New API:
- gst_gl_context_get_config()
- gst_gl_context_request_config()
A GL context configuration is a GstStructure that has some well-known
names for common values that can also be extended in platform-specific
ways if necessary.
Wrapped OpenGL contexts may be able to retrieve the GL context
configuration depending on the platform. If that information is
available, GstGLContext will attempt to create an context that matches
the shared OpenGL context config unless gst_gl_context_request_config()
has been called.
A new environment variable 'GST_GL_CONFIG' will be read to influence the
configuration chosen. The environment variable will only be used as a
fallback if gst_gl_context_request_config() has not been called.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/809>
This meta hold one buffer of the same codec data as the parent memory. This
extra frame luma will be used as the alpha values for the final combined
frame. This is notably used to support VP8/VP9 alpha as defined in WebM and
matroska specification.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1128>
These work the same way as the corresponding properties on queue and
allow to control the internal buffer size of the appsrc in a more
flexible way.
Unlike in queue the max-buffers and max-time properties are 0 (i.e.
disabled) by default for backwards compatibility reasons.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1133>
`@` references are used to reference function parameters, struct members
or enum variants _within_ the current type/function. It cannot and
should not be used to reference to types outside that.
Since C has no notion of member functions it makes little sense to
prefix these with `@`; most of the documentation here was referencing
functions on _different_ types anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1090>
If the alsasink thread starts the write loop but another thread pauses
the underlying alsa device, the sink thread will endlessly loop.
snd_pcm_writei() will return 0 if the state is SND_PCM_STATE_PAUSED
and the loop will never make any progress.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1097>
Table 1.10 – "Levels for the AAC Profile" only goes to 5 max channels
/ 7 max channel post amendmend, so I assume the number of channels
should not include LFE, otherwise there's no valid level for 5.1 resp.
7.1 (post amendmend)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/680>
Based upon valgrind finding:
Conditional jump or move depends on uninitialised value(s)
at 0x4AFF589: read_rtp_header_extensions (gstrtpbasedepayload.c:1197)
by 0x4AFF9E5: gst_rtp_base_depayload_set_headers
(gstrtpbasedepayload.c:1298)
by 0x4AFFEE0: gst_rtp_base_depayload_do_push
(gstrtpbasedepayload.c:1413)
by 0x4AFFF53: gst_rtp_base_depayload_push
(gstrtpbasedepayload.c:1448)
by 0x4AFDEBA: gst_rtp_base_depayload_handle_buffer
(gstrtpbasedepayload.c:801)
by 0x4AFE41E: gst_rtp_base_depayload_chain_list
(gstrtpbasedepayload.c:899)
by 0x48F262C: gst_pad_chain_data_unchecked (gstpad.c:4414)
by 0x48F3333: gst_pad_push_data (gstpad.c:4655)
by 0x48F3DF8: gst_pad_push_list (gstpad.c:4814)
by 0x4AFAD87: gst_rtp_base_payload_push_list
(gstrtpbasepayload.c:1978)
by 0x72B3154: gst_rtp_vp8_pay_handle_buffer (gstrtpvp8pay.c:672)
by 0x4AF7031: gst_rtp_base_payload_chain (gstrtpbasepayload.c:868)
Uninitialised value was created by a heap allocation
at 0x483C77F: malloc (in
/usr/lib/x86_64-linux-gnu/valgrind/vgpreload_memcheck-amd64-linux.so)
by 0x4B8BA78: g_malloc (gmem.c:106)
by 0x4BA3A9D: g_slice_alloc (gslice.c:1069)
by 0x488D777: _sysmem_new_block (gstallocator.c:413)
by 0x488DB28: default_alloc (gstallocator.c:512)
by 0x488D3E8: gst_allocator_alloc (gstallocator.c:310)
by 0x4AE97E3: gst_rtp_buffer_set_extension_data (gstrtpbuffer.c:856)
by 0x4AF9EC6: set_headers (gstrtpbasepayload.c:1757)
by 0x489FE4D: gst_buffer_list_foreach (gstbufferlist.c:287)
by 0x4AFA87A: gst_rtp_base_payload_prepare_push
(gstrtpbasepayload.c:1915)
by 0x4AFAD06: gst_rtp_base_payload_push_list
(gstrtpbasepayload.c:1970)
by 0x72B3154: gst_rtp_vp8_pay_handle_buffer (gstrtpvp8pay.c:672)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1075>
e.g. if we have:
video-x/raw,format=I420 ! compositor ! video/x-raw,format=BGRA
This will currently produce a warning as the alpha-ness of the chosen
'best' format (I420) will be different from the value restricted by the
downstream caps filter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1059>
The converter might have a non-passthrough mix-matrix. The converter
can determine whether it should pass through, so let it, then remove it
if it's indeed a passthrough.
FIXME: Not converting when we need to but the config is invalid (e.g.
because the mix-matrix is not the right size) produces garbage. An
invalid config should cause a GST_FLOW_NOT_NEGOTIATED.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1070>
The users of this API need to be able to differentiate between EINTR
and ERROR. For example, in rtspsrc, gst_rtsp_conninfo_connect()
behaves differently when gst_rtsp_connection_connect_with_response_usec()
returns an ERROR or EINTR. The former is an element error while the
latter is simple a GST_ERROR since it was a user cancellation of the
connection attempt.
Due to this, rtspsrc was incorrectly emitting element errors while
going to NULL, which would or would not reach the application in
a racy manner.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1069>
While preparing a blist for pushing, some RTP header extension may
request caps change for a specific buffer in the list. When this
happens, depayloader should immediately push those buffers from the list
that precede the currently processed buffer (for which the caps change
was requested) and only then apply the new caps to the src pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
Some header extensions may need to read information from the payloader's
sink caps. Introduce gst_rtp_header_extension_update_from_sinkcaps ()
that passes the caps to the extension, which can then use it to update
its internal state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
The way pad->priv->input_buffer reference was managed was pretty
spurious:
- it was overridden without unrefing it, which could potentially lead to
leaks.
- we were unreffing it while keeping the pointer around, which could
potentially lead to use-after-free or double-free.
As priv->input_buffer is actually no longer used outside of the
aggregate() method, remove it from pad->priv to simplify the code and
prevent the issues desribed above.
Fix a single buffer leak when shutting down the pipeline as the buffer
returned from gst_aggregator_pad_drop_buffer() was never unreffed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1061>
This code path is meant to convert the current buffer to the new format
on update. It was using priv->input_buffer as input which is either
priv->buffer or a converted version of it.
Use priv->buffer instead as priv->input_buffer may no longer be a valid
reference.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1061>
If the `render_rect` for a dispmanx display is set after calling
`window_resize` the resize defaults to the dp_width and dp_height to
determine the location of the render rectangle instead of the correct
dimensions that should be set on the window_egl.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1056>
I found a rather reproducible race in a WebKit LayoutTest when a player
was intantiated and a VP8/9 video was loaded, then torn down after
getting the video dimensions from the caps.
The crash occurs during the handling of the first frame by gstvpxdec.
The following actions happen sequentially leading to a crash.
(MT=Main Thread, ST=Streaming Thread)
MT: Sets pipeline state to NULL, which deactivates vpxdec's srcpad,
which in turn sets its FLUSHING flag.
ST: gst_vpx_dec_handle_frame() -- which is still running -- calls
gst_video_decoder_allocate_output_frame(); this in turn calls
gst_video_decoder_negotiate_unlocked() which fails because the
srcpad is FLUSHING. As a direct consequence of the negotiation
failure, a pool is NOT set.
gst_video_decoder_negotiate_unlocked() still assumes there is a
pool, crashing in a critical in gst_buffer_pool_acquire_buffer()
a couple statements later.
This patch fixes the bug by returning != GST_FLOW_OK when the
negotiation fails. If the srcpad is FLUSHING, GST_FLOW_FLUSHING is
returned, otherwise GST_FLOW_ERROR is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1031>
