Age | Commit message (Collapse) | Author | Files | Lines |
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Allow renegotiation to happen when buffers have returned after an allocation
query. As the allocation query is serialized, all buffers from the pool
should have returned and we can stop it to create a new one for the
new format
https://bugzilla.gnome.org/show_bug.cgi?id=682770
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Similar to set_format but it uses TRY_FMT instead of S_FMT
https://bugzilla.gnome.org/show_bug.cgi?id=682770
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g_inet_socket_address_get_address() does not give
us a ref to the address, so don't unref it.
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Previously we advanced the in_data pointer by bps for every channel, and then
later again for block_size*bps. This caused us to be one sample further than
expected if an input buffer covered two analysis frames. And in the end lead
to completely bogus values reported by level.
https://bugzilla.gnome.org/show_bug.cgi?id=746065
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Make sure the state change won't hang trying to shut down pads
by making sure the streaming has stopped before chaining up.
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to gst_core_audio_get_channel_layout().
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CID #1226474
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CID 1212156
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We just need to save the ebit information in case there is an error decoding.
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https://bugzilla.gnome.org/show_bug.cgi?id=745704
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If gst_pad_push() fails, inform and return flow error.
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Value set in ret will be overwritten just before exiting the function.
CID #1226469
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These are outside the expected range of sequence numbers and should be
clipped, especially for RTSP they might belong to packets from before a seek
or a previous stream in general.
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When the sdp media attribute framesize are converted to caps
the <payload> should not be included.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
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streams
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We were skipping the filter step while returning template caps, for
example.
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For output device, we should not update the buffer with flags and
timestamp when we dequeue. The information in the v4l2_buffer is not
meaningful and it breaks the case where the buffer is rendered at
multiple places.
https://bugzilla.gnome.org/show_bug.cgi?id=745438
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The properties were there before, but not used anywhere.
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Fix missing index in syncword searching
https://bugzilla.gnome.org/show_bug.cgi?id=745585
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To get a multiple of bpf use a subtraction and not an addition
https://bugzilla.gnome.org/show_bug.cgi?id=745684
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Use the object lock instead of the splitmux lock to protect
internal property variables, so they're not locked when
switching to a new file.
https://bugzilla.gnome.org/show_bug.cgi?id=744420
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See https://bugzilla.gnome.org/show_bug.cgi?id=745539
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We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.
https://bugzilla.gnome.org/show_bug.cgi?id=745599
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finish_frame() assumes that there is an output buffer.
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Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
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https://bugzilla.gnome.org/show_bug.cgi?id=745587
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This is helpful to provide statistics in the format defined in
http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.
https://bugzilla.gnome.org/show_bug.cgi?id=745587
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Duration accumulation can cause rounding errors and generate wrong
duration with different buffers that share the same timestamp.
https://bugzilla.gnome.org/show_bug.cgi?id=745192
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... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
that return PTS or DTS based on stream type.
https://bugzilla.gnome.org/show_bug.cgi?id=745192
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And also create only one, there's no need yet to create all 32 until
we implement RFC2762.
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This reverts commit 1591adf4cd843d13d8622a30c619425691a84128.
https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
It's the beginning of an implementation of RFC 2762, which is needed for
large multicast groups. The implementation is not yet complete but why
not leave what is there and implement RFC 2762 instead?
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rtpsession declares an array of maps to store srrcs but only the
the key 0 is being used. This patch replaces the array of maps
for just one map and remove useless parameters in rtpsession
https://bugzilla.gnome.org/show_bug.cgi?id=745586
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In gst_avi_demux_handle_src_query, there is not needed code.
We already check about stream is vbr or not at the upper line.
o, we don't need to check this condition becase stream is not
vbr 100% in this case.
https://bugzilla.gnome.org/show_bug.cgi?id=745276
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Just avoid using the deprecated function entirely,
it's easy enough. Defining the macro is not enough.
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gdk_pixbuf_new_from_inline() has been deprecated in favour
of GResource.
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The ringbuffer does allow renegotiation, so we do not have to report
fixed caps once it is acquired (based on a similar patch for the sink
side by Ilya Konstantinov <ilya.konstantinov@gmail.com>).
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Once osxaudiosink's device is open, it fixates on the initial caps and
refuses to accept new caps. This is erroneous since the Audio Unit is
can accept a new ASBD, and GstAudioRingBuffer supports reconfiguration
as well.
https://bugzilla.gnome.org/show_bug.cgi?id=743925
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Ensure gst_v4l2_buffer_pool_release_buffer() releases the associated
GstV4l2MemoryGroup. In particular, this allows for closing the DMABUF
handles prior to instantiating new ones.
https://bugzilla.gnome.org/show_bug.cgi?id=745443
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