With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.
Simplify the command handling, only continue looping when we have not received
another command or when the previous loop was successfull.
Avoid looping on a disconnected socket.
If the lock is not released before emitting a signal, it may cause a deadlock
if any other function in the element is called.
Also removed an unused timestamp parameter
https://bugzilla.gnome.org/show_bug.cgi?id=649617
Since the segment duration is given in terms of the
GST_MATROSKA_ID_TIMECODESCALE we should only convert it into
nanoseconds when we are sure that any scale specified in the file has
been read.
https://bugzilla.gnome.org/show_bug.cgi?id=650258
If the bitrates for all but one audio/video streams are known, and the
total stream size and duration can be determined, this calculates the
unkown bitrate as (stream size / duration) - (sum of known bitrates).
While this is not guaranteed to be very accurate, it should be good
enough for most purposes.
For example, this is useful for H.263 + AAC streams where no 'btrt' atom
is available for the video portion.
https://bugzilla.gnome.org/show_bug.cgi?id=619548
This parses the 'damr' atom if present, and exports the maximum bitrate
of the stream using the mode set field to determine the highest bitrate
frame type that might be present.
https://bugzilla.gnome.org/show_bug.cgi?id=620186
Since the segment duration is given in terms of the
GST_MATROSKA_ID_TIMECODESCALE we should only convert it into
nanoseconds when we are sure that any scale specified in the file has
been read.
https://bugzilla.gnome.org/show_bug.cgi?id=650258
Otherwise wavenc will fail if upstream decides to set equivalent caps or caps
with additional information later.
Thanks to Alexander Schremmer for finding this bug.
A duration tag gets inserted only for streamable=false, so only
update/write the duration later if we actually inserted that tag,
otherwise we write garbage into other tags.
https://bugzilla.gnome.org/show_bug.cgi?id=649060
Refuse h264 caps without stream-format and codec_data fields for
now, to avoid creating broken files. This might cause some pipelines
that worked previously to fail. However, the move from -bad to -good
is our only chance to fix this up, so make it strict for now. We can
always change it back to be less strict in future.
https://bugzilla.gnome.org/show_bug.cgi?id=647919
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.
g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.
Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
We use -DG_DISABLE_ASSERT for the pre-releases, which makes these
warnings pop up in cases that were previously covered by g_assert_not_reached()
and the like:
tvtime/greedyh.c:801:14: warning: 'scanline' may be used uninitialized in this function
matroska-mux.c:501:19: warning: 'context' may be used uninitialized in this function
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.
gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
This is needed for automatic transcoding using encodebin. Our typefinder
does not always add a variant to the found caps, and encodebin needs
an *exact* match to the caps on the source pad template, so we need
to add the variant-less video/quicktime caps to the template as well
for encodebin to be able to find it. Add unit test for this as well.
https://bugzilla.gnome.org/show_bug.cgi?id=642879
... and not only when sort-of feeling like it.
In any case, if it turns out all really is in order,
and presumably DTS == PTS, then no ctts will be produced anyway.
That is, all sorts of problems arise with re-ordered input timestamps that
tend to defy automagic handling for every case, so allow for a few variations
that can be tried depending on circumstances.
Also try to document accordingly.
Also fixes#638288.
