Luis de Bethencourt [Tue, 2 Jun 2015 17:09:48 +0000 (18:09 +0100)]
goom2k1: remove variables not needed anymore
https://bugzilla.gnome.org/show_bug.cgi?id=742875
Luis de Bethencourt [Tue, 2 Jun 2015 16:52:46 +0000 (17:52 +0100)]
goom2k1: rebase to use the audiovisualizer class
Rebase to have goom2k1 using the common GstAudioVisualizer class
https://bugzilla.gnome.org/show_bug.cgi?id=742875
Luis de Bethencourt [Tue, 2 Jun 2015 16:29:36 +0000 (17:29 +0100)]
goom: rebase to use the audiovisualizer class
Edward Hervey [Tue, 2 Jun 2015 14:27:24 +0000 (16:27 +0200)]
check: Use GST_CHECK_MAIN () macro everywhere
Makes source code smaller, and ensures we go through common initialization
path (like the one that sets up XML unit test output ...)
Sebastian Dröge [Tue, 26 May 2015 12:47:31 +0000 (14:47 +0200)]
rtpsession: Only schedule a timer when we actually have to send RTCP
Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
feedback is actually pending and no regular RTCP has to be sent).
This improves CPU usage and battery life quite a lot.
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Fri, 22 May 2015 10:44:03 +0000 (13:44 +0300)]
rtpsession: Remove useless goto
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Thu, 21 May 2015 09:54:47 +0000 (12:54 +0300)]
examples: Set RTP profile to AVPF for rtpaux examples
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Mon, 4 May 2015 14:41:50 +0000 (16:41 +0200)]
rtspsrc: Set RTP profile on the rtpsession objects
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Thu, 21 May 2015 11:13:56 +0000 (14:13 +0300)]
rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Mon, 4 May 2015 09:51:41 +0000 (11:51 +0200)]
rtpsession: Only put RRs and full SDES into regular RTCP packets
If we may suppress the packet due to the rules of RFC4585 (i.e. when
below the t-rr-int), we can send a smaller RTCP packet without RRs
and full SDES. In theory we could even send a minimal RTCP packet
according to RFC5506, but we don't support that yet.
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Mon, 4 May 2015 11:51:50 +0000 (13:51 +0200)]
rtpsession: Keep track of tp/tn and t_rr_last separately
Otherwise we can't properly schedule RTCP in feedback profiles as we need to
distinguish the time when we last checked for sending RTCP (tp) but might have
suppressed it, and the time when we last actually sent a non-early RTCP
packet.
This together with the other changes should now properly implement RTCP
scheduling according to RFC4585, and especially allow us to send feedback
packets a lot if needed but only send regular RTCP packets every once in a
while.
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Sebastian Dröge [Mon, 4 May 2015 09:42:08 +0000 (11:42 +0200)]
rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
And modify our RTCP scheduling algorithm accordingly. We now can send more
RTCP packets if needed for feedback, but will throttle full RTCP packets by
rtcp-min-interval (t-rr-int from RFC4585).
In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
statically set to 1s or 0s by RFC4585. Tmin defines how often we should
send RTCP packets at most.
https://bugzilla.gnome.org/show_bug.cgi?id=746543
Olivier Crête [Sat, 30 May 2015 21:41:05 +0000 (17:41 -0400)]
mulawdec: Let baseclass estimate bitrate
This makes playback directly from a file work with the right caps.
Tim-Philipp Müller [Wed, 27 May 2015 15:31:23 +0000 (16:31 +0100)]
dynudpsink: keep GCancellable fd around instead of re-creating it constantly
And create it only when starting the element.
Tim-Philipp Müller [Wed, 27 May 2015 14:55:56 +0000 (15:55 +0100)]
udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly
Otherwise we constantly create/close event file descriptors,
every time we call g_socket_condition_timed_wait() or
g_socket_send_message(s)(), i.e. a lot. Which is not
particularly good for performance.
Can't create GCancellable in ::start() here because it's used
in client_new() which may be called via the add-client action
signal which may be called before the element is up and running.
Tim-Philipp Müller [Tue, 19 May 2015 17:13:16 +0000 (18:13 +0100)]
udpsrc: keep GCancellable fd around instead of re-creating it constantly
Otherwise we constantly create/close event file descriptors,
every single time we call g_socket_condition_timed_wait() or
g_socket_receive_message(), i.e. twice per packet received!
This was not particularly good for performance.
Also only create GCancellable on start-up.
Luis de Bethencourt [Tue, 26 May 2015 14:33:37 +0000 (15:33 +0100)]
matroska: overwritten value assignment
curpos is set and immediately after, set again. Remove the redundant
assignment.
https://bugzilla.gnome.org/show_bug.cgi?id=749909
Tim-Philipp Müller [Sat, 23 May 2015 12:47:17 +0000 (13:47 +0100)]
rtpvrawdepay: don't shadow existing outbuf variable
And fix unref of the wrong one which will contain NULL
in an error code path.
Tim-Philipp Müller [Sat, 23 May 2015 12:23:22 +0000 (13:23 +0100)]
rtpvrawdepay: map/unmap output frame only once, not for every input packet
Map output buffer after creating it and keep it mapped
until we're done with it instead of mapping/unmapping
it for every single input buffer.
Thiago Santos [Mon, 25 May 2015 11:47:47 +0000 (08:47 -0300)]
qtdemux: remove fixme from 2006
It has been verified by use over time.
Thiago Santos [Sat, 23 May 2015 17:36:41 +0000 (14:36 -0300)]
qtdemux: fix reverse playback of fragmented media
qtdemux creates a samples array and gets the timestamps for buffers by
accumulating their durations. When doing reverse playback of fragments,
accumulating samples will lead to wrong timestamps as the timestamps
should go decreasing from fragment to fragment and the accumulation
will produce wrong results.
In this case, when receiving a discont for fragmented reverse playback,
the previous samples information should be flushed before new data
is processed.
Jimmy Ohn [Fri, 22 May 2015 16:03:18 +0000 (01:03 +0900)]
splitfilesrc: Implement binary search in find_part_for_offset
Implement binary search using gst_util_array_binary_search
https://bugzilla.gnome.org/show_bug.cgi?id=749690
Sebastian Dröge [Thu, 21 May 2015 10:26:53 +0000 (13:26 +0300)]
rtpsession: Don't crash if we receive FIR/PLI from a source we don't know
Santiago Carot-Nemesio [Thu, 21 May 2015 07:35:58 +0000 (09:35 +0200)]
rtpsession: Fix collection of statistics
Stats should be collected on the media rtp source not in the
sender one.
https://bugzilla.gnome.org/show_bug.cgi?id=749669
Edward Hervey [Mon, 20 Apr 2015 08:07:30 +0000 (10:07 +0200)]
multifilesink: Add a new max-duration file switching mode
This new mode ensures that files will never exceed a certain duration
based on incoming buffer PTS (and duration if present)
Note:
* You need timestamped buffers (duh). If some of the incoming buffers don't
have PTS, then it will just accept them in the current file
Edward Hervey [Fri, 17 Apr 2015 14:18:32 +0000 (16:18 +0200)]
multifilesink: streamline the file-switch code a bit
Use the same functions regardless of the mode we are using
Edward Hervey [Thu, 2 Apr 2015 12:35:18 +0000 (13:35 +0100)]
multifilesink: add "aggregate-gops" property to process GOPs as a whole
This property can be used in combination with next-file=max-size
(and perhaps a future next-file=max-duration) to make sure that
each file part starts cleanly with a key frame and the appropriate headers.
In order for this property to work correctly, upstream elements should make
sure than any headers that need to be written in a standalone file are:
1) in the streamheader caps field
2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER
that are just before the keyframe buffer
This is useful for MPEG-TS/MPEG-PS file segmenting in
combination with mpegtsmux or mpegpsmux.
Original patch by: Tim-Philipp Müller <tim@centricular.com>
Sebastian Dröge [Wed, 20 May 2015 13:37:22 +0000 (16:37 +0300)]
rtspsrc: Use single-include header for the RTSP library
Tim-Philipp Müller [Fri, 24 Oct 2014 22:47:21 +0000 (23:47 +0100)]
udp: don't use soon-to-be-deprecated g_cancellable_reset()
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."
https://bugzilla.gnome.org/show_bug.cgi?id=739132
Stefan Sauer [Mon, 18 May 2015 18:13:01 +0000 (20:13 +0200)]
Revert "doc: Workaround gtkdoc issue"
This reverts commit
1797c8f8b12d7f4c7a9444c94f34f4d08ec85945.
This is fixed by the gtk-doc 1.23 release.
<para> cannot contain <refsect2>:
http://www.docbook.org/tdg/en/html/para.html
http://www.docbook.org/tdg/en/html/refsect2.html
Nicola Murino [Mon, 18 May 2015 14:40:21 +0000 (16:40 +0200)]
rtpg726pay: fix caps leak
https://bugzilla.gnome.org/show_bug.cgi?id=749544
Nicola Murino [Mon, 18 May 2015 14:34:13 +0000 (16:34 +0200)]
rtpg726depay: don't leak input buffer
https://bugzilla.gnome.org/show_bug.cgi?id=749543
Sebastian Dröge [Mon, 18 May 2015 14:38:31 +0000 (17:38 +0300)]
rtpsource: Queue bad packets instead of dropping them
So we can send them out once we found the next, consecutive sequence number in
case one is following.
Sebastian Dröge [Mon, 18 May 2015 14:38:14 +0000 (17:38 +0300)]
rtpsource: Use g_queue_foreach() to unref all buffers in queues
Sebastian Dröge [Mon, 18 May 2015 14:19:31 +0000 (17:19 +0300)]
rtpsource: Refactor seqnum comparison code a bit
Sebastian Dröge [Mon, 18 May 2015 14:08:53 +0000 (17:08 +0300)]
rtpsource: Allow sequence number wraparound during probation
Sebastian Dröge [Mon, 18 May 2015 14:07:23 +0000 (17:07 +0300)]
rtpsource: Make sequence number comparison code more readable
... by using gst_rtp_buffer_compare_seqnum() and signed integers
instead of implictly using effects of integer over/underflows.
Sebastian Dröge [Wed, 22 Apr 2015 16:54:06 +0000 (18:54 +0200)]
rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
It might just be a late retransmission or spurious packet from elsewhere, but
resetting everything would mean that we will cause a noticeable hickup. Let's
get some confidence first that the sequence numbers changed for whatever
reason.
https://bugzilla.gnome.org/show_bug.cgi?id=747922
Nicolas Dufresne [Sun, 17 May 2015 03:37:06 +0000 (23:37 -0400)]
doc: Workaround gtkdoc issue
With gtkdoc 1.22, the XML generator fails when a itemizedlist is
followed by a refsect2. Workaround the issue by wrapping the
refsect2 into para.
Stefan Sauer [Fri, 23 Jan 2015 12:57:40 +0000 (13:57 +0100)]
qtdemux: avoid wrong warnings on unknown node types
Add 'name' and 'mean' fourccs, as we handle them. Right now each use would
trigger a warning.
Nicola Murino [Fri, 8 May 2015 17:13:00 +0000 (19:13 +0200)]
rtpg726depay: add block_align to output caps
It is needed to correctly negotiate caps with matroskamux
and most other muxers.
https://bugzilla.gnome.org/show_bug.cgi?id=749129
Sebastian Dröge [Tue, 12 May 2015 10:41:58 +0000 (13:41 +0300)]
audiofxbasefirfilter: Fix time-domain convolution with >1 channels
input_samples is the number of frames, but we used it as the number of
samples.
https://bugzilla.gnome.org/show_bug.cgi?id=747204
Sebastian Dröge [Tue, 12 May 2015 09:13:16 +0000 (12:13 +0300)]
vp[89]enc: Properly convert between GStreamer and encoder timebase
... by switching numerator and denominator when scaling.
https://bugzilla.gnome.org/show_bug.cgi?id=749122
Sebastian Dröge [Mon, 11 May 2015 10:33:26 +0000 (13:33 +0300)]
vp[89]enc: Don't set timebase from the framerate
The framerate very often is just an indication of the ideal framerate, not the
actual framerate of the stream. By just using the framerate, we confuse the
rate control algorithm algorithm as multiple frames will map to the same PTS
or have durations of 0.
https://bugzilla.gnome.org/show_bug.cgi?id=749122
Mark Nauwelaerts [Sun, 10 May 2015 12:21:04 +0000 (14:21 +0200)]
tests: wavpackparse: fix unit test
See also https://bugzilla.gnome.org/show_bug.cgi?id=738237
Tim-Philipp Müller [Sun, 10 May 2015 10:05:00 +0000 (11:05 +0100)]
docs: update example pipelines in element docs
Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
Piotr Drąg [Sat, 9 May 2015 17:48:55 +0000 (19:48 +0200)]
po: update POTFILES.in
https://bugzilla.gnome.org/show_bug.cgi?id=749163
Tim-Philipp Müller [Sun, 10 May 2015 09:52:18 +0000 (10:52 +0100)]
splitmuxsrc: minor error message clean-up
Don't put filename in error message shown to user.
Guillaume Desmottes [Thu, 7 May 2015 14:25:36 +0000 (16:25 +0200)]
flacparse: fix buffer leak when stored to seektable
Fix a leak with the
validate.file.playback.change_state_intensive.samples_multimedia_cx_flac_Yesterday_flac
scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=749072
Paul Hyunil [Thu, 7 May 2015 08:10:37 +0000 (17:10 +0900)]
qtdemux: fix example pipeline in docs
The gst-launch script for example launch line to test qtdemux is
missing a queue before the decodebins, otherwise the gst-launch-1.0
command won't work.
https://bugzilla.gnome.org/show_bug.cgi?id=749054
Sebastian Dröge [Thu, 7 May 2015 12:51:45 +0000 (14:51 +0200)]
Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
This reverts commit
d22ec496328e6ba8edbf2d071d5608b2af2831e8.
Application code might expect that it only gets external sources on those
signals, and get confused by this. If anything we would need to add new
signals.
Sebastian Dröge [Wed, 25 Mar 2015 14:27:34 +0000 (15:27 +0100)]
rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
Without this it seems impossible for an application to easily get notified
about the internal ssrcs that are created, e.g. sender sources, and also
to know when they are active and produce RTCP packets.
https://bugzilla.gnome.org/show_bug.cgi?id=746747
Guillaume Desmottes [Mon, 4 May 2015 17:26:14 +0000 (19:26 +0200)]
jpegdec: fix frame leaks in handle_frame() implementation
handle_frame() is supposed to consume @frame, so if we don't call
gst_video_decoder_drop_frame() or gst_video_decoder_finish_frame() we have to
release it manually.
https://bugzilla.gnome.org/show_bug.cgi?id=748909
Sebastian Dröge [Mon, 4 May 2015 14:50:38 +0000 (16:50 +0200)]
rtspsrc: Fix up last commit
Sebastian Dröge [Mon, 4 May 2015 14:46:02 +0000 (16:46 +0200)]
rtspsrc: Only do RTX when using a feedback profile
Sebastian Dröge [Mon, 4 May 2015 11:50:31 +0000 (13:50 +0200)]
rtpsession: The stats min_interval is in seconds, not nanoseconds
We have to scale it to compare it against our clock times.
Sebastian Dröge [Mon, 4 May 2015 09:38:27 +0000 (11:38 +0200)]
rtpsession: Only return TRUE if early feedback was requested already and it's early enough
Luis de Bethencourt [Thu, 30 Apr 2015 14:42:34 +0000 (15:42 +0100)]
matroska: remove unused property enum items
Tim-Philipp Müller [Thu, 30 Apr 2015 11:13:59 +0000 (12:13 +0100)]
qtdemux: fix buffer leak on eos in push mode
Based on patch by Guillaume Desmottes.
scenario: validate.http.playback.seek_with_stop.raw_h264_1_mp4
https://bugzilla.gnome.org/show_bug.cgi?id=748617
Sebastian Dröge [Wed, 29 Apr 2015 17:41:29 +0000 (19:41 +0200)]
qtdemux: Check for sizes of the rdrf (redirect) atom before accessing the data and use g_strndup() instead of g_strdup()
Thanks to Ralph Giles for reporting this.
Sebastian Dröge [Wed, 29 Apr 2015 13:52:27 +0000 (15:52 +0200)]
rtspsrc: Only enable retransmissions if there is retransmission info in the SDP
Otherwise we're going to send early RTCP and NACKs in non-feedback sessions
too, which will confuse servers.
https://bugzilla.gnome.org/show_bug.cgi?id=748627
Ravi Kiran K N [Wed, 11 Feb 2015 12:39:24 +0000 (18:09 +0530)]
dvdemux: extract recording time
Extracts the recorded time of the dv file from
the metadata and puts it into the global tags.
https://bugzilla.gnome.org/show_bug.cgi?id=743657
Guillaume Desmottes [Tue, 28 Apr 2015 13:59:25 +0000 (15:59 +0200)]
matroskademux: fix seek event leak
gst_matroska_demux_handle_seek_event() doesn't consume the
event so we have to unref it.
https://bugzilla.gnome.org/show_bug.cgi?id=748584
Sebastian Dröge [Tue, 28 Apr 2015 13:42:49 +0000 (15:42 +0200)]
matroska-demux: Send pending tags when adding a new pad
We might've parsed those tags before already and tried to push them to
non-existing pads before. Now let's do it for real.
Sebastian Dröge [Thu, 23 Apr 2015 16:57:37 +0000 (18:57 +0200)]
rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
We need to convert the size to bits for our calculations.
https://bugzilla.gnome.org/show_bug.cgi?id=747863
Sebastian Dröge [Thu, 23 Apr 2015 16:53:39 +0000 (18:53 +0200)]
rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
https://bugzilla.gnome.org/show_bug.cgi?id=747863
Sebastian Dröge [Tue, 14 Apr 2015 16:41:07 +0000 (18:41 +0200)]
rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
https://bugzilla.gnome.org/show_bug.cgi?id=747863
Sebastian Dröge [Thu, 23 Apr 2015 16:49:37 +0000 (18:49 +0200)]
rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
https://bugzilla.gnome.org/show_bug.cgi?id=747863
Sebastian Dröge [Mon, 27 Apr 2015 14:36:27 +0000 (16:36 +0200)]
rtpjitterbuffer: Fix RTX unit test
The calculations were a bit off everywhere, even before the changes done
recently to the delay for RTX of expected future packets. It only worked by
accident, but now the calculations are all correct again. Hopefully.
Luis de Bethencourt [Mon, 27 Apr 2015 10:22:11 +0000 (11:22 +0100)]
Rename property enums from ARG_ to PROP_
Property enum items should be named PROP_ for consistency and readability.
Ilya Konstantinov [Fri, 24 Apr 2015 23:49:58 +0000 (02:49 +0300)]
rtpjitterbuffer: Fix "stats" property docs
https://bugzilla.gnome.org/show_bug.cgi?id=748436
Tim-Philipp Müller [Sun, 26 Apr 2015 16:54:52 +0000 (17:54 +0100)]
Remove obsolete Android build cruft
This is not needed any longer.
Thiago Santos [Fri, 24 Apr 2015 16:55:08 +0000 (13:55 -0300)]
videocrop: print the property values when set
Instead of printing the currently used values. The log is meant
to show what the properties changed to, not what is being currently
used.
Luis de Bethencourt [Fri, 24 Apr 2015 16:01:10 +0000 (17:01 +0100)]
remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
Tim-Philipp Müller [Thu, 23 Apr 2015 23:30:35 +0000 (00:30 +0100)]
level: fix infinite loop for very low interval values
https://bugzilla.gnome.org/show_bug.cgi?id=745515
Tim-Philipp Müller [Thu, 23 Apr 2015 15:08:54 +0000 (16:08 +0100)]
tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
Make sure the test environment is set up.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
Tim-Philipp Müller [Thu, 23 Apr 2015 15:08:32 +0000 (16:08 +0100)]
configure: bump automake requirement to 1.14 and autoconf to 2.69
This is only required for builds from git, people can still
build tarballs if they only have older autotools.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
Tim-Philipp Müller [Thu, 23 Apr 2015 15:06:57 +0000 (16:06 +0100)]
Update .gitignore
Jesper Larsen [Thu, 23 Apr 2015 07:55:59 +0000 (09:55 +0200)]
rtspsrc: Fix RTCP caps leak
https://bugzilla.gnome.org//show_bug.cgi?id=748353
Sebastian Dröge [Wed, 22 Apr 2015 18:24:20 +0000 (20:24 +0200)]
rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
delay. If jitter is very low, this should prevent unnecessary retransmission
requests to some degree.
https://bugzilla.gnome.org/show_bug.cgi?id=748041
Sebastian Dröge [Wed, 22 Apr 2015 17:41:07 +0000 (19:41 +0200)]
rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
https://bugzilla.gnome.org/show_bug.cgi?id=748041
Miguel París Díaz [Mon, 13 Apr 2015 09:20:40 +0000 (11:20 +0200)]
rtpjitterbuffer: Add "rtx-next-seqnum" property
If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
future packets based on when they are estimated to arrive.
See also https://bugzilla.gnome.org/show_bug.cgi?id=748041
https://bugzilla.gnome.org/show_bug.cgi?id=739868
Sebastian Dröge [Wed, 22 Apr 2015 17:29:34 +0000 (19:29 +0200)]
rtxreceive: Put debug output for retransmission requests at the right place
Before it was only ever printed once for every time a ssrc was associated with
a specific stream.
Wim Taymans [Wed, 22 Apr 2015 16:05:24 +0000 (18:05 +0200)]
v4l2: don't add the same interlace mode twice
Some drivers modify the interlace mode to progressive, no matter what
input you give them, make sure that we don't add the same interlace mode
twice.
Luis de Bethencourt [Tue, 21 Apr 2015 15:34:21 +0000 (16:34 +0100)]
equalizer: fix dynamic changes on bands
When we are in passthrough, the transform function doesn't run and if the
passthrough check is in this function it will never be deactivated. Fix this by
checking directly whenever a gain is changed.
Also set the passthrough to TRUE at init because the gains default to 0, so we
can passthrough until any gain property is changed.
https://bugzilla.gnome.org/show_bug.cgi?id=748068
Sebastian Dröge [Wed, 22 Apr 2015 08:30:52 +0000 (10:30 +0200)]
Remove INSTALL file
autotools automatically generate this, and when using different versions
for autogen.sh there will always be changes to a file tracked by git.
Sebastian Dröge [Wed, 22 Apr 2015 08:30:14 +0000 (10:30 +0200)]
Remove LICENSE_readme
It's completely outdated and just confusing, better if people are
forced to look at the actual code in question than trusting this file.
Luis de Bethencourt [Tue, 21 Apr 2015 14:21:33 +0000 (15:21 +0100)]
v4l2: cast unused return to void
Quell unchecked return value defect by casting the return value to void and
making it explicit it is going to be ignored.
CID #206031
Thiago Santos [Fri, 17 Apr 2015 16:08:02 +0000 (13:08 -0300)]
vp8dec: optimize vpx image to gstbuffer copy when strides match
Solving this FIXME. Copy the full plane when strides are the same
Thiago Santos [Thu, 16 Apr 2015 18:11:05 +0000 (15:11 -0300)]
vp9dec: optimize vpx image to gstbuffer copy when strides match
Solving this FIXME. Copy the full plane when strides are the same
Vincent Penquerc'h [Fri, 17 Apr 2015 12:32:54 +0000 (13:32 +0100)]
ac3parse: fix memory leak
Alex O'Konski [Fri, 17 Apr 2015 06:51:46 +0000 (06:51 +0000)]
icydemux: Fix segfault if metadata-interval is 0
Prevents an extra unref of GstBuffer when passing a non-icy stream through
icydemux with metadata-interval set to 0.
Reproducible with:
gst-launch-1.0 filesrc location=~/testsong.mp3 ! \
'application/x-icy,metadata-interval=(int)0' ! icydemux ! decodebin ! wavenc ! \
filesink location=~/testsong.wav
https://bugzilla.gnome.org/show_bug.cgi?id=748024
Ravi Kiran K N [Fri, 17 Apr 2015 06:24:23 +0000 (11:54 +0530)]
audiofx: fix typo in example pipelines
Fix typo in example pipelines
https://bugzilla.gnome.org/show_bug.cgi?id=748022
Ilya Konstantinov [Wed, 15 Apr 2015 15:22:37 +0000 (18:22 +0300)]
osxaudio: fix spelling in debug message
https://bugzilla.gnome.org//show_bug.cgi?id=747936
Luis de Bethencourt [Thu, 16 Apr 2015 15:33:44 +0000 (16:33 +0100)]
tests: selectable amount of bands in equalizer demo
Adding an option in the equalizer demo to make the number of bands selectable.
Sebastian Dröge [Thu, 16 Apr 2015 13:31:25 +0000 (15:31 +0200)]
rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
https://bugzilla.gnome.org/show_bug.cgi?id=747394
Arun Raghavan [Mon, 6 Apr 2015 07:26:50 +0000 (12:56 +0530)]
rtpsession: Track RTX ssrc caps
This is needed so that we can generate SR for RTX stream correctly (the
clock rate is required).
https://bugzilla.gnome.org/show_bug.cgi?id=747394
Sebastian Dröge [Tue, 14 Apr 2015 11:56:38 +0000 (13:56 +0200)]
rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
https://bugzilla.gnome.org/show_bug.cgi?id=747394
Luis de Bethencourt [Thu, 16 Apr 2015 15:01:50 +0000 (16:01 +0100)]
tests: switch equalizer demo to play from uri
Switch the equalizer-nbands demo to use uridecodebin, so users can listen to
something more pleasant than white noise. If anybody misses the white noise
a uri handler to audiotestsrc can be used.
Luis de Bethencourt [Thu, 16 Apr 2015 10:17:38 +0000 (11:17 +0100)]
tests: improve readability of equalizer demo
Rename variable name to make it more readable, add comments for the three
scales created per block, and set the window title.