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Author SHA1 Message Date
Thibault Saunier c1d6094bc4 webrtcsrc: Do not pass raw caps in the transceiver 
That was not making sense.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 18:25:44 +03:00
Thibault Saunier 0e447a9316 webrtcsrc: Fix caps used when creating transceiver 
We used to pass all media keys and attributes to the caps which
incorrect. Instead we should be using only the keys from the map
and remove all information related to rtcp which is irrelevant
to create the transceiver.

This also simplifies the code.

New caps look like:

```
Caps(
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 96,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP8",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 102,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 104,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 106,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 108,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 127,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 39,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 98,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "0",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 100,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "2",
    },
)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 18:25:44 +03:00
Seungha Yang 1dc96548c4 fallbacksrc: Don't apply fallback-audio-caps to the main audio stream 
Intended behavior is configuring audio convert/resample elements
only for the fallback stream and also fallback-audio-caps is set.
Video and image stream are doing it as intended already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 17:55:25 +03:00
Guillaume Desmottes 403ac0c188 fallbackswitch: document the pad priority ordering 
I just wasted lots of time trying to figure out why my higher priority
pad wasn't used...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 17:55:16 +03:00
Sanchayan Maity 48cc8570e4 videofx: border: Do not advertise I420 for non-zero border radius 
In certain cases, roundedcorners would negotiate to I420 even when user
supplied a non-zero border radius.

For example, the below pipeline leads to I420 being negotiated even
though a non-zero border radius was given. Ideally, this pipeline
should have failed at the negotiation stage.

```bash
gst-launch-1.0 -v \
   videotestsrc num-buffers=1000 pattern=white ! \
   video/x-raw,width=320,height=180 ! \
   roundedcorners border-radius-px=10 ! videobox border-alpha=0 top=-10 left=-10 right=-10 bottom=-10 fill=yellow ! \
   compositor name=comp sink_0::xpos=960   sink_0::ypos=0  sink_0::width=320 sink_0::height=180 sink_0::alpha=1.0 sink_1::xpos=960 sink_1::ypos=180  sink_1::width=320 sink_1::height=180 sink_1::alpha=1.0  \
   sink_2::xpos=960 sink_2::ypos=360  sink_2::width=320 sink_2::height=180 sink_2::alpha=1.0 sink_3::xpos=0 sink_3::ypos=0  sink_3::width=960 sink_3::height=720 sink_3::alpha=1.0 ! \
   video/x-raw,width=1280,height=720! x264enc ! mp4mux ! filesink location=test.mp4 \
   videotestsrc num-buffers=1000 pattern=red ! \
   video/x-raw,width=320,height=180 ! roundedcorners border-radius-px=10 ! comp. \
      videotestsrc num-buffers=1000 pattern=blue ! \
   video/x-raw,width=320,height=180 ! roundedcorners border-radius-px=10 ! comp. \
      videotestsrc num-buffers=1000 pattern=green ! \
   video/x-raw,width=960,height=720 ! roundedcorners border-radius-px=10 ! comp.
```

If border radius is non-zero, we should not really allow negotiation
to select I420. Fix this by returning only A420 for border-radius > 0
in `transform_caps` instead of returning both like earlier.

Another example of a simpler pipeline like below which would earlier work

```bash
gst-launch-1.0 videotestsrc pattern=red ! videoconvert ! video/x-raw,width=1923,height=1087,format=I420 ! roundedcorners border-radius-px=40 ! video/x-raw,format=I420 ! videoconvert ! gtksink
```

now fails with

```bash
WARNING: erroneous pipeline: could not link roundedcorners0 to videoconvert1, roundedcorners0 can't handle caps video/x-raw, format=(string)I420
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 17:55:08 +03:00
Antonio Kevo 19ffa05bb4 fmp4: Use updated start_pts when checking stream filled 
After calculating the earliest pts, the fragment_start_pts and
chunk_start_pts in State are updated. However, when checking if the
stream is filled, the previous start_pts (set to None) is used instead.
This means that chunk_filled and fragment_filled will be false the first
time aggregate() is called, assuming timeout is false, all_eos is false,
and the sinkpad is not EOS. This requires aggregate() having to be
called a second time before the first fragment is sent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 17:55:02 +03:00
Sebastian Dröge 2b7f87b0b7 Update Cargo.lock 2023-05-09 20:47:56 +03:00
Sebastian Dröge 573307b32e Update version to 0.10.7 2023-05-09 20:44:27 +03:00
Sebastian Dröge d750e801bc Update CHANGELOG.md for 0.10.7 2023-05-09 20:44:27 +03:00
Sebastian Dröge 6e40c73a95 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1203>
 2023-05-09 16:26:41 +03:00
Sebastian Dröge fb0aca9586 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1203>
 2023-05-09 16:00:00 +03:00
Sebastian Dröge 41ea793fd8 Update to AWS SDK 0.27 and async-tungstenite 0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1203>
 2023-05-09 16:00:00 +03:00
Nick Steel fbd6c89612 spotify: check cached creds username before use 
If a username was specified, only use cached credentials that match
that username.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1203>
 2023-05-09 15:57:47 +03:00
Jan Beich 9dc92da832 gtk4: unbreak wayland, x11egl, x11glx features on non-Linux 
As the features are non-default leave the responsibility to filter by
platform to consumers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1203>
 2023-05-09 15:56:49 +03:00
François Laignel 91fe56468a net/webrtc: src: add signal "request-encoded-filter" 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-05-02 15:22:43 +02:00
François Laignel 2b6a908911 net/webrtc: sink: add signal "request-encoded-filter" 
The new "request-encoded-filter" signal is emitted when the encoder and related
elements are added to the pipeline. When defined, the element returned by the
signal is inserted between the encoder and the payloader.

The transformation can be reverted using the [insertable streams API] on the
receiver side.

[insertable streams API]: https://developer.mozilla.org/en-US/docs/Web/API/Insertable_Streams_for_MediaStreamTrack_API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-05-02 15:22:43 +02:00
François Laignel 0e6b9df932 net/webrtc: sink: abort stats collection before stopping the Signaller 
In some rare cases, the webrtc-test entered a deadlock while executing
`WebRTCSink::unprepare`. Attaching gdb to a blocked instance showed:

* `gstrswebrtc::signaller:👿:Signaller::stop()` parked, waiting for a
  `Condvar` in `Signaller::stop()`. This was most likely awaiting for the
  receive task to complete while it was locked in `element.end_session()`.
  This code path is triggered from `unprepare` with the `State` `Mutex` locked.
* `webrtcsink:👿:WebRtcSink::process_stats` waiting for a contended `Mutex`,
  which is also the `State` `Mutex`. This prevented completion of the signal
  `gst_webrtc_bin_get_stats`.

This commit aborts the task in charge of periodically collecting stats and
ensures any remaining iteration completes before requesting the Signaller to
stop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-05-02 15:22:43 +02:00
François Laignel 2cb1fd7fc1 net/webrtc: src: don't set stun-server on webrtcbin when our property is None 
... otherwise an error occurs about the stun-server address being an empty
string which doesn't comply with the expected address format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-04-30 13:04:26 +02:00
Sebastian Dröge b27d4c2550 ci: Follow 0.20 branch of the gstreamer-rs repo for the CI template 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:31:24 +03:00
Sebastian Dröge 96f355ba96 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Lily Foster 7a0f6272ce meson: support rust cross-compiling with cargo wrapper 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Sebastian Dröge d7ec86defe deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Sebastian Dröge a29769789f Update async-tungstenite and AWS SDK dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Sebastian Dröge 1db07fe451 aws: Update to AWS SDK 0.55/0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
John King 0c83f82b7f spotify: fix credentials cache 
Cache Spotify response instead of username and password.
This should resolve frequent "New login to Spotify" emails.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Guillaume Desmottes 1daac1e301 spotify: move Settings to common module 
Will be used to implement the lyrics source element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Guillaume Desmottes 60e89fd46d spotifyaudiosrc: use Settings Default to define default props 
Makes it easier to change one property's default value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Sebastian Dröge 5c580709ee Fix a couple of new Rust 1.69 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Lily Foster b98aa4e17c meson: avoid passing the --features flag to wrapper when empty 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:56:44 +03:00
Edward Hervey a76330e76c rtpgccbwe: Don't process empty lists 
The structure parsing could result in an empty vector. Don't do any processing
since the loss code assumes it's non-empty for average estimates which would
result in weird/invalid results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:56:37 +03:00
Arun Raghavan 1f26b9e336 ffv1dec: Drop rank for now 
We'll keep the rank lower than avdec_ffv1, at least until we're
comfortable with support for the entire range of possible inputs working
well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:56:27 +03:00
Sebastian Dröge 33be56bd26 net: ndi: Update to libloading 0.8 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:56:13 +03:00
Sebastian Dröge 7726008d52 deny: Update for miniz_oxide dependency duplication 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:56:08 +03:00
Mathieu Duponchelle 8a72ec563e tttocea608: specify raw 608 field 
The element can only output field=0 raw 608 data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:55:46 +03:00
Mathieu Duponchelle 701f53b9fe textwrap: add some logs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 11:55:13 +03:00
François Laignel d2db786136 net/webrtc: backport the serial-sorted WebRtcSink pad request 
This is a partial backport of [#58439204] to get predictable track order.
With this commit, we are sure the `mid`s sequence in the Sdp offer will reflect
the order by which the `webrtcsink` pads were requested.

[#58439204]: 584392049c
 2023-04-20 15:09:43 +02:00
Sebastian Dröge 88548f41b6 Update Cargo.lock 2023-04-06 11:24:25 +03:00
Sebastian Dröge fdd9b3f638 Update CHANGELOG.md for 0.10.6 2023-04-06 11:24:25 +03:00
Sebastian Dröge 48ffd4eb49 Update versions to 0.10.6 2023-04-06 11:24:25 +03:00
Sebastian Dröge df471188dc Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Sebastian Dröge 0d8a5245b0 ndisrc: Fix copying of raw video frames with different NDI/GStreamer strides 
And also don't copy each line twice for single-plane formats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Sebastian Dröge 2e2ec4efdc deny: Update for older versions of the windows bindings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Tim-Philipp Müller 198477e63b git: replace LICENSE file symlinks with copies 
Git will de-duplicate the contents for us anyway, and
symlinks can cause problems with some versions of git
and also on Windows.

https://github.com/mesonbuild/meson/issues/11646
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4326

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Sebastian Dröge b2a0591676 fallbacksrc: Don't check caps when linking to the fallbackswitch 
Downstream might have different caps requirements and linking might
fail. Instead of having linking fail, give upstream an opportunity to
reconfigure and otherwise have a normal negotiation error during data
flow.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/334

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Guillaume Desmottes 329956de87 uriplaylistbin: example: add queues 
Prevent pipeline starvation with some media such as
https://assets.onestream.live/studio/Videos/1080p/osl-interval-1080p-8.mp4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Vivia Nikolaidou edc366f612 livesync: Actually assume zero upstream latency when query fails 
The code said "assuming zero" but left latency at None instead of
Some(0), failing to unwrap the value later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Mathieu Duponchelle 525e3afe70 webrtcsink: fix calculation of fec_ratio with multiple encoders 
In this context, the bitrate variable is for all encoders, but the
max_bitrate field is per encoder. To calculate a proper FEC ratio, we
need to scale max_bitrate to the number of encoders.

+ Also clamp the fec-percentage that we set on the transceiver for extra
  safety

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
Sebastian Dröge 6f63abe653 deny: Update for duplicated old dependencies in dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:40 +00:00
David Revay 5bb65d3e33 chore(webrtcsink): fix max-bitrate blurb and nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:39 +00:00
Vivia Nikolaidou f15fd82f83 webrtcsink: Add ice-transport-policy option 
Can be used to force relay ICE candidates, ensuring TURN server is used.
Proxy to the corresponding setting in webrtcbin,

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:39 +00:00