This will still cause some timestamp jitter, but giving a hint as to the duration
rather than nothing seems to be a better idea.
Also, this allows some scenarios (like remuxing with asfmux) to estimate the total
duration using the accumulated packet duration (which will be correct).
The simple index entries also contain the number of packets one needs
to retrieve at a given position to get a full keyframe. We therefore
use that information to retrieve all those packets in one buffer when
working in pull-mode.
In gst_asf_demux_chain_headers, when 'goto wrong_type' was called
asfdemux tried to free a const pointer that had been cast to a
normal pointer variable.
We weren't taking the preroll into account previously, meaning that we
were always seeking preroll nanoseconds too early... resulting in a lot
of dropped packets (which are before the start time).
This brings quit a bit closer to as-fast-as-possible seeking in asf files.
Post global tags only after we've added our source pads, so that
tag events get sent downstream in addition to tag messages posted
on the bus. This makes sure tags can be picked up automatically
when transcoding, but also by tagreadbin/playbin2. Fixes#519721.
While we're at it, also add a container-format tag.
When we receive a DISCONT as input, don't clear our complete state but simply
mark a discont that will be put on the next buffer. The code will be able to
handle and throw away incomplete data.
Add some more debug info.
Remove an unused variable.
Let's not put every single mp3 frame in our index, a few frames per
second should be more than enough. For now use an index interval
of 100ms-500ms depending on the upstream size, to keep the index at
a reasonable size. Factor out the code that adds the index entry
into a separate function for better code readability.
While technically upstream may be seekable even if it doesn't know
the exact size, I can't think of a use case where this distincation
is relevant in practice, so for now just assume we're not seekable
if upstream doesn't provide us with a size. Makes sure we don't
build a seek index when streaming internet radio with sources that
pretend to be seekable until you try to actually seek.
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_class_init),
(gst_x264_enc_header_buf), (gst_x264_enc_encode_frame):
Use hyphen in property name, perform safety buffer size check
prior to mem access, and some more parentheses in macro.
Original commit message from CVS:
* ext/apexsink/Makefile.am:
Link against -lgcrpyto for RSA_new and RSA_free.
* ext/faac/gstfaac.c:
* ext/x264/gstx264enc.c:
Fix compiler warnings.
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_set_src_caps):
Construct source caps in more conventional (and correct) manner.
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_reset),
(gst_x264_enc_chain), (gst_x264_enc_encode_frame):
* ext/x264/gstx264enc.h:
Do not deal with duplicated input (timestamps). If needed,
a generic element can do so.
Do not manipulate input timestamps on the way out,
since that shifts the timeline and A/V sync.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Integrate new properties into documentation.
* ext/x264/gstx264enc.c: (gst_x264_enc_class_init),
(gst_x264_enc_init), (gst_x264_enc_init_encoder),
(gst_x264_enc_set_property), (gst_x264_enc_get_property):
Fix up API prior to eventual plugin move.
API: GstX264Enc:pass (provides more options, and changed to enum)
