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* ccconverter: fix framerate caps negotiation from non-cdp to cdpMatthew Waters2021-07-021-3/+13
| | | | | | | | | We can only convert from non-cdp to cdp within the confines of valid cdp framerates. The existing caps negotiation code was allowing any framerate to convert to a cdp output which is incorrect and would hit an assertion later. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2372>
* webrtcbin: Add a test for setting codec preferences as part of ↵Sebastian Dröge2021-06-251-0/+102
| | | | | | "on-new-transceiver" when setting the remote offer Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
* webrtc: Use fail_unless_equals_string() for string assertionsSebastian Dröge2021-06-251-5/+5
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
* webrtcbin test: Fix race in new testOlivier Crête2021-06-241-0/+8
| | | | | | | Pull a buffer from a sink to make sure that the caps are already set before trying to update them. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2348>
* d3d11: Add support for GRAY and more YUV formatsSeungha Yang2021-06-231-72/+129
| | | | | | | | | | | | | | | | | | | | By this commit, following formats will be newly supported by d3d11 elements * Y444_{8, 12, 16}LE formats: Similar to other planar formats. Such Y444 variants are not supported by Direct3D11 natively, but we can simply map each plane by using R8 and/or R16 texture. * P012_LE: It is not different from P016_LE, but defining P012 and P016 separately for more explicit signalling. Note that DXVA uses P016 texture for 12bits encoded bitstreams. * GRAY: This format is required for some codecs (e.g., AV1) if monochrome is supported * 4:2:0 planar 12bits (I420_12LE) and 4:2:2 planar 8, 10, 12bits formats (Y42B, I422_10LE, and I422_12LE) Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2346>
* tests: msdkh264dec: Run test only if factory is available.Víctor Manuel Jáquez Leal2021-06-211-7/+9
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2336>
* wpe: Make wpesrc!video pad an always padThibault Saunier2021-06-081-13/+8
| | | | | | There should always be a `video` pad no matter what. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2273>
* d3d11: Add support for YV12 and NV21 formatsSeungha Yang2021-06-031-4/+4
| | | | | | Handle UV swapped 4:2:0 8bits formats Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2290>
* test: camerabin: Fix buffer size calculationNicolas Dufresne2021-06-021-1/+5
| | | | | | | | | | | We were assunming that GStreamer size for RGB (24bit packed) data was width x height x 3, but GStreamer defaults to specific alignment. Use GstVideoInfo API in order to obtain the buffer size. This fixes failure seen when trying to merge: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/998 which make us negoaite 1x1 instead of 16x16 in this test. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2297>
* Use gst_buffer_new_memdup()Tim-Philipp Müller2021-05-242-12/+19
| | | | | | | | Update for function rename in core. https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/827 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2281>
* Use new gst_buffer_new_copy()Tim-Philipp Müller2021-05-232-24/+13
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2279>
* wpe: Implement audio supportPhilippe Normand2021-05-193-0/+177
| | | | | | | The wpesrc bin now exposes "sometimes" audio src pads, one for every PCM audio stream created by WPEWebKit. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252>
* examples: va: Update the VA examples because of the new va lib.He Junyan2021-05-183-9/+13
| | | | | | | Because we introduce the new va lib, the va examples need to include new header files and add more library linkage. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2196>
* webrtc test: Print content of error GstMessageOlivier Crête2021-05-131-0/+8
| | | | | | Makes it easier to interpret the result of the CI! Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
* webrtcbin tests: Add test for intersection src pad capsOlivier Crête2021-05-131-0/+113
| | | | | | | This checks that the codec preferences are intersected also with what the src pad can handle. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
* webrtc test: Add explicit test clockOlivier Crête2021-05-131-0/+8
| | | | | | This way the test clock is not linked to the multiple harnesses Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
* webrtc test: Add test for codec preferences negotiationOlivier Crête2021-05-131-2/+77
| | | | | | | Validate that it does the intersection with the caps from the sink pad and rejects the offer creation otherwise. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
* webrtcbin: Implement caps queries on sinkpad based on codec preferencesOlivier Crête2021-05-131-0/+38
| | | | | | Also includes a unit test. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
* webrtcbin tests: Use properties to access the inside of the transceiver objectOlivier Crête2021-05-131-7/+11
| | | | | | This will allow hiding the insides from unsafe application access. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
* webrtcbin: Add unit test for closing of data channelsJohan Sternerup2021-05-121-0/+178
| | | | | | | | Add test for verifying that the data channel "close" action signal triggers an SCTP_RESET_STREAMS request that is propagated to the other side and eventually leads to both sides closing properly. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
* Use gst_element_request_pad_simple...François Laignel2021-05-057-15/+15
| | | | | | Instead of the deprecated gst_element_get_request_pad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2180>
* tests/netsim: Set src caps before creating buffersDoug Nazar2021-04-211-1/+3
| | | | | | | GstHarness requires the source pad caps to be set before buffer allocations. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2179>
* d3d11: Add support for BGRx and RGBx formatsSeungha Yang2021-04-211-4/+4
| | | | | | | For such formats, we can re-use existing BGRA/RGBA implementations but ignoring alpha channel Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2174>
* webrtcbin test: Don't fail if data channel is createdOlivier Crête2021-04-211-0/+2
| | | | | | In tests that voluntarily create a data channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2168>
* tests/avtp: increase timeout of test_depayloader_fragmented_bigDoug Nazar2021-04-141-3/+6
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2160>
* check: fix dash_mpdparser_check_mpd_client_set_methods test.Doug Nazar2021-04-141-2/+2
| | | | | | | Setting guint64 valist properties without type specifier fails on 32bit archs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2161>
* line21enc: fix remove-caption-meta property testDoug Nazar2021-04-131-4/+43
| | | | | | | It's possible for the same address to be allocated to the decoded metadata. Switch test to actual detect if it was removed. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2159>
* tests: fix shm test deadlockDoug Nazar2021-04-131-2/+2
| | | | | | | | Stopping the consumer first would occasionally allow the producer to fill the shm segment causing it to block in send() and unable to be stopped. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2158>
* check: Fix test dash_mpdparser_xlink_periodDoug Nazar2021-04-131-1/+1
| | | | | | | | | | | Test used http://404/ERROR/XML.period as an invalid url. Curl now interprets that as an 32bit int and tries an actual connect which timesout. Use .invalid as an IANA reserved domain for invalid DNS. curl -v http://404/ERROR/XML.period * Trying 0.0.1.148:80... Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2157>
* webrtcbin test: Wait for set-local-desc & set-remote-desc to continueOlivier Crête2021-04-121-10/+12
| | | | | | | | To avoid racing betwen the SDPs being set and the next step of the test, let's wait for setting the SDP both locally and remotely to succeed. of the test Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* webrtcbin test: Add for the case where a second m-line is renegotiatedOlivier Crête2021-04-121-9/+102
| | | | | | | | | This is for the case where there answerer forces a specific media type for a m-line, but he origin offer only has the other media type. In this case, we will create a second transceiver on receiving the offer and add the desired media type using renegotiation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* webrtc test: Verify that forcing different kinds on peers failsOlivier Crête2021-04-121-0/+69
| | | | | | | If the offer contains an audio kind and a video kind, forcing them both at m-line zero will fail. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* webrtc tests: Verify that create-offer is rejected when neededOlivier Crête2021-04-121-0/+80
| | | | | | | Verify that it gets rejected if a m-line at index 1 is requested but there is no m-line 0. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* webrtcbin test: Add test for various cases where get_request_pad is meant to ↵Olivier Crête2021-04-121-0/+93
| | | | | | | | fail This should ensure that the recently added code works. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* webrtcbin: Try to match an existing transceiver on pad requestOlivier Crête2021-04-121-0/+1
| | | | | | This should avoid creating extra transceivers that are duplicated. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* webrtcbin test: Test adding a stream to a stream+datachannelOlivier Crête2021-04-121-1/+71
| | | | | | | This use-case was previously broken by the expectation of having a 1-1 match between the pad id and the m-line index Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
* h2645parser: Catch overflows in AVC/HEVC NAL unit length calculationsSebastian Dröge2021-03-242-0/+120
| | | | | | | | | | | | | | | | | | | Offset and size are stored as 32 bit guint and might overflow when adding the nal_length_size, so let's avoid that. For the size this would happen if the AVC/HEVC NAL unit size happens to be stored in 4 bytes and is 4294967292 or higher, which is likely corrupted data anyway. For the offset this is something for the caller of these functions to take care of but is unlikely to happen as it would require parsing on a >4GB buffer. Allowing these overflows causes all kinds of follow-up bugs in the h2645parse elements, ranging from infinite loops and memory leaks to potential memory corruptions. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2103>
* gst: don't use volatile to mean atomicMatthew Waters2021-03-221-2/+2
| | | | | | | | | | | | volatile is not sufficient to provide atomic guarantees and real atomics should be used instead. GCC 11 has started warning about using volatile with atomic operations. https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719 Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2098>
* tests/webrtc: check for more sdp things across the boardMatthew Waters2021-03-191-245/+501
| | | | | | | | | | | e.g. - test for a=setup:$val and direction attributes in all tests - test number of media sections - test number of formats in each m= section (for audio/video) - test no duplicate formats Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2093>
* examples: Add an mpegtsmux example of prog-map usage.Jan Schmidt2021-03-182-1/+100
| | | | | | | | Add an example of how to construct the prog-map structure for the MPEG-TS muxers and assign streams to programs, and set PCR and PMT PIDs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2039>
* cccombiner: implement schedulingMathieu Duponchelle2021-03-171-92/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Prior to that, cccombiner's behaviour was essentially that of a funnel: it strictly looked at input timestamps to associate together video and caption buffers. This patch instead exposes a "schedule" property, with a default of TRUE, to control whether caption buffers should be smoothly scheduled, in order to have exactly one per output video buffer. This can involve rewriting input captions, for example when the input is CDP sequence counters are rewritten, time codes are dropped and potentially re-injected if the input video frame had a time code meta. Caption buffers may also get split up in order to assign captions to the correct field when the input is interlaced. This can also imply that the input will drift from synchronization, when there isn't enough padding in the input stream to catch up. In that case the element will start dropping old caption buffers once the number of buffers in its internal queue reaches a certain limit (configurable). The property is exposed so that existing users of cccombiner can revert back to the original behaviour, but should eventually be removed, as that behaviour was simply inadequate. This commit also disallows changing the input caption type, as this would needlessly complicate implementation, and removes the corresponding test. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2076>
* va: example: multiple-vpp: test sharpen with dynamic controllerVíctor Manuel Jáquez Leal2021-03-162-4/+52
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2082>
* va: example: multiple-vpp: test direction changeVíctor Manuel Jáquez Leal2021-03-121-1/+57
| | | | | | | | | | If the driver supports it (iHD, so far) and the parameter -d is set, the direction of the video will be changed randomly. In the code you can select, at compilation time, if the direction change is done by element's property or by pipeline events. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2074>
* zxing: update to support version 1.1.1Stéphane Cerveau2021-03-121-18/+15
| | | | | | | | Support new API in 1.1.1 Update the supported input video format. Update tests to use parse_launch Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2037>
* play: tests: Switch user-agent test to a real HTTP serverPhilippe Normand2021-03-092-7/+175
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2061>
* play: tests: Refactor to use new Message bus APIPhilippe Normand2021-03-092-599/+524
| | | | | | | | | Instead of relying on an extra GMainLoop, the messages are poped from the player bus and handled synchronously. This should avoid flaky behaviors. Fixes #608 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2061>
* webrtc: don't generate duplicate rtx payloads when bundle-policy is setMatthew Waters2021-03-091-0/+94
| | | | | | | | | | | | | | | | It was possible to generate a SDP that had an RTX payload type that matched one of the media payload types when providing caps via codec_preferences without any sink pads. Fixes m=video 9 UDP/TLS/RTP/SAVPF 96 ... a=rtpmap:96 VP8/90000 a=rtcp-fb:96 nack pli a=fmtp:96 apt=96 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
* tests: Add negotiation tests for the interlace elementsVivia Nikolaidou2021-03-082-0/+217
| | | | | | Many complicated cases exist. Would be good to have some checks. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2062>
* webrtc ice: Add 'min/max-rtp-port' props for setting RTP port rangeIlya Kreymer2021-03-011-1/+74
| | | | | | | | | default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored) add 'gathering_started' flag to avoid changing ports after gathering has started validity checks: min port <= max port enforced, error thrown otherwise include tests to ensure port range is being utilized (by @hhardy) Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
* va: vpp: implement GstColorBalance interfaceVíctor Manuel Jáquez Leal2021-02-252-5/+37
| | | | | | And modify multiple-vpp example to use it with -r parameter. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2033>
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