Commit graph

25739 commits

Author SHA1 Message Date
Yeongjin Jeong
454147d269 vp9parser: Add new API for parsing superframe info
Some elements are using their own implementations for superframe parsing.
To reduce redundant code, we need to add API to the parser.
2020-03-02 01:31:38 +09:00
Guillaume Desmottes
09367da35c transcodebin: mark properties as GST_PARAM_MUTABLE_READY
They are all used in the READY to PAUSED transition so should not be
changed after.
2020-02-28 16:57:30 +00:00
Guillaume Desmottes
de4ea94766 transcodebin: force decoding if a filter is defined
Filter operates on raw data so don't allow decodebin to produce
encoded data if one is defined.

My use case here is keeping the video stream untouched but apply a filter
on the audio one, while keeping the same audio format.
2020-02-28 16:57:30 +00:00
Guillaume Desmottes
4b6164339f transcodebin: logs when inserting, or not, a filter
It's not easy atm to figure out from the logs if a filter has actually be
inserted or not.
2020-02-28 16:57:30 +00:00
Seungha Yang
9575d835ba d3d11decoder: Do not print warning message if retry count is in expected range
gst_d3d11_result() will print warning message when HRESULT != S_OK.
However, since the retry is trivial stuff, check hr == E_PENDING first
and do not warn it.
2020-02-28 13:21:04 +09:00
Seungha Yang
418e6991c1 d3d11decoder: Check decoder status report
... and if h/w decoder reports error, increase error count.
2020-02-28 13:21:00 +09:00
Olivier Crête
26ac42f7c0 fakevideosink: Align max-lateness/processing-deadline to GstVideoSink
To emulate correctly the timing video of a real sink, let's set those
properties just like a real video sink.
2020-02-27 23:25:44 +00:00
Jan Schmidt
8e3472faee webrtc: Use the dtlssrtenc rtp-sync property
Instead of synchronising at the ICE transport, do clock sync for the
RTP stream at the DTLS transport via the dtlssrtpenc rtp-sync
property. This avoids delaying RTCP while waiting until it is time
to output an RTP packet when rtcp-mux is enabled.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
2020-02-27 12:30:32 +00:00
Jan Schmidt
0c72a41767 gstdtlsrtpenc: Add rtp-sync property
Add an rtp-sync property which synchronises RTP streams
to the pipeline clock before passing them to funnel for
merging with RTCP.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
2020-02-27 12:30:32 +00:00
Nirbheek Chauhan
a06ddd182d dash: Don't use sscanf + glib format modifiers
We do not have a way to know the format modifiers to use with string
functions provided by the system. `G_GUINT64_FORMAT` and other string
modifiers only work for glib string formatting functions. We cannot
use them for string functions provided by the stdlib. See:
https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description

F.ex.
```
 ../ext/dash/gstxmlhelper.c: In function 'gst_xml_helper_get_prop_unsigned_integer_64':
../ext/dash/gstxmlhelper.c:473:40: error: unknown conversion type character 'l' in format [-Werror=format=]
     if (sscanf ((gchar *) prop_string, "%" G_GUINT64_FORMAT,
                                        ^~~
In file included from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32,
                 from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32,
                 from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/glib-2.0/glib.h:30,
                 from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27,
                 from ../ext/dash/gstxmlhelper.h:26,
                 from ../ext/dash/gstxmlhelper.c:22:
/builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here
 #define G_GUINT64_FORMAT "llu"
                            ^
../ext/dash/gstxmlhelper.c:473:40: error: too many arguments for format [-Werror=format-extra-args]
     if (sscanf ((gchar *) prop_string, "%" G_GUINT64_FORMAT,
                                        ^~~
```

In the process, we're also following the DASH MPD spec more closely
now, which specifies that ranges must follow RFC 2616 section 14.35.1:
https://tools.ietf.org/html/rfc2616#page-138
2020-02-27 09:42:33 +00:00
Sebastian Dröge
cc8b90967b dtls: Set a random serial number and issuer/subject in the self-signed certificates
This is also what Chrome and Firefox are doing, citing privacy concerns.
Also putting OpenWebRTC from Sweden as issuer/subject is rather
confusing.
2020-02-27 08:27:19 +00:00
Guillaume Desmottes
2cb7c66ac7 transcodebin: consider 'any' as no restriction
gstreamer-rs set 'any' as default restriction which actually means 'no
restriction' so handle it as the absence of restriction.
2020-02-26 13:12:37 +00:00
Seungha Yang
f2e322197e d3d11window: Set DXGI_PRESENT_ALLOW_TEARING only in fullscreen mode
The DXGI_PRESENT_ALLOW_TEARING flag might cause unexpected tearing
side effect. Setting it in fullscreen mode only seems to be
the correct usage as in the Microsoft's direct3d examples.
2020-02-26 13:18:16 +09:00
Guillaume Desmottes
54d8360baa transcodebin: fix caps leak
encodecaps was leaked if the profile has restrictions.
2020-02-26 03:23:20 +00:00
Jan Alexander Steffens (heftig)
91a033a85e
rtmp2: Allow setting flash-version
In case the application has to deal with fussy servers. User agent
sniffing is so last decade.

Adds a property to set the Flash version on both the sink and the src.
The default stays the same (IIRC, Flash plugin for Linux from 2009).
2020-02-25 15:10:28 +01:00
Jan Schmidt
69ef74d96a h265parser: Skip unused bits in an SEI.
Alternative approach to 18b54f8d34
that skips all bits in a broken SEI correctly.
2020-02-25 09:33:47 +00:00
Stéphane Cerveau
98f8bb4e7e Revert "h265parser: Skip unused SEI bits differently"
This reverts commit 18b54f8d34.
2020-02-25 09:33:47 +00:00
Stéphane Cerveau
5881c1ad2d tests: add picture timing SEI parsing 2020-02-25 09:33:47 +00:00
Dylan Yip
0c798c8771 h265parse: Fix offset by one error in pic timing SEI
Offset by one error causes a free/malloc error when parsing pic timing
SEI messages.
2020-02-25 09:33:47 +00:00
Seungha Yang
4643dc1c12 d3d11decoder: Ensure the written bitstream buffer size is 128 bytes aligned
DXVA spec is saying that the size of bitstream buffer provided by hardware decoder
should be 128 bytes aligned. And also the host software decoder should
align the size of written buffer to 128 bytes. That means if the slice
(or frame in case of VP9) size is not aligned with 128 bytes,
the rest of non 128 bytes aligned memory should be zero-padded.

In addition to aligning implementation, some variables are renamed
to be more intuitive by this commit.
2020-02-24 14:46:27 +00:00
Haihao Xiang
acc14225c9 msdk: libva: Don't set the hint if MFX_VERSION is lower than 1025
MFX_MEMTYPE_VIDEO_MEMORY_ENCODER_TARGET is defined since MFX_VERSION
1025.
2020-02-24 01:26:58 +00:00
Jan Alexander Steffens (heftig)
02a6a794ec
rtmp2: Expose connection stats as property
Save the stats before we destroy the connection, so we can still
retrieve them afterwards.
2020-02-21 19:26:35 +01:00
Jan Alexander Steffens (heftig)
f1a9a3146a
rtmp2: Add gst_rtmp_connection_get_stats and _get_null_stats
The former uses a thread-safe way of getting statistics from the
connection without having to protect the fields with a lock.

The latter produces a zeroed statistics structure for use when no
connection exists.
2020-02-21 19:26:35 +01:00
Jan Alexander Steffens (heftig)
5d720eb59e
rtmp2: Count outgoing bytes and acked bytes
For statistics.
2020-02-21 19:26:33 +01:00
Jan Alexander Steffens (heftig)
0c344a7efb rtmp2sink: Add a property for the outgoing chunk size 2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
f7bb2cdeb7 rtmp2: Add gst_rtmp_connection_set_chunk_size 2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
63ec837824 rtmp2: Handle outgoing set chunk/window size properly
Apply outgoing sizes only after writing the chunk to the peer. This is
important particularly for the set chunk size and allows exposing it
without threading issues.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
a566461294 rtmp2: Chunk messages as buffers in loop thread
Move output chunking from gst_rtmp_connection_queue_message into
gst_rtmp_connection_start_write, which effectively moves it from the
streaming thread into the loop thread.

This allows us to handle the outgoing chunk-size message (which is
generated by changing the future chunk-size property) properly, which
could come from any other thread.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
9a13df9ba5 rtmp2: Consistently use GstBuffer for RTMP chunks 2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
b03780233e rtmp2: Add gst_rtmp_chunk_stream_serialize_all
Serializes an RTMP message into a series of chunks, all in one buffer.

Similar to what gst_rtmp_connection_queue_message does to serialize
into a GByteArray.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
cb7f0c4be7 rtmp2: Add gst_rtmp_output_stream_write_all_buffer_async
Similar to gst_rtmp_output_stream_write_all_bytes_async, but takes a
GstBuffer instead of a GBytes. It can also return the number of bytes
written, which might be lower in case of an error.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
286a3829b6 rtmp2: Improve handling incoming set chunk/window size
Reject out-of-spec sizes and warn about suspiciously small sizes.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
14fd7e0884 rtmp2: Lock self->lock before OBJECT_LOCK
OBJECT_LOCK is used to protect property access only. self->lock is
used to access the RtmpConnection, mostly between the streaming thread
and the loop thread.

To avoid deadlocks involving these two locks, we obey a lock order:
If both self->lock and OBJECT_LOCK are needed, self->lock must be locked
first. Clarify this.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
6583e00d50 rtmp2: Reject oversized messages
We only have 24 bits for the size, so reject anything larger.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
0044e7a1ba rtmp2: Count in_bytes_acked instead of in_bytes_unacked
This is nicer for statistics.
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
11a1de0053 rtmp2: rtmpconnection: Use more appropriate size types
- guint32 for chunk size and window size
- guint64 for running counters
2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
279e3c333c rtmp2: Add a g_return_val_if_fail 2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
03c3257f0f rtmp2: Replace explicit unref with g_main_context_invoke_full 2020-02-21 15:20:41 +00:00
Jan Alexander Steffens (heftig)
baad4fd91b rtmp2: rtmpconnection: Use GST_*_OBJECT logging
GstRtmpConnection isn't a GstObject with a name or path, but we still
get the GObject's type and address.
2020-02-21 15:20:41 +00:00
Marc Leeman
424c593871 rist: fix two minor memory leaks 2020-02-21 12:16:31 +01:00
Marc Leeman
6da6b6f3f0 rtpmanagerbad: fix two minor memory leaks 2020-02-21 12:16:28 +01:00
Marc Leeman
a710fbc12b rtpmanagerbad: reduce lock in rtpsink 2020-02-21 12:16:21 +01:00
Marc Leeman
61b062a12e rtpmanagerbad: documentation comment fix 2020-02-21 12:16:17 +01:00
Jan Schmidt
499be261cd webrtc: Configure transportsendbin latency internally
Add latency configuration logic to transportsendbin to
isolate it from the overall pipeline latency. That means that
it configures minimum latency internally based on the
latency query, and sends a latency event upstream that
matches.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1209
2020-02-21 13:42:05 +11:00
Seungha Yang
fbe7917a94 d3d11decoder: Add padding space on decoder output view when it's not aligned
Most H/W decoders have required alignment and dxva is also the case.
2020-02-20 17:32:42 +09:00
Seungha Yang
fe72bf6053 d3d11decoder: Register elements per GPU device with capability check
This implementation is similar to what we've done for nvcodec plugin.
Since supported resolution, profiles, and formats are device dependent ones,
single template caps cannot represent them, so this modification
will help autoplugging and fallback.

Note that the legacy gpu list and list of resolution to query were
taken from chromium's code.
2020-02-18 11:58:45 +00:00
Seungha Yang
13586bc77a d3d11device: Fix typo
s/vender/vendor
2020-02-18 11:58:45 +00:00
Seungha Yang
8ead80eecd d3d11device: Adjust debug level for when _new() fails
gst_d3d11_device_new might be used to enumerate device.
2020-02-18 11:58:45 +00:00
Matthew Waters
bd31caf0b0 vkswapper: keep a reference on the input buffer until present is finished
Otherwise, there may be a very small period of time where the buffer can
be freed while being presented.
2020-02-18 15:52:22 +11:00
Jan Schmidt
96a407334d webrtc: Merge ICE candidates to local descriptions
When emitting ICE candidates, also merge them to the local and
pending description so they show up in the SDP if those are
retrieved from the current-local-description and
pending-local-description properties.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/676
2020-02-17 14:23:56 +00:00