For vbr audio streams we need to use the number of blocks to calculate the
timestamps.
When the allocation of additional index memory fails, don't throw away what
we had before.
Various cleanups.
Implement scanning of the file when we can parse the index.
Some refactoring of common code.
Cleanups and comments.
Remove some reimplemented code.
Remove index massage code and put a FIXME where we should do something
equivalent later.
Remove some duplicate counters.
Be smarter when updateing the current the timestamp and offset in the stream
because we can reuse previously calculated values when simply go forward one
step.
Correctly set metadata on outgoing buffers.
Add a new function and datastructure to parse and hold the index entries on a
per stream base. Also avoid doing too much work trying to figure out the
timestamps and durations as we can trivially do that later.
Less information in the entries makes them 2 times smaller and not doing too
much work makes this code about 12 times faster than the regular case.
Hook in the new function alongside the existing function for comparison until
the rest of the code is updated to handle the new index datastructure.
For calculating the durations of each sample, we are supposed to add each
duration modulo 1<<32 so make the elapsed time counter a uint32.
Fixes#595942
If it looks like we would be allocating a silly size for our sample
index, just bail out instead of trying to allocate it. Helps with
broken or fuzzed files where we might end up trying to malloc a
couple of hundred MBs otherwise.
Make sure we don't read beyond the atom boundary. Note that the code
behaves slightly differently in the corner case where there is not
enough atom data for the specified number of samples (n_samples_time)
in the atom, but still enough data to fill the pre-allocated index of
n_samples entries: before we would just stop parsing the stts data
and continue, whereas now we will likely error out. This should not
be a problem in practice though. We could maintain the old behaviour
by doing reads with a size check inside the loop if needed.
Use GstByteReader to parse stsz and stsc chunks, and check size of
available data before parsing it, instead of blindly assuming there
will be enough data. Fixes crashes with some fuzzed/broken files.
At the end, Dirac streams have an EOS packet with 0 length.
Don't ever sit in an infinite loop when processing one. Allows
muxing Dirac into mkv to complete successfully.
Add a property to allow control over what event causes a file
to finish being written and a new file start. The default is
the same as before -- each buffer causes a new file to be
written. Added is a case where buffers are written to the
same file until a discontinuity in the stream.
Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and
ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux.
Fixes#594490
When we receive a reordered packet with the same timestamp as the previous one
(which can happen for fragmented packets) don't consider the packet as lost but
instead wait for the reordered packet to arrive.
Switch the warning-level, so that a reordering does not get a warning, only
an actual produced lost-packet.
Fixes#594251
In gst_rtspsrc_parse_digest_challenge(), rtspsrc does a g_strndup of the auth
header items and then passes them to gst_rtsp_connection_set_auth_param()
without freeing.
Fixes#594133
When receiving a sync-packet, all sessions with the same cname will be compared
and synced together. In this process, there could still be references to a
session that has been shut down in the meanwhile.
This patch makes sure that these references are removed when shutting down a
session, so that the syncing can be done safely.
Fixes#594283
The priv->clock_rate variable could become -1 between when its checked to not
be -1 and when its used, causing an assertion. Fixed by taking the mutex
earlier in the chain() function.
Fixes#593955
This reintroduces the fix for bug #593391. It also applies it in
gst_rtp_session_sync_rtcp() which has very similar code to
gst_rtp_session_send_rtcp().
When we construct a timestamp that would result in a timestamp that is earlier
than when the packet was received, reset the skew calculation as this is
probably a sign that the sender restarted or paused.
Fixes#593354
Recent changes in gst-plugins-good/gst/effectv prevents it from being compiled
with gcc 3. The problem is that the new code uses preprocessor conditionals
within a macro call which does not work with older versions of gcc.
Fixes bug #593688.
In case of non-interleaved (= sequentially payloaded) streams,
the AU-Index serves little purpose (that is not already covered by
RTP fields). (Broken) Payloaders might consider this field then
to be disregarded and have non spec compliant values, e.g. each
RTP packet having AU-Index 2 (rather than 0). As such, ensure/force
simple sequential sending of non-interleaved streams.
3GPP specs define a number of tags along with precise layout. While these
are normally expected to be found in a container whose major brand is a
3GPP brand, this may also happen when a 3GPP brand is only mentioned as a
compatible brand. Apply some checks, heuristic and fallbacks to extract
such tags as well.
Handle large, invalid or otherwise unusual chunk sizes.
Verify some chunk sizes to be at least the size they are
expected to be and round up some sizes to even number for
e.g. offset administration, which must also be properly
tracked in push mode.
