=== release 0.10.31 === 2012-02-21 Tim-Philipp Müller * configure.ac: releasing 0.10.31, "Faster" 2012-02-20 12:22:12 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: force baseline profile is profile-level-id is unspecified If profile-level-id is missing or invalid, we want any upstream encoder to default to baseline profile, so specify that in the caps we pass upstream. If the caps contain no profile restriction, an encoder may default to high or main profile. 2012-02-17 17:21:53 +0000 Tim-Philipp Müller * gst/equalizer/gstiirequalizer.c: equalizer: fix switching from passthrough to non-passthrough when parameters change commit b5bf0294 moved the if(need_new_coefficients) set_passthrough(equ) after the if(is_passthrough) return FLOW_OK shortcut, so the passthrough mode would never get updated even if the coefficients change. Fixes equalizer-test doing .. nothing. 2012-02-16 17:14:20 +0800 Gary Ching-Pang Lin * sys/v4l2/v4l2_calls.c: v4l2src: failure to query some optional controls is not a fatal error Don't post a (fatal) error message on the bus just because we failed to query some control. Fixes issue with built-in Suyin Corp webcam for HP notebook (usbid 064e:e28a) on OpenSuse 12.1, where querying red/blue balance fails. https://bugzilla.gnome.org/show_bug.cgi?id=670197 2012-02-16 12:59:10 +0000 Tuukka Pasanen * sys/v4l2/v4l2_calls.c: v4l2src: fix for webcamstudio vloopback Because vlooback emits 25 - ENOTTY and no EINVAL v4l2src thought it can't handle this and does not work. https://bugzilla.gnome.org/show_bug.cgi?id=669455 2012-02-13 12:06:37 +0100 Mark Nauwelaerts * tests/check/elements/flacparse.c: tests: flacparse: check and compare intended data 2012-02-09 22:12:14 +0100 Mark Nauwelaerts * tests/check/elements/mpegaudioparse.c: tests: mpegaudioparse: remove stray declaration 2012-02-09 10:11:48 +0100 Marc Leeman * gst/udp/gstmultiudpsink.c: multiudpsink: typo fix (bytes send -> bytes sent) 2012-02-07 14:10:44 -0800 Ralph Giles * ext/shout2/gstshout2.c: shout2send: send video/webm through libshout. This requires SHOUT_FORMAT_WEBM, added in libshout 2.3.0, so video/webm support is contingent on that symbol being defined. Also an indentation change required by the pre-commit hook. https://bugzilla.gnome.org/show_bug.cgi?id=669590 2012-01-28 11:13:16 +0100 Nicola Murino * gst/matroska/matroska-demux.c: matroskademux: avoid posting invalid duration for each frame https://bugzilla.gnome.org/show_bug.cgi?id=666583 2012-02-05 13:40:13 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.30.3 pre-release 2012-02-03 22:05:59 +0530 Arun Raghavan * ext/pulse/plugin.c: pulseaudiosink: Lower rank to prevent autoplugging pulseaudiosink breaks visualisations in its current form, so let's prevent it from being autoplugged for the time being. The best we can hope to do in the 0.10 series is query the list of available sinks and their formats, and expose these as the bin's sinkpad caps. While this is not a comprehensive solution, it will make sure that we're only trying to support compressed formats if we're certain that one exists. The long-term fix for this will be in the form of proper upstream renegotiation support in the 0.11/1.0 series. https://bugzilla.gnome.org/show_bug.cgi?id=666361 2012-02-03 14:53:31 +0000 Vincent Penquerc'h * ext/flac/gstflacenc.c: flacenc: fix event leak when there is no peer on the src pad 2012-02-02 12:27:09 +0000 Vincent Penquerc'h * gst/flv/gstflvmux.c: flvmux: specify we only accept raw AAC in template caps No header seems to be added, and the codec ID is the same as used for raw by flvdemux, so raw seems the only supported case. https://bugzilla.gnome.org/show_bug.cgi?id=665394 2012-02-02 12:25:21 +0000 Vincent Penquerc'h * gst/flv/gstflvdemux.c: flvdemux: specify we only output raw AAC in template caps https://bugzilla.gnome.org/show_bug.cgi?id=665394 2012-01-30 14:52:37 +0000 Vincent Penquerc'h * gst/rtp/gstrtpmp2tpay.c: rtpmp2tpay: do not try to flush a packet when no data is available https://bugzilla.gnome.org/show_bug.cgi?id=668874 2010-06-11 08:36:33 +0200 Pascal Buhler * gst/rtp/gstrtph264depay.c: rtph264depay: Exclude NALu size from payload length on truncated packets. https://bugzilla.gnome.org/show_bug.cgi?id=667846 2012-01-28 13:05:09 +0000 Vincent Penquerc'h * gst/videobox/gstvideobox.c: videobox: avoid wrapping opaque to transparent 2012-01-25 15:21:44 +0000 Jayakrishnan M * ext/cairo/Makefile.am: cairo: fix build, make sure libgstvideo can be found https://bugzilla.gnome.org/show_bug.cgi?id=668648 2012-01-25 13:19:12 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpsession.c: rtpmanager: don't pretend our random hostnames are fully-qualified domain names 2012-01-23 13:15:46 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/rtpsession.c: rtpmanager: don't reveal the user's username, hostname or real name by default Send a randomly made-up user@hostname as CNAME and don't send a NAME at all by default. https://bugzilla.gnome.org/show_bug.cgi?id=668320 2012-01-20 17:06:42 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: simplify internal src event debug logging ... which avoids almost superfluous obtaining of rtsp element. 2012-01-20 17:03:50 +0100 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: avoid NULL string comparison 2012-01-20 17:02:15 +0100 Mark Nauwelaerts * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: prevent out-of-bound array access 2012-01-20 17:01:37 +0100 Mark Nauwelaerts * gst/isomp4/atomsrecovery.c: isomp4: recovery: add sanity check ... on possibly bogus/corrupt input data. 2012-01-20 16:58:28 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroska-demux: remove redundant variable 2012-01-20 16:57:52 +0100 Mark Nauwelaerts * gst/deinterlace/gstdeinterlace.c: deinterlace: fix arithmetic for unsigned comparison 2012-01-20 16:55:06 +0100 Mark Nauwelaerts * gst/imagefreeze/gstimagefreeze.c: imagefreeze: add various missing break 2012-01-20 16:49:14 +0100 Mark Nauwelaerts * gst/alpha/gstalphacolor.c: alphacolor: remove redundant statement 2012-01-20 16:48:49 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: improve upstream peer duration querying ... to avoid accepting unhandled duration query result. 2012-01-20 16:47:36 +0100 Mark Nauwelaerts * ext/pulse/pulsesrc.c: pulsesrc: additional error condition checking 2012-01-20 16:46:21 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: additional error condition checking 2012-01-20 16:44:21 +0100 Mark Nauwelaerts * ext/jpeg/gstjpegenc.c: jpegenc: check _alloc_buffer result and perform fallback alloc if needed ... rather than carrying on with NULL buffer. 2012-01-13 18:11:36 +0000 Vincent Penquerc'h * ext/pulse/pulsesrc.c: pulsesrc: fix wrong error check pa_stream_* functions return negative on error, despite the defines for error codes being positive. I only got to repro the error twice, so I'm not sure 100% sure this fixes the issue (the negative var being uninitialized after returning from pa_stream_get_latency). 2012-01-16 17:51:18 +0000 Vincent Penquerc'h * gst/cutter/gstcutter.c: cutter: fix leak of unused GValue 2012-01-16 16:10:08 +0000 Vincent Penquerc'h * tests/check/elements/autodetect.c: tests: fix autodetect test not testing correctly for state change success State change to PAUSED can be done async, so if this happens, we need to wait for the change to be done (or failed). 2012-01-16 15:42:46 +0000 Vincent Penquerc'h * gst/rtp/gstrtph263ppay.c: rtph263ppay: fix caps leak 2012-01-16 12:13:50 +0000 Vincent Penquerc'h * gst/deinterlace/gstdeinterlace.c: deinterlace: make interlacedness test deterministic If the interlaced flag is not present in the caps, we assume the data is not interlaced, instead of leaving the boolean uninitialized. 2012-01-13 17:43:49 +0000 Vincent Penquerc'h * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: oss4: fix caps leaks 2012-01-13 17:25:59 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2src.c: v4l2src: fix caps leak 2012-01-13 15:57:20 +0000 Vincent Penquerc'h * tests/check/elements/videocrop.c: tests: fix caps leak in videocrop test 2012-01-13 10:32:59 +0000 Tim-Philipp Müller * gst/rtpmanager/gstrtpptdemux.c: rtpptdemux: plug pad leak in error code path Based on patch by: Stig Sandnes Don't leak srcpad if there are no caps. https://bugzilla.gnome.org/show_bug.cgi?id=667820 2011-10-04 10:00:02 +0200 Stig Sandnes * sys/osxvideo/cocoawindow.m: osxvideo: Fix leak of NSOpenGLPixelFormat object https://bugzilla.gnome.org/show_bug.cgi?id=667818 2011-09-05 10:43:19 +0200 Havard Graff * sys/v4l2/gstv4l2src.c: v4l2src: Don't assert when the interface is not implemented. Simply return FALSE instead. https://bugzilla.gnome.org/show_bug.cgi?id=667817 2012-01-12 00:18:39 +0200 Raimo Järvi * sys/waveform/gstwaveformsink.c: * sys/waveform/gstwaveformsink.h: waveformsink: Fix mingw warnings https://bugzilla.gnome.org/show_bug.cgi?id=667719 2012-01-12 18:23:42 +0000 Vincent Penquerc'h * gst/rtpmanager/gstrtpssrcdemux.c: gstrtpssrcdemux: fix element leak 2012-01-12 14:19:22 +0000 Vincent Penquerc'h * gst/matroska/matroska-read-common.c: matroska: do not leak attachment buffers 2012-01-12 10:30:11 +0000 Vincent Penquerc'h * ext/flac/gstflacenc.c: flacenc: do not drop the first data buffer on the floor (and leak it either) 2012-01-11 18:45:33 -0300 Reynaldo H. Verdejo Pinochet * Android.mk: Temporarily disabling multifile for the Android build There is a hard dependency on inotify comming from gio. We are not currently bundling inotify with the Android dist so I'm disabling multifile for now until someone gets around to sort this out. This change fixes building on Android 2012-01-11 01:45:34 +0000 Tim-Philipp Müller * tests/check/pipelines/wavenc.c: tests: fix wavenc test on big endian wavenc only accepts little-endian PCM, but most of our elements such as audiotestsrc only produce or process audio in native endianness, so we need to plug a converter before wavenc on big endian systems. 2012-01-05 19:25:33 +0000 Vincent Penquerc'h * gst/isomp4/gstqtmux.c: isomp4: fix caps leak 2012-01-05 19:08:03 +0000 Vincent Penquerc'h * gst/isomp4/gstqtmux.c: isomp4: remove dead assignment 2012-01-04 19:40:14 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 11f0cd5 to cb5da59 2012-01-04 17:59:55 +0000 Tim-Philipp Müller * tests/check/elements/qtmux.c: tests: fix some leaks and remove files when done in qtmux test 2011-12-14 10:14:20 +0100 Peter Seiderer * gst/multifile/gstmultifilesink.c: multifilesink: post better error message when we run out of disk space Map write errno ENOSPC to GST_RESOURCE_ERROR_NO_SPACE_LEFT. 2011-12-27 11:50:03 +0000 Tim-Philipp Müller * gst/udp/gstudpsrc.c: udpsrc: fix valgrind warning https://bugzilla.gnome.org/show_bug.cgi?id=666644 2011-12-21 13:22:03 +0100 John Ogness * gst/udp/gstudpsrc.c: udpsrc: drop dataless UDP packets It is allowed to send/receive UDP packets with no data. When such a packet is available, select() will return with success but ioctl(FIONREAD) will return 0. But a read() must still occur in order to clear off the UDP packet from the queue. This patch will read the dataless packet from the socket. If select() was woken for other reasons (and FIONREAD returns 0), this may result in a UDP packet getting accidentally dropped. But since UDP is not reliable, this is acceptable. NOTE: This patch fixes a nasty bug where sending a dataless UDP packet to a udpsrc instance will cause an infinite loop. https://bugzilla.gnome.org/show_bug.cgi?id=666644 Signed-off-by: John Ogness 2011-12-21 20:50:21 +0100 Nicola Murino * ext/jpeg/gstjpegdec.c: jpegdec: fix peer_caps leak https://bugzilla.gnome.org/show_bug.cgi?id=666688 2011-12-25 14:23:29 +0000 Tim-Philipp Müller * gst/flv/gstflvmux.c: flvmux: don't try to push already-freed buffers Fixes unit test. 2011-09-09 11:42:09 +0100 Vincent Penquerc'h * gst/audioparsers/gstac3parse.c: ac3parse: let bsid 9 and 10 through Files with 9 and 10 happen, and seem to comply with the <= 8 format, so let them through. The spec says nothing about 9 and 10. https://bugzilla.gnome.org/show_bug.cgi?id=658546 2011-12-16 19:15:38 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: properly determine final duration ... which can be authoratively obtained from our own written timestamps. 2011-12-19 13:56:30 +0100 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: only write full metadata at start ... rather than having (potentially) unnecessary duplicates written all over, or even contradictory varying filesize info, or duration info that will not be rewritten upon header rewrite. 2011-12-21 17:43:10 +0100 Branko Subasic * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: do not consider duration of non-finalized file ... to avoid it clamping requested seek position. Non-finalized file case, determined by whether _parse_blockgroup_or_simpleblock ever updates the segment duration. Fixes #652195. 2011-12-21 15:06:57 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: improve decision to fall back to scanning when seeking ... which is basically iff not streaming and no entry found in index 2011-12-13 18:18:45 +0100 Mark Nauwelaerts * gst/matroska/matroska-read-common.c: matroskademux: filter bogus index entries with missing block number ... to avoid contradictory information resulting in seeks sending more downstream than needed for the corresponding segment. 2011-12-13 18:15:18 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: cater for safer arithmetic with global start time 2011-12-13 17:02:01 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: tweak final closing segment sending ... to avoid it interfering with (sparse) stream syncing. 2011-12-12 11:54:56 +0100 Sebastian Dröge * gst-libs/gst/glib-compat-private.h: glib-compat: Add license boilerplate for LGPL 2011-12-12 15:15:46 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: mind (un)signed in some timestamp arithmetic ... to avoid ending up with invalid (negative) duration. 2011-02-09 15:31:22 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: increase parse tolerance for fuzzy file cases 2011-12-12 10:38:20 +0000 Tim-Philipp Müller * Makefile.am: build: dist glib-compat-private.h properly Add missing slash. 2011-12-12 10:18:14 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: use atexit, g_atexit has been deprecated in glib master 2011-12-12 02:52:13 +0000 Tim-Philipp Müller * ext/dv/gstdvdemux.c: * ext/flac/gstflacdec.c: * ext/wavpack/gstwavpackparse.c: * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/isomp4/gstqtmoovrecover.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtspsrc.c: * gst/videomixer/videomixer2.c: * gst/wavparse/gstwavparse.c: Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly GStaticRecMutex is part of our API/ABI, not much we can do here in 0.10 for most of these. 2011-12-12 02:41:37 +0000 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: * tests/icles/equalizer-test.c: * tests/icles/gdkpixbufsink-test.c: * tests/icles/test-oss4.c: * tests/icles/videocrop-test.c: tests: g_thread_init() is deprecated in glib master It's not needed any longer. 2011-12-12 02:38:37 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpclientsink.c: * gst/rtpmanager/gstrtpsession.c: * sys/oss4/oss4-mixer.c: * tests/icles/v4l2src-test.c: Use g_thread_try_new() instead of g_thread_crate() with newer glib versions 2011-12-12 02:31:36 +0000 Tim-Philipp Müller * gst/alpha/gstalpha.c: * gst/alpha/gstalpha.h: alpha: use new glib API for static mutex if available 2011-12-12 02:30:45 +0000 Tim-Philipp Müller * Makefile.am: * ext/jack/gstjackaudioclient.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/soup/gstsouphttpclientsink.c: * gst-libs/gst/glib-compat-private.h: * gst/audiofx/audiochebband.c: * gst/audiofx/audiocheblimit.c: * gst/audiofx/audiofirfilter.c: * gst/audiofx/audioiirfilter.c: * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: * gst/equalizer/gstiirequalizer.c: * gst/imagefreeze/gstimagefreeze.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/shapewipe/gstshapewipe.c: * gst/udp/gstmultiudpsink.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: * sys/oss4/oss4-mixer.c: * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2xoverlay.c: * sys/ximage/gstximagesrc.c: Work around deprecated thread API in glib master Add private replacements for deprecated functions such as g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly to avoid the deprecation warnings. We'll change these over to the new API once we depend on glib >= 2.32. 2011-12-12 10:24:45 +0100 Sebastian Dröge * configure.ac: configure: Require GLib >= 2.24 All other modules require this already and nobody is testing with older versions anyway. 2011-12-11 18:40:31 +0000 Tim-Philipp Müller * ext/gdk_pixbuf/gstgdkpixbufsink.c: gdkpixbufsink: fix inverted pixel-aspect-ratio Spotted by Mike Morrison. https://bugzilla.gnome.org/show_bug.cgi?id=665882 2011-12-11 17:55:14 +0000 Tim-Philipp Müller * ext/pulse/pulseaudiosink.c: pulseaudiosink: don't leak pad template 2011-12-10 15:13:07 +0000 Tim-Philipp Müller * configure.ac: * gst/deinterlace/tvtime-dist.c: * gst/videobox/gstvideoboxorc-dist.c: * gst/videomixer/blendorc-dist.c: * po/eo.po: * win32/common/config.h: 0.10.30.2 pre-release 2011-12-10 14:48:57 +0000 Tim-Philipp Müller * ext/soup/gstsouphttpclientsink.c: soup: fix start/stop race in souphttpclientsink Fix crash or hang in generic/states unit test when doing stop() right after start(). Create main loop in the start function already and not just in the thread function, so that stop() always has a valid main loop to quit on. Also, calling g_main_loop_quit() before g_main_loop_run() won't work and result in the stop function waiting for the thread to join forever. Therefore, wait for the thread to be ready and get the main loop running in the start() function, to be sure stop() always works. 2011-12-10 13:35:08 +0000 Tim-Philipp Müller * tests/files/Makefile.am: tests: dist test file used in matroskaparse unit test 2011-12-10 12:32:32 +0000 Tim-Philipp Müller * tests/check/elements/rgvolume.c: tests: fix up rgvolume test for basetransform event caching Some tests assumed that tag events would always pushed through immediately, which isn't the case any longer, so push a newsegment event and an empty buffer first. 2011-12-10 02:21:02 +0000 Tim-Philipp Müller * po/LINGUAS: * po/eo.po: * po/ja.po: * po/lv.po: * po/sr.po: po: update translations 2011-12-09 15:50:28 +0000 Tim-Philipp Müller * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: jack: don't leak client name when freeing the element And add gtk-doc chunks for the new property. https://bugzilla.gnome.org/show_bug.cgi?id=665872 2011-12-09 15:45:03 +0000 Nicolas Baron * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: jack: add "client-name" property to jackaudiosink and jackaudiosrc https://bugzilla.gnome.org/show_bug.cgi?id=665872 2011-12-08 11:00:45 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroskamux: stream-format=raw goes with aac caps, not mp3 caps 2011-12-02 12:07:24 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2src: do not ignore the highest frame interval https://bugzilla.gnome.org/show_bug.cgi?id=665387 2011-12-02 11:59:03 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2src: do not ignore the largest resolution The 'max' value isn't an STL style "one after the end" bound, but the largest allowed value. https://bugzilla.gnome.org/show_bug.cgi?id=665387 2011-12-06 16:47:25 +0100 Stefan Sauer * gst/multifile/gstmultifilesink.h: docs: add add the two enum values that were just added too 2011-12-06 16:14:54 +0100 Stefan Sauer * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/multifile/gstmultifilesink.h: multifilesink: expose the enum property docs for splitting mode. Fixes #665666. 2011-12-05 12:15:21 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2object.c: v4l2: replace deprecated GST_CLASS_LOCK 2011-11-24 13:58:01 +0100 Sebastian Rasmussen * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Ceil jpeg dimensions, instead of floor A JPEG image inside an RTP stream has a preceeding RFC2435 header that conveys width/height. The dimensions in this header are limited to be multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must already indirectly have image data dimensions that are rounded up in order to contain enough data to render the image. Therefore this fix safely rounds the image dimensions in the RFC2435 header up to the closest multiple of 8. 2011-12-04 12:50:57 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: ensure we only check for sample/block mixup at start Otherwise we might trigger at some point within the file, but the check is only making sense for the second block. 2011-12-03 18:14:59 +0000 Vincent Penquerc'h * gst/matroska/matroska-parse.c: matroskaparse: warn if accumulating headers after they were pushed https://bugzilla.gnome.org/show_bug.cgi?id=665412 2011-10-25 12:54:43 -0700 David Schleef * gst/matroska/matroska-parse.c: matroskaparse: fix parsing Mark more parts as belonging to streamheaders. 2011-12-03 17:30:10 +0000 Vincent Penquerc'h * gst/flv/gstflvdemux.c: flvdemux: fix discontinuity threshold check when timestamps go backwards Since unsigned types are used, a negative value would show as very, very positive. Fixes A/V sync on some... less than well made files where timestamps go backwards. 2011-12-02 12:01:22 +0000 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2src: add a comment about a "hidden" assumption on rank values https://bugzilla.gnome.org/show_bug.cgi?id=665387 2011-12-01 14:13:05 +0000 Tim-Philipp Müller * tests/check/Makefile.am: tests: fix up LIBS order som more` 2011-12-01 13:22:42 +0000 Tim-Philipp Müller * gst/matroska/matroska-mux.c: matroska-mux: fix name of new property and the unit test https://bugzilla.gnome.org/show_bug.cgi?id=654379 2011-09-25 14:57:56 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: add basic buffer list handling We assume for now that all buffers in a buffer list should end up in the same file (so we can group GOPs in buffer lists, for example). Could optimise this a bit to avoid the memcpy. 2011-09-23 18:43:35 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: multifilesink: write stream-headers when switching to the next file in max-size mode 2011-09-23 18:31:01 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: add new 'max-size' mode for switching to the next file 2011-09-23 17:49:05 +0100 Tim-Philipp Müller * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: multifilesink: add "max-file-size" property for new next-file mode 2011-12-01 13:38:06 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Don't forget SSA subtitles in last commit 2011-12-01 13:34:52 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: matroskademux: Only check for markup and escape if necessary for plaintext subtitles Otherwise we break USF and ASS/SSA subtitles. 2011-12-01 13:23:33 +0100 Alessandro Decina * gst/multifile/Makefile.am: multifile: fix build in uninstalled setup Add -base libs includes to CFLAGS, fix order of LIBS . 2011-12-01 13:08:01 +0100 Alessandro Decina * tests/check/elements/multifile.c: tests: fix g_mkdtemp presence check in multifile tests g_mkdtemp was added in glib 2.30 even though the doc claims it was added in 2.26. 2011-07-17 23:56:04 +0200 Alessandro Decina * gst/multifile/Makefile.am: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * tests/check/Makefile.am: * tests/check/elements/multifile.c: multifilesink: add flag to cut after a force key unit event 2011-12-01 12:47:26 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Copy all buffer flags when creating a subtitle buffer copy after postprocessing This also copies the caps. Otherwise we could end up pusing the first buffer without any caps, which causes downstream to not get notified about the caps. Fixes bug #664892. 2011-10-11 02:07:13 +0200 Alexey Fisher * gst/matroska/matroska-mux.c: matroskamux: make default framerate optional per stream there is at least two use cases where default frame rate should or may be disabled: - vp8 stream with altref frame enabled. If default frame rate is enabled, some players will missinterprete it (critical!) - for webm container, to reduce micro overhead - for stream with variable frame rate. Signed-off-by: Alexey Fisher 2011-11-30 22:13:11 +0100 Stefan Sauer * gst/effectv/gstripple.c: rippletv: fix CLAMP end-values 2011-11-30 19:25:37 +0000 Tim-Philipp Müller * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.interfaces: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-aasink.xml: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-alpha.xml: * docs/plugins/inspect/plugin-alphacolor.xml: * docs/plugins/inspect/plugin-annodex.xml: * docs/plugins/inspect/plugin-apetag.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-audioparsers.xml: * docs/plugins/inspect/plugin-auparse.xml: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-cacasink.xml: * docs/plugins/inspect/plugin-cairo.xml: * docs/plugins/inspect/plugin-cutter.xml: * docs/plugins/inspect/plugin-debug.xml: * docs/plugins/inspect/plugin-deinterlace.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-efence.xml: * docs/plugins/inspect/plugin-effectv.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-esdsink.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-flv.xml: * docs/plugins/inspect/plugin-flxdec.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-gdkpixbuf.xml: * docs/plugins/inspect/plugin-goom.xml: * docs/plugins/inspect/plugin-goom2k1.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-halelements.xml: * docs/plugins/inspect/plugin-icydemux.xml: * docs/plugins/inspect/plugin-id3demux.xml: * docs/plugins/inspect/plugin-imagefreeze.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-isomp4.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-level.xml: * docs/plugins/inspect/plugin-matroska.xml: * docs/plugins/inspect/plugin-monoscope.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-multifile.xml: * docs/plugins/inspect/plugin-multipart.xml: * docs/plugins/inspect/plugin-navigationtest.xml: * docs/plugins/inspect/plugin-oss4.xml: * docs/plugins/inspect/plugin-ossaudio.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-pulseaudio.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-rtsp.xml: * docs/plugins/inspect/plugin-shapewipe.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-smpte.xml: * docs/plugins/inspect/plugin-soup.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speex.xml: * docs/plugins/inspect/plugin-taglib.xml: * docs/plugins/inspect/plugin-udp.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-videobox.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-videofilter.xml: * docs/plugins/inspect/plugin-videomixer.xml: * docs/plugins/inspect/plugin-wavenc.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-wavparse.xml: * docs/plugins/inspect/plugin-ximagesrc.xml: * docs/plugins/inspect/plugin-y4menc.xml: docs: update docs 2011-11-30 19:00:42 +0000 Tim-Philipp Müller * gst/multifile/Makefile.am: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/patternspec.c: * gst/multifile/patternspec.h: splitfilesrc: specify filenames via normal wildcards instead of regular expressions Less cracktastic in the end. 2011-10-10 18:28:11 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: check bytes actually read, just in case Handle corner case where we try to read beyond the end of the last file part, in which case we want to return a short read. If we get fewer bytes than expected for any other file part, we should just error out, since something fishy's going on then. 2011-10-06 08:33:19 +0100 Tim-Philipp Müller * gst/multifile/gstsplitfilesrc.c: splitfilesrc: set offsets on buffers Looks like some parsers (in some versions at least) expect the offsets to be set, and behave weird if that's not the case (e.g. off-by-one in h264parse). 2011-07-28 20:19:56 +0100 Tim-Philipp Müller * configure.ac: * gst/multifile/Makefile.am: * gst/multifile/gstmultifile.c: * gst/multifile/gstsplitfilesrc.c: * gst/multifile/gstsplitfilesrc.h: multifile: add splitfilesrc element Add new splitfilesrc element that presents multiple files (selectable via a location regex) as one single contiguous file. 2011-11-29 17:34:10 -0300 Thiago Santos * ext/pulse/pulseaudiosink.c: Revert "pulseaudiosink: fix caps leak" This reverts commit d6a9de9e2aedc8b66ab3219902b5a37e8d65ada2. setcaps functions aren't supposed to take ownership of the caps passed 2011-11-28 12:58:44 +0000 Vincent Penquerc'h * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gstcairooverlay.c: * ext/cairo/gstcairorender.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacdec.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstswitchsink.c: * ext/gconf/gstswitchsrc.c: * ext/gdk_pixbuf/gstgdkpixbuf.c: * ext/gdk_pixbuf/gstgdkpixbufsink.c: * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: * ext/hal/gsthalaudiosrc.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/raw1394/gstdv1394src.c: * ext/raw1394/gsthdv1394src.c: * ext/shout2/gstshout2.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpsrc.c: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * ext/taglib/gsttaglibmux.c: * ext/wavpack/gstwavpackdec.c: * ext/wavpack/gstwavpackenc.c: * ext/wavpack/gstwavpackparse.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/audiofx/audiopanorama.c: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: * gst/autodetect/gstautoaudiosrc.c: * gst/autodetect/gstautovideosink.c: * gst/autodetect/gstautovideosrc.c: * gst/avi/gstavidemux.c: * gst/avi/gstavimux.c: * gst/avi/gstavisubtitle.c: * gst/cutter/gstcutter.c: * gst/debugutils/breakmydata.c: * gst/debugutils/cpureport.c: * gst/debugutils/efence.c: * gst/debugutils/gstcapsdebug.c: * gst/debugutils/gstcapssetter.c: * gst/debugutils/gstnavigationtest.c: * gst/debugutils/gstnavseek.c: * gst/debugutils/gstpushfilesrc.c: * gst/debugutils/gsttaginject.c: * gst/debugutils/progressreport.c: * gst/debugutils/rndbuffersize.c: * gst/debugutils/testplugin.c: * gst/deinterlace/gstdeinterlace.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstop.c: * gst/effectv/gstquark.c: * gst/effectv/gstradioac.c: * gst/effectv/gstrev.c: * gst/effectv/gstripple.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flv/gstflvdemux.c: * gst/flv/gstflvmux.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/goom2k1/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/imagefreeze/gstimagefreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstrtpxqtdepay.c: * gst/isomp4/qtdemux.c: * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: * gst/law/mulaw-decode.c: * gst/law/mulaw-encode.c: * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: * gst/matroska/matroska-parse.c: * gst/matroska/webm-mux.c: * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrgvolume.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpac3pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpceltdepay.c: * gst/rtp/gstrtpceltpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpdvdepay.c: * gst/rtp/gstrtpdvpay.c: * gst/rtp/gstrtpg722depay.c: * gst/rtp/gstrtpg722pay.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg726depay.c: * gst/rtp/gstrtpg726pay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729pay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpj2kdepay.c: * gst/rtp/gstrtpj2kpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpjpegpay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmparobustdepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpmpvpay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqcelpdepay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirendepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/shapewipe/gstshapewipe.c: * gst/smpte/gstsmpte.c: * gst/smpte/gstsmptealpha.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videocrop/gstaspectratiocrop.c: * gst/videocrop/gstvideocrop.c: * gst/videofilter/gstgamma.c: * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer2.c: * gst/wavenc/gstwavenc.c: * gst/wavparse/gstwavparse.c: * gst/y4m/gsty4mencode.c: * sys/directsound/gstdirectsoundsink.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/osxvideo/osxvideosink.m: * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosrc.c: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2src.c: * sys/waveform/gstwaveformsink.c: * sys/ximage/gstximagesrc.c: * tests/check/elements/qtmux.c: various: fix pad template leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664 2011-11-28 11:47:11 +0100 Chad * gst/debugutils/gsttaginject.c: taginject: set gap-aware The element does not modify the data anyway. 2011-11-26 21:39:33 +0100 Stefan Sauer * gst/equalizer/gstiirequalizer.c: equalizer: also sync the parameters for the filter bands 2011-11-26 16:06:59 +0000 Tim-Philipp Müller * gst/matroska/matroska-ids.c: matroskademux: initialise seen_markup_tag field on subtitle stream context 2011-11-25 19:28:55 -0300 Thiago Santos * gst/isomp4/gstqtmuxmap.c: ismlmux: Use iso-fragmented as variant type Using 'iso' conflicts with mp4mux variant type, ismlmux now uses iso-fragmented Fixes #656823 2011-11-24 12:05:33 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: Implement GstStreamVolume interface PulseAudio 1.0 supports per-source-output volumes, and this exposes the functionality via the GstStreamVolume interface. When compiled against pre-1.0 PulseAudio, the interface is not implemented, and the "volume" or "mute" properties are not available. This bit of ugliness will go away when we can depend on PulseAudio 1.0 or greater. https://bugzilla.gnome.org/show_bug.cgi?id=595055 2011-09-10 21:21:38 -0700 Arun Raghavan * ext/pulse/pulsesrc.c: pulsesrc: Trivial comment copy-paste-o fix 2011-11-14 12:43:27 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulseaudiosink: Remove redundant code 2011-11-14 12:41:41 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulseaudiosink: Clean up refcounting in event probe Makes sure we don't leak a refcount if the object is disposed before a NEWSEGMENT turns up. 2011-11-24 16:31:38 +0000 Vincent Penquerc'h * gst/flv/gstflvdemux.c: flvdemux: fix seeking Which I accidentally broke when fixing flv videos breaking on spurious timestamp discontinuities in broken files. https://bugzilla.gnome.org/show_bug.cgi?id=631430 2011-11-25 13:13:47 +0100 Stefan Sauer * gst/effectv/gstradioac.c: * gst/effectv/gstradioac.h: effectv: repair color modes in radioactv by taking rgb,bgr into account 2011-11-25 11:44:49 +0100 Stefan Sauer * gst/effectv/gstradioac.c: radioactv: add one more set of caps It also work in this format. Avoids the need for conversion. 2011-11-25 11:44:18 +0100 Stefan Sauer * gst/effectv/gstradioac.c: * gst/effectv/gstshagadelic.c: effecttv: fix reverse negotiation The plugins were using _fixed_caps_ and thus not adjusting to new upstream sizes. Spotted by Tim Müller. 2011-11-25 11:43:16 +0100 Stefan Sauer * gst/effectv/gstwarp.c: warptv: remove not needed ifdef 2011-11-25 10:15:35 +0100 Stefan Sauer * gst/effectv/gstripple.c: rippletv: clean up the rendering code a bit This is corrrupts the memoy when resizing. Add a FIXME to make it resizeable once that is solved. 2011-11-24 20:42:49 +0100 Stefan Sauer * gst/effectv/gstquark.c: * gst/effectv/gststreak.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: effecttv: fix reverse negotiation The plugins were using _fixed_caps_ and thus not adjusting to new upstream sizes. Spotted by Tim Müller. 2011-11-24 14:14:53 -0300 Thiago Santos * gst/multifile/gstmultifilesink.c: multifilesink: Fix leak of filename strings Do not forget to free the filename strings when deleting the list of files. 2011-11-24 14:11:33 -0300 Thiago Santos * tests/check/elements/multifile.c: multifile: fix build of tests Tests fail to build because g_mkdtemp is available from glib since 2.26. This patch adds a condition around the redefinition of g_mkdtemp on the tests to only build it if glib is older than 2.26. 2011-09-27 16:49:45 +0100 Vincent Penquerc'h * gst/wavparse/gstwavparse.c: wavparse: skip id32 tags This allows decoding at least one sample where something has stuffed some ID3 tag before the (supposedly initial) FMT\ . https://bugzilla.gnome.org/show_bug.cgi?id=660249 2011-10-31 17:06:18 +0000 Vincent Penquerc'h * gst/effectv/gstedge.c: edgetv: trivial comment fix for clarity https://bugzilla.gnome.org/show_bug.cgi?id=661841 2011-10-31 17:04:23 +0000 Vincent Penquerc'h * gst/effectv/gstedge.c: edgetv: don't leave bits of the output buffer uninitialized Let's initialize them to zero. It looks alright, but then it also looks alright with v3, or with the corresponding pixels from the source. I don't know what the original intent would be, and the original effectv source also has this bug/feature. https://bugzilla.gnome.org/show_bug.cgi?id=661841 2011-11-24 10:25:02 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstamrparse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparse: Use the sinkpad template caps as fallback, not the srcpad ones 2011-11-24 09:59:40 +0100 Sebastian Dröge * gst/audioparsers/gstmpegaudioparse.c: mpegaudioparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:57:57 +0100 Sebastian Dröge * gst/audioparsers/gstflacparse.c: flacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:55:47 +0100 Sebastian Dröge * gst/audioparsers/gstdcaparse.c: dcaparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:53:18 +0100 Sebastian Dröge * gst/audioparsers/gstamrparse.c: amrparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:49:27 +0100 Sebastian Dröge * gst/audioparsers/gstamrparse.c: amrparse: Mark some more functions as static 2011-11-24 09:48:33 +0100 Sebastian Dröge * gst/audioparsers/gstac3parse.c: ac3parse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-24 09:44:58 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Mark some functions as static and remove unused function declarations 2011-11-24 09:43:14 +0100 Sebastian Dröge * gst/audioparsers/gstaacparse.c: aacparse: Implement ::get_sink_caps vfunc to propagate downstream caps constraints upstream 2011-11-23 00:57:39 +0000 Tim-Philipp Müller * tests/check/Makefile.am: * tests/check/elements/.gitignore: * tests/check/elements/matroskaparse.c: * tests/files/pinknoise-vorbis.mkv: tests: add basic unit test for matroskaparse 2011-11-23 00:56:26 +0000 Tim-Philipp Müller * gst/matroska/matroska-parse.c: matroskaparse: don't leak stream headers https://bugzilla.gnome.org/show_bug.cgi?id=664548 2011-11-16 19:08:05 +0100 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: ensure to free allocated padded data 2011-11-16 18:57:38 +0100 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: reset tag setter interface when appropriate 2011-11-16 18:57:21 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: reset tag setter interface when appropriate 2011-11-14 15:34:57 +0000 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstflacparse.h: flacparse: detect when a file lies about fixed block size If the sample/block number happens to be the same as the block size, we assume variable block size, and thus counters in samples in the headers. This can only get us a false positive for a block size of 1, which is invalid. We can get false negatives more often though (eg, if not starting at the start of the stream), but then that's already GIGO. 2011-09-02 19:20:07 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: gstrtpsession: Add special mode to use FIR as repair as Google does https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-09-01 17:47:38 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: rtpsession: Send FIR requests in response to key unit requests with all-headers=TRUE https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-09-01 16:25:21 -0400 Olivier Crête * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: rtpsession: Put the PLI requests in each RTPSource Also refactor a bit and put all the keyframe request code in one place inside rtpsession.c https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-08-31 14:35:33 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Hack to FIR because Google doesn't set the sender ssrc correctly https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-08-30 19:06:13 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Process received Full Intra Requests Process FIR requests according to RFC 5104 https://bugzilla.gnome.org/show_bug.cgi?id=658419 2011-11-07 18:43:26 +0000 Sjoerd Simons * sys/v4l2/gstv4l2object.c: v4l2: Set pixel-aspect-ratio to 1/1 We don't currently support setting the pixel-aspect-ratio from V4L2. So simply set it to be 1/1 in the caps to prevent negotiation failures when fixating to weird values (e.g. when the downstream caps has pixel-aspect-ratio = [ MIN, MAX ] ) https://bugzilla.gnome.org/show_bug.cgi?id=663580 2011-11-11 10:06:25 -0300 Thiago Santos * ext/pulse/pulseaudiosink.c: pulseaudiosink: fix caps leak 2011-11-11 14:55:48 +0100 Mark Nauwelaerts * ext/pulse/pulsesink.c: pulsesink: do not leak clientname when setting up property 2011-11-11 18:05:35 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulse: Chain up dispose() in pulseaudiosink 2011-11-08 15:35:26 +0000 Vincent Penquerc'h * gst/avi/gstavidemux.c: avidemux: fix wrong stride when inverting uncompressed video Such frames have a stride multiple of 4, see http://lscube.org/pipermail/ffmpeg-issues/2010-April/010247.html. This showed up on a sample using a odd width of 24 bit video. https://bugzilla.gnome.org/show_bug.cgi?id=652288 2011-11-09 10:32:06 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: minimal sanity check on creation datetime 2011-11-02 12:58:12 -0400 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Return the sink pad template as sink caps, not the src's https://bugzilla.gnome.org/show_bug.cgi?id=577784 2009-03-15 19:26:48 -0400 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Also implement size/framerate restrictions in getcaps https://bugzilla.gnome.org/show_bug.cgi?id=577784 2009-03-04 20:50:19 -0500 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes https://bugzilla.gnome.org/show_bug.cgi?id=577784 2011-11-08 14:31:34 +0100 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: also set segment stop at startup rather than only post seek ... so as to ensure consistent playback with or without seek, especially in presence of some bogus edit list entries. 2011-11-02 17:02:54 +0000 Raul Gutierrez Segales * gst/flv/Makefile.am: gst/flv/: add amfdefs.h to noinst_HEADERS https://bugzilla.gnome.org/show_bug.cgi?id=663334 2011-10-03 17:50:43 +0100 Vincent Penquerc'h * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: detect large pts gaps and resync Should work on multiple gaps, but tested on only one. https://bugzilla.gnome.org/show_bug.cgi?id=631430 2011-08-22 10:40:45 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: fix off by one between granpos and last_stop 2011-10-07 19:41:35 +0100 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: flacparse: fix last frame timestamp in fixed block size mode The last block may have a different block size, so we should not use it to scale or we'll end up with a wrong timestamp. See comment and quote from the FLAC format documentation in the code. Fixes looped playback of FLAC files (via about-to-finish). https://bugzilla.gnome.org/show_bug.cgi?id=661215 2011-10-27 15:52:47 +0100 Vincent Penquerc'h * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttextoverlay.h: cairotextoverlay: add a 'silent' property to skip rendering https://bugzilla.gnome.org/show_bug.cgi?id=662856 2011-11-07 12:00:12 +0100 René Stadler * gst/matroska/ebml-write.c: matroskamux: fix regression causing malformed files This was caused by me in 1b213d. It seems I was too focused on 0.11 when I did this and tested the wrong branch. The problem was reported by Alexey Fisher. 2011-11-03 23:28:31 +0000 Tim-Philipp Müller * gst/rtp/gstrtpvrawdepay.c: rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN Fixes compiler warning on mingw32 2011-10-31 16:18:32 +0100 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: avoid shortcut evaluation when adding paired mp4 tag Fixes (part of) #638711. 2011-10-31 15:43:25 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: do not use unoffical V_MJPEG codec id ... but as not spec'ed especially, consider it a VfW compatibility case. Fixes #659837. 2011-10-30 19:30:14 +0000 Tim-Philipp Müller * ext/flac/gstflacenc.h: flacenc: remove dead code from header We require a new-enough libflac that this condition will never apply. 2011-10-28 09:57:36 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegdec.c: jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg jpegdec (using libjpeg 6.2/8) can't decode some lossless types of JPEG. https://bugzilla.gnome.org/show_bug.cgi?id=556648 2011-10-28 12:30:33 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: elaborate some debug statements 2011-10-11 20:56:51 +0400 Stas Sergeev * gst/flv/gstflvdemux.c: flvdemux: be careful with negative cts Fixes #661477. 2011-10-06 13:04:54 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: tune non-update seek handling cases Fixes #661049. 2011-10-28 10:40:36 +0200 Sebastian Dröge * gst/videomixer/videomixer2.c: videomixer2: Use the clip function instead of the prepare_buffer function 2011-10-28 09:36:17 +0200 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/gstcollectpads2.c: * gst/videomixer/gstcollectpads2.h: * gst/videomixer/videomixer2.h: * gst/videomixer/videomixer2pad.h: videomixer2: Use collectpads2 from core 2011-10-28 00:41:45 +1100 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: deinterlace: Don't pointlessly hold object lock over caps operations Avoids a deadlock when getcaps is recursive due to the getcaps being reflected upstream/downstream. The lock isn't actually protecting anything here. 2011-10-27 00:37:03 +1100 Jan Schmidt * gst/flv/amfdefs.h: * gst/flv/gstflvmux.c: flvmux: add some comments and defines to clarify code. 2011-10-10 15:36:14 +0200 René Stadler * gst/matroska/ebml-write.c: matroska: refactor ebml-write to be more 0.11 friendly Switching to a more 0.11-friendly pattern, where getting the buffer's data pointer and setting the size many times is less natural. This is of course in preparation to the upcoming port of the plugin. 2011-10-11 21:45:46 +0200 René Stadler * gst/matroska/ebml-write.c: matroska: remove stale floatcast include GDOUBLE_TO_BE was moved to core a long time ago. 2011-10-11 22:10:27 +0200 René Stadler * gst/matroska/matroska-mux.c: matroskamux: fix possible crash with malformed dirac codec_data Since size is unsigned, we need to safeguard against wrapping below zero. 2011-10-21 22:33:34 +0200 René Stadler * gst/equalizer/gstiirequalizer.c: equalizer: remove avoidable call to gst_object_set_name 2011-10-21 22:32:38 +0200 René Stadler * gst/deinterlace/gstdeinterlace.c: deinterlace: remove avoidable call to gst_object_set_name 2011-10-16 20:30:25 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: increase arbitrary resolution limits Apparently libpng can technically do up to 2^31-1 rows and columns. However it imposes an (arbitrary) default limit of 1 million (that could theoretically be lifted by using some additional API). Moved array allocation to the heap now. 2011-10-16 20:25:41 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: don't unconditionally allocate 4096 pointers on the stack Instead allocate as many as needed (on the stack still). 2011-10-16 20:05:28 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: ensure setcaps was called before chain function This is needed to properly error out for e.g. "fakesrc ! pngenc ! fakesink". 2011-10-16 19:44:27 +0200 René Stadler * ext/libpng/gstpngenc.c: pngenc: validate input buffer size Just for safety; of course such mismatch represents a bug in another element. 2011-10-16 19:41:28 +0200 René Stadler * ext/libpng/Makefile.am: * ext/libpng/gstpngenc.c: * ext/libpng/gstpngenc.h: pngenc: make setcaps more robust, use gstvideo functions A setcaps function needs to actually verify the caps carefully. In this case, it was possible to e.g. link a video decoder with YUV+RGB template caps to pngenc. That would cause a crash when the decoder pushes a YUV buffer. Same thing when pushing a valid buffer that exceeds the resolution limits. Also, missing framerate caps field would cause a glib critical warning due to invalid GValue. This fails hard now. 2011-10-21 10:01:43 +0200 René Stadler * gst/matroska/matroska-read-common.c: ebml: small correction to previous commit Signal a short read with UNEXPECTED, exactly like the peek_bytes function. 2011-10-19 13:09:51 +0200 Edward Hervey * gst/matroska/matroska-read-common.c: ebml: Fix push-based behaviour The 'peek' method was completely wrong (!?) 2011-10-18 18:31:17 +0530 Arun Raghavan * ext/pulse/pulseaudiosink.c: pulse: Get caps correctly on pad block Instead of always going upstream, we should first see if already got caps from a setcaps() call. https://bugzilla.gnome.org/show_bug.cgi?id=661262 2011-10-18 12:25:14 +0100 Tim-Philipp Müller * ext/wavpack/gstwavpackenc.c: wavpackenc: don't unref buffer with gst_object_unref() 2011-10-18 12:05:01 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: only use is_pcm for 1.0 of pulseaudio 2011-10-18 11:58:57 +0200 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: only disable trickmodes for !pcm Only disable trickmodes when we are not dealing with raw PCM samples. 2011-10-14 10:56:16 +0530 Arun Raghavan * gst/videomixer/videomixer2.c: videomixer2: Fix a leak Buffers weren't being unref'ed in one case inside, causing memory usage to blow up. 2011-10-14 09:10:01 +0200 Marc Leeman * gst/rtp/gstrtpvrawdepay.c: set colour masks for video/x-raw-rgb in rtpvrawdepay 2011-10-13 16:59:50 +0530 Arun Raghavan * gst/videomixer/videomixer2.c: videomixer2: Fix incorrect gst_buffer_replace() call This got exposed when gst_buffer_replace() was changed from a macro to a function. 2011-10-12 11:26:50 +0200 Edward Hervey * gst/rtp/gstrtpvrawpay.c: rtpvrawpay: Only use 24 LSB for depth=24 RGB caps ... and indent the masks for clarity 2011-10-11 14:58:43 +0200 René Stadler * gst/matroska/matroska-mux.c: matroskamux: fix segment handling, so we actually use running time gst_matroska_mux_best_pad adjusts the buffer timestamp to running time using the segment stored in the pad's collect data. However, the event handler didn't pass the newsegment event on to collectpads' handler, so this segment was never updated at all. Re-fixes bug #432612. 2011-10-10 19:01:23 +0100 Sjoerd Simons * gst/rtp/gstrtpg722pay.c: gstrtpg722pay: Compensate for clockrate vs. samplerate difference The RTP clock-rate used for G722 is 8000, even though the samplerate is 16000. Compensate for this by pretending G722 has 8 bits per sample instead of the 4 bits as if it were a codec that ran at half the speed, but with twice the number of bits. Fixes #661376 2011-09-27 19:25:53 +0100 Sjoerd Simons * ext/jpeg/gstjpegdec.c: jpegdec: Implement upstream negotiation Add upstream negotiation for jpegdec. Fixes #660275 2011-10-10 19:02:58 +0100 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroska-demux: don't leak audio codec_data buffer 2011-10-10 13:20:04 +0200 Stefan Sauer * tests/examples/cairo/Makefile.am: tests: add missing PLUGIN_ASE_LIBS to LDADD 2011-10-09 21:31:27 +0200 Mark Nauwelaerts * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: speexenc: only push header buffers following initial events 2011-10-09 11:18:18 -0300 Thiago Santos * gst/isomp4/atomsrecovery.c: qtmux: Fix memory leak on atoms recovery function Remember to free the ftyp data after writing it to a file. Fixes #660969 2011-09-21 18:45:42 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: improve segment handling with non-zero starting timestamp ... as well as related items, such as seeking and position reporting. https://bugzilla.gnome.org/show_bug.cgi?id=659808 2011-09-29 18:41:53 +0400 Stas Sergeev * sys/v4l2/gstv4l2object.c: * sys/ximage/gstximagesrc.c: v4l2, ximagesrc: fix some printf format compiler warnings https://bugzilla.gnome.org/show_bug.cgi?id=660150 2011-09-30 12:42:22 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: Refactor bitrate check test Refactor bitrate check test to accomodate multiple tests for bitrate 2011-09-30 13:02:31 -0300 Thiago Santos * gst/isomp4/atoms.c: qtmux: update esds atom under wave atom for aac bitrates AAC in mov format puts an ESDS atom inside of a WAVE atom in STSD atom, we need to update the bitrate on this ESDS. This patch fixes it. 2011-09-30 12:41:52 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/fourcc.h: qtmux: Also update btrt atom When rewriting bitrates, also update the btrt atom under stsd 2011-09-30 10:55:53 -0300 Thiago Santos * tests/check/elements/qtmux.c: tests: qtmux: add tests for bitrate average calculation Adds tests to make sure qtmux/mp4mux sets average bitrate correctly 2011-09-28 11:41:49 -0300 Thiago Santos * gst/isomp4/atoms.c: * gst/isomp4/atoms.h: * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmux.h: qtmux: Calculate average bitrate for streams Calculate and use average bitrate for streams when no bitrate tag was received 2011-09-28 10:41:14 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: Avoid a buffer metadata copy if possible If first_ts is 0 there is no need to subtract, so we might skip some copying to make the buffer metadata writable. 2011-09-29 23:21:46 +0100 Tim-Philipp Müller * ext/speex/gstspeexenc.c: speexenc: initialise variable before adding to it 2011-09-29 17:21:22 +0200 Mark Nauwelaerts * ext/speex/gstspeexdec.c: * ext/speex/gstspeexdec.h: speexdec: port to audiodecoder 2011-09-29 16:33:01 +0200 Mark Nauwelaerts * ext/speex/gstspeexenc.h: speexenc: clean up some unused remnants 2011-09-29 17:32:23 +0200 Mark Nauwelaerts * ext/speex/Makefile.am: * ext/speex/gstspeexenc.c: * ext/speex/gstspeexenc.h: speexenc: port to audioencoder 2011-09-28 16:09:58 +0200 Mark Nauwelaerts * ext/flac/Makefile.am: * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: port to audioencoder 2011-09-27 15:59:24 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: * gst/matroska/matroska-ids.h: * gst/matroska/matroska-parse.c: matroskademux: ensure minimal alignment for audio/x-raw-* buffers Since matroskademux will attempt to push unaligned buffers, downstream might have trouble with those, especially if downstream uses ORC, such as audioconvert. Ensure we push buffers aligned to the basic type at least for those raw buffers. https://bugzilla.gnome.org/show_bug.cgi?id=659798 2011-09-28 00:10:09 +0300 Raimo Järvi * gst/goom2k1/goom_core.c: goom2k1: Fix compiler warnings on 64 bit mingw-w64 Fixes bug #660294. 2011-09-25 15:13:39 +0100 Tim-Philipp Müller * ext/soup/Makefile.am: * ext/soup/gstsoup.c: * ext/soup/gstsouphttpclientsink.c: * ext/soup/gstsouphttpclientsink.h: * ext/soup/gstsouphttpsink.c: * ext/soup/gstsouphttpsink.h: soup: rename souphttpsink to souphttpclientsink To avoid confusion, and because we might want a server sink at some point too. https://bugzilla.gnome.org/show_bug.cgi?id=659947 2011-09-23 16:39:46 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsink.c: * ext/soup/gstsouphttpsink.h: souphttpsink: don't create unused second sink pad object The base class will create the sink pad. 2011-09-23 15:36:36 +0200 Julien Isorce * gst/audioparsers/gstac3parse.c: ac3parse: correctly check for ac3/e-ac3 switch https://bugzilla.gnome.org/show_bug.cgi?id=659943 2011-09-20 13:38:53 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: improve downstream flow return feedback to upstream ... although basertpdepay does not really make it easy/possible to do so all the way. 2011-09-20 12:11:47 +0100 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: * sys/ximage/gstximagesrc.h: ximagesrc: add xid and xname properties to allow capturing a particular window A particular window may be selected using the new xid (X-Window XID, eg a pointer) and xname (window title) properties. If both are specified, the XID is used in preference, falling back to xname if not found. Default (if none of xid and xname are specified, or if no such window is found) is to capture the root window. https://bugzilla.gnome.org/show_bug.cgi?id=546932 2011-08-02 17:39:44 +0100 Tim-Philipp Müller * tests/check/elements/qtmux.c: tests: add unit test to make sure encodebin picks mp4mux for variant=iso https://bugzilla.gnome.org/show_bug.cgi?id=651496 2011-09-19 12:15:11 +0200 Ha Nguyen * gst/rtpmanager/gstrtpbin.c: rtpbin: Fix a leaked clock for each buffering message Fixes bug #659237. 2011-09-19 12:11:32 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: parse embedded ID32 tags 2011-09-02 13:41:41 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: rtpsession: avoid source premature timing out Use slightly adjusted sender interval to determine sender timeout rather than our own sender side interval (which may have been forced small). 2011-08-25 12:40:52 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: avoid timing out source too quickly ... following a PAUSE/PLAY cycle, particularly applicable when operating with a short RTCP interval (possibly forced so server-side). 2011-08-24 14:37:52 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer/rtpbin: relax dropping rtcp packets ... to at least having it trigger a/v synchronization, possibly without using provided values which are still not considered sane (as previously dropped). 2011-08-24 14:34:23 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: some more reset when clearing pt map ... which in particular caters for some more reset following a possible rtsp PLAY. 2011-08-21 21:58:38 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: do not set elements to PLAYING when doing seek in PAUSED 2011-09-01 14:47:48 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: only reset skew on gap if input ts available 2011-08-18 14:12:21 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: check some more for possible rtp timestamp discontinuity ... when operating in non slave mode, and reset if detected. This should avoid some (large) bogus outgoing timestamp due to jumps in rtp time, as result of PAUSE/PLAY or seek or ... 2011-08-08 12:48:50 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: switch to rtp time based syncing when guessed appropriate 2011-08-08 12:15:20 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: alternative inter-stream syncing methods ... at least if not syncing to NPT time: * either sync using RTCP SR data (as currently) * only perform the above once using initial RTCP SR packets * discard RTCP and sync by equating provided stream's clock-base rtptime, as provided by jitterbuffer (typically obtained from RTP-Info in RTSP). 2011-08-08 12:11:24 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: also provide clock-base to sync signal 2011-08-08 12:09:41 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: allow configurable rtcp stream syncing interval ... rather than necessarily syncing at each RTCP SR. 2011-08-01 08:35:01 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpsession.c: rtpsession: trigger reconsideration if rtcp interval set 2011-08-01 08:32:24 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: configure rtcp interval if provided ... in PLAY response. 2011-09-16 16:53:22 +0300 Lasse Laukkanen * gst/isomp4/gstqtmux.c: isomp4: Fix allowing zero duration tracks https://bugzilla.gnome.org/show_bug.cgi?id=637486 2011-09-05 10:11:18 +0100 Vincent Penquerc'h * gst/udp/gstudpnetutils.c: udpsrc: error out when no protocol is specified in the uri It is certainly better than to crash. https://bugzilla.gnome.org/show_bug.cgi?id=658178 2011-09-19 09:37:58 +0200 Vincent Penquerc'h * ext/speex/gstspeexenc.c: speexenc: do not use invalid buffer timestamps 2011-03-29 12:09:18 +0530 Arun Raghavan * ext/pulse/Makefile.am: * ext/pulse/plugin.c: * ext/pulse/pulseaudiosink.c: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.h: pulse: New pulseaudiosink element to handle format changes This introduces a new bin which wraps around pulsesink and depending on the formats supported by the sink, plugs in/out a decodebin2 as required. This allows users to switch sinks on the stream and adapts accordingly (for example, you could watch a movie in passthrough mode on your receiver which supports AC3 decode, then plug out and switch to a non-digital profile to continue uninterrupted on analog output). The bin is required because doing the same with playbin2/playsink will require API changes that cannot be made in 0.10. With 0.11/1.0, we should be able to ask for upstream caps renegotiation to deal with all this. https://bugzilla.gnome.org/show_bug.cgi?id=657179 2011-09-16 15:03:23 +0200 Branko Subasic * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-read-common.c: matroskademux: Avoid sending EOS when in paused state Changed the ebml reader's gst_ebml_peek_id_length() function so that it returns the actual reason for why the peek failed, instead of (almost) always returning GST_FLOW_UNEXPECTED. This prevents the pulling task from sending EOS when doing a flushing seek. 2011-09-15 15:53:47 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: fix stuttering A/V Someone got had by implicit promotion to unsigned in ops with a signed and an unsigned value. https://bugzilla.gnome.org/show_bug.cgi?id=659153 2011-09-14 16:37:12 +0100 Vincent Penquerc'h * gst/debugutils/gstnavseek.c: navseek: toggle pause/play on space bar A useful thing to have. https://bugzilla.gnome.org/show_bug.cgi?id=659065 2011-09-14 14:46:00 +0200 David Svensson Fors * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: configurable timestamp gap handling matroskademux performs segment tricks to skip gaps in streams, notably at start for non 0 based files. There may however be cases when full presentation (including intermediate gaps) is desired, so a property allows to configure as of which gap to act (or not at all). API: GstMatroskaDemux::max-gap-time Fixes #659009. 2011-09-12 09:21:47 -0300 Thiago Santos * tests/check/elements/flvmux.c: tests: flvmux: Fix flvmux's tests after fix for request pads handling Now that flvmux doesn't release its request pads on PAUSED->READY the test doesn't need to re-request them for every reuse test start. 2011-09-09 09:12:56 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: Fix ctts generation for streams that don't start at 0 timestamps Subtract the first timestamp of a stream from all input buffers to get 0-based timestamps for creating a sane ctts table. Without this patch the ctts could have larger values than needed, causing the playback to have a delay at startup. As the first timestamp is only found after a few buffers are queued (due to possible reordered buffers), once we find the first timestamp we subtract it from all buffers on the queue, from that point on, all buffers have their timestamps subtract when they are collected. https://bugzilla.gnome.org/show_bug.cgi?id=658659 2011-09-12 07:55:19 +0200 Alessandro Decina * gst/flv/gstflvmux.c: flvmux: don't release request pads going PAUSED->READY Don't release request pads but just reset them. This makes pipelines using flvmux reusable. 2011-09-09 12:35:50 +0100 Vincent Penquerc'h * gst/audioparsers/gstac3parse.c: ac3parse: use bsid 9 and 10 to control sample rate See http://matroska.org/technical/specs/codecid/index.html The spec is silent about this though... https://bugzilla.gnome.org/show_bug.cgi?id=658546 2011-09-07 14:13:03 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: ensure some initial state variable setup ... which might otherwise be skipped if the PLAY command is issued before the OPEN command had a chance to actually be acted upon. Fixes #657376. 2011-09-08 15:02:05 +0200 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: tweak gap handling ... so as to avoid buffers before and after gap to have identical running time. 2011-09-08 13:28:24 +0200 Guillaume Desmottes * sys/v4l2/gstv4l2object.c: v4l2: use GST_RESOURCE_ERROR_BUSY if v4l2_ioctl fails with EBUSY https://bugzilla.gnome.org/show_bug.cgi?id=658543 2011-09-07 08:54:17 -0300 Thiago Santos * gst/isomp4/gstqtmux.c: qtmux: remove one G_UNLIKELY for user property Using G_UNLIKELY on user properties isn't nice, specially when that is the default option. 2011-03-15 11:03:53 +0100 Andoni Morales Alastruey * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: handle GstForceKeyUnit event ... by starting a new cluster after forwarding event. Fixes #644154. 2011-09-07 14:27:36 +0200 Sebastian Dröge * tests/check/elements/cmmldec.c: * tests/check/elements/cmmlenc.c: cmml: Use complete cmml caps in the unit test 2011-09-07 14:26:01 +0200 Sebastian Dröge * tests/check/elements/qtmux.c: qtmux: Use complete MPEG caps in the unit test 2011-09-07 14:18:58 +0200 Stefan Sauer * docs/plugins/Makefile.am: docs: cleanup makefiles Remove commented out parts that we don't need. Remove "the wingo addition" - no so useful after all. Narrow down file-globs for plugin docs. 2011-08-29 14:12:22 +0200 Konstantin Miller * ext/soup/gstsouphttpsrc.c: souphttpsrc: Don't handle HTTP response 407 as error if proxy authentication data is available Fixes bug #657422. 2011-09-07 12:11:39 +0200 Sebastian Dröge * gst/audioparsers/gstac3parse.c: ac3parse: Add Converter to the classification because it can convert between different alignments This allows decodebin2 to let it negotiate properly. 2011-09-07 12:10:48 +0200 Sebastian Dröge * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstdcaparse.c: * gst/audioparsers/gstflacparse.c: * gst/audioparsers/gstmpegaudioparse.c: audioparsers: Improve src template caps Remove the parsed/framed fields and add all fields to the template caps that always exist. 2011-09-06 15:59:49 +0200 Mark Nauwelaerts * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: parse codec_data to determine number of samples per frame Fixes #656734. 2011-09-06 21:24:46 +0200 Stefan Sauer * common: Automatic update of common submodule From a39eb83 to 11f0cd5 2011-09-06 15:40:32 +0200 Stefan Sauer * common: Automatic update of common submodule From 605cd9a to a39eb83 2011-09-06 15:05:37 +0200 Mark Nauwelaerts * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: make default duration check less sensitive Frame duration might vary for 1 usecond, in this case matroskamux decides to create BLOCKGROUP instead of SIMPLEBLOCK. Convert duration to timecodescale which is (typically) less precise, and then also allow the difference of 1/-1 to arrange for less sensitive check. Based on patch by Alexey Fisher Fixes #653080. 2011-09-06 13:18:40 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4gdepay.c: rtpmp4gdepay: improve bogus interleaved index compensating Patch by Fixes #654585. 2011-09-06 10:33:21 +0200 Sebastian Dröge * ext/soup/gstsouphttpsrc.c: souphttpsrc: Allow positive, non-1.0 segment rates Only negative rates are not supported. Fixes bug #658305. 2011-09-05 15:50:56 +0200 Mark Nauwelaerts * tests/check/elements/parser.c: tests: parsers: provide more real data when testing draining of garbage 2011-09-05 15:50:04 +0200 Mark Nauwelaerts * gst/audioparsers/gstamrparse.c: amrparse: fix and streamline valid frame checking ... to handle various combinations of sync or not, and sufficient data or not as might be expected. Fixes #650714. 2011-09-05 14:49:32 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: fragmented support; avoid adjustment for keyframe seek ... since all index data may not yet be available at that time. 2011-09-05 14:48:02 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: fragmented support; mark all audio track samples as keyframe 2011-09-05 14:46:29 +0200 Brian Li * gst/isomp4/qtdemux.c: qtdemux: fragmented support; properly init return variable value Fixes #655918. 2011-09-05 13:31:20 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: add gtk-doc for new short-header property 2011-09-05 13:18:39 +0200 Marc Leeman * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: allow sending short RTSP requests to a server Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by GStreamer, but do accept the short header as sent by Live555. This patch makes the extending the request optional by adding a property (short-header). Fixes #655805. API: GstRTSPSrc:short-header 2009-03-04 14:51:09 -0500 Olivier Crête * gst/rtp/gstrtph263ppay.c: rtph263ppay: Set H263-2000 if thats what the other side wants The static caps states this element supports H263-2000, but setcaps never sets it, so it was lie. See https://bugzilla.gnome.org/show_bug.cgi?id=577784 2011-08-30 19:02:51 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Initialise the last_keyframe_request variable 2011-08-31 16:04:24 +0200 Peter Korsgaard * gst/udp/gstmultiudpsink.c: multiudpsink: make add/remove/clear/get-stats action signals http://bugzilla.gnome.org/show_bug.cgi?id=657830 Signed-off-by: Peter Korsgaard 2011-08-30 13:33:49 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: push mode; perform some extra checks prior to upstream seeking 2011-08-30 13:28:21 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: push mode; fix buffered streaming That is, in case where no seek is peformed to moov, but preceding limited mdat is buffered. 2011-08-29 15:13:56 +0200 Mark Nauwelaerts * gst/isomp4/qtdemux.c: qtdemux: avoid overflow wraparound in timestamp when adding durations Do some type juggling to avoid overflow, while still allowing for 'negative' durations (which would need a wraparound effect). 2011-08-25 23:37:47 +0100 Vincent Penquerc'h * sys/v4l2/v4l2src_calls.c: v4l2src: make this work more than once in a row We used to skip frame rate setup if the camera was already setup with the requested frame rate. This breaks some cameras though, causing them to not output data (several models of Thinkpad cameras have this problem at least). So, don't skip. https://bugzilla.gnome.org/show_bug.cgi?id=638300 2011-08-23 12:12:15 +0100 Vincent Penquerc'h * gst/audioparsers/gstaacparse.c: aacparse: only require two frames in a row when we do not have sync This avoids a single bit error dropping two frames unnecessarily. The two consecutive frames check is still required when we don't have sync. https://bugzilla.gnome.org/show_bug.cgi?id=657080 2011-08-23 21:41:15 +0530 Arun Raghavan * ext/pulse/pulsesink.c: pulsesink: Trivial indentation fix 2011-07-21 17:23:28 -0400 Monty Montgomery * ext/flac/gstflacdec.c: flacdec: Correct sample number rounding resulting in timestamp jitter flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer. Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample. This corrects the time->sample convesion 2011-08-20 14:48:20 -0700 David Schleef * gst/debugutils/breakmydata.c: breakmydata: element is not passthrough 2011-07-13 11:20:34 -0700 David Schleef * gst/multifile/gstmultifilesrc.c: multifilesrc: quiet debugging 2011-07-10 21:40:20 -0700 David Schleef * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: * gst/deinterlace/gstdeinterlacemethod.c: * gst/deinterlace/gstdeinterlacemethod.h: * gst/deinterlace/tvtime/greedy.c: * gst/deinterlace/tvtime/greedyh.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/scalerbob.c: * gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc: * gst/deinterlace/tvtime/vfir.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: change field handling through methods This likely breaks stuff. The good: all of the methods now create field images aligned with input frames, without timestamp mangling. The bad: this touches a lot of code, much of which is hairy and in need of cleanup. However, at this point we can reasonably create a PSNR-based test. 2011-08-21 14:41:14 +0200 Alessandro Decina * gst/multifile/gstmultifilesink.c: multifilesink: reset ->streamheaders to NULL on _stop Fixes invalid memory access reusing multifilesink 2011-08-18 13:37:39 +0200 David Henningsson * ext/pulse/pulsesink.c: pulsesink: Allow writes in bigger chunks There's no use in splitting the incoming data down to the segsize limit - by writing as much as possible in one chunk, we increase performance and avoid PulseAudio unnecessary rewinds. Signed-off-by: David Henningsson 2011-08-08 22:14:28 +0100 Vincent Penquerc'h * gst/matroska/matroska-demux.c: matroskademux: ensure no-more-pads is always emitted In particular, do so even if failing to read while prerolling, such as when reading from a partial file (eg, while it is being downloaded). This fixes a wedge in playbin2. https://bugzilla.gnome.org/show_bug.cgi?id=651965 2011-08-16 17:27:13 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: avoid timestamp/offset tracking going out of sync The libFLAC API is callback based, and we must only call it to output data when we know we have enough input data. For this reason, a single processing step is done when receiving a buffer. However, if there were metadata buffers still pending, a step intended for the first audio frame might end up writing that leftover metadata. Since a single step is done per buffer, this will cause every buffer to be written one step late. This would add some latency (a bufferfull's worth), possibly lose a buffer when seeking or the like, and also cause timestamp and offset to be applied to the wrong buffer, as updates to the "current" segment last_stop (from incoming buffer timestamp) will be applied to an output buffer originating from the previous incoming buffer. This fixes the issue by ensuring that, upon receiving the first audio frame, processing is done till all metadata is processed, so the next "single step" done will be for the audio frame. After this, we should keep to 1 input buffer -> 1 output buffer and so avoid getting out of sync. https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 15:32:07 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: bail on reserved value Now that we look at the right bits, we can test against the reserved value as we do for other fields. https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 15:27:43 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: fix bit twiddling Right shifting a 8 bit value by 8 bits is twice too much to get the high 4 bits. https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 15:22:46 +0100 Vincent Penquerc'h * ext/flac/gstflacdec.c: flacdec: warn if we see a variable block size where unsupported https://bugzilla.gnome.org/show_bug.cgi?id=650960 2011-08-16 18:25:29 +0100 Vincent Penquerc'h * gst/spectrum/gstspectrum.c: spectrum: avoid crashing by resetting the correct number of channels https://bugzilla.gnome.org/show_bug.cgi?id=656606 2011-08-16 13:16:22 +0100 Vincent Penquerc'h * gst/audioparsers/gstflacparse.c: flacparse: fix off by one in frame size check Yes, I was tracking another bug and the small test file I generated to test with improbably just happened to trigger this, with a second and last frame of 1615 bytes. https://bugzilla.gnome.org/show_bug.cgi?id=656649 2011-08-14 20:46:01 +0100 Tim-Philipp Müller * gst/id3demux/id3v2.3.0.html: * gst/id3demux/id3v2.4.0-frames.txt: * gst/id3demux/id3v2.4.0-structure.txt: id3demux: remove specs from git as well now that parsing code is in -base 2011-07-14 15:42:36 +0200 Mark Nauwelaerts * configure.ac: * gst/id3demux/Makefile.am: * gst/id3demux/gstid3demux.c: * gst/id3demux/id3tags.c: * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: id3demux: use -base provided id3 tag parsing https://bugzilla.gnome.org/show_bug.cgi?id=654388 2011-08-13 16:51:22 +0100 Tim-Philipp Müller * ext/jack/gstjackaudiosrc.c: jackaudiosrc: fix error message code And also post 'not found' error if jackd is not even installed. 2011-08-12 16:32:58 +0200 Stefan Kost * gst/isomp4/qtdemux.c: qtdemux: initialize bitrate variable and reset for each loop Don't check eventually unset variable and don't accidentially use values from last cycle. 2011-08-09 11:28:17 +0200 Edward Hervey * gst/rtsp/gstrtspsrc.c: rtspsrc: Properly error out if SDP contains no streams Also fixes unitialized variable error on macosx. 2011-08-09 09:05:31 +0100 Vincent Penquerc'h * sys/ximage/gstximagesrc.c: ximagesrc: clear flags on buffer reuse This will ensure a logically new buffer does not keep flags from a previous use of that buffer (eg, DISCONT would be set on the first buffer, and mistakenly kept when reused). https://bugzilla.gnome.org/show_bug.cgi?id=653709 2011-08-08 10:54:26 +0100 Vincent Penquerc'h * sys/v4l2/gstv4l2object.c: v4l2: take care not to change the current format where appropriate Some drivers are buggy are will change the current format when processing VIDIOC_TRY_FMT. Save and restore the current format to ensure the format is kept unchanged. https://bugzilla.gnome.org/show_bug.cgi?id=649067 2011-08-07 12:23:26 +0200 Sjoerd Simons * sys/v4l2/v4l2src_calls.c: v4l2src: Use fraction compare util function. Use the fraction compare utility to compare function, not the handcrafted one. The handcrafted one is buggy as it doesn't take into account rounding error. For example comparing a framerate of 20/1 on a camera configured as 30/1 fps would yield true: 1 == (1 * 20)/30 and not re-configure the camera. Fixes #656104 2011-08-03 22:50:05 +1000 Jan Schmidt * gst/matroska/matroska-read-common.c: * gst/matroska/matroska-read-common.h: * gst/matroska/matroska.c: matroska: Register new debug category Register the matroskareadcommon debug category when the plugin is loaded to avoid assertion output when debug is turned on. 2011-07-29 13:03:55 +0200 Philippe Normand * gst/isomp4/qtdemux.c: qtdemux: soften assertion check on stream size https://bugzilla.gnome.org/show_bug.cgi?id=655570 2011-08-03 10:09:42 +0200 Robert Krakora * gst/rtp/gstrtpjpegpay.c: rtpjpegpay: Add support for H.264 payload in MJPEG container See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf Fixes bug #655530. 2011-08-02 22:05:08 -0400 Tristan Matthews * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: jackaudiosink: Don't call g_alloca() in process_cb g_alloca() is not RT-safe, so instead we should allocate the memory needed in advance. Fixes #655866 2011-08-02 23:42:58 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * sys/v4l2/gstv4l2object.c: docs: fix two more Since: tags 2011-07-31 04:19:00 +0300 Mart Raudsepp * gst/deinterlace/gstdeinterlace.c: deinterlace: Fix Since tags for fieldanalysis related new properties commit c1b100cf9c is after 0.10.29 and 0.10.30 was a branched release. So fix Since tags from 0.10.29 to 0.10.31 for the new properties. 2011-07-29 13:05:42 +0100 Tim-Philipp Müller * ext/pulse/pulsesink.c: pulsesink: fix variable-set-but-not-used compiler warning with older pulse versions 2011-07-29 12:07:24 +0200 Mark Nauwelaerts * gst/rtpmanager/rtpsession.c: rtpsession: properly init rtcp_min_interval 2011-03-09 11:04:36 +0530 Arun Raghavan * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.c: pulsesink: Add support for compressed formats This adds support for various compressed formats (AC3, E-AC3, DTS and MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF, HDMI and Bluetooth). The acceptcaps() function allows bins to probe for what formats the sink being connected to support. This only works after the element is set to at least READY. If the underlying sink changes and the format we are streaming is not available, we emit a message that will allow upstream elements/bins to block and renegotiate a new format. 2011-03-01 15:34:46 +0530 Arun Raghavan * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulsesink: Use the extended stream API if available This uses the new extended API for creating streams. This will allow us to support compressed formats natively in pulsesink as well. 2011-07-29 00:07:52 +0530 Arun Raghavan * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: Add a source-output-index property This exposes the source output index of the record stream that we open so that clients can use this with the introspection if they want (to move the stream, for example). 2011-07-28 14:44:57 +0200 Mark Nauwelaerts * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: keep a ref on the src pad while using it Prevent a possible race if clear_ssrc() is called between getting the pad and doing the push. Based on patch by https://bugzilla.gnome.org/show_bug.cgi?id=650916 2011-05-24 11:29:57 +0300 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/gstrtpssrcdemux.h: rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit We need to keep the lock held because we don't want a push before the "new-ssrc-pad" handler has completed. But we may want to push an event from inside that handler, hence the recursive mutex. https://bugzilla.gnome.org/show_bug.cgi?id=650916 2011-05-24 11:17:25 +0300 Olivier Crête * gst/rtpmanager/gstrtpssrcdemux.c: rtpssrcdemux: Use PADs lock https://bugzilla.gnome.org/show_bug.cgi?id=650916 2011-07-27 18:15:20 +0100 Sjoerd Simons * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: Cope with FU-A E bit not being set Some h264 payloaders are unfortunately buggy and don't correctly set the E bit in FU-A NAL when they have ended. Work around this by assuming such a fragmentation unit has ended when there was no packet loss and a new NAL is started 2011-04-12 17:01:47 +0530 Arun Raghavan * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: Support switching alignment on-the-fly This allows switching of alignment for E-AC3 streams at run-time. This is requested by downstream elements via a custom event. https://bugzilla.gnome.org/show_bug.cgi?id=650313 2011-04-09 12:26:56 +0530 Arun Raghavan * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: * tests/check/elements/ac3parse.c: ac3parse: Add support for IEC 61937 alignment When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading requires each buffer to contain 6 blocks from each substream. This adds code to collect all the frames needed to meet this requirement before pushing out a buffer. https://bugzilla.gnome.org/show_bug.cgi?id=650313 2011-06-08 15:57:37 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: Always send application requested feedback in immediate mode Send as many application requested feedback messages in immediate mode, even if they have already been sent. https://bugzilla.gnome.org/show_bug.cgi?id=654583 2011-06-08 14:48:01 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Don't let the computed RTP bandwidth fall too low If it falls too low, the computed RTCP bandwidth will be near zero and the RTCP thread will be stopped. https://bugzilla.gnome.org/show_bug.cgi?id=654583 2011-04-25 16:13:38 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: Wait longer to timeout SSRC collision Using the current RTCP interval to timeout SSRC collision can lead to collisions being timed out immediately if a BYE packet is sent because it is sent immediately, so the interval is 0. This is not what we want. So just set a static 10 times the default RTCP interval, it should be enough https://bugzilla.gnome.org/show_bug.cgi?id=648642 2011-07-19 13:38:01 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: set SOURCE flag at init time Fixes #654816. 2011-07-18 16:46:27 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: rtph264depay: Complete merged AU on marker bit The marker bit on a RTP packet means the AU has been completed, so push it out immediately to reduce the latency. https://bugzilla.gnome.org/show_bug.cgi?id=654850 2011-07-18 20:27:38 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit An access unit could contain multiple NAL units, in that case, only the last RTP packet of the last NALU should have its marker bit set. https://bugzilla.gnome.org/show_bug.cgi?id=654850 2011-07-20 08:52:58 +0200 Alessandro Decina * gst/multipart/multipartmux.c: multipart: fix compiler warning 2011-07-19 12:05:51 +0200 Mark Nauwelaerts * gst/auparse/gstauparse.c: auparse: avoid hanging on invalid short input ... as in such case there is no srcpad yet on which to forward EOS. 2011-07-18 15:13:33 -0300 Thiago Santos * ext/pulse/pulsesrc.c: pulsesrc: Fix default value leaking Remember to free the default value of client name, avoiding a leak 2011-07-18 14:24:48 +0200 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: rtph264depay: reset upon FLUSH_STOP ... which is particularly needed when merging NAL units, where not resetting would lead to output of an older (pre-flush) AU (with unintended timestamp). 2011-07-18 14:30:51 +0200 Mark Nauwelaerts * gst/multifile/gstmultifilesink.c: multifilesink: do not use g_slist_free_full ... as that is only in GLib 2.28, which is not yet required at this time. 2011-07-18 09:38:26 +0200 Alessandro Decina * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesink.h: * tests/check/elements/multifile.c: multifilesink: add max-files property Add max-files property to limit the number of files saved on disk. API: multifilesink::max-files 2011-07-17 23:36:55 +0200 Alessandro Decina * gst/multifile/gstmultifilesink.c: multifilesink: refactor file opening and closing code 2011-07-16 19:38:51 +0200 Alexey Fisher * gst/matroska/matroska-demux.c: matroskademux: fix pixel-aspect-ratio if header has only one display variable Current matroska demux calculates the pixel aspect ratio only if both DisplayHeight and DisplayWidth are set, but it is legal to use only one variable if the other is equal to PixelWidth or PixelHeight, at least the mkclean utility is doing that. So this makse mkcleaned files play correctly. https://bugzilla.gnome.org/show_bug.cgi?id=654744 2011-07-16 23:47:50 +0100 Antoine Jacoutot * gst/goom/plugin_info.c: goom: fix build on PPC on openbsd A missing sys/param.h include results in: /usr/include/sys/proc.h:64: error: 'MAXLOGNAME' undeclared here (not in a function) /usr/include/sys/proc.h:285: error: 'MAXCOMLEN' undeclared here (not in a function) when compiling goom on openbsd/ppc. We can just remove the two sys/ includes here, they are not needed for anything. https://bugzilla.gnome.org/show_bug.cgi?id=654749 2011-07-14 20:10:02 -0400 Olivier Crête * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic Partially reverts 397dc60b 2011-03-04 15:41:22 -0500 Olivier Crête * gst/rtp/Makefile.am: * gst/rtp/gstrtph264pay.c: rtph264pay: Implement getcaps Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level) 2011-07-12 15:04:38 +0200 Mark Nauwelaerts * gst/rtsp/gstrtspsrc.c: rtspsrc: fix seeking regression ... introduced when shuffling around code for the async implementation by setting state of source (and udp sources) in _play before downstream flushing is undone. 2011-07-11 15:23:41 +0300 René Stadler * gst/audioparsers/gstac3parse.c: * gst/audioparsers/gstac3parse.h: ac3parse: fix buffer duration on blocks-per-frame change The gst_base_parse_set_frame_rate call was predicated on a change to sample rate, duration or profile. However, the block count per frame can also change between packets, which would result in incorrect buffer durations. 2011-07-09 19:23:41 -0700 David Schleef * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: multifilesrc: Improve looping Add start-index and stop-index properties. 2011-06-16 13:57:03 +0100 Jonny Lamb * gst/multifile/gstmultifilesrc.c: * gst/multifile/gstmultifilesrc.h: multifile: add loop property to multifilesrc Fixes: #652727 Signed-off-by: Jonny Lamb Signed-off-by: David Schleef 2009-11-20 10:07:43 +0100 Philip Jägenstedt * sys/directsound/gstdirectsoundsink.c: directsoundsink: 16-bit audio is signed, 8-bit is unsigned. Pretending to handle 8-bit signed causes distorted audio when actually given such audio, which you will get if passing 8-bit unsigned through audioconvert ! audioresample, as audioresample only handles 8-bit signed. Fixes #605834. Signed-off-by: David Schleef 2011-07-07 18:27:36 +0200 Alexey Fisher * gst/matroska/matroska-demux.c: matroskademux: handle blocks with duration=0 Some video frames, for example alt-ref frame in VP8, will be never displayed. This is why it has duration=0. This patch allow to use this duration. Bug: 654175 Signed-off-by: Alexey Fisher 2011-07-06 17:18:05 -0700 David Schleef * gst/isomp4/gstqtmux.c: * gst/isomp4/gstqtmuxmap.c: qtmux: Add direct dirac mapping 2011-06-29 20:59:26 +0300 René Stadler * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: prevent race condition causing ref leak Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the deferred call to be run before returning. This causes a race when READY->NULL is executed shortly after, which stops the mainloop. This leaks the element reference which is passed as userdata for the callback (introduced in commit 7cf996, bug #614765). The correct fix is to wait in READY->NULL for all outstanding calls to be fired (since libpulse doesn't provide a DestroyNotify for the userdata). We get rid of the reference passing from 7cf996 altogether, since finalization from the callback would anyways lead to a deadlock. Re-fixes bug #614765. 2011-07-04 08:58:14 +0300 René Stadler * ext/pulse/pulsesink.c: pulsesink: small cleanup of copy-paste code 2011-06-29 19:50:42 +0300 René Stadler * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: remove unused member variable and misleading log message Wim changed it in commit 8bfd80 so that pa_defer_ran is not read anywhere. The log message used to annotate a mainloop_wait call which is gone. 2011-07-04 12:58:38 -0700 David Schleef * gst/goom/gstgoom.c: goom: Don't answer lantency queries before negotiation 2011-07-04 14:30:09 +0200 Mark Nauwelaerts * ext/jpeg/gstjpegdec.c: jpegdec: avoid crashing on invalid input without components 2011-07-04 11:25:28 +0200 Mark Nauwelaerts * gst/flv/gstflvmux.c: flvmux: pass along segment info to collectpads ... so it can track this and be subsequently used to determine running time etc. 2011-07-04 11:24:23 +0200 Mark Nauwelaerts * gst/flv/gstflvdemux.c: flvdemux: indicate raw format in aac caps 2011-07-03 19:51:32 -0700 David Schleef * ext/pulse/plugin.c: pulse: Increase ranks to PRIMARY + 10 So that pulsesrc/pulsesink get chosen over other possible PRIMARY src/sinks by autoaudiosink. Presumably, if pulse is available, it is always preferred over another src/sink. Fixes: #647540. 2011-06-30 18:47:48 -0700 David Schleef * gst/multipart/multipartmux.c: multipartmux: Add \r\n to tail of pushed buffers Clients such as Firefox require the \r\n after the payload. 2011-06-16 14:52:51 +0200 Branko Subasic * gst/matroska/ebml-read.c: * gst/matroska/matroska-demux.c: matroskademux: avoid looping when searching for clusters Fixes some bugs that results in the demuxer looping when seaching for clusters in non-finalized files. https://bugzilla.gnome.org/show_bug.cgi?id=652195 2011-06-10 18:54:48 +0530 Debarshi Ray * gst/matroska/matroska-parse.c: matroskaparse: fix reference counting of parse->streamheader https://bugzilla.gnome.org/show_bug.cgi?id=652286 Signed-off-by: David Schleef 2011-06-29 14:39:52 -0700 David Schleef * ext/jpeg/gstjpegenc.c: jpegenc: Don't round up size of encoded buffers For some reason, in code dating to 2001, encoded jpeg buffers were rounded up to multiples of 4 bytes. With the added bonus that the extra bytes are unwritten, causing valgrind issues. Oops. I can't think of any reason why JPEG buffers need to be multiples of 4 bytes, so I removed the padding. There might be some code somewhere that depends on this behavior, so if this needs to be reverted, please fix the valgrind issues. 2011-06-29 12:05:04 +0200 Mark Nauwelaerts * gst/isomp4/gstqtmux.c: qtmux: free date tag 2011-06-28 12:26:37 +0200 Jonas Larsson * gst/audioparsers/gstaacparse.c: aacparse: not so greedy minimum frame size Fixes #653559. 2011-06-25 11:39:23 -0700 David Schleef * configure.ac: configure: remove non-pkg-config check for shout Fixes: 653327 2011-06-20 18:49:57 +0200 Andoni Morales Alastruey * ext/raw1394/gst1394clock.c: dv1394src: make the internal clock thread safe Fixes: #653091. 2011-06-24 11:54:29 +0200 Miguel Angel Cabrera Moya * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: return correct type when assertion fails 2011-06-23 11:28:27 -0700 David Schleef * common: Automatic update of common submodule From 69b981f to 605cd9a 2011-02-02 16:18:54 +0530 Arun Raghavan * configure.ac: * ext/pulse/pulsesink.c: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulse: Drop support for PA versions before 0.9.16 This drops support fof PulseAudio versions prior to 0.9.16, which was released about 1.5 years ago. Testing with very old versions is not feasible and we don't want to maintain 2 independent code-paths. 2011-06-21 15:15:06 +0200 Mark Nauwelaerts * gst/rtp/gstrtpmp4adepay.c: rtpmp4adepay: fix output buffer timestamps in case of multiple frames 2011-06-20 16:47:36 -0400 Olivier Crête * gst/rtpmanager/rtpsession.c: rtpsession: The signal has 5 arguments, not 4 2011-06-18 13:43:02 +0100 Tim-Philipp Müller Bump git version after unplanned 0.10.30 release Merge branch '0.10.30' Conflicts: configure.ac docs/plugins/inspect/plugin-1394.xml docs/plugins/inspect/plugin-aasink.xml docs/plugins/inspect/plugin-alaw.xml docs/plugins/inspect/plugin-alpha.xml docs/plugins/inspect/plugin-alphacolor.xml docs/plugins/inspect/plugin-annodex.xml docs/plugins/inspect/plugin-apetag.xml docs/plugins/inspect/plugin-audiofx.xml docs/plugins/inspect/plugin-audioparsers.xml docs/plugins/inspect/plugin-auparse.xml docs/plugins/inspect/plugin-autodetect.xml docs/plugins/inspect/plugin-avi.xml docs/plugins/inspect/plugin-cacasink.xml docs/plugins/inspect/plugin-cairo.xml docs/plugins/inspect/plugin-cutter.xml docs/plugins/inspect/plugin-debug.xml docs/plugins/inspect/plugin-deinterlace.xml docs/plugins/inspect/plugin-dv.xml docs/plugins/inspect/plugin-efence.xml docs/plugins/inspect/plugin-effectv.xml docs/plugins/inspect/plugin-equalizer.xml docs/plugins/inspect/plugin-esdsink.xml docs/plugins/inspect/plugin-flac.xml docs/plugins/inspect/plugin-flv.xml docs/plugins/inspect/plugin-flxdec.xml docs/plugins/inspect/plugin-gconfelements.xml docs/plugins/inspect/plugin-gdkpixbuf.xml docs/plugins/inspect/plugin-goom.xml docs/plugins/inspect/plugin-goom2k1.xml docs/plugins/inspect/plugin-gstrtpmanager.xml docs/plugins/inspect/plugin-halelements.xml docs/plugins/inspect/plugin-icydemux.xml docs/plugins/inspect/plugin-id3demux.xml docs/plugins/inspect/plugin-imagefreeze.xml docs/plugins/inspect/plugin-interleave.xml docs/plugins/inspect/plugin-isomp4.xml docs/plugins/inspect/plugin-jack.xml docs/plugins/inspect/plugin-jpeg.xml docs/plugins/inspect/plugin-level.xml docs/plugins/inspect/plugin-matroska.xml docs/plugins/inspect/plugin-mulaw.xml docs/plugins/inspect/plugin-multifile.xml docs/plugins/inspect/plugin-multipart.xml docs/plugins/inspect/plugin-navigationtest.xml docs/plugins/inspect/plugin-oss4.xml docs/plugins/inspect/plugin-ossaudio.xml docs/plugins/inspect/plugin-png.xml docs/plugins/inspect/plugin-pulseaudio.xml docs/plugins/inspect/plugin-replaygain.xml docs/plugins/inspect/plugin-rtp.xml docs/plugins/inspect/plugin-rtsp.xml docs/plugins/inspect/plugin-shapewipe.xml docs/plugins/inspect/plugin-shout2send.xml docs/plugins/inspect/plugin-smpte.xml docs/plugins/inspect/plugin-soup.xml docs/plugins/inspect/plugin-spectrum.xml docs/plugins/inspect/plugin-speex.xml docs/plugins/inspect/plugin-taglib.xml docs/plugins/inspect/plugin-udp.xml docs/plugins/inspect/plugin-video4linux2.xml docs/plugins/inspect/plugin-videobox.xml docs/plugins/inspect/plugin-videocrop.xml docs/plugins/inspect/plugin-videofilter.xml docs/plugins/inspect/plugin-videomixer.xml docs/plugins/inspect/plugin-wavenc.xml docs/plugins/inspect/plugin-wavpack.xml docs/plugins/inspect/plugin-wavparse.xml docs/plugins/inspect/plugin-ximagesrc.xml docs/plugins/inspect/plugin-y4menc.xml win32/common/config.h 2011-06-17 10:37:33 +0100 Tim-Philipp Müller * sys/sunaudio/gstsunaudiosink.c: * sys/sunaudio/gstsunaudiosink.h: sunaudio: fix typo in comment 2011-06-17 03:07:09 +0300 Stefan Kost * gst/audiofx/audioecho.c: audioecho: fix param flags If the parameter cannot be changed in paused&playing, it is not controlable. Set the appropriate mutability flag instead.