +2015-12-04 12:25:11 +0100 Wim Taymans + + * gst/tcp/gstmultisocketsink.c: + multisocketsink: let downstream know we support metadata + Let downstream know that we support GstNetControlMessage metadata API. + +2015-12-03 16:38:45 +0100 Edward Hervey + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Avoid pushing buffers before segment start + In the case where the stream doesn't have a framerate set and the frames + don't have a duration set, we still want to use the clipping path to + make sure we don't push buffers outside of the segment. + The problem was the previous iteration was setting a duration of 2s, which + meant that any buffer which was less than 2s before the segment start would + end up getting pushed. + Instead, use a saner 40ms (25fps single frame duration) to figure out whether + the frame could be within the segment or not + +2015-12-02 20:19:43 -0800 Reynaldo H. Verdejo Pinochet + + * gst-libs/gst/allocators/Makefile.am: + * gst-libs/gst/app/Makefile.am: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/fft/Makefile.am: + * gst-libs/gst/pbutils/Makefile.am: + * gst-libs/gst/rtp/Makefile.am: + * gst-libs/gst/rtsp/Makefile.am: + * gst-libs/gst/sdp/Makefile.am: + * gst-libs/gst/tag/Makefile.am: + * gst-libs/gst/video/Makefile.am: + Drop usage of deprecated g-ir-scanner --strip-prefix flag + +2015-12-02 18:16:05 +0000 Tim-Philipp Müller + + * gst/playback/gstdecodebin2.c: + decodebin2: fix "Attempt to unlock mutex that was not locked" + Introduced in commit ee44337f, caused the decodebin + test_text_plain_streams unit test to abort. + https://bugzilla.gnome.org/show_bug.cgi?id=752651 + +2015-11-16 14:50:58 +0100 Edward Hervey + + * gst/playback/gstrawcaps.h: + playback: Expose XSUB formats by default + This is a workaround, we should remove this once we have a proper + decoder + +2015-11-16 14:50:30 +0100 Edward Hervey + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: Also consider XSUB as a subtitle format + +2015-11-16 14:49:55 +0100 Edward Hervey + + * gst-libs/gst/pbutils/descriptions.c: + pbutils: Add description for XSUB subpicture format + +2015-11-16 14:49:19 +0100 Edward Hervey + + * gst-libs/gst/riff/riff-media.c: + riff: 'DXSA' is the same as 'DXSB' + Which is subpicture/x-xsub + +2015-07-21 09:58:56 +0200 Edward Hervey + + * gst/playback/gststreamsynchronizer.c: + streamsynchronizer: Rename GstStream => GstSyncStream + Avoid clashes with future GstStream from core + +2015-12-02 09:00:31 -0500 Evan Callaway + + * gst-libs/gst/rtsp/gstrtspdefs.c: + * gst-libs/gst/rtsp/gstrtspdefs.h: + rtspconnection: Update capitalization of x-sessioncookie + Some servers incorrectly parse header names with strict case-sensitivity. For + compatibility with these systems change X-Sessioncookie to x-sessioncookie. + https://bugzilla.gnome.org/show_bug.cgi?id=758921 + +2015-12-02 16:16:22 +0200 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + decodebin: Update buffering messages when removing an element that had buffering pending + Otherwise we'll remove that element while keeping its buffering message in our + list, and because of that never ever report buffering 100% as that element + will always be at a lower percentage. + This fixes e.g. seeking over Period boundaries in DASH and various other + issues when buffering happens between group switches. + Also use a new mutex for protecting the buffering messages. The object lock is + already used by gst_object_has_as_ancestor() and we need to use it now for + checking if the buffering message sender has the to-be-removed element as + ancestor. + +2015-12-02 09:52:19 +0100 Wim Taymans + + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + multisocketsink: keep on reading when we stop sending + When we stop sending because we need more data, still keep a GSource + around to receive data from the clients. + Also handle read and write in the same go. + +2015-12-01 19:57:10 +0200 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Post latency message on the bus after set_caps() + The latency is only known once the caps are known, and might change + whenever the caps are changing. + https://bugzilla.gnome.org/show_bug.cgi?id=758911 + +2015-09-25 14:47:48 +0200 Michael Olbrich + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: Post latency message on the bus after set_caps() + Any latency query before this will not get the correct latency so a new + latency query should be triggered once the audio sink know its own latency. + Without this the initial latency query from the pipeline arrives too early + sometimes and the resulting latency is too short. + https://bugzilla.gnome.org/show_bug.cgi?id=758911 + +2015-11-06 14:21:14 +0000 Thomas Bluemel + + * gst/playback/gstdecodebin2.c: + [PATCH] Fix a race condition accessing the decode_chain field. + Make sure that any access to the GstDecodeBin's decode_chain + field is protected using the EXPOSE_LOCK. Also add a simple + reference counter to the GstDecodeChain structure so that when + the type_found signal fires it can hold onto the decode chain + even while the EXPOSE_LOCK is not held. This should fix a + race condition if the type_found signal fires right in the + middle of a state change that messes with the same decode + chain. + https://bugzilla.gnome.org/show_bug.cgi?id=755260 + +2015-08-20 17:30:38 +0100 Vincent Penquerc'h + + * gst/playback/gstdecodebin2.c: + decodebin: early out on pad-added when the pad is inactive + The pad may be recently deactivated if the element is switched + back down very quickly. + https://bugzilla.gnome.org/show_bug.cgi?id=752651 + +2015-08-20 17:29:36 +0100 Vincent Penquerc'h + + * gst/playback/gstdecodebin2.c: + decodebin: lock the expose lock around decode_chain use + Helps with a crash in decodebin when quickly switching states. + https://bugzilla.gnome.org/show_bug.cgi?id=752651 + +2015-11-28 14:24:55 +0000 Luis de Bethencourt + + * gst-libs/gst/pbutils/codec-utils.c: + codec-utils: accept wrong version field in OpusHead header + Some Opus files found on the wild have 0 in the version field of the + OpusHead header, instead of the correct value of 1. The files still + play, don't make this error fatal. + https://bugzilla.gnome.org/show_bug.cgi?id=758754 + +2015-11-26 11:33:02 +0000 William Manley + + * gst-libs/gst/allocators/gstfdmemory.c: + allocators: add debug category for fd memory and allocator + Debugging can now be viewed by setting GST_DEBUG=fdmemory:9 + https://bugzilla.gnome.org/show_bug.cgi?id=758744 + +2015-11-20 20:18:34 +0000 Tim-Philipp Müller + + * tests/check/libs/tag.c: + tests: tags: add unit test for ID3v2 PRIVATE_DATA tag extraction + https://bugzilla.gnome.org/show_bug.cgi?id=730926 + +2014-09-29 14:17:39 +0530 Ravi Kiran K N + + * gst-libs/gst/tag/gstid3tag.c: + * gst-libs/gst/tag/id3v2frames.c: + id3v2frames: Handle private frames + Handle PRIV ID3 tag having owner information (string) + and binary data, add to tag messages list. + https://bugzilla.gnome.org/show_bug.cgi?id=730926 + +2015-11-20 19:15:22 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/id3v2.c: + tags: id3: make sure to register private-id3v2-frame tag before using it + +2015-11-17 17:07:37 +0100 Ognyan Tonchev + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * tests/check/libs/rtspconnection.c: + rtspconnection: Add support for parsing custom headers + https://bugzilla.gnome.org/show_bug.cgi?id=758235 + +2015-11-15 02:58:54 -0800 Reynaldo H. Verdejo Pinochet + + * gst-libs/gst/pbutils/encoding-profile.c: + * gst-libs/gst/pbutils/encoding-target.c: + * gst-libs/gst/rtsp/gstrtspmessage.c: + * gst-libs/gst/sdp/gstsdpmessage.c: + * tests/examples/encoding/encoding.c: + Remove unnecessary NULL checks before g_free() + g_free() is NULL-safe + +2015-11-17 09:06:34 +0900 Vineeth TM + + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + xvimagesink/ximagesink: Fix structure memory leak + https://bugzilla.gnome.org/show_bug.cgi?id=758204 + +2015-11-12 14:39:17 +0000 Luis de Bethencourt + + * gst-libs/gst/pbutils/codec-utils.c: + codec-utils: guint8 can't hold value over 255 + channels is a guint8, so the max value is 255 and checking if it value is + > 256 will never be false. + CID 1338687, CID 1338688 + +2015-11-12 14:18:03 +0000 Luis de Bethencourt + + * gst-libs/gst/audio/audio-converter.c: + audio-converter: remove unneeded check for unsigned < 0 + Commit ff6d1a2a25b247688f38e117782a6b43d525706a changed sample's type from + gint to gsize (and renamed it to in_samples). gsize is an unsigned long, + which means it can never be a negative value and the check making sure that + in_samples is >= 0 is never going to be false. Removing it. + CID 1338689 + +2015-11-11 14:44:55 +0900 Vineeth TM + + * tests/check/libs/video.c: + tests:video: Fix overlay rectangle and buffer leak + Created overlay rectangle is not being freed in video tests + pix2 buffer is being created and not freed + https://bugzilla.gnome.org/show_bug.cgi?id=757927 + +2015-11-11 14:37:21 +0900 Vineeth TM + + * gst-libs/gst/pbutils/encoding-target.c: + pbutils:encoding-target: Fix string memory leak + https://bugzilla.gnome.org/show_bug.cgi?id=757926 + +2015-11-11 15:02:39 +0900 Vineeth TM + + * gst-libs/gst/audio/audio-quantize.c: + audio-quantize: Fix dither_buffer memory leak + https://bugzilla.gnome.org/show_bug.cgi?id=757928 + +2015-11-11 00:59:16 +1100 Jan Schmidt + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: Re-init on new caps + If we get new input caps, then reset the decoder + ready for new headers and fresh data. Makes + chained oggs work when reusing the decoder. + +2015-11-02 23:12:19 +1100 Matthew Waters + + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/gstvideoaffinetransformationmeta.c: + * gst-libs/gst/video/gstvideoaffinetransformationmeta.h: + * win32/common/libgstvideo.def: + videometa: add GstVideoAffineTransformationMeta + Adds a simple 4x4 affine transformations meta for passing arbitrary + transformations on buffers. + Based on patch by Matthieu Bouron + https://bugzilla.gnome.org/show_bug.cgi?id=731791 + +2015-11-10 09:52:24 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-converter.c: + * gst-libs/gst/audio/audio-converter.h: + * gst/audioconvert/gstaudioconvert.c: + audio-converter: add output size argument + Make it possible to have a different number of output samples than input + samples when we, for example, want to add resampling later. + +2015-11-07 00:43:55 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: Check API arguments and assert if needed + +2015-11-06 19:31:47 +0100 Edward Hervey + + * gst/playback/gstdecodebin2.c: + decodebin: Properly deactivate ghostpads + Just setting the ghostpad as flushing wasn't enough. It needs to be + consistent on the internal proxypad also, otherwise you end up in + situations where: + * a pending buffer on the target pad triggers the sticky event + propagation + * the default implementation sees that the proxypad is not flushing, + so it tries to push it to the other pad (the actual ghostpad) + * the ghostpad is flushing, so returns FALSE + * the push_event function sees that pushing the event failed... + * ... and pending buffer push returns GST_FLOW_ERROR, instead of + GST_FLOW_FLUSHING + By using gst_pad_set_active(FALSE), we ensure that both the ghostpad + and the proxypad are flushing/deactivated. The situation above will + no longer occur, and a GST_FLOW_FLUSHING will be returned. + +2015-11-06 18:11:41 +0000 Tim-Philipp Müller + + * gst/audioconvert/gstaudioconvertorc-dist.c: + * gst/audioconvert/gstaudioconvertorc-dist.h: + * gst/audioconvert/gstaudioconvertorc.orc: + * gst/audioconvert/plugin.c: + audioconvert: fix build + Don't include file that is no longer generated, and remove some + files that are no longer needed because they have moved into the + lib. Fixes distcheck. + +2015-11-06 18:00:41 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-converter.c: + audio-converter: require interleaved samples and no resampling + We can't yet do resampling or anything other than interleaved audio. + +2015-11-06 17:54:21 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiopack-dist.c: + * gst-libs/gst/audio/gstaudiopack-dist.h: + audio: update ORC dist files + +2015-11-06 17:49:00 +0100 Wim Taymans + + * docs/plugins/Makefile.am: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/audio-converter.c: + * gst-libs/gst/audio/audio-converter.h: + * gst-libs/gst/audio/audio.h: + * gst-libs/gst/audio/gstaudiopack.orc: + * gst/audioconvert/Makefile.am: + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvert.h: + * tests/check/Makefile.am: + * win32/common/libgstaudio.def: + audio-converter: move audio converter to audio libs + Move the audio-converter helper to the audio library. + +2015-11-06 17:39:33 +0100 Wim Taymans + + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/audio-channel-mix.c: + * gst-libs/gst/audio/audio-channel-mix.h: + * gst-libs/gst/audio/audio.h: + * gst/audioconvert/Makefile.am: + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvert.c: + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/gstchannelmix.h: + * win32/common/libgstaudio.def: + audio-channel-mix: move channel mixer to audio libs + Move the channel mixer code to the audio library + +2015-11-06 17:29:22 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-channels.c: + * gst-libs/gst/audio/audio-info.c: + * gst-libs/gst/audio/audio.c: + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/gstaudioconvert.c: + * gst/audioconvert/gstchannelmix.c: + audio: add debug categories + +2015-11-06 16:42:35 +0100 Wim Taymans + + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/gstchannelmix.h: + channelmix: don't limit channelpositions + Don't set a limit on the channel positions, just like the metadata. + +2015-11-06 16:03:20 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/gstchannelmix.h: + channelmix: simplify API a little + Remove the format and layout from the mix_samples function and use the + format when creating the channel mixer object. Also use a flag to handle + the unlikely case of non-interleaved samples like we do elsewhere. + +2015-11-06 15:50:34 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/gstchannelmix.h: + channelmix: GstChannel -> GstAudioChannel + Rename GstChannel to GstAudioChannel + +2015-11-06 13:02:19 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-quantize.c: + * gst-libs/gst/audio/audio-quantize.h: + audio-quantize: update docs + Update docs + Add another flag for the quantizer + +2015-11-06 12:46:36 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvert.c: + * gst/audioconvert/gstaudioconvertorc.orc: + * gst/audioconvert/gstchannelmix.c: + audioconvert: cleanups and add some docs + Add docs for the internal audioconvert object before moving it to the + audio library. + Remove get_sizes and implement the trivial logic in the element. + Remove some unused orc functions + +2015-11-06 12:46:12 +0100 Wim Taymans + + * win32/common/libgstaudio.def: + defs: update defs + +2015-11-06 12:37:14 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiopack-dist.c: + * gst-libs/gst/audio/gstaudiopack-dist.h: + audio: update orc files + +2015-11-06 12:10:48 +0100 Wim Taymans + + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/audio-quantize.c: + * gst-libs/gst/audio/audio-quantize.h: + * gst-libs/gst/audio/audio.h: + * gst-libs/gst/audio/gstaudiopack.orc: + * gst/audioconvert/Makefile.am: + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvert.c: + * gst/audioconvert/gstaudioconvert.h: + * gst/audioconvert/gstaudioquantize.c: + * gst/audioconvert/gstaudioquantize.h: + * gst/audioconvert/gstfastrandom.h: + audioconvert: move audio quantize code to libs + Move the audio quantize code from audioconvert to the audio library. + work on making an audio converter helper function similar to the video + converter. + Fold fastrandom directly into the quantizer, add some ORC code to + optimize this later. + +2015-11-05 12:42:56 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-channels.c: + * gst-libs/gst/audio/audio-channels.h: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst/audioconvert/gstaudioconvert.c: + * win32/common/libgstaudio.def: + audio-channels: rename get_default_mask + Rename _get_default_mask() to _get_fallback_mask() to make it more + clear that the function only provides a fallback if nothing else can be + done. Also clarify this in the documentation. + API: gst_audio_channel_get_fallback_mask() + +2015-11-05 11:34:07 +0100 Thibault Saunier + + * gst/volume/gstvolume.c: + volume: Do not try to get binding value array if we are not processing any sample + In some conditions we might process empty buffers, calling + gst_control_binding_get_value_array in that case will lead + to the assertion: + (lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed + +2015-11-05 10:40:18 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-channels.c: + * gst-libs/gst/audio/audio-channels.h: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst/audioconvert/gstaudioconvert.c: + * win32/common/libgstaudio.def: + audio-channels: make method to get default channel-mask + Add a new method to get the default channel-mask. + Use the new method on audiodecoder and audioconvert. + API: gst_audio_channel_get_default_mask() + +2014-11-10 11:11:37 +0100 Andreas Frisch + + * tests/check/libs/video.c: + tests: Add a test for video blending over transparent frames + And fix the test_overlay_blend test where we blend over a + transparent frame and where expecting wrong results + https://bugzilla.gnome.org/show_bug.cgi?id=681447 + +2013-11-30 01:59:55 +0100 Arnaud Vrac + + * gst-libs/gst/video/video-blend.c: + video: blend using OVER operation + Also support all premultiplied/non-premultiplied source/destination + configurations + https://bugzilla.gnome.org/show_bug.cgi?id=681447 + +2015-11-03 16:51:47 +0200 Sebastian Dröge + + * ext/ogg/gstoggstream.c: + oggdemux: Create full Opus caps with all fields + https://bugzilla.gnome.org/show_bug.cgi?id=757152 + +2015-11-03 18:30:09 +0200 Sebastian Dröge + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/pbutils/Makefile.am: + * gst-libs/gst/pbutils/codec-utils.c: + * gst-libs/gst/pbutils/codec-utils.h: + * win32/common/libgstpbutils.def: + codec-utils: Add utilities for Opus caps and the OpusHead header + https://bugzilla.gnome.org/show_bug.cgi?id=757152 + +2015-11-03 11:11:57 +0200 Sebastian Dröge + + * ext/ogg/gstoggmux.c: + oggmux: Use GstAudioClippingMeta for Opus for accurate end clipping + ... instead of relying on the segment. For the clipping at the start we assume + a proper value in the OpusHead, as generated by opusparse or opusenc. + Transmuxing in general is not guaranteed to produce the correct values, or + even have a OpusHead (e.g. when having RTP input). + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-11-03 10:58:35 +0200 Sebastian Dröge + + * ext/ogg/Makefile.am: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstoggstream.h: + oggdemux: Add GstAudioClippingMeta for Opus for accurate start/end clipping + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-11-02 16:19:42 +0200 Sebastian Dröge + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/audio/audio.h: + * gst-libs/gst/audio/gstaudiometa.c: + * gst-libs/gst/audio/gstaudiometa.h: + * win32/common/libgstaudio.def: + audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-11-02 11:19:23 +0200 Sebastian Dröge + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstoggstream.h: + oggdemux: Allow start clipping for Opus + The granulepos does not have the pre-skip subtracted while timestamps do, + and the last granulepos will be shorter by the number of samples that should + be dropped because of padding in the end. + As such, extrapolating the granule of the beginning of the first frame will + lead to a negative value, which is not a problem but intentional. + https://bugzilla.gnome.org/show_bug.cgi?id=757153 + +2015-11-03 16:38:09 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudiopack-dist.c: + * gst-libs/gst/audio/gstaudiopack-dist.h: + audio: update disted orc backup files + +2015-11-03 14:08:25 +0000 Luis de Bethencourt + + * gst-libs/gst/audio/gstaudioclock.c: + audioclock: use GST_STIME_FORMAT for GstClockTimeDiff + GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can + handle negative values better. + https://bugzilla.gnome.org/show_bug.cgi?id=757480 + +2015-11-03 13:44:39 +0000 Luis de Bethencourt + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Print GstClockTimeDiff as a signed integer in debug logs + +2015-11-03 11:59:09 +0100 Wim Taymans + + * gst-libs/gst/audio/audio-format.c: + * gst-libs/gst/audio/audio-format.h: + * gst-libs/gst/audio/gstaudiopack.orc: + * gst/audioconvert/audioconvert.c: + audio-format: add TRUNCATE_RANGE flag + Add a TRUNCATE_RANGE flag for unpack functions to fill the least + significate bits with 0 (as did the old code). Also add functions + that don't truncate. Use the TRUNC flag in audioconvert for + backwards compatibility for now. + +2015-11-03 11:57:32 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiopack.orc: + audiopack: improve pack functions + Avoid shifts by using convh functions. + +2015-11-03 11:44:54 +0100 Wim Taymans + + * gst/audioconvert/gstaudioconvertorc.orc: + * tests/check/elements/audioconvert.c: + audioconvert: change multiplier for int<->float conversion + Use (1 << 31) as the multiplier for int<->float conversions. This makes + sure that int->float conversions always end up with floats between + [-1.0, 1.0]. + For the conversion from float to int, this multiplier will give the complete + int range after we perform clipping. + Change the unit test to take this into consideration. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301 + +2015-11-02 17:32:55 +0000 Luis de Bethencourt + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: use GST_STIME_ARGS for GstClockTimeDiff + No need to use G_GINT64_FORMAT for potentially negative values of + GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS. + Plus it creates more readable values in the logs. + https://bugzilla.gnome.org/show_bug.cgi?id=757480 + +2015-11-02 16:36:35 +0000 Luis de Bethencourt + + * ext/ogg/gstoggmux.c: + oggmux: Print GstClockTimeDiff as a signed integer in debug logs + +2015-11-02 16:09:52 +0000 Luis de Bethencourt + + * ext/ogg/gstoggdemux.c: + oggdemux: Use GstClockTimeDiff and print signed integer in debug logs + Use GstClockTimeDiff and Clock macros to print signed integer time + differences in the debug logs. + https://bugzilla.gnome.org/show_bug.cgi?id=757480 + +2015-11-02 14:06:39 +0000 Luis de Bethencourt + + * tests/examples/seek/scrubby.c: + examples: use GST_STIME_FORMAT for GstClockTimeDiff + GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can + handle negative values better. + https://bugzilla.gnome.org/show_bug.cgi?id=757480 + +2015-11-02 17:14:51 +0200 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiometa.h: + audio: Fix parameters to gst_buffer_get_audio_downmix_meta() in macro + +2015-11-02 15:54:19 +0100 Wim Taymans + + * gst/audiotestsrc/gstaudiotestsrc.c: + audiotestsrc: increase freq limit + Raise the frequency limit and try to negotiate to a samplerate of 4*freq + when larger then the default samplerate. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450 + +2015-11-02 15:46:22 +0100 Wim Taymans + + * gst/audiotestsrc/gstaudiotestsrc.c: + audiotestsrc: add support for unlimited number of channels + Raise the channel limit and set the channel-mask for > 2 channels. + +2015-11-02 13:19:09 +0100 Wim Taymans + + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/audiotestsrc/gstaudiotestsrc.h: + audiotestsrc: add support for all formats + Use the pack functions to also support the other audio formats we + have. + +2015-11-02 12:09:42 +0000 Luis de Bethencourt + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: subtract time difference with GST_CLOCK_DIFF + To ensure the subtraction of two GstClockTime values (which are guint64) + can be negative. Use GST_CLOCK_DIFF which returns a gint64. + CID 1338049 + +2015-11-02 11:34:56 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: Do not force user to provide an encoding profile name + And use the profile called `default` if none provided. + +2015-11-02 11:30:07 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-target.c: + encoding-target: Do not unconditionally break when searching for a target + Otherwise the loop is useless! + Fixes CID 1338051 + +2015-10-24 20:08:47 +0300 Sebastian Dröge + + * gst/audioresample/gstaudioresample.c: + audioresample: Clip input buffers to the segment before handling them + https://bugzilla.gnome.org/show_bug.cgi?id=757068 + +2015-10-24 20:05:10 +0300 Sebastian Dröge + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: Clip input buffers to the segment before handling them + https://bugzilla.gnome.org/show_bug.cgi?id=757068 + +2015-10-24 20:02:13 +0300 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiofilter.c: + audiofilter: Clip input buffers to the segment before handling them + https://bugzilla.gnome.org/show_bug.cgi?id=757068 + +2015-11-01 23:05:10 +0000 Tim-Philipp Müller + + * gst/audioconvert/gstaudioconvertorc-dist.c: + * gst/audioconvert/gstaudioconvertorc-dist.h: + audioconvert: update orc backup code to fix build without orc + +2015-10-26 21:32:41 +0100 Csaba Toth + + * gst/tcp/gstmultisocketsink.c: + multisocketsink: fix "client-removed" signal on 64-bit platforms and with bindings + The client-removed signal used G_INT_TYPE instead of G_SOCKET_TYPE + in its definition leading to problems on platforms where the size + of a pointer is larger than the size of an integer, It would also + not work at all with dynamic language bindings. + https://bugzilla.gnome.org/show_bug.cgi?id=757155 + +2015-10-28 18:36:41 +0100 Joan Pau Beltran + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: fix handling of Bayer format 'gbrg' + Due to a typo, videotestsrc did not handle the Bayer + format 'gbrg' properly and reported it as invalid, + causing negotiation errors. + https://bugzilla.gnome.org/show_bug.cgi?id=757264 + +2015-10-30 17:36:48 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvertorc.orc: + * gst/audioconvert/gstaudioquantize.c: + * gst/audioconvert/gstaudioquantize.h: + audioconvert: rework audioconvert + Rewrite audioconvert to try to make it more clear what steps are + executed during conversion. + Add passthrough step that just does a memcpy when possible. + Add ORC optimized dither and quantization functions. + Implement noise-shaping on S32 samples only and allow for arbitrary + noise shaping coefficients if we want this later. + +2015-10-30 17:33:32 +0100 Wim Taymans + + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/gstchannelmix.h: + channelmix: fix up API a little + don't use gpointer * for something that should be gpointer. + +2015-10-28 11:40:42 +0100 Wim Taymans + + * gst/audioconvert/gstaudioquantize.c: + audioquantize: make helper for add with saturation + +2015-10-29 16:52:31 +0200 Sebastian Dröge + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Print another time difference as a signed integer instead of a huge unsigned one + +2015-10-29 16:01:26 +0200 Sebastian Dröge + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Print GstClockTimeDiff as a signed integer in debug logs + +2015-10-29 00:01:01 +0530 Nirbheek Chauhan + + * tools/gst-device-monitor.c: + tools: gst-device-monitor: fix two memory leaks + The removed GList link needs to be freed too, and + the G_OPTION_REMAINING arguments need to be freed. + +2015-10-28 15:50:44 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-target.c: + encoding-target: Add a GST_ENCODING_TARGET_PATH envvar to find target files + +2015-10-28 15:47:00 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-target.c: + encoding-target: Allow having encoding target without a category set + There was already some code to handle that, but the support was not + complete in those code paths. + +2015-10-27 12:56:48 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-target.c: + encoding-target: Create directory before trying to save encoding targets + +2015-10-27 12:50:26 +0100 Thibault Saunier + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: Allow specifying the target category in the serialized encoding target + +2015-10-27 17:28:06 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvert.c: + * gst/audioconvert/gstaudioconvert.h: + * gst/audioconvert/gstaudioquantize.c: + * gst/audioconvert/gstaudioquantize.h: + audioconvert: make the quantizer a reusable object + Turn the quantizer into a reusable object. + +2015-10-27 13:24:31 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/gstchannelmix.h: + audioconvert: make the channel mixer a separate reusable object + A first attempt at making the channel mixer a separate object. + +2015-10-28 11:32:57 +0100 Wim Taymans + + * gst/audioconvert/gstaudioquantize.c: + audioquantize: fix 8-pole noise shaping + Fix the 8-pole noise shaping error update. We were mixing errors from + different channels. + +2015-10-27 15:44:06 +0200 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + decodebin: Send SEEK events directly to adaptive streaming demuxers + This makes sure that they will always get SEEK events, even if we're currently + in the middle of a group switch (i.e. switching to another + representation/bitrate/etc). + https://bugzilla.gnome.org/show_bug.cgi?id=606382 + +2015-10-06 15:20:51 +0200 Guillaume Desmottes + + * gst/playback/gstdecodebin2.c: + decodebin: fix event leak + As stated in GST_PAD_PROBE_HANDLED's documentation, we are + supposed to unref the event before returning. + Fixes an event leak in the validate.hls.playback.play_15s.hls_bibbop + validate scenario. + https://bugzilla.gnome.org/show_bug.cgi?id=754459 + +2015-10-23 19:13:05 +0300 Sebastian Dröge + + * gst/audioconvert/gstaudioconvertorc-dist.c: + * gst/audioconvert/gstaudioconvertorc-dist.h: + audioconvert: Update disted orc files + +2015-10-23 16:58:17 +0200 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/audioconvert.h: + * gst/audioconvert/gstaudioconvertorc.orc: + * gst/audioconvert/gstaudioquantize.c: + * gst/audioconvert/gstchannelmix.c: + audioconvert: use pack/unpack functions + Rework the converter to use the pack/unpack functions + Because the unpack functions can only unpack to 1 format, add a separate + conversion step for doubles when the unpack function produces int. + Do conversion to S32 in the quantize function directly. + Tweak the conversion factor for doing float->int conversion slightly to + get the full range of negative samples, use clamp to make sure we don't + exceed our int range on the positive axis (see also #755301) + +2015-10-23 12:02:28 +0300 Sebastian Dröge + + * gst/playback/gstplaybin2.c: + playbin: Send upstream events directly to playsink + Send event directly to playsink instead of letting GstBin iterate + over all sink elements. The latter might send the event multiple times + in case the SEEK causes a reconfiguration of the pipeline, as can easily + happen with adaptive streaming demuxers. + What would then happen is that the iterator would be reset, we send the + event again, and on the second time it will fail in the majority of cases + because the pipeline is still being reconfigured + +2015-10-23 17:25:50 +0900 Eunhae Choi + + * tests/check/gst/typefindfunctions.c: + tests: typefindfunctions: fix error leaks + https://bugzilla.gnome.org/show_bug.cgi?id=757008 + +2015-09-23 18:47:52 +0200 Thibault Saunier + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: Force alpha downstream if foreground color contains alpha + Otherwise the foreground color won't be fully represented in the + outputted frames. + https://bugzilla.gnome.org/show_bug.cgi?id=755482 + +2015-10-22 12:07:44 +0800 Pavel Bludov + + * gst-libs/gst/video/video-overlay-composition.h: + video: overlay-composition: fix rectangle and composition cast macros + Closing parenthesis was missing in two cases. + https://bugzilla.gnome.org/show_bug.cgi?id=756893 + +2015-10-21 14:34:56 +0100 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From b99800a to b319909 + +2015-10-20 17:29:42 +0300 Sebastian Dröge + + * configure.ac: + Use new GST_ENABLE_EXTRA_CHECKS #define + https://bugzilla.gnome.org/show_bug.cgi?id=756870 + +2015-10-21 14:25:47 +0300 Sebastian Dröge + + * README: + * common: + Automatic update of common submodule + From 9aed1d7 to b99800a + +2015-10-20 12:08:23 +0300 Sebastian Dröge + + * gst-libs/gst/rtp/gstrtpbuffer.h: + rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1 + +2015-10-20 03:58:26 +1100 Matthew Waters + + * gst/playback/gstdecodebin2.c: + decodebin: track the exposable pads through connect_pad + The logic introduced by + [d50b713: decodebin: set the decode pad target before setting elements to PAUSED] + to expose pads would only ever be able to possibly expose one (the last) pad per element. + Make it so that any exposable pads are able to be exposed rather than just the + last pad returned by connect_element. + https://bugzilla.gnome.org/show_bug.cgi?id=742924 + +2015-10-20 03:52:24 +1100 Matthew Waters + + * gst/playback/gstdecodebin2.c: + decodebin: return the possibly new chain in analyze_new_pad + In the case of analyzing a demuxer chain, analyze_new_pad may create + a new GstDecodeChain. This was not propagated to the calling function which as + of [d50b713f decodebin: set the decode pad target before setting elements to PAUSED] + is now required to be able to expose the correct pad. + https://bugzilla.gnome.org/show_bug.cgi?id=742924 + +2015-10-19 15:32:19 +0530 Rajat Verma + + * gst/playback/gstplaysink.c: + playsink: relink text_pad in case of reconfiguration + In case of reconfiguration, text_pad should be re-connected with + stream synchronizer sink pad. Otherwise we'll leave an unlinked pad around if + there always was a streamsynchronizer text pad. + https://bugzilla.gnome.org/show_bug.cgi?id=756804 + +2015-09-14 15:25:11 +0900 eunhae choi + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: fix issue about eos handling during flushing + If the flush-start is arrived during _eos_wait() in basesink, + the 'eos' flag is overwritten to TRUE after exiting the _eos_wait(). + To resolve the overwritten issue, + the subclass doing the _eos_wait() call should return the right value. + If the eos flag is set to TRUE again, it will cause error(enter the eos flow) + of the following state changing from PAUSED to PLAYING in basesink. + https://bugzilla.gnome.org/show_bug.cgi?id=754980 + +2015-10-17 22:25:22 +0300 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstsubtitleoverlay.c: + decodebin/playbin/playsink/subtitleoverlay: Post async-done on state change failures + https://bugzilla.gnome.org/show_bug.cgi?id=756611 + +2015-10-17 22:20:31 +0300 Sebastian Dröge + + * gst/playback/gstplaysink.c: + playsink: Immediately error out if state change fails + Otherwise we chain up to the parent class' change_state function and might + override the failure with SUCCESS. + https://bugzilla.gnome.org/show_bug.cgi?id=756611 + +2015-10-17 21:47:07 +0300 Sebastian Dröge + + * gst/playback/gstplaybin2.c: + * gst/playback/gsturidecodebin.c: + playbin/uridecodebin: Always post async-done immediately if we're a live pipeline + Not only if the base class told us, but also if one of our own elements did. + https://bugzilla.gnome.org/show_bug.cgi?id=756611 + +2015-10-16 03:40:43 +1100 Matthew Waters + + * gst/playback/gstdecodebin2.c: + decodebin: set the decode pad target before setting elements to PAUSED + Otherwise caps and context queries will disappear into nothing and therefore + fail. With autoplug-query now actually working, users (such as playbin) can + proxy these queries to the selected video sink and be able to select an + more appropriate configuration. + https://bugzilla.gnome.org/show_bug.cgi?id=731204 + +2015-10-17 20:36:27 +0300 Sebastian Dröge + + * gst-libs/gst/video/video.c: + video: Add out annotations to the out parameters of gst_video_calculate_display_ratio() + https://bugzilla.gnome.org/show_bug.cgi?id=754567 + +2015-10-16 10:48:50 +1100 Matthew Waters + + * win32/common/libgstrtp.def: + win32 update exports for new rtp symbols + +2015-07-22 11:31:05 +0200 Stian Selnes + + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/rtp/gstrtpbuffer.h: + * tests/check/libs/rtp.c: + rtpbuffer: Add map flag to skip padding + Encrypted RTP buffers may contain encrypted padding, hence it's + necessary to have an option to relax the validation in order to + successfully map the buffer. + When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set + gst_rtp_buffer_map() will map the buffer like if padding is not + present. + https://bugzilla.gnome.org/show_bug.cgi?id=752705 + +2015-10-15 22:40:50 +0300 Sebastian Dröge + + * gst-libs/gst/rtp/gstrtpbuffer.c: + Revert "rtpbuffer: increase logging level when map fails" + This reverts commit e3c8a820176ba39dfae85944fa9c6ae202ec681d. + It causes too much noise in the logs. + +2015-10-15 15:32:58 +0200 Miguel París Díaz + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtpbuffer: increase logging level when map fails + https://bugzilla.gnome.org/show_bug.cgi?id=756641 + +2015-10-15 10:01:38 +0900 Vineeth TM + + * gst/playback/gstplaysink.c: + playsink: Fix volume element leak + In case sink implements a streamvolume interface, volume element is being got + from the sink. But this is transfer full. So the memory should be freed before + setting it to NULL. This was resulting in major memory leaks + https://bugzilla.gnome.org/show_bug.cgi?id=755867 + +2015-10-14 00:32:11 +0300 Sebastian Dröge + + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + alsa: Use 8 bit pointer type for byte-based pointer arithmetic + Usually these loops only run once, so there's no problem here. But sometimes + they run twice, and by adding the number of bytes to a 16 bit pointer type we + would advance twice as much as we should. + Also use snd_pcm_frames_to_bytes() in alsasrc to calculate + the number of bytes to skip, same as we do in alsasink. + Thanks to Lucio A. Hernandez for reporting. + +2015-10-12 14:02:58 +0300 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioencoder.c: + * tests/check/libs/audioencoder.c: + Revert "audioencoder: timestamp headers same as first buffer and use duration 0" + This reverts commit dd4d6d9ed54c2a63a7e45661519d9965417707c5. + It breaks ogg muxing and the vorbisenc unit test. + +2015-08-28 11:44:19 +0200 Havard Graff + + * gst-libs/gst/audio/gstaudioencoder.c: + * tests/check/libs/audioencoder.c: + audioencoder: timestamp headers same as first buffer and use duration 0 + https://bugzilla.gnome.org/show_bug.cgi?id=754224 + +2015-08-28 11:25:22 +0200 Havard Graff + + * tests/check/libs/audioencoder.c: + audioencoder-tests: port to use GstHarness + https://bugzilla.gnome.org/show_bug.cgi?id=754223 + +2015-08-27 17:28:30 +0200 Havard Graff + + * tests/check/libs/audiodecoder.c: + audiodecoder-test: port to using GstHarness + https://bugzilla.gnome.org/show_bug.cgi?id=754196 + +2015-10-04 18:36:00 +0100 Sebastian Dröge + + * sys/xvimage/xvimagepool.c: + xvimagesink: Put error message into debug output instead of just throwing it away + +2015-10-02 22:19:52 +0300 Sebastian Dröge + + * configure.ac: + * gst-libs/gst/rtsp/gstrtspconnection.c: + Update GLib dependency to 2.40.0 + +2014-03-15 17:35:56 +0100 Sebastian Rasmussen + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * tests/check/libs/rtpbasepayload.c: + rtpbasepayload: Implement video SDP attributes + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726472 + +2015-09-25 15:17:53 +0300 Vivia Nikolaidou + + * tools/gst-play.c: + gst-play: Removed erroneous comment + The "fall through" comment was wrong. Removed. + https://bugzilla.gnome.org/show_bug.cgi?id=755440 + +2015-09-22 23:12:10 +0300 Vivia Nikolaidou + + * tools/gst-play.c: + gst-play: Add keyboard shortcut '0' to seek to beginning + https://bugzilla.gnome.org/show_bug.cgi?id=755440 + +2015-08-25 16:24:12 +0900 Vineeth T M + + * gst/videorate/gstvideorate.c: + videorate: remove unnecessary break statement + Trivial patch to remove unncessary break statement used after + goto statement. + https://bugzilla.gnome.org/show_bug.cgi?id=754054 + +2015-08-20 15:59:15 +0900 Vineeth TM + + * gst-libs/gst/tag/mklicensestables.c: + * tests/examples/encoding/encoding.c: + * tests/examples/playback/playback-test.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/scrubby.c: + * tests/icles/stress-playbin.c: + * tests/icles/test-effect-switch.c: + * tools/gst-device-monitor.c: + * tools/gst-discoverer.c: + * tools/gst-play.c: + gstreamer: base: Fix memory leaks when context parse fails. + When g_option_context_parse fails, context and error variables are not getting free'd + which results in memory leaks. Free'ing the same. + And replacing g_error_free with g_clear_error, which checks if the error being passed + is not NULL and sets the variable to NULL on free'ing. + https://bugzilla.gnome.org/show_bug.cgi?id=753852 + +2015-06-24 23:55:35 +0200 Mathieu Duponchelle + + * gst/encoding/gstencodebin.c: + encodebin: Fix special case + Allows to run such a command line : + gst-launch-1.0 uridecodebin uri=file:///home/meh/Music/sthg.mp4 ! \ + encodebin profile-string="audio/x-wav|1" ! filesink location=sthg.wav + Previously the code failed because wavenc is considered as a muxer. + We still want encodebin to audio/x-wav as an AudioEncodingProfile, + so this simple fix allows that. + Ability to mux raw streams in containers such as matroskamux + is a different issue. + https://bugzilla.gnome.org/show_bug.cgi?id=751470 + +2015-09-29 10:12:28 +0530 Rajat Verma + + * gst/playback/gstdecodebin2.c: + decodebin: free hidden groups at time of switching groups + hidden groups should be freed at time of switching groups to avoid memory use + from balloning up. + https://bugzilla.gnome.org/show_bug.cgi?id=755770 + +2015-10-02 10:07:33 +0300 Sebastian Dröge + + * win32/common/libgstpbutils.def: + win32: Update exports for new audiovisualizer symbols + +2015-10-02 15:04:34 +1000 Jan Schmidt + + * tests/check/Makefile.am: + * tests/check/libs/baseaudiovisualizer.c: + tests: Add baseaudiovisualizer test, moved from -bad + +2015-10-02 15:05:26 +1000 Jan Schmidt + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: Don't fixate framerate if downstream didn't provide one + intersection with a downstream that accepts any video/x-raw caps + with no further detail won't create a framerate field. If it's + not in the caps, don't fixate it, just set it to 30/1 + +2015-10-01 21:53:20 +0200 Stefan Sauer + + * docs/plugins/gst-plugins-base-plugins-docs.sgml: + * docs/plugins/gst-plugins-base-plugins-sections.txt: + * docs/plugins/gst-plugins-base-plugins.args: + * docs/plugins/gst-plugins-base-plugins.hierarchy: + * docs/plugins/inspect/plugin-alsa.xml: + docs: add alsamidisrc to docs + +2015-10-01 21:43:21 +0200 Antonio Ospite + + * ext/alsa/Makefile.am: + * ext/alsa/gstalsamidisrc.c: + * ext/alsa/gstalsamidisrc.h: + * ext/alsa/gstalsaplugin.c: + midi: add an ALSA MIDI sequencer source + The alsamidisrc element allows to get input event from ALSA MIDI + sequencer devices, and possibly convert them to sound using some + downstream element like fluiddec. + Fixes #738687 + +2015-10-01 15:27:55 +0100 Luis de Bethencourt + + * gst-libs/gst/pbutils/gstaudiovisualizer.c: + visual: make private all variable subclasses don't need + Subclasses don't need access to all variables. Making them private. + https://bugzilla.gnome.org/show_bug.cgi?id=742875 + +2015-10-01 11:55:59 +0100 Luis de Bethencourt + + * ext/libvisual/Makefile.am: + * ext/libvisual/gstaudiovisualizer.c: + * ext/libvisual/gstaudiovisualizer.h: + * ext/libvisual/visual.h: + * gst-libs/gst/pbutils/Makefile.am: + * gst-libs/gst/pbutils/gstaudiovisualizer.c: + * gst-libs/gst/pbutils/gstaudiovisualizer.h: + visual: merge audiovisalizer base classes + Move the audiovisualizer base class to pbutils, so it can be used by plugins + from other modules + https://bugzilla.gnome.org/show_bug.cgi?id=742875 + +2015-10-01 12:48:52 +0100 Tim-Philipp Müller + + * gst/typefind/gsttypefindfunctions.c: + typefinding: minor clean-up + Remove unnecessary brackets from IS_MPEGTS_HEADER macro. + +2015-10-01 12:32:33 +0100 Pankaj Darak + + * gst/typefind/gsttypefindfunctions.c: + typefinding: mpeg-ts detection improvement + Allow AFC to be 0 for null pid packets. + https://bugzilla.gnome.org/show_bug.cgi?id=726117 + +2015-09-30 18:18:15 +0100 Tim-Philipp Müller + + * tests/check/elements/subparse.c: + tests: subparse: add unit test for closing tag detection + should be handled like + https://bugzilla.gnome.org/show_bug.cgi?id=755875 + +2015-09-30 18:17:13 +0100 Tim-Philipp Müller + + * gst/subparse/gstsubparse.c: + subparse: detect closing tags even if there's a space after the slash + should be handled like + https://bugzilla.gnome.org/show_bug.cgi?id=755875 + +2015-09-23 11:59:22 -0400 Perry Hung + + * gst-libs/gst/app/Makefile.am: + app: pass PKG_CONFIG_PATH for gir files for libgstapp as well + gir include search directories should respect PKG_CONFIG_PATH, + just like we do everywhere else. Makes g-i pick up the right + paths when using ./configure --with-pkg-config-path= + https://bugzilla.gnome.org/show_bug.cgi?id=755494 + +2015-09-25 23:51:06 +0200 Sebastian Dröge + + * configure.ac: + Back to development + === release 1.6.0 === -2015-09-25 Sebastian Dröge +2015-09-25 23:15:20 +0200 Sebastian Dröge + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.6.0 + * docs/plugins/inspect/plugin-adder.xml: + * docs/plugins/inspect/plugin-alsa.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-audioconvert.xml: + * docs/plugins/inspect/plugin-audiorate.xml: + * docs/plugins/inspect/plugin-audioresample.xml: + * docs/plugins/inspect/plugin-audiotestsrc.xml: + * docs/plugins/inspect/plugin-cdparanoia.xml: + * docs/plugins/inspect/plugin-encoding.xml: + * docs/plugins/inspect/plugin-gio.xml: + * docs/plugins/inspect/plugin-libvisual.xml: + * docs/plugins/inspect/plugin-ogg.xml: + * docs/plugins/inspect/plugin-pango.xml: + * docs/plugins/inspect/plugin-playback.xml: + * docs/plugins/inspect/plugin-subparse.xml: + * docs/plugins/inspect/plugin-tcp.xml: + * docs/plugins/inspect/plugin-theora.xml: + * docs/plugins/inspect/plugin-typefindfunctions.xml: + * docs/plugins/inspect/plugin-videoconvert.xml: + * docs/plugins/inspect/plugin-videorate.xml: + * docs/plugins/inspect/plugin-videoscale.xml: + * docs/plugins/inspect/plugin-videotestsrc.xml: + * docs/plugins/inspect/plugin-volume.xml: + * docs/plugins/inspect/plugin-vorbis.xml: + * docs/plugins/inspect/plugin-ximagesink.xml: + * docs/plugins/inspect/plugin-xvimagesink.xml: + * gst-libs/gst/video/video-orc-dist.c: + * gst-plugins-base.doap: + * win32/common/_stdint.h: + * win32/common/config.h: + Release 1.6.0 + +2015-09-25 22:50:51 +0200 Sebastian Dröge + + * po/af.po: + * po/az.po: + * po/bg.po: + * po/ca.po: + * po/cs.po: + * po/da.po: + * po/de.po: + * po/el.po: + * po/en_GB.po: + * po/eo.po: + * po/es.po: + * po/eu.po: + * po/fi.po: + * po/fr.po: + * po/gl.po: + * po/hr.po: + * po/hu.po: + * po/id.po: + * po/it.po: + * po/ja.po: + * po/lt.po: + * po/lv.po: + * po/nb.po: + * po/nl.po: + * po/or.po: + * po/pl.po: + * po/pt_BR.po: + * po/ro.po: + * po/ru.po: + * po/sk.po: + * po/sl.po: + * po/sq.po: + * po/sr.po: + * po/sv.po: + * po/tr.po: + * po/uk.po: + * po/vi.po: + * po/zh_CN.po: + Update .po files 2015-09-24 18:06:58 +0200 Sebastian Dröge diff --git a/NEWS b/NEWS index e04f31844..a4bffc6a6 100644 --- a/NEWS +++ b/NEWS @@ -1,64 +1,2 @@ -This is GStreamer 1.6.0 - -The GStreamer team is proud to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! - -This release has been in the works for more than a year and is packed with new -features, bug fixes and other improvements. - -See http://gstreamer.freedesktop.org/releases/1.6/ for the full list of -changes. - -Highlights - -- Stereoscopic 3D and multiview video support -- Trick mode API for key-frame only fast-forward/fast-reverse playback etc. -- Improved DTS (decoding timestamp) vs. PTS (presentation timestamp) handling - to account for negative DTS -- New GstVideoConverter API for more optimised and more correct conversion of - raw video frames between all supported formats, with rescaling -- v4l2src now supports renegotiation -- v4l2transform can now do scaling -- V4L2 Element now report Colorimetry properly -- Easier chunked recording of MP4, Matroska, Ogg, MPEG-TS: new splitmuxsink - and multifilesink improvements -- Content Protection signalling API and Common Encryption (CENC) support for - DASH/MP4 -- Many adaptive streaming (DASH, HLS and MSS) improvements -- New PTP and NTP network client clocks and better remote clock tracking - stability -- High-quality text subtitle overlay at display resolutions with glimagesink - or gtkglsink -- RECORD support for the GStreamer RTSP Server -- Retransmissions (RTX) support in RTSP server and client -- RTSP seeking support in client and server has been fixed -- RTCP scheduling improvements and reduced size RTCP support -- MP4/MOV muxer acquired a new "robust" mode of operation which attempts to - keep the output file in a valid state at all times -- Live mixing support in aggregator, audiomixer and compositor was improved a - lot -- compositor now supports rescaling and converting inputs streams on the fly -- New audiointerleave element with proper input synchronisation and live input - support -- Blackmagic Design DeckLink capture and playback card support was rewritten - from scratch; 2k/4k support; mode sensing -- KLV metadata support in RTP and MPEG-TS -- H.265 video encoder (x265), decoders (libav, libde265) and RTP payloader and - depayloaders -- New DTLS plugin and SRTP/DTLS support -- OpenGL3 support, multiple contexts and context propagation, 3D video, - transfer/conversion separation, subtitle blending -- New OpenGL-based QML video sink, Gtk GL video sink, CoreAnimation - CAOpenGLLayerSink video sink -- gst-libav switched to ffmpeg as libav-provider, gains support for - 3D/multiview video, trick modes, and the CAVS codec -- GstHarness API for unit tests -- gst-editing-services got a completely new ges-launch-1.0 interface, improved - mixing support and integration into gst-validate -- gnonlin has been deprecated in favor of nle (Non Linear Engine) in - gst-editing-services -- gst-validate has a new plugin system, an extensive default testsuite, - support for concurrent test runs and valgrind support -- cerbero build tool for SDK binary packages gains new 'bundle-source' command -- Various improvements to the Android, iOS, OS X and Windows platform support +This is GStreamer 1.7.1 diff --git a/RELEASE b/RELEASE index d133af621..4cccb28f7 100644 --- a/RELEASE +++ b/RELEASE @@ -1,18 +1,17 @@ -Release notes for GStreamer Base Plugins 1.6.0 +Release notes for GStreamer Base Plugins 1.7.1 -The GStreamer team is proud to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! +The GStreamer team is pleased to announce the first release of the unstable +1.7 release series. The 1.7 release series is adding new features on top of +the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release +series of the GStreamer multimedia framework. The unstable 1.7 release series +will lead to the stable 1.8 release series in the next weeks. Any newly added +API can still change until that point. -This release has been in the works for more than a year and is packed with new -features, bug fixes and other improvements. - - -See -http://gstreamer.freedesktop.org/releases/1.6/ -for the full list of changes. +Binaries for Android, iOS, Mac OS X and Windows will be provided separately +during the unstable 1.7 release series. @@ -62,10 +61,43 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg) Bugs fixed in this release - * 752148 : Drawing from paths passed to cairo does not work with PANGOCAIRO_BACKEND=coretext - * 754344 : libs: build rtp after audio - * 754833 : dmabuf & fdmemory: fix allocator_alloc documentation - * 755392 : video: bugs with gst_video_frame_copy and videoconvert (with test scripts) + * 681447 : video overlay composition: fix video blending over transparent frame + * 705579 : Playbin prevents plugins requesting a GstContext to work properly + * 726117 : typefinding: issue in MPEG-TS detection logic for streams with Null Pids + * 726472 : rtpbasepayload: Implement video SDP attributes + * 727970 : videorate: remove dead code + * 730926 : tags: add GST_TAG_PRIVATE_DATA and expose ID3 private frame ( " PRIV " ) data + * 731791 : videometa: add GstVideoAffineTransformationMeta + * 738687 : midi: add alsamidisrc, an ALSA MIDI sequencer source + * 749596 : rtsp-over-http authentication failure + * 751470 : encodebin: Fix special case. + * 752651 : decodebin: segfault on setting to NULL + * 753852 : gstreamer: base: Fix memory leaks when context parse fails. + * 754054 : videorate: remove unnecessary break statement + * 754196 : audiodecoder-test: port to using GstHarness + * 754223 : audioencoder-tests: port to use GstHarness + * 754450 : audiotestsrc: remove frequency and channel number limit + * 755260 : decodebin: Fix a race condition accessing the decode_chain field. + * 755301 : audioconvert: Integer- > Float conversion creates values slightly smaller than -1.0 + * 755440 : gst-play: Add keyboard shortcut '0' to seek to beginning + * 755482 : videotestsrc: Force alpha downstream if foreground color contains alpha + * 756804 : playsink: text_sink dynamic reconnection is not working + * 757008 : tests: typefindfunctions: Fix error leak + * 757068 : audio{filter,convert,resample}: Clip input buffers to the segment before handling them + * 757351 : audioconvert: Latest audioconvert outputs noise + * 757480 : Use GST_STIME_FORMAT and GST_STIME_ARGS with GstClockTimeDiff + * 757926 : pbutils:encoding-target: Fix string memory leak + * 757927 : tests:video: Fix overlay rectangle and buffer leak + * 757928 : audio-quantize: Fix dither_buffer memory leak + * 758235 : rtspconnection: add support for parsing custom headers + * 758744 : allocators: Add logging category for GstFdMemory + * 758911 : audiobasesink/src: send latency message on setcaps + * 758922 : rtspconnection should optionally make HTTP requests with abs_path instead of absoluteURI + * 759126 : appsrc: issues with duration query handling + * 759329 : convertframe: Support video crop when convert frame + * 759356 : encodebin: Implement an encoding profile serialization format + * 742875 : [API] new audiovisualizer base class + * 758754 : oggdemux: failing to play an Opus sample file ==== Download ==== @@ -102,6 +134,50 @@ subscribe to the gstreamer-devel list. Contributors to this release - * Aurélien Zanelli + * Andreas Frisch + * Antonio Ospite + * Arnaud Vrac + * Csaba Toth + * Edward Hervey + * Eunhae Choi + * Evan Callaway + * Guillaume Desmottes + * Havard Graff + * Jan Schmidt + * Joan Pau Beltran + * Julien Isorce + * Kazunori Kobayashi + * Koop Mast + * Luis de Bethencourt + * Mathieu Duponchelle + * Matthew Waters + * Michael Olbrich + * Miguel París Díaz + * Nicolas Dufresne + * Nirbheek Chauhan + * Ognyan Tonchev + * Pankaj Darak + * Pavel Bludov + * Perry Hung + * Philippe Normand + * Rajat Verma + * Ravi Kiran K N + * Reynaldo H. Verdejo Pinochet * Sebastian Dröge + * Sebastian Rasmussen + * Song Bing + * Stefan Sauer + * Stian Selnes + * Thiago Santos + * Thibault Saunier + * Thomas Bluemel + * Tim-Philipp Müller + * Vincent Penquerc'h + * Vineeth T M + * Vineeth TM + * Vivia Nikolaidou + * William Manley + * Wim Taymans + * Xavier Claessens + * eunhae choi   \ No newline at end of file diff --git a/configure.ac b/configure.ac index d3302d596..9c52aeb47 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/prerelease -AC_INIT([GStreamer Base Plug-ins],[1.7.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base]) +AC_INIT([GStreamer Base Plug-ins],[1.7.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base]) AG_GST_INIT @@ -56,10 +56,10 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 700, 0, 700) +AS_LIBTOOL(GST, 701, 0, 701) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.7.0.1 +GST_REQ=1.7.1 dnl *** autotools stuff **** diff --git a/docs/plugins/inspect/plugin-adder.xml b/docs/plugins/inspect/plugin-adder.xml index 8a975c538..dfb48df19 100644 --- a/docs/plugins/inspect/plugin-adder.xml +++ b/docs/plugins/inspect/plugin-adder.xml @@ -3,10 +3,10 @@ Adds multiple streams ../../gst/adder/.libs/libgstadder.so libgstadder.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-alsa.xml b/docs/plugins/inspect/plugin-alsa.xml index 5f3ceae28..fd899b5a5 100644 --- a/docs/plugins/inspect/plugin-alsa.xml +++ b/docs/plugins/inspect/plugin-alsa.xml @@ -3,10 +3,10 @@ ALSA plugin library ../../ext/alsa/.libs/libgstalsa.so libgstalsa.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-app.xml b/docs/plugins/inspect/plugin-app.xml index b3f978cd7..d0c9e320b 100644 --- a/docs/plugins/inspect/plugin-app.xml +++ b/docs/plugins/inspect/plugin-app.xml @@ -3,10 +3,10 @@ Elements used to communicate with applications ../../gst/app/.libs/libgstapp.so libgstapp.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audioconvert.xml b/docs/plugins/inspect/plugin-audioconvert.xml index 4008f8996..4594f6a31 100644 --- a/docs/plugins/inspect/plugin-audioconvert.xml +++ b/docs/plugins/inspect/plugin-audioconvert.xml @@ -3,10 +3,10 @@ Convert audio to different formats ../../gst/audioconvert/.libs/libgstaudioconvert.so libgstaudioconvert.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiorate.xml b/docs/plugins/inspect/plugin-audiorate.xml index 30ecde202..2dee643d0 100644 --- a/docs/plugins/inspect/plugin-audiorate.xml +++ b/docs/plugins/inspect/plugin-audiorate.xml @@ -3,10 +3,10 @@ Adjusts audio frames ../../gst/audiorate/.libs/libgstaudiorate.so libgstaudiorate.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audioresample.xml b/docs/plugins/inspect/plugin-audioresample.xml index 4cba92b9b..635c3c6e3 100644 --- a/docs/plugins/inspect/plugin-audioresample.xml +++ b/docs/plugins/inspect/plugin-audioresample.xml @@ -3,10 +3,10 @@ Resamples audio ../../gst/audioresample/.libs/libgstaudioresample.so libgstaudioresample.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiotestsrc.xml b/docs/plugins/inspect/plugin-audiotestsrc.xml index dccac7095..c8a585b26 100644 --- a/docs/plugins/inspect/plugin-audiotestsrc.xml +++ b/docs/plugins/inspect/plugin-audiotestsrc.xml @@ -3,10 +3,10 @@ Creates audio test signals of given frequency and volume ../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so libgstaudiotestsrc.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-cdparanoia.xml b/docs/plugins/inspect/plugin-cdparanoia.xml index 8ee6f9ddd..549b5f86e 100644 --- a/docs/plugins/inspect/plugin-cdparanoia.xml +++ b/docs/plugins/inspect/plugin-cdparanoia.xml @@ -3,10 +3,10 @@ Read audio from CD in paranoid mode ../../ext/cdparanoia/.libs/libgstcdparanoia.so libgstcdparanoia.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-encoding.xml b/docs/plugins/inspect/plugin-encoding.xml index 9fce4e9b5..b1a69aa51 100644 --- a/docs/plugins/inspect/plugin-encoding.xml +++ b/docs/plugins/inspect/plugin-encoding.xml @@ -3,10 +3,10 @@ various encoding-related elements ../../gst/encoding/.libs/libgstencodebin.so libgstencodebin.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-gio.xml b/docs/plugins/inspect/plugin-gio.xml index 5c3b33582..4e27ece8e 100644 --- a/docs/plugins/inspect/plugin-gio.xml +++ b/docs/plugins/inspect/plugin-gio.xml @@ -3,10 +3,10 @@ GIO elements ../../gst/gio/.libs/libgstgio.so libgstgio.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-libvisual.xml b/docs/plugins/inspect/plugin-libvisual.xml index fad7b9b73..35c30ca2a 100644 --- a/docs/plugins/inspect/plugin-libvisual.xml +++ b/docs/plugins/inspect/plugin-libvisual.xml @@ -3,10 +3,10 @@ libvisual visualization plugins ../../ext/libvisual/.libs/libgstlibvisual.so libgstlibvisual.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-ogg.xml b/docs/plugins/inspect/plugin-ogg.xml index 3c8f90b31..b7b663b2d 100644 --- a/docs/plugins/inspect/plugin-ogg.xml +++ b/docs/plugins/inspect/plugin-ogg.xml @@ -3,10 +3,10 @@ ogg stream manipulation (info about ogg: http://xiph.org) ../../ext/ogg/.libs/libgstogg.so libgstogg.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-pango.xml b/docs/plugins/inspect/plugin-pango.xml index 08cdbd93e..8989a992f 100644 --- a/docs/plugins/inspect/plugin-pango.xml +++ b/docs/plugins/inspect/plugin-pango.xml @@ -3,10 +3,10 @@ Pango-based text rendering and overlay ../../ext/pango/.libs/libgstpango.so libgstpango.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-playback.xml b/docs/plugins/inspect/plugin-playback.xml index 0f648d92f..69a709acb 100644 --- a/docs/plugins/inspect/plugin-playback.xml +++ b/docs/plugins/inspect/plugin-playback.xml @@ -3,10 +3,10 @@ various playback elements ../../gst/playback/.libs/libgstplayback.so libgstplayback.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-subparse.xml b/docs/plugins/inspect/plugin-subparse.xml index e6294d0be..75ed13d02 100644 --- a/docs/plugins/inspect/plugin-subparse.xml +++ b/docs/plugins/inspect/plugin-subparse.xml @@ -3,10 +3,10 @@ Subtitle parsing ../../gst/subparse/.libs/libgstsubparse.so libgstsubparse.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-tcp.xml b/docs/plugins/inspect/plugin-tcp.xml index d1cd20a47..15e551709 100644 --- a/docs/plugins/inspect/plugin-tcp.xml +++ b/docs/plugins/inspect/plugin-tcp.xml @@ -3,10 +3,10 @@ transfer data over the network via TCP ../../gst/tcp/.libs/libgsttcp.so libgsttcp.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-theora.xml b/docs/plugins/inspect/plugin-theora.xml index 72d600de1..65e731952 100644 --- a/docs/plugins/inspect/plugin-theora.xml +++ b/docs/plugins/inspect/plugin-theora.xml @@ -3,10 +3,10 @@ Theora plugin library ../../ext/theora/.libs/libgsttheora.so libgsttheora.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-typefindfunctions.xml b/docs/plugins/inspect/plugin-typefindfunctions.xml index 6881527af..103e1e0aa 100644 --- a/docs/plugins/inspect/plugin-typefindfunctions.xml +++ b/docs/plugins/inspect/plugin-typefindfunctions.xml @@ -3,10 +3,10 @@ default typefind functions ../../gst/typefind/.libs/libgsttypefindfunctions.so libgsttypefindfunctions.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videoconvert.xml b/docs/plugins/inspect/plugin-videoconvert.xml index 7e702c516..071d0e02a 100644 --- a/docs/plugins/inspect/plugin-videoconvert.xml +++ b/docs/plugins/inspect/plugin-videoconvert.xml @@ -3,10 +3,10 @@ Colorspace conversion ../../gst/videoconvert/.libs/libgstvideoconvert.so libgstvideoconvert.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videorate.xml b/docs/plugins/inspect/plugin-videorate.xml index b5b8f53b2..8ed4bba71 100644 --- a/docs/plugins/inspect/plugin-videorate.xml +++ b/docs/plugins/inspect/plugin-videorate.xml @@ -3,10 +3,10 @@ Adjusts video frames ../../gst/videorate/.libs/libgstvideorate.so libgstvideorate.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videoscale.xml b/docs/plugins/inspect/plugin-videoscale.xml index 3f020c099..dfaf58b62 100644 --- a/docs/plugins/inspect/plugin-videoscale.xml +++ b/docs/plugins/inspect/plugin-videoscale.xml @@ -3,10 +3,10 @@ Resizes video ../../gst/videoscale/.libs/libgstvideoscale.so libgstvideoscale.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videotestsrc.xml b/docs/plugins/inspect/plugin-videotestsrc.xml index aebf8007d..a5593662d 100644 --- a/docs/plugins/inspect/plugin-videotestsrc.xml +++ b/docs/plugins/inspect/plugin-videotestsrc.xml @@ -3,10 +3,10 @@ Creates a test video stream ../../gst/videotestsrc/.libs/libgstvideotestsrc.so libgstvideotestsrc.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-volume.xml b/docs/plugins/inspect/plugin-volume.xml index 15f34711f..726cf0647 100644 --- a/docs/plugins/inspect/plugin-volume.xml +++ b/docs/plugins/inspect/plugin-volume.xml @@ -3,10 +3,10 @@ plugin for controlling audio volume ../../gst/volume/.libs/libgstvolume.so libgstvolume.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-vorbis.xml b/docs/plugins/inspect/plugin-vorbis.xml index 300e3fa10..ffeb07642 100644 --- a/docs/plugins/inspect/plugin-vorbis.xml +++ b/docs/plugins/inspect/plugin-vorbis.xml @@ -3,10 +3,10 @@ Vorbis plugin library ../../ext/vorbis/.libs/libgstvorbis.so libgstvorbis.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-ximagesink.xml b/docs/plugins/inspect/plugin-ximagesink.xml index b6a068ff7..e84bc89c6 100644 --- a/docs/plugins/inspect/plugin-ximagesink.xml +++ b/docs/plugins/inspect/plugin-ximagesink.xml @@ -3,10 +3,10 @@ X11 video output element based on standard Xlib calls ../../sys/ximage/.libs/libgstximagesink.so libgstximagesink.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-xvimagesink.xml b/docs/plugins/inspect/plugin-xvimagesink.xml index 42eafa7a5..d8834d32e 100644 --- a/docs/plugins/inspect/plugin-xvimagesink.xml +++ b/docs/plugins/inspect/plugin-xvimagesink.xml @@ -3,10 +3,10 @@ XFree86 video output plugin using Xv extension ../../sys/xvimage/.libs/libgstxvimagesink.so libgstxvimagesink.so - 1.7.0.1 + 1.7.1 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap index 71b1581ea..4ca4ae20b 100644 --- a/gst-plugins-base.doap +++ b/gst-plugins-base.doap @@ -34,6 +34,36 @@ A wide range of video and audio decoders, encoders, and filters are included. + + + 1.7.1 + master + + 2015-12-24 + + + + + + + 1.6.2 + 1.6 + + 2015-12-14 + + + + + + + 1.6.1 + 1.6 + + 2015-10-30 + + + + 1.6.0 diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h index 1864535f4..06e809ad0 100644 --- a/win32/common/_stdint.h +++ b/win32/common/_stdint.h @@ -1,8 +1,8 @@ #ifndef _GST_PLUGINS_BASE__STDINT_H #define _GST_PLUGINS_BASE__STDINT_H 1 #ifndef _GENERATED_STDINT_H -#define _GENERATED_STDINT_H "gst-plugins-base 1.6.0" -/* generated using gnu compiler gcc-5 (Debian 5.2.1-17) 5.2.1 20150911 */ +#define _GENERATED_STDINT_H "gst-plugins-base 1.7.1" +/* generated using gnu compiler gcc-5 (Debian 5.3.1-4) 5.3.1 20151219 */ #define _STDINT_HAVE_STDINT_H 1 #include #endif diff --git a/win32/common/audio-enumtypes.c b/win32/common/audio-enumtypes.c index 63a12eccf..c113912de 100644 --- a/win32/common/audio-enumtypes.c +++ b/win32/common/audio-enumtypes.c @@ -6,7 +6,10 @@ #include "audio.h" #include "audio-format.h" #include "audio-channels.h" +#include "audio-channel-mix.h" +#include "audio-converter.h" #include "audio-info.h" +#include "audio-quantize.h" #include "gstaudioringbuffer.h" /* enumerations from "audio-format.h" */ @@ -97,12 +100,14 @@ gst_audio_pack_flags_get_type (void) { static volatile gsize g_define_type_id__volatile = 0; if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GEnumValue values[] = { + static const GFlagsValue values[] = { {GST_AUDIO_PACK_FLAG_NONE, "GST_AUDIO_PACK_FLAG_NONE", "none"}, + {GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, "GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE", + "truncate-range"}, {0, NULL, NULL} }; GType g_define_type_id = - g_enum_register_static ("GstAudioPackFlags", values); + g_flags_register_static ("GstAudioPackFlags", values); g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); } return g_define_type_id__volatile; @@ -191,6 +196,49 @@ gst_audio_channel_position_get_type (void) return g_define_type_id__volatile; } +/* enumerations from "audio-channel-mix.h" */ +GType +gst_audio_channel_mix_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_AUDIO_CHANNEL_MIX_FLAGS_NONE, "GST_AUDIO_CHANNEL_MIX_FLAGS_NONE", + "none"}, + {GST_AUDIO_CHANNEL_MIX_FLAGS_NON_INTERLEAVED, + "GST_AUDIO_CHANNEL_MIX_FLAGS_NON_INTERLEAVED", "non-interleaved"}, + {GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN, + "GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN", "unpositioned-in"}, + {GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT, + "GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT", "unpositioned-out"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstAudioChannelMixFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +/* enumerations from "audio-converter.h" */ +GType +gst_audio_converter_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_AUDIO_CONVERTER_FLAG_NONE, "GST_AUDIO_CONVERTER_FLAG_NONE", "none"}, + {GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE, + "GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE", "source-writable"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstAudioConverterFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + /* enumerations from "audio-info.h" */ GType gst_audio_flags_get_type (void) @@ -227,6 +275,67 @@ gst_audio_layout_get_type (void) return g_define_type_id__volatile; } +/* enumerations from "audio-quantize.h" */ +GType +gst_audio_dither_method_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_AUDIO_DITHER_NONE, "GST_AUDIO_DITHER_NONE", "none"}, + {GST_AUDIO_DITHER_RPDF, "GST_AUDIO_DITHER_RPDF", "rpdf"}, + {GST_AUDIO_DITHER_TPDF, "GST_AUDIO_DITHER_TPDF", "tpdf"}, + {GST_AUDIO_DITHER_TPDF_HF, "GST_AUDIO_DITHER_TPDF_HF", "tpdf-hf"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstAudioDitherMethod", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_audio_noise_shaping_method_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_AUDIO_NOISE_SHAPING_NONE, "GST_AUDIO_NOISE_SHAPING_NONE", "none"}, + {GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK, + "GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK", "error-feedback"}, + {GST_AUDIO_NOISE_SHAPING_SIMPLE, "GST_AUDIO_NOISE_SHAPING_SIMPLE", + "simple"}, + {GST_AUDIO_NOISE_SHAPING_MEDIUM, "GST_AUDIO_NOISE_SHAPING_MEDIUM", + "medium"}, + {GST_AUDIO_NOISE_SHAPING_HIGH, "GST_AUDIO_NOISE_SHAPING_HIGH", "high"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstAudioNoiseShapingMethod", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_audio_quantize_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_AUDIO_QUANTIZE_FLAG_NONE, "GST_AUDIO_QUANTIZE_FLAG_NONE", "none"}, + {GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED, + "GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED", "non-interleaved"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstAudioQuantizeFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + /* enumerations from "gstaudioringbuffer.h" */ GType gst_audio_ring_buffer_state_get_type (void) diff --git a/win32/common/audio-enumtypes.h b/win32/common/audio-enumtypes.h index a323a87db..1b51e1c88 100644 --- a/win32/common/audio-enumtypes.h +++ b/win32/common/audio-enumtypes.h @@ -20,12 +20,28 @@ GType gst_audio_pack_flags_get_type (void); GType gst_audio_channel_position_get_type (void); #define GST_TYPE_AUDIO_CHANNEL_POSITION (gst_audio_channel_position_get_type()) +/* enumerations from "audio-channel-mix.h" */ +GType gst_audio_channel_mix_flags_get_type (void); +#define GST_TYPE_AUDIO_CHANNEL_MIX_FLAGS (gst_audio_channel_mix_flags_get_type()) + +/* enumerations from "audio-converter.h" */ +GType gst_audio_converter_flags_get_type (void); +#define GST_TYPE_AUDIO_CONVERTER_FLAGS (gst_audio_converter_flags_get_type()) + /* enumerations from "audio-info.h" */ GType gst_audio_flags_get_type (void); #define GST_TYPE_AUDIO_FLAGS (gst_audio_flags_get_type()) GType gst_audio_layout_get_type (void); #define GST_TYPE_AUDIO_LAYOUT (gst_audio_layout_get_type()) +/* enumerations from "audio-quantize.h" */ +GType gst_audio_dither_method_get_type (void); +#define GST_TYPE_AUDIO_DITHER_METHOD (gst_audio_dither_method_get_type()) +GType gst_audio_noise_shaping_method_get_type (void); +#define GST_TYPE_AUDIO_NOISE_SHAPING_METHOD (gst_audio_noise_shaping_method_get_type()) +GType gst_audio_quantize_flags_get_type (void); +#define GST_TYPE_AUDIO_QUANTIZE_FLAGS (gst_audio_quantize_flags_get_type()) + /* enumerations from "gstaudioringbuffer.h" */ GType gst_audio_ring_buffer_state_get_type (void); #define GST_TYPE_AUDIO_RING_BUFFER_STATE (gst_audio_ring_buffer_state_get_type()) diff --git a/win32/common/config.h b/win32/common/config.h index 78833158a..e9feeec43 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -59,6 +59,9 @@ /* system wide data directory */ #define GST_DATADIR PREFIX "\\share" +/* Define if extra runtime checks should be enabled */ +#undef GST_ENABLE_EXTRA_CHECKS + /* Extra platform specific plugin suffix */ #undef GST_EXTRA_MODULE_SUFFIX @@ -87,7 +90,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2015-09-25" +#define GST_PACKAGE_RELEASE_DATETIME "2015-12-24" /* Define if static plugins should be built */ #undef GST_PLUGIN_BUILD_STATIC @@ -185,9 +188,6 @@ /* Define if the GNU gettext() function is already present or preinstalled. */ #undef HAVE_GETTEXT -/* Define to enable glib GIO unix (used by gio-unix-2.0). */ -#undef HAVE_GIO_UNIX_2_0 - /* Define to 1 if you have the `gmtime_r' function. */ #undef HAVE_GMTIME_R @@ -337,7 +337,7 @@ #define PACKAGE_NAME "GStreamer Base Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Base Plug-ins 1.6.0" +#define PACKAGE_STRING "GStreamer Base Plug-ins 1.7.1" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-base" @@ -346,7 +346,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "1.6.0" +#define PACKAGE_VERSION "1.7.1" /* directory where plugins are located */ #ifdef _DEBUG @@ -380,7 +380,7 @@ #undef USE_TREMOLO /* Version number of package */ -#define VERSION "1.6.0" +#define VERSION "1.7.1" /* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most significant byte first (like Motorola and SPARC, unlike Intel). */ diff --git a/win32/common/pbutils-enumtypes.c b/win32/common/pbutils-enumtypes.c index 99a6f1df0..0f0bc9310 100644 --- a/win32/common/pbutils-enumtypes.c +++ b/win32/common/pbutils-enumtypes.c @@ -11,6 +11,7 @@ #include "install-plugins.h" #include "missing-plugins.h" #include "gstdiscoverer.h" +#include "gstaudiovisualizer.h" /* enumerations from "install-plugins.h" */ GType @@ -89,3 +90,46 @@ gst_discoverer_serialize_flags_get_type (void) } return g_define_type_id__volatile; } + +/* enumerations from "gstaudiovisualizer.h" */ +GType +gst_audio_visualizer_shader_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_AUDIO_VISUALIZER_SHADER_NONE, "GST_AUDIO_VISUALIZER_SHADER_NONE", + "none"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE, "GST_AUDIO_VISUALIZER_SHADER_FADE", + "fade"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP", "fade-and-move-up"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN", + "fade-and-move-down"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT", + "fade-and-move-left"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT", + "fade-and-move-right"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT", + "fade-and-move-horiz-out"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN", + "fade-and-move-horiz-in"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT", + "fade-and-move-vert-out"}, + {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN, + "GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN", + "fade-and-move-vert-in"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstAudioVisualizerShader", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} diff --git a/win32/common/pbutils-enumtypes.h b/win32/common/pbutils-enumtypes.h index 1ca0476db..27754b0ee 100644 --- a/win32/common/pbutils-enumtypes.h +++ b/win32/common/pbutils-enumtypes.h @@ -17,6 +17,10 @@ GType gst_discoverer_result_get_type (void); #define GST_TYPE_DISCOVERER_RESULT (gst_discoverer_result_get_type()) GType gst_discoverer_serialize_flags_get_type (void); #define GST_TYPE_DISCOVERER_SERIALIZE_FLAGS (gst_discoverer_serialize_flags_get_type()) + +/* enumerations from "gstaudiovisualizer.h" */ +GType gst_audio_visualizer_shader_get_type (void); +#define GST_TYPE_AUDIO_VISUALIZER_SHADER (gst_audio_visualizer_shader_get_type()) G_END_DECLS #endif /* __PB_UTILS_ENUM_TYPES_H__ */ -- 2.34.1