When playing a live stream, make the recommended buffering threshold at most the
hold-back distance from live. If we start 3 seconds from the live edge, there's
no point trying to buffer more - we'll just hit the live edge and have to wait
for more data to be available anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a field to the DownloadRequest that reports the most recent time at which
data arrived. Update it in the DownloadHelper.
Add a method to retrieve the GST_BUFFER_OFFSET() for the DownloadRequest's data
buffer (if any).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
After cancelling a DownloadRequest, the download helper may not do so
immediately, so we can't assert on the in_use flag. Also, since there's no
refcount on the preload hint struct in the download request callback data, make
sure no callbacks will be dispatched when we're going to free the preload hint
struct.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Implement fulfilment of HTTP requests from the active preload downloads by
finding any preload request that can provide the requested data and feeding
bytes from the internal DownloadRequest to the caller provided target
DownloadRequest.
Doesn't yet calculate timestamps to make the target request have a sensible
apparent bitrate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add download_request_take_buffer_range() and
download_request_get_bytes_available() methods.
download_request_take_buffer_range() takes bytes from the front of the request
that satisfy the requested start/end byterange, and puts any remaining bytes
back into the DownloadRequest
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a helper that submits and handles blocking preload requests for future
PART/MAP data from live playlists. Add handling in the hlsdemux stream to submit
preload requests when hitting the end of the available segments in a live
playlist.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a flag to hlsdemux to enable or disable LL-HLS handling.
When LL-HLS is enabled and an LL-HLS playlist is loaded, use the part-hold-back
threshold to choose a starting segment.
For live streams that aren't LL-HLS, use the provided hold-back attribute, or
fall back to landing 3 segments from the end.
Make the gst_hls_media_playlist_seek() method able to choose a partial segment
within 2 target durations of the end of the playlist when requested.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Fix an off-by-one in gst_hls_media_playlist_sync_to_playlist() that would ignore
the first fragment in the reference playlist. The error was harmless, since we
expect the reference playlist to be older than the playlist we're
synchronising (so the first/oldest segment in the reference playlist will likely
not exist in the new playlist), so this is just for correctness.
Also fix a segment leak in gst_hls_media_playlist_advance_fragment() when
ignoring the partial_only segment.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Add a function for synchronising current position with the contents of a
playlist that is specifically for that and can handle synchronising to a partial
segment.
gst_hls_media_playlist_seek() will be used only when performing external seek
requests, to find the best segment or partial segment at which to resume
playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Fixes for stream_time recalculation and handling in partial segments.
Disallow bitrate switching when in the middle of partial segments - only at a
full segment (or right before the first partial segment of a segment).
It's possible but more difficult to switch bitrates in the middle of a partial
segment group, since they are less likely to have aligned keyframes. In any
case, the seek code can't do that right now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.
This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
In theory, `dispose()` functions should be idempotent and should be
prepared not to crash or cause a double-free if an unref done from
inside caused a recursive call to `dispose()` of the same object.
https://developer.gnome.org/gobject/stable/howto-gobject-destruction.html
This patch modifies the `dispose()` method to honor these constraints.
Since the double `dispose()` call won't actually occur in qtdemux (there
is no cycle detection mechanism that could invoke it to work that way),
this is more of a code cleanup than a user-facing problem fix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3822>
Without this, the plugin cannot be loaded in a devenv because the
RPATH is not added to the plugin dylib. This RPATH will be stripped on
install, which is what we want.
When deploying apps, people are supposed to use `macdeployqt` to
create an AppBundle that bundles Qt for you and sets the RPATHs
correctly to point to that bundled Qt.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3708>
Deserialize socket control messages as GstSocketTimestampMessage only
if (level, type) is (SOL_SOCKET, SCM_TIMESTAMPNS).
Without this patch, messages with types SCM_RIGHTS or SCM_CREDENTIALS
could be deserialized as GstSocketTimestampMessage instead of
GUnixFDMessage or GUnixCredentialsMessage from gio.
Fixes#1736
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3777>
AVC-Intra is a range of H.264-compliant intra-only codecs from
Panasonic. The codes and descriptions have been taken from VLC.
The (encumbered) sample I have here produces byte-stream H.264,
including SPS and PPS and no `avcC` box.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>
When calculating the presentation offset for CMAF input in live
playback, subtract the stream_time of the fragment from the
calculated presentation offset, so that the first fragment
is played at running time zero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3680>
This is recommended by various specifications for such framerates, while
for integer framerates we continue using centiframes to allow for some
more accuracy.
Using N means that no rounding error accumulates, eventually leading to
outputting a packet with a different duration.
Some tools such as MediaInfo determine that a stream is variable
framerate if any packet has a different duration than the others, and
there is no reason I can see for not using the full 4 bytes of
resolution that the mp4 timescale offers.
Example problematic pipeline:
```
videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
```
This results in a media file that MediaInfo detects as variable
framerate because the 5000th packet has duration 99 instead of 100.
With this patch, the timescale is 60000 and all packets have duration
1001.
Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>
This reverts the decision from
https://bugzilla.gnome.org/show_bug.cgi?id=754230
where it was decided that we rather play safe and only use the `tfdt` if
it is "significantly different" to the sum of sample durations.
As the specification says
If the time expressed in the track fragment decode time (‘tfdt’) box
exceeds the sum of the durations of the samples in the preceding
movie and movie fragments, then the duration of the last sample
preceding this track fragment is extended such that the sum now
equals the time given in this box.
we have to use the `tfdt` in general to allow for it to signal gaps in
the stream.
A muxer producing fragments might not yet know the full duration of the
last sample of a previous fragment if the next fragment starts with a
gap, and knowing the actual start of the next fragment would potentially
require to violate latency requirements.
Additionally, the existence of `tfdt` allows to avoid accumulating
rounding errors from summing up the durations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>
The rtpjitterbuffer test drop_messages_interval uses a GstClockTime for
the message drop interval. This property is defined as a guint. On
systems with 64-bit time_t but 32-bit uint, this can cause the
g_object_set function to fail to read the arguments properly.
Fixes: #1656
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3580>
