This allows correct handling of wrapping around backwards during the
first wraparound period and avoids the infamous "Cannot unwrap, any
wrapping took place yet" error message.
It allows makes sure that for actual timestamp jumps a valid value is
returned instead of 0, which then allows the caller to handle it
properly. Not having this can have the caller see the same timestamp (0)
for a very long time, which for example can cause rtpjitterbuffer to
output the same timestamp for a very long time.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1500
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3202>
Adding loopback capture mode for specified PID.
Note that this feature requires Windows 10 build 20348
(Windows 11/Windows Server 2022 or later),
and any process loopback related properties will not be exposed
if OS does not support it.
Example launch lines:
* wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID>
Captures audio generated by an application (specified by PID)
and its child process
* wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID>
Captures desktop audio excluding PID and its child process
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>
If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>
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>
When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.
For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
The scenario is what we try in the tests:
- we have a segment with .stop set
- some frame(s) flow
- we get a CAPS event
- we get an EOS (before getting buffers after the CAPS event)
in that case, without that patch, the segment is not properly closed
which is not correct. In this patch we keep track of previous caps until
a new buffer arrives, this way in that situation we set previous caps
again, and close the segment with the previous buffer.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1352
in this specific case
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>
The implementation was inconsistent between create and destroy. EGLImage
creation and destruction is requires for EGL 1.5 and up, while
otherwise the KHR version is only available if EGL_KHR_image_base
feature is set. Not doing these check may lead to getting a function
pointer to a stub, which is notably the case when using apitrace.
Fixes#1389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2925>
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().
Fixes#1294
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
