When calculating the seek range for a live stream, use the same hold-back logic
as when choosing a starting segment, including low-latency segments if
enabled. Permits seeking closer to the live edge when re-synching or catching
up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When playing at the live edge of a live playlist, and a download fails, we don't
expect there to be a next fragment. That case is handled lower down anyway, so
don't retry infinitely on spurious http errors at the live edge.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
gst_hls_demux_stream_has_next_fragment() can be called with a NULL
current_segment if we're past the end of the current playlist. In that case,
just return FALSE instead of hitting a critical in the playlist code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When playing LL-HLS playlists in LL-HLS mode, update the playlist more often (on
the partial segment interval) or else we end up downloading them in bursts and
playing further from the live edge than intended.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
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>
If we keep the old events they can be end up being passed to the app, that could
discard the protection information because it has been seen before.
Drive by improvement: use g_queue_clear_full instead of foreach+clear for
protection events.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3547>
jpegdec is capable to parse input frames, but if jpegparse is before,
there's no need to reparse frames. This patch configure jpegdec as
packetized, skipping parsing, if negotiated sink caps has the boolean
field 'parsed' as true.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2464>
According to comment in gst_pulsering_stream_latency_cb, latency updates
happen every 100 ms. The code in gst_pulsering_stream_latency_cb does
not care about the actual state of the ringbuffer, pbuf->acquired or
not.
Thus, every 100 ms the ringbuf->segdone may be set, even though the
object itself might be in 'destroyed' state. On next
gst_pulseringbuffer_acquire the segdone is not touched, so playback may
resume with invalid/wrong segdone value. This finally leads to a period
of silence playing after resuming the pipeline.
The problem was found on 'not-yet-released'-hardware and so far was not
reproducible on desktop computer.
Removing the callback as long as the ringbuffer is not in 'acquired'
state solves the problem reliably on the hardware device that the issue
was detected on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3082>
The stream selection is done on the currently outputting tracks, but in order to
(de)activate the backing streams we can only do it if the input and output
period are identical.
Fixes crash when doing stream selection during period migration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3525>
This regression was introduce by fix for making buffer pool thread safe. When
we renegotiate, the pool will be setup after we set the format. But the code
has been simplified to only get the pool once before, which caused a null
pointer deref.
Fixes 94ba019 ("v4l2: Fix SIGSEGV on 'change state' during 'format change'")
Related to !3481Fixes#1626
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3513>
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.
This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
- Based heavily on the existing Qt5 integration however:
- The sharing of OpenGL resources is slightly different
- The integration with the scengraph is a bit different
- Wayland, XCB and KMS have been smoke tested. Android, MacOS/iOS,
Windows may or may not work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3281>
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.
Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.
Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>
In current tile representation, only tiles with power of two
width and height in bytes are supported. This limitation
prevents adding more complex tiles formats.
In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
replace if with an array of GstVideoTileInfo. Each plane tiles are then
described with their pixels width/height, line stride and total size.
The helper gst_video_format_info_get_tile_sizes() that depends on the
deprecated API is also being removed. This can simply be removed as it wasn't
in any stable release yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
The only case where we definitely need to write a new trun is when the
data_offset value does not match the end of the list of entries.
Needing multiple trun atoms is required when interleaving multiple
streams together.
All other cases can be covered by adding more entries to the existing
trun atom.
Fixes playback of fragemented mp4 in ffplay and chrome.
Using e.g. mp4mux fragment-duration=1000 fragment-mode=dash-or-mss
and
mp4mux fragment-duration=1000 fragment-mode=first-moov-then-finalise
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3426>
Postpone the cleanup of any consecutive sequence of lost frames
which starts at frame 0, until frame 100 is dequeued from driver.
This allows fluster tests JVT/CVWP2_TOSHIBA_E, JVC/CVWP3_TOSHIBA_E
and HEVC/POC_A_Bossen_3 that sends out-of-order frames to successfully
complete (e.g., test of Amphion vpu driver).
Fixes#1569
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3398>
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, wavparse calls the typefinder helper
except that means it runs all typefinders.
Since it only cares about checking for DTS, we should only run the
audio/x-dts typefinder (if present). Commit 858e516 did not really
fix things.
Use the new type helper with the caps to fix this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3417>
Set udpsrc seqnums on all events sent to the udpsrc's, and before
forwarding events out of rtspsrc set the latest seek seqnum on them if
any.
Also produce a consistent seqnum in rtspsrc from the very beginning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
Instead of returning a "const gchar" or a "gchar" that should not be freed, always
return a duplicated string as those functions were used together with g_strdup anyway.
This is needed to prepare support for returning modified strings in next commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
This is small regression from commit f7abd81a.
When calling `gst_element_query()` no pad is associated with that query, but the
current code always forwards the query to the associated pad, which is NULL in
previous case. This patch checks for the pad before forwarding the query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3404>
In a few cases throughout qtdemux, the results of QT_UINT32 were being
stored in a signed integer, which could cause subtle bugs in the case of
an integer overflow, even allowing the the result to equal a negative
number!
This patch prevents this by simply storing the results of this function
call properly in an unsigned integer type. Additionally, we fix up the
length checking with stsd parsing to prevent cases of child atoms
exceeding their parent atom sizes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3344>
Unlike the legacy elements, GstAdaptiveDemuxStream is a GObject now,
so a bunch of things that were actually stream methods on the
parent demux object can directly become stream methods now.
Move the stream class out to a header of its own.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
Sometimes g_input_stream_read_all_finish() can return
0 bytes, but still succeed (return TRUE) and have more
data available later. Only finish the transfer
if it returns 0 bytes *and* FALSE with no error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
The cancelled flag was only set in the stream finalize()
method, after all activity on the stream has stopped anyway.
Replace uses of cancelled with checks on the stream state.
Remove the replaced flag, which was checked but never set
to TRUE anywhere any more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
When matching segments across playlists with Program-Date-Times,
use the difference in segment PDTs to adjust the stream time
that's being transferred. This can fix cases where the
segment boundaries don't align across different streams
and the first download gets thrown away once the PTS
is seen and found not to match.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3309>
Check whether the init file / MAP data for a segment
is different to the current data and trigger an
update if so. Previously, the header would only
be checked in HLS after switching bitrate or
after a seek / first download.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3307>
Previously the minimum buffering threshold was hardcoded to a specific
value (10s). This is suboptimal this an actual value will depend on the actual
stream being played.
This commit sets the low watermark threshold in time to 0, which is an automatic
mode. Subclasses can provide a stream `recommended_buffering_threshold` when
update_stream_info() is called.
Currently implemented for HLS, where we recommended 1.5 average segment
duration. This will result in buffering being at 100% when the 2nd segment has
been downloaded (minus a bit already being consumed downstream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3240>
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, right now we call the typefinder helper
which runs all typefinders.
Speed up this type finding process by specifying the extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3294>
We've seen occasional crashes in the `wavparse` module associated with
referencing a buffer in `gst_wavparse_chain` that's already been freed. The
reference is stolen when the buffer is transferred to the adapter with
`gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
the buffer, the buffer could be freed during interaction with the adapter in
`gst_wavparse_stream_headers`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>
in certain ways.
In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000 52 49 46 46 e4 fd 00 00 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt |
00000010 12 00 00 00 01 00 01 00 80 3e 00 00 00 7d 00 00 |.........>...}..|
00000020 02 00 10 00 64 61 74 61 |....data|
00000028
```
(Note that the original file is much larger. This was the smallest sub-file
I could find that would generate the crash.)
Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
This is a regression that was introduced in
cca2f555d1 (yes, 9 years ago).
The only place where a demuxer streaming thread should be stopped is when the
sinkpad is deactivated from pull mode (i.e. PAUSED->READY).
Attempting to stop the task in this function would cause this to happen when a
FLUSH_STOP or STREAM_START event is received... which can cause deadlocks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3109>
If the SETUP request returns an IPv6 server address in the Transport
field, we would generate an incorrect URI, and multiudpsink would fail
to initialize:
```
rtspsrc gstrtspsrc.c:9780:dump_key_value:<source> key: 'Transport', value: 'RTP/AVP;unicast;source=fe80::dc27:25ff:fe5e:bd13:8080;client_port=62696-62697;server_port=4000-4001'
...
rtspsrc gstrtspsrc.c:4595:gst_rtspsrc_stream_configure_udp_sinks:<source> configure RTP UDP sink for fe80::dc27:25ff:fe5e:bd13:8080:4000
...
multiudpsink gstmultiudpsink.c:1229:gst_multiudpsink_configure_client:<udpsink0> error: Invalid address family (got 23)
```
We can't look at stream->is_ipv6 because we can't rely on the server
returning the right value there. In the issue reported about this,
server reported itself as `KuP RTSP Server/0.1`, and the SDP was:
```
c=IN IP4
m=video 54608 RTP/AVP 96
a=rtpmap:96 H264/90000
```
So we need to parse the string value and figure out the family
ourselves.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1058
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1819>
