When dealing with live streams, the function was assuming that all segments of
the playlist had valid stream_time. But that isn't TRUE, for example in the case
of failing to synchronize playlists.
Fixes losing sync due to not being able to match playlist on updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
The name is already passed via the signal parameters so it doesn't have
to be retrieved again via GstObject API, which would crash on other
GObjects. Child proxy child objects can be any kind of GObject and the
code here otherwise handles this correctly already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6951>
On various 32 bit systems, time_t is actually 64 bits while long is
still only 32 bits. The macro would wrongly trigger its assertion in
this case if a value with more than 68 years worth of seconds is
converted.
Examples are various newer 32 bit platforms and old ones that are
compiled with -D_TIME_BITS=64.
Also statically assert that time_t is either 32 or 64 bits. Other values
might need adjustments in the macro.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6919>
The prime_fds for multi planes may be the same. For example, on Intel's
platform, the NV12 surface may have the same FD for the plane0 and the
plane1. Then, the DRM_IOCTL_GEM_CLOSE will close the same handle twice
and get an "Invalid argument 22" error the second time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6916>
To simplify the description, I'm assuming we only have two streams: video and audio.
For the video stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(1) => blocked waiting in gst_stream_synchronizer_wait
- FLUSH_START => unblocked
- FLUSH_STOP => stream->wait reset to false
- NEW_SEGMENT(2) => not waiting, since stream->wait is false
Then for the audio stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(2) => blocked waiting in gst_stream_synchronizer_wait for ever.
Note: The first NEW_SEGMENT event and the FLUSH_START, FLUSH_STOP events of the audio stream
are dropped before being received by the streamsynchronizer element, because the decodebin audio pad src
is not yet linked to the playsink audio pad sink.
To fix this deadlock, we don't reset stream->wait to false in the FLUSH_STOP event when it is not
waiting for the EOS of the other streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6887>
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6857>
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6857>
This mirrors the behaviour in vp8enc / vp9enc and is generally more
useful than using any framerate from the caps as it provides some degree
of accuracy if the stream doesn't have timestamps perfectly according to
the framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6909>
Turns out AudioConvertHostTimeToNanos and AudioGetCurrentHostTime are macOS-only APIs, which prevents apps using
GStreamer on iOS from being accepted into App Store.
This commit replaces those functions with a manual version of what they do - mach_absolute_time() for the current time,
and data from mach_timebase_info() at the beginning to convert host timestamps to nanoseconds.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6899>
_get_osfhandle() expects valid fd and CRT will abort program
if given paramerter is invalid. The fd can be invalidated
in various way, file was deleted by other process after
we open a file. To avoid it, our own exception
handler must be installed so that _get_osfhandle() can return
INVALID_HANDLE_VALUE if fd is invalid.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6879>
If the muxer times out because of the latency deadline it can happen
that some pads have no caps yet. In that case skip creation of streams
for these pads and create updated section tables once the first buffer
arrives later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6859>
This makes sure that for sparse streams (KLV, DVB subtitles, ...) the
muxer does not wait until the next buffer is available for them but
times out on the latency deadline and outputs data.
For non-live pipelines it will still be necessary for upstream to
correctly produce gap events for sparse streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6859>
__STDC_NO_ATOMICS doesn't seem to exist. In fact the only compiler
I've found that sets any of those is msvc, but it sets
__STDC_NO_ATOMICS__, not __STDC_NO_ATOMICS.
__STDC_NO_ATOMICS__ is the one documented by C11 standard.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6856>
This fixes the code regarding dropping "ghost frames", that is to say input
frames which ended up not producing any decoded frame.
The iteration itself makes sense.. but it was stopping at the "input" frame and
not the decoded frame we just got back.
When dealing with I-frame codecs, ffmpeg will decode frames in separate frames,
so there is no guarantee that they are decoding in order.
Fixes playback issues with such codecs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6845>
There's nothing requiring <= 64 channels except for getting the reorder
map and creating a channel mixing matrix, but those won't be possible to
call anyway as channel positions can only express up to 64 channels.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6828>
Certain V4L2 fourccs don't (yet) have DRM counter parts, in which case
we can't create DMA_DRM caps for them. This is usually the case for
specific tilings, which are represented as modifiers for DMA formats.
While using these tilings is generally preferable - because of e.g.
lower memory usage - it can result in additional conversion steps when
interacting with DMA based APIs such as GL, Vulkan or KMS. In such cases
using a DMA compatible format usually ends up being the better option.
Before the addition of DMA_DRM caps, this was what playbin3 ended up
requesting in various cases - e.g. prefering NV12 over NV12_4L4 - but
the addition of DMA_DRM caps seems to confuse the selection logic.
As a simple and quite robust solution, assume that peers supporting
DMA_DRM caps always prefer these and reorder the caps accordingly.
In the future we plan to have a translation layer for cases where
there is a matching fourcc+modifier pair for a V4L2 fourcc, ensuring
optimal results.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6824>
Doing it in gst_play_new() means that bindings that directly call
g_object_new() with the GType wouldn't end up initializing both.
This affects at least the Python and GJS bindings.
gst_init() is nonetheless only called from gst_play_new() once because
calling it from class_init would likely lead to problems as that's
called from somewhere in the middle of GObject.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6821>
