mirrors/gst-plugins-rs
mirrors/gst-plugins-rs
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git synced 2024-12-23 10:30:40 +00:00
Code Issues Projects Releases Wiki Activity
2628 commits 14 branches 101 tags 30 MiB
Commit graph

2628 commits

This branch
This branch
All branches
Author SHA1 Message Date
Arun Raghavan
b05c21680d 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/1219>
 2023-05-23 09:27:00 -04:00
Thibault Saunier
04e35e86d6 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/1214>
 2023-05-18 18:23:56 +03:00
Thibault Saunier
e73d7082a6 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/1214>
 2023-05-18 18:23:56 +03:00
Seungha Yang
3406e604cd 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/1213>
 2023-05-17 23:49:09 +09:00
Guillaume Desmottes
7ebf2d7a4f 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/1208>
 2023-05-15 16:13:20 +02:00
Sanchayan Maity
067d47f0ec 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/1211>
 2023-05-15 12:19:09 +05:30
François Laignel
7ba0073052 use Pad builders for optional name definition 
Also, apply auto-naming in the following cases

* When building from a non wildcard-named template, the name of the template is
  automatically assigned to the Pad. User can override with a specific name by
  calling `name()` on the `PadBuilder`.
* When building with a target and no name was provided via the above, the
  GhostPad is named after the target.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/448
Auto-naming discussion: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1255#note_1891181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
 2023-05-12 12:55:31 +02:00
François Laignel
8e93d294e5 Update to argumentless {Bin,Pipeline}::new 
See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/449

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
 2023-05-12 12:55:31 +02:00
Sebastian Dröge
32d59c31d8 fmp4: examples: Update to dash-mpd 0.9 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1210>
 2023-05-12 09:45:40 +03:00
Seungha Yang
773fcd0780 transcriberbin: Add "language-code" property 
Proxy the child transcriber element's property so that transcriberbin
can apply the property with required state management

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1205>
 2023-05-10 19:12:01 +00:00
Antonio Kevo
cf21bfabf2 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/1207>
 2023-05-10 16:09:25 +02:00
Sebastian Dröge
8b2b12e767 Update CHANGELOG.md for 0.10.7 2023-05-09 20:48:43 +03:00
François Laignel
680d5221db net/webrtc: src: add signal "request-encoded-filter" 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
 2023-05-09 12:02:15 +02:00
François Laignel
092ae1fec8 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/1202>
 2023-05-09 11:17:32 +02:00
François Laignel
dc5ddd3022 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/1202>
 2023-05-09 10:26:11 +02:00
François Laignel
eca269cbf2 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/1202>
 2023-05-09 10:26:07 +02:00
Sebastian Dröge
de1c8ece43 deny: Update 2023-05-08 18:44:12 +03:00
Sebastian Dröge
cb5b527d74 Update to AWS SDK 0.27 and async-tungstenite 0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1199>
 2023-05-02 15:30:00 +03:00
Guillaume Desmottes
76c8279101 fmp4: define minBufferTime in example mpd 
Required to validate the manifest with https://beta.conformance.dashif.org/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1196>
 2023-04-27 14:16:40 +02:00
Sebastian Dröge
05ee55d617 fmp4: Update example to dash-mpd 0.8 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1193>
 2023-04-25 08:54:48 +03:00
Nick Steel
3d01c9b363 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/1192>
 2023-04-24 15:29:22 +01:00
Jan Beich
8d6751c88d 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/1191>
 2023-04-22 12:58:29 +00:00
Lily Foster
8e4fd2c167 meson: support rust cross-compiling with cargo wrapper 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1188>
 2023-04-21 16:55:37 +00:00
Sebastian Dröge
41ba4b2bc3 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
 2023-04-21 10:48:10 +00:00
Sebastian Dröge
5451035215 Update async-tungstenite and AWS SDK dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
 2023-04-21 10:48:10 +00:00
John King
2bd2e501d9 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/1183>
 2023-04-21 09:12:47 +02:00
Sebastian Dröge
cc3646640e Fix a couple of new Rust 1.69 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1186>
 2023-04-20 16:47:45 +03:00
Lily Foster
760e97c7e7 meson: avoid passing the --features flag to wrapper when empty 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1184>
 2023-04-18 16:26:11 -04:00
Edward Hervey
721d17e181 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/1181>
 2023-04-15 19:35:27 +02:00
Mathieu Duponchelle
dbdb9bc164 webrtcsink: fix navigation data channel 
At some point, presumably recently, the data channel stopped being
requested in Ready, making webrtcbin refuse to create it.

There was quite a lot of churn recently so I couldn't pinpoint the
breaking commit easily.

Fix by simply restoring the correct behavior of requesting the channel
after going to the Ready state

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/341

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1180>
 2023-04-14 14:26:22 +02:00
Sebastian Dröge
47159ad3c2 Make sure to keep around and drop bus watches after usage in all the examples 2023-04-14 12:46:43 +03:00
Arun Raghavan
aabfb61834 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/1174>
 2023-04-13 15:58:49 +00:00
Mathieu Duponchelle
f1fd8d84c3 webrtc: extract a BaseWebRTCSink 
For documentation purposes, AwsKVSWebRTCSink should not inherit from
another element.

+ Mark base class as plugin API and update plugin cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
 2023-04-13 15:06:59 +00:00
Guillaume Desmottes
367b98bfcb fmp4: dash_vod example: reformat 
Not sure why rustfmt updated those because of my previous change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1177>
 2023-04-13 09:50:15 +02:00
Guillaume Desmottes
371ac83169 fmp4: dash_vod example: use dash-mpd to generate the manifest 
Maybe a bit overkill for such simple example but more exemplary for
actual applications.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1177>
 2023-04-13 09:50:11 +02:00
Loïc Le Page
dba91bceca webrtc: fix documentation after signaller interface changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1175>
 2023-04-12 20:19:22 +02:00
Thibault Saunier
8f2273328b webrtcsrc: Return bool en 'end-session' as required 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1172>
 2023-04-12 12:17:56 +00:00
Sebastian Dröge
5dcdf645d6 net: ndi: Update to libloading 0.8 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1173>
 2023-04-12 11:03:05 +03:00
Sebastian Dröge
011f0d5fee deny: Update for miniz_oxide dependency duplication 2023-04-11 11:14:08 +03:00
Mathieu Duponchelle
355f925954 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/1166>
 2023-04-11 09:26:24 +10:00
Mathieu Duponchelle
f366c20869 awstranscriber: fix what we send over for translations 
Prior to this commit, we were sending over words concatenated together
with no separators, for instance "Idon'twanttobeanemperor".

The translation service seems clever enough to translate the contents
anyway, but there is no reason to make its task harder than necessary,
and it didn't re-add separators when the target language was the same as
the source language, which resulted in less than ideal output.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
 2023-04-10 20:47:12 +00:00
Mathieu Duponchelle
408fd2030c awstranscriber: slight debug improvement 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
 2023-04-10 20:47:12 +00:00
Mathieu Duponchelle
4fcbb6ae61 textwrap: add some logs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
 2023-04-10 20:47:12 +00:00
Guillaume Desmottes
3eca8c60e3 ci: check for typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
 2023-04-10 13:35:32 +02:00
Guillaume Desmottes
403004a85e fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
 2023-04-10 13:35:32 +02:00
Mathieu Duponchelle
a455819871 webrtcsink: fix tracking of signaller state 
For the signaller to get stopped, we need to remember that we started it
in the first place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
3368f55a88 webrtcsink: don't return value from error closure 
the signal doesn't expect a return value, which meant we were panicking
as soon as the signaller tried to report an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
58c8c0edc7 webrtc: signaller iface: fix session-ended vs end-session confusion 
Session ending is bidirectional: the signaller can tell the sink that a
session was ended, and the sink can tell the signaller to end a session.

As such, two signals are needed, before this patch the second case was
not working as in essence the sink was telling itself that a session was
ended, and obviously failing to even find it when trying to end it again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Tim-Philipp Müller
0b5cf4e5fd ci: add check for symlinks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
 2023-04-08 23:20:54 +01:00
Tim-Philipp Müller
7c30430320 webrtc-api: replace LICENSE file symlink with copy 
As in !1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
 2023-04-08 17:22:37 +01:00