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Author SHA1 Message Date
Sebastian Dröge
1e6fb1c1a4 Update CHANGELOG.md for 0.10.9 2023-06-19 20:42:58 +03:00
Sebastian Dröge
ca54e26edc deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:56:41 +03:00
Sebastian Dröge
b0a3939267 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:55:22 +03:00
Sebastian Dröge
7829a07629 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:55:07 +03:00
Sebastian Dröge
12ca3808a0 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:54:43 +03:00
Mathieu Duponchelle
927c3e9bdf webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:48:08 +03:00
François Laignel
f3ae457cfa mp4, fmp4: fix byte order for opus extension 
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.

In `write_dops`, `to_le_bytes` variants were used.

Related to [2].

[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2
[2] https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:46:31 +03:00
Mathieu Duponchelle
dbd8946608 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:46:23 +03:00
Seungha Yang
719455815f mccparse: Map timecode to PTS directly without offset 
Assumes that caption stream's timeline starts from zero,
and maps timecode time_since_daily_jam() to PTS directly without
subtracting the first seen timecode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:45:01 +03:00
Guillaume Desmottes
638ffc3c7c fallbackswitch: add 'stop-on-eos' property 
Fix the following use case:
- main input of fallbackswitch is finite (a media file)
- fallback input is infinite (videotestsrc)
- main input is done and send eos, which is propagated downstream
- fallbackswitch switches to fallback, sending STREAM_START which reset
  EOS downstream (aggregator does that)
- fallback input keeps pushing buffers forever.

Solve it by adding a 'stop-on-eos' property so fallbackswitch stops
pushing property once the main input is eos.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:44:16 +03:00
Guillaume Desmottes
0998f9a303 fallbackswitch: remove unused SinkState::eos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:44:08 +03:00
Guillaume Desmottes
06c5d8766d fallbackswitch: log when handling events 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:44:02 +03:00
Sebastian Dröge
aa799bc26c webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:43:36 +03:00
Sebastian Dröge
bea00c7413 Use MPL as license specifier for plugins only requiring GStreamer < 1.20 
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:42:12 +03:00
Sebastian Dröge
47a213b322 Update Cargo.lock 2023-06-07 01:02:13 +03:00
Sebastian Dröge
d0d97bfc03 Update CHANGELOG.md for 0.10.8 2023-06-07 01:01:28 +03:00
Sebastian Dröge
361152d884 Update versions to 0.10.8 2023-06-07 00:54:32 +03:00
Mathieu Duponchelle
1edf4a144e net/aws/transcriber: track discont offset in input stream 
and add it up to subsequent transcripts.

This ensures synchronization is maintained even after the input stream
experiences a discontinuity and a gap in its timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:16:55 +02:00
Sebastian Dröge
5bbd7002a7 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:49:45 +03:00
Guillaume Desmottes
14168930cd uriplaylistbin: use thiserror 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:45:35 +03:00
Guillaume Desmottes
52b30a37ed uriplaylistbin: example: display iterations 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:45:28 +03:00
Guillaume Desmottes
7e9cb37892 uriplaylistbin: example display when leaving because of eos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:45:21 +03:00
Guillaume Desmottes
febc28863e uriplaylistbin: prevent deadlock when notifying property changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:45:15 +03:00
Sebastian Dröge
245990a078 fmp4mux: Don't wait for more data if a stream has no GOP starting before fragment end 
Simply don't output anything for this stream and only include it in the
future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:43:56 +03:00
Sebastian Dröge
c9d2ba306b fmp4mux: Consider a stream filled if the earliest GOP starts after the current chunk 
There's not going to be any buffer to output for this stream in the
current chunk.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:43:50 +03:00
Sebastian Dröge
ea202411f9 Fix a couple of trivial clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:43:39 +03:00
Edward Hervey
18773a9df1 rtpgccbwe: Improve packet handling 
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:43:33 +03:00
Sebastian Dröge
e8e247d1ed net: Update to AWS SDK 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:41:20 +03:00
Sebastian Dröge
70f92ddbf7 whipsink: Request pads with webrtcbin's pad templates and not our own 
It's invalid to request pads with a pad template that is not part of the
element, and results in a critical warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:40:01 +03:00
Mathieu Duponchelle
da51c3a58b webrtcsink: further refactor connection to stats signals 
- Stop passing webrtcbin around without using it

- Stop using glib::closure as clippy complains when using a unit type
  default-return

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:38:19 +03:00
Mathieu Duponchelle
2bb0a666a8 webrtcsink: fix stats_sigid logic 
First off, we just created the session, we know stats_sigid is None
at this point.

Second, don't first assign the result of connecting on-new-ssrc to the
field, then the result of connection twcc-stats, that simply doesn't
make sense.

Finally, actually check that stats_sigid *is* None before connecting
twcc-stats, as I understand it this must have been the original
intention / behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:36:51 +03:00
Mathieu Duponchelle
77f003f699 webrtcsink: don't panic in twcc-stats callback 
If webrtcbin was disposed of at this point, simply return

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/345
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:36:44 +03:00
François Laignel
b8b718fe62 webrtcsink: remove unneeded mut 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:34:55 +03:00
Arun Raghavan
b72a0a2177 Revert "fmp4: Return a running time in get_next_time()" 
This reverts commit 04bb7b4db0.

As Sebastian points out, the chunk PTS is already in running time, so
this was wrong from the start.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/363
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:34:30 +03:00
Sebastian Dröge
e4702d1378 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1215>
 2023-05-18 18:25:44 +03:00
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