- 23 Nov, 2018 1 commit
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Xavier Claessens authored
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- 21 Nov, 2018 2 commits
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Nicolas Dufresne authored
The previous patch did not even compile on any possible platform or C standard. That commit also didn't have a proper commit message. Android ships Linux with a different signature for ioctl. They first released an ioctl with int as request type, and later "fixed" it by adding an override with unsign, which is still not matching Linux and BSD implementation which uses unsigned long int.
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Xavier Claessens authored
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- 19 Nov, 2018 2 commits
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Xavier Claessens authored
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PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it supports. We were previously exposing a maximum rate of INT_MAX, which is incorrect, but worked because nothing was really using a rate greater than 384000 kHz. While playing DSD data, we hit a case where there might be very high sample rates (>1MHz), and pulsesink fails during stream creation with such streams because it erroneously advertises that it supports such rates. Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in the caps string. Instead, we fix up the rate to what we actually support whenever we use our macro caps.
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- 15 Nov, 2018 1 commit
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This patch enables matroskademux to receive seeks before it reaches GST_MATROSKA_READ_STATE_DATA. Closes gstreamer/gst-plugins-good#514 This also enables receiving seeks in the element READY state. When such a seek is received, it is stored to be later handled when GST_MATROSKA_READ_STATE_DATA is reached.
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- 13 Nov, 2018 2 commits
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Reset RTPSession when rtpsession changes state from PAUSED to READY. Without this change, a stored last_rtptime in RTPSource could interfere with RTP timestamp generation in RTCP Sender Report. Fixes #510
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- 12 Nov, 2018 2 commits
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Mathieu Duponchelle authored
Fixes #516
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This commit adds a .gitlab-ci.yml file, which uses a feature to fetch the config from a centralized repository. The intent is to have all the gstreamer modules use the same configuration. The configuration is currently hosted at the gst-ci repository under the gitlab/ci_template.yml path. Part of gstreamer-project#29
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- 08 Nov, 2018 1 commit
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Nicolas Dufresne authored
If there was no interlace-mode field in the caps. Read back the value selected by the driver. This way, if the driver does not support progressive, then it will automatically negotiate the returned mode unless this mode is not supported by GStreamer. This method was already used for colorimetry. Just like colorimetry, the interlace mode is not longer probed by v4l2src dues to performance issues. Fixes #511
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- 07 Nov, 2018 3 commits
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Return the size / total duration as a ballpark estimate. gstreamer/gst-plugins-base#60
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Return the upstream size over the duration as a first estimate. gstreamer/gst-plugins-base#60
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Otherwise signal handlers from bindings will take ownership of them as they are still floating, and we won't own a reference inside rtpbin anymore. Fixes gstreamer/gst-plugins-good#515
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- 05 Nov, 2018 4 commits
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Decreasing timestamps break rtmpsink. With contributions from Olivier Crête. https://bugzilla.gnome.org/show_bug.cgi?id=796382
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Matthew Waters authored
Remove the git directory
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Haihao Xiang authored
This fixes gstreamer/gst-plugins-good#513
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- 01 Nov, 2018 1 commit
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If ctts (CompositionOffsetBox) has larger sample_offset (offset between PTS and DTS) than (2 * duration) of the stream, assume the ctts box to be corrupted and ignore the box. https://bugzilla.gnome.org/show_bug.cgi?id=797262
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- 28 Oct, 2018 6 commits
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Olivier Crête authored
This element doesn't support planar audio yet.
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Nirbheek Chauhan authored
Without these dependencies, the enumtype may not be generated when the test is built, which will cause a compile failure.
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Nirbheek Chauhan authored
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Olivier Crête authored
Make it possible to modify the SDES in a packet at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=763502
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- 27 Oct, 2018 3 commits
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This fixes a bug where in some files mehd.fragment_duration is one unit less than the actual duration of the fragmented movie, as explained below: mehd.fragment_duration is computed by scaling the end timestamp of the last frame of the movie in (in nanoseconds) by the movie timescale. In some situations, the end timestamp is innacurate due to lossy conversion to fixed point required by GstBuffer upstream. Take for instance a movie with 3 frames at exactly 3 fps. $ gst-launch-1.0 -v videotestsrc num-buffers=3 \ ! video/x-raw, framerate="(fraction)3/1" \ ! x264enc \ ! fakesink silent=false dts: 999:59:59.333333334, pts: 1000:00:00.000000000, duration: 0:00:00.333333333 dts: 999:59:59.666666667, pts: 1000:00:00.666666666, duration: 0:00:00.333333334 dts: 1000:00:00.000000000, pts: 1000:00:00.333333333, duration: 0:00:00.333333333 The end timestamp is calculated by qtmux in this way: end timestamp = last frame DTS + last frame DUR - first frame DTS = = 1000:00:00.000000000 + 0:00:00.333333333 - 999:59:59.333333334 = = 0:00:00.999999999 qtmux needs to round this timestamp to the declared movie timescale, which can ameliorate this distortion, but it's important that round-neareast is used; otherwise it would backfire badly. Take for example a movie with a timescale of 30 units/s. 0.999999999 s * 30 units/s = 29.999999970 units A round-floor (as it was done before this patch) would set fragment_duration to 29 units, amplifying the original distorsion from 1 nanosecond up to 33 milliseconds less than the correct value. The greatest distortion would occur in the case where timescale = framerate, where an entire frame duration would be subtracted. Also, rounding is added to tkhd duration computation too, which potentially has the same problem. https://bugzilla.gnome.org/show_bug.cgi?id=793959
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In some cases, a bind error occurs during operation. Printing the information about the problem is critical for finding the conflict https://bugzilla.gnome.org/show_bug.cgi?id=797340
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- 24 Oct, 2018 5 commits
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Add HEVC encoder support. https://bugzilla.gnome.org/show_bug.cgi?id=797141
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Nicolas Dufresne authored
We can now use the new GstFAllocator to ask the allocator not to close the wrapped FD. This way the dup is no longer needed.
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Nicolas Dufresne authored
There is no specific needs to duplicate the FD. Unlike the exportation, we don't depend on code that will call close. This will make debugging easyer since the traced FD will match the exporter.
- 22 Oct, 2018 2 commits
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Sebastian Dröge authored
qtmux: Add property for providing a threshold after which we create an edit list for gaps at the start https://bugzilla.gnome.org/show_bug.cgi?id=797290
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Sebastian Dröge authored
It was previously set to the display aspect ratio, e.g. 4x3, 16x9, etc. but should be set to the display size. This is a regression from e655d47d (1.5.1) and was correct before that. https://bugzilla.gnome.org/show_bug.cgi?id=797318
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- 21 Oct, 2018 1 commit
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Previous commit created caps with incorrect aac codec data that did not match the audio channel. https://bugzilla.gnome.org/show_bug.cgi?id=797256
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- 20 Oct, 2018 2 commits
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g_ptr_array_find_with_equal_func was introduced in glib 2.54 which is a higher version than our minimum required one. https://bugzilla.gnome.org/show_bug.cgi?id=797239
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GHashTable iteration order changed in recent GLib, and tests were relying on that. https://mail.gnome.org/archives/desktop-devel-list/2018-October/msg00016.html
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- 19 Oct, 2018 2 commits
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... before the old streams is not exposed yet for MSS stream. In case of DASH, newly configured streams will be exposed whenever demux got moov without delay. Meanwhile, since there is no moov box in MSS stream, the caps will act like moov. Then, there is delay for exposing new pads until demux got the first moof. So, following scenario is possible only for MSS but not for DASH, STREAM-START -> CAPS -> (configure stream but NOT EXPOSED YET) -> STREAM-START-> CAPS (configure stream again). In above scenario, we can reuse old stream without any stream reconfigure. https://bugzilla.gnome.org/show_bug.cgi?id=797239
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GPtrArray has less overhead than linked list and the length also can be auto updated by using it. https://bugzilla.gnome.org/show_bug.cgi?id=797239
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