Adds a new plugin for ASIO devices.
Although there is a standard low-level audio API, WASAPI, on Windows,
ASIO is still being broadly used for audio devices which are aiming to
professional use case. In case of such devices, ASIO API might be able
to show better quality and latency performance depending on manufacturer's
driver implementation.
In order to build this plugin, user should provide path to
ASIO SDK as a build option, "asio-sdk-path".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2309>
Normally uri is get from user input and invalid user input should not
be treated as critical error. Moved gst_uri_is_valid outside of
g_return_val_if_fail.
NULL uri is checked inside of gst_uri_is_valid and is correctly
treated as critical error, removed unneeded checks of NULL uri outside
of gst_uri_is_valid function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/816>
... for user to be able to set the number of required samples.
For instance, our default value is 240 samples
(about 5ms latency in case that sample rate is 48000), which might
be larger than actual buffer size of audio capture device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2307>
Prevent a condition where splitmuxsink won't go back to NULL state
after a child element fails to change state by making sure that
a READY->READY state change doesn't fail, and by returning
GST_FLOW_ERROR or GST_FLOW_FLUSHING upstream to shut down streaming
as quickly as possible.
This can happen after (for example) setting an invalid filename
on the sink element. In that case, the READY->PAUSED transition
fails, but with internal elements still in the NULL state. Trying
to set splitmuxsink back to NULL then ends up trying to bring
those NULL elements up to READY with a READY->READY transition,
(which fails, prevent splitmuxsink from getting to NULL)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>
The print_ref_pic_list_b now not only needs to trace the ref_pic_list_b0/1,
but also need to trace the ref_frame_list_0_short_term. We need to pass the
name directly to it rather than an index to refer to ref_pic_list_b0/1.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2425>
When driver return error on update plane request, kmssink
disables the scaling and retries plane update.
While doing so kmssink was matching the source rectangle dimensions
to the target rectangle dimensions which were calculated
as per scaling but this is incorrect, instead what we want here is
that target rectangle dimensions should match the source rectangle
dimensions as scaling is disabled now and so we match result
rectangle dimensions with source rectangle dimensions.
While at it, also match the result rectangle coordinates for
horizontal and vertical offsets with source rectange coordinates,
as since there is no scaling being done so no recentering is
required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2415>
Some cut VP9 streams begin with a non key frame. The current code
just bail out the parse_process_frame() if not a key frame. Because
of this, we do not set the valid caps before we push the data of the
first frame(even this first frame will be discarded by the downstream
decoder because it is not a key frame).
The pipeline such as:
gst-launch-1.0 filesrc location=some.ivf ! ivfparse ! vp9parse !
vavp9dec ! fakesink
will get a negotiation error and the pipeline can not continue. The
correct behaviour should be: the decoder discard the first frame and
continue to decode later frames successfully.
So, when the parse does not have valid stream info(should be the first
frame case), we should continue and report caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2427>
We may need to drop the slices such as RASL pictures with the NoRaslOutputFlag, so
the current picture of h265decoder may be freed. We should not assign the frame->
output_buffer too early until we really output it. Or, the later coming slices will
allocate another picture and trigger the assert of:
gst_video_decoder_allocate_output_frame_with_params:
assertion 'frame->output_buffer == NULL' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2421>
There is currently no way for users to receive incoming events from
appsink while keeping them properly serialized with the buffers flow.
This can be especially useful when application is injecting custom
downstream events into the pipeline and needs to know when they reached
appsink.
Solving this by adding a new signal notifying about new incoming events
and a set of action signals and method to pull those events.
The API is actually pulling the samples and events all together as they
are actually fetched from the same queue.
Having a specific API to pull only events would have the side effect of
discarding samples (and pulling samples would discard events) making
this API not convenient for users.
Partially fix#247
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
In H265, the stream may have odd bit depth such as 9 or 11. And
the bit depth of luma and chroma may differ. For example, the
stream with luma depth of 8 and chroma depth of 9 should use the
10 bit rtformat as the decoded picture format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2420>
The problem is that EGLNativeWindowSurface and wl_egl_surface are the
same object underneath, so we must recreate both together. As an
optimization, the EGLNativeWindowSurface wrapper is only re-created
if the window_handle changed.
On Mesa, this would cause crash, which will be fixed by:
https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/11979
And will lead to proper errors in the future or on other GL stack. This
issue was encounter using a permanent GstGLDisplay after cycling one of
multiple independent pipelines through NULL state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1230>
In low_latency mode, try to bump the picture as soon as possible
without the frames disorder:
1. We can directly output the continuous non-reference frame.
2. Consider max_num_reorder_frames, which is special useful for
I-P mode.
3. Consider the leading pictures with negative POC.
4 Output small POC pictures when non-reference frame comes.
4. Output the POC increment<=2 pictures. This is not 100% safe,
but in practice this condition can be used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2373>
The picture->ref field will change from time to time according to decoder's
state and reference sliding window. We need another flag to record whether
the picture is a reference picture when it is created, and this can help
the bumping check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2373>
Accord to spec, we should not add the current picture into the DPB
when we check whether it needs to bump, so the checks of the IDR and
the "memory_management_control_operation equal to 5" are no needed.
And the spec also says that the DPB only needs to bump when there is
no empty frame buffer left(We handle the IDR cases in other places).
We need to follow that and the max_num_reorder_frames is useless.
We also minus 1 in has_empty_frame_buffer because the current frame
has not been added yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2373>
When current frame memory_management_control_operation equal to 5, that
means we need to drain the dpb and the current picture act as an IDR frame.
So it should have smaller poc than the later pictures to ensure the output
order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2373>
* If we have an index table for non-framed essence, we can handle it
* The demuxer has a state which indicates whether it will next fetch a KLV or
data contained *within* a KLV.
* The position on Essence Tracks always correspond to the next entry to fetch,
demuxer offset will be skipped accordingly whenever we switch between
partitions (in case of resyncs). A copy of the main clip/custom KLV for that
partition is kept to track the position within the essence of that partition.
* For clip/custom-wrapped raw audio, if the edit rate is too small (and would
cause plenty of tiny buffers to be outputted), specify a minimum number of edit
units per buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
* For pull-based, this avoids pulling content if it's not needed (ex: skipping filler
packet, not downloading the content if we only need to know if/where an essence
packet is, etc...). Allows reducing i/o usage to the minimum.
* This also allows doing sub-klv position tracking, and opens the way for
non-frame-wrapping handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
In order to figure out the exact start position (backed by a keyframe) accross
all tracks, we first figure out the backing keyframe position, and *then* seek
to that position.
Avoids ending up in situations where we would properly seek to the backing
keyframe on video ... but not on the audio streams (they would have been set to
the original non-keyframe position). Fixes key-unit seeking.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
The picture essence coding matching was wrong. Use the proper "base" MXFUL for
video mpeg compression for matching.
Also handle the case where some old files would put the essence container label
in the essence coding field
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
* Streamline offset <=> entry handling. Historically the demuxer didn't support
information from index tables and stored the discovered information in an array
per track. When index table support was added, a parallel system was setup for
that relationship. This commit unifies this into one system with the
`find_edit_entry()` and `find_entry_for_offset()` functions.
* By extension, per-track offset entry tables are only created/used if no index
table is present for those tracks.
* Use index table information as-is. The index table system from MXF is quite
complex and there are various ways to use the information contained
within. Instead of converting that information we store the data from the tables
as-is and extract the needed information when needed.
* Handle index tables without entries (i.e. all content package units are of the
same size).
* Allow collecting index table segments as we go instead of only once if a
random-index-pack is present. This also improves support of some files in
push-mode.
* When searching for keyframe entries, use the keyframe_offset if
present (speeds up searching).
* For interleaved content (i.e. several tracks in the sample essence container),
we use a system to be able to identify the position of each track in the delta
entries of index tables.
* Handle temporal offset only on tracks which *do* need it (as specified in the
delta entries of the index tables). If present, those offsets are stored in a
pre-processed table which allows computing PTS from DTS with a simple offset.
* Add a quirk for files which are known to be have wrongly stored temporal
offsets.
* Overall opens the way to handle more types of MXF files, especially those with
non-frame-wrapping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
This is similar to how the same issue was handled in qtdemux.
In order for the "DTS <= PTS" constraint to be respected, we calculate the
maximum temporal reordering that can happen (via index tables).
If there is a non-0 temporal reordering, we:
* Shift all outgoing PTS by that amount
* Shift segment for that stream by that amount
* Don't modify DTS (i.e. they might end up having negative running-time, before
the start of the segment)
Also ensure all entries have a valid PTS set, previously this wouldn't be set
for entries with a temporal offset of 0.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/584
(and maybe a lot of other issues)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
120 bytes in 5 blocks are definitely lost in loss record 7,615 of 9,510
at 0x484486F: malloc (vg_replace_malloc.c:380)
by 0x58A2938: g_malloc (gmem.c:106)
by 0x58BA1F4: g_slice_alloc (gslice.c:1069)
by 0x588F059: g_list_prepend (glist.c:335)
by 0x5B9C5C0: select_best_master_clock (gstptpclock.c:756)
by 0x5B9CA8E: cleanup_cb (gstptpclock.c:1930)
by 0x589AD20: g_timeout_dispatch (gmain.c:4889)
by 0x589A4CE: UnknownInlinedFun (gmain.c:3337)
by 0x589A4CE: g_main_context_dispatch (gmain.c:4055)
by 0x58EE4E7: g_main_context_iterate.constprop.0 (gmain.c:4131)
by 0x5899A92: g_main_loop_run (gmain.c:4329)
by 0x5B9BA4C: ptp_helper_main (gstptpclock.c:1980)
by 0x58C8C31: g_thread_proxy (gthread.c:826)
576 bytes in 24 blocks are definitely lost in loss record 8,782 of 9,510
at 0x484486F: malloc (vg_replace_malloc.c:380)
by 0x58A2938: g_malloc (gmem.c:106)
by 0x58BA1F4: g_slice_alloc (gslice.c:1069)
by 0x588F059: g_list_prepend (glist.c:335)
by 0x5B9C5C0: select_best_master_clock (gstptpclock.c:756)
by 0x5B9EFA0: handle_announce_message (gstptpclock.c:934)
by 0x5B9EFA0: handle_ptp_message (gstptpclock.c:1765)
by 0x5B9EFA0: have_stdin_data_cb (gstptpclock.c:1851)
by 0x589A4CE: UnknownInlinedFun (gmain.c:3337)
by 0x589A4CE: g_main_context_dispatch (gmain.c:4055)
by 0x58EE4E7: g_main_context_iterate.constprop.0 (gmain.c:4131)
by 0x5899A92: g_main_loop_run (gmain.c:4329)
by 0x5B9BA4C: ptp_helper_main (gstptpclock.c:1980)
by 0x58C8C31: g_thread_proxy (gthread.c:826)
by 0x5DA4298: start_thread (pthread_create.c:481)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/852>
Currently the decoder selection is very naive: The type with the highest
rank that matches the current caps is used. This works well for software
decoders. The exact supported caps are always known and the static caps are
defined accordingly.
With hardware decoders, e.g. vaapi, the situation is different. The decoder
may reject the caps later during a caps query. At that point, a new decoder
is created. However, the same type is chosen an after several tries,
decodebin fails.
To avoid this, do the caps query while adding the decoder and try again
with other decoder types if the query fails:
1. create the decoder from the next matching type
2. add and link the decoder
3. change the decoder state to READY
4. do the caps query
if it fails then remove the decoder again and go back to 1.
5. expose the source pad
6. sync the decoder state with the parent.
This way, the decoder is already part of the pipeline when the state change
to READY happens. So context handling should work as before.
Exposing the source pad after the query was successful is important:
Otherwise the thread from the decoder source pad may block in a blocked pad
downstream in the playsink waiting for other pads to be ready.
The thread now blocks trying to set the state back to NULL while holding
the SELECTION_LOCK. Other streams may block on the SELECTION_LOCK and the
playsink never unblocks the pad. The result is a deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1201>
