- Dec 24, 2015
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Dec 21, 2015
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Update the -Bsymbolic check with the version glib has. This version works with clang. https://bugzilla.gnome.org/show_bug.cgi?id=759713
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- Dec 19, 2015
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The EOS event can be propagated to the downstream elements when is_eos flag remains set even after leaving the flushing state. This fix allows this element to normally restart the streaming after receiving the flush event by clearing the is_eos flag. https://bugzilla.gnome.org/show_bug.cgi?id=759110
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- Dec 18, 2015
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Thiago Santos authored
audiosink and videosink string variables are unused
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Matthew Waters authored
Unconditionally adding the template caps when proxying the caps query will play havoc with decoders that attempt to choose an output format based on some caps features. Creating a sink that does not include those caps features and a decoder/parser/etc that preferentially chooses some specific caps feature when available, will always return the decoder/parser/etc template caps and choose a feature that downstream will be unable to support. Fix by limiting the addition of the template caps to when the result is actually empty. https://bugzilla.gnome.org/show_bug.cgi?id=758212
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- Dec 17, 2015
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Sebastian Dröge authored
It's meant to be used for external plugins that can then all be disabled via --disable-external. gio-unix-2.0 however is just an optional dependency for the TCP unit test. Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part there needs to be an AM_CONDITIONAL for the feature with FALSE.
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- Dec 16, 2015
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Sebastian Dröge authored
This reverts commit 77dc09c3. It can cause the FLUSH_START/STOP events to go to the sink elements, which then causes state changes and various other problems. We shouldn't really flush downstream here, the idea is to do *draining*. Apart from that the testcase for the original bug here works without this commit now.
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Luis de Bethencourt authored
This should make easier to parse the debug logs. s/fnctl/fcntl
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Vincent Penquerc'h authored
Since the loops increasing count from 0 are always run at least once (if count < 1), count will always be at least one when compared to the drop/dup conditions. Coverity 1139674
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Wim Taymans authored
Rework the main processing loop. We now create an audio processing chain from small core functions. This is very similar to how the video-converter core works and allows us to statically calculate an optimal allocation strategy for all possible combinations of operations. Make sure we support non-interleaved data everywhere. Add functions to calculate in and out frames and latency.
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Wim Taymans authored
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Sebastian Dröge authored
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- Dec 14, 2015
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Nicolas Dufresne authored
This reverts commit 3642e9a3.
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This is a workaround for broken pulse module.
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Sebastian Dröge authored
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Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we are using a proxy server. Also, send Host header for compatability with HTTP/1.1 and some HTTP/1.0 servers. https://bugzilla.gnome.org/show_bug.cgi?id=758922
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gst_rtsp_connection_connect_with_response accepts a response pointer which it fills with the response from setup_tunneling if the connection is configured to be tunneled. The motivation for this is to allow the caller to inspect the response header to determine if additional authentication is required so that the connection can be retried with the appropriate authentication headers. The function prototype of gst_rtsp_connection_connect has been preserved for compatability with existing code and wraps gst_rtsp_connection_connect_with_response. https://bugzilla.gnome.org/show_bug.cgi?id=749596
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Sebastian Dröge authored
CID 1139615
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Wim Taymans authored
Allow an array of sample blocks to be passed to the channel mix and quantizer functions to support non-interleaved formats.
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Wim Taymans authored
Make it possible to pass an array of sample blocks when dealing with non-interleaved formats.
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- Dec 12, 2015
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Luis de Bethencourt authored
Support media using the aliases defined in http://www.fourcc.org/ that are exact duplicates of already known codes.
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Luis de Bethencourt authored
Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h, like gst_riff_create_audio_caps() does.
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- Dec 11, 2015
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Julien Isorce authored
It lets us know easily which pool is activated or inactivated during the negotiation. https://bugzilla.gnome.org/show_bug.cgi?id=720597
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Tim-Philipp Müller authored
Depayloaders will look at rtpbuffer->buffer for the discont flag. When we set the discont flag on a buffer in the rtp base depayloader and we have to make the buffer writable, make sure the rtpbuffer actually contains the newly-flagged buffer, not the original input buffer. This was introduced with the addition of the process_rtp_packet vfunc, but would only trigger if the input buffer wasn't flagged already and was not writable already.
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Tim-Philipp Müller authored
The problem was triggered only when the input buffers were not writable, so add extra ref to test this code path.
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Tim-Philipp Müller authored
When we detect a discont and the input buffer isn't already flagged as discont, handle_buffer() does a gst_buffer_make_writable() on the input buffer in order to set the flag. This assumed it had ownership of the input buffer though, which it didn't. This would still work fine in most scenarios, but could lead to crashes or mini object unref criticals in some cases when a discont is detected, e.g. when using pcapparse in front of a depayloader. This problem was introduced in bc14cdf5.
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- Dec 10, 2015
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Wim Taymans authored
Add a property and logic to send a GstNetworkMessage event containing the message that was received from a client. This can be used to implement simply bidirectional communication.
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Wim Taymans authored
Add a property and logic to send a GstNetworkMessageDispatched event upstream to notify that a buffer has been sent. This can be used to keep track of what client received what buffers.
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Wim Taymans authored
Add a property to handle GstNetworkMessage events. These events contain a buffer that is sent on the socket to allow for simple bidirectional communication.
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- Dec 09, 2015
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Wim Taymans authored
Improve the converter API to allow for an max input and output number of samples and return the number of consumed/produced samples.
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- Dec 08, 2015
- Dec 07, 2015
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Nicolas Dufresne authored
From b319909 to 86e4663
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- Dec 04, 2015
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Wim Taymans authored
Let downstream know that we support GstNetControlMessage metadata API.
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- Dec 03, 2015
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In the case where the stream doesn't have a framerate set and the frames don't have a duration set, we still want to use the clipping path to make sure we don't push buffers outside of the segment. The problem was the previous iteration was setting a duration of 2s, which meant that any buffer which was less than 2s before the segment start would end up getting pushed. Instead, use a saner 40ms (25fps single frame duration) to figure out whether the frame could be within the segment or not
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Reynaldo H. Verdejo Pinochet authored
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