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examples: webrtc: Update dependencies in Rust examples

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6078>
This commit is contained in:
Sebastian Dröge 2024-02-09 10:44:27 +02:00 committed by GStreamer Marge Bot
parent 6bb47a125f
commit 0871d1edc4
10 changed files with 663 additions and 889 deletions
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508
subprojects/gst-examples/webrtc/janus/rust/Cargo.lock generated
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14
subprojects/gst-examples/webrtc/janus/rust/Cargo.toml
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@ -11,15 +11,15 @@ clap = { version = "4", features = ["derive"] }
anyhow = "1" anyhow = "1"
url = "2" url = "2"
rand = "0.8" rand = "0.8"
async-tungstenite = { version = "0.24", features = ["gio-runtime"] } async-tungstenite = { version = "0.25", features = ["gio-runtime"] }
gst = { package = "gstreamer", version = "0.21" } gst = { package = "gstreamer", version = "0.22" }
gst-webrtc = { package = "gstreamer-webrtc", version = "0.21" } gst-webrtc = { package = "gstreamer-webrtc", version = "0.22" }
gst-sdp = { package = "gstreamer-sdp", version = "0.21" } gst-sdp = { package = "gstreamer-sdp", version = "0.22" }
serde = "1" serde = "1"
serde_derive = "1" serde_derive = "1"
serde_json = "1.0.53" serde_json = "1.0.53"
http = "1.0" http = "1.0"
glib = "0.18" glib = "0.19"
gio = "0.18" gio = "0.19"
log = "0.4.8" log = "0.4.8"
env_logger = "0.10" env_logger = "0.11"

2
subprojects/gst-examples/webrtc/janus/rust/src/janus.rs
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@ -489,7 +489,7 @@ impl JanusGateway {
); );
let encode_bin = let encode_bin =
gst::parse_bin_from_description_with_name(bin_description, false, "encode-bin")?; gst::parse::bin_from_description_with_name(bin_description, false, "encode-bin")?;
pipeline.add(&encode_bin).expect("Failed to add encode bin"); pipeline.add(&encode_bin).expect("Failed to add encode bin");

2
subprojects/gst-examples/webrtc/janus/rust/src/main.rs
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@ -62,7 +62,7 @@ impl App {
} }
fn new() -> Result<Self, anyhow::Error> { fn new() -> Result<Self, anyhow::Error> {
let pipeline = gst::parse_launch( let pipeline = gst::parse::launch(
"webrtcbin name=webrtcbin stun-server=stun://stun.l.google.com:19302 \ "webrtcbin name=webrtcbin stun-server=stun://stun.l.google.com:19302 \
videotestsrc pattern=ball ! videoconvert ! queue name=vqueue", videotestsrc pattern=ball ! videoconvert ! queue name=vqueue",
)?; )?;

491
subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/Cargo.lock generated
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8
subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/Cargo.toml
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@ -10,10 +10,10 @@ async-std = "1"
clap = { version = "4", features = ["derive"] } clap = { version = "4", features = ["derive"] }
anyhow = "1" anyhow = "1"
rand = "0.8" rand = "0.8"
async-tungstenite = { version = "0.24", features = ["async-std-runtime", "async-native-tls"] } async-tungstenite = { version = "0.25", features = ["async-std-runtime", "async-native-tls"] }
gst = { package = "gstreamer", version = "0.21" } gst = { package = "gstreamer", version = "0.22" }
gst-webrtc = { package = "gstreamer-webrtc", version = "0.21" } gst-webrtc = { package = "gstreamer-webrtc", version = "0.22" }
gst-sdp = { package = "gstreamer-sdp", version = "0.21" } gst-sdp = { package = "gstreamer-sdp", version = "0.22" }
serde = "1" serde = "1"
serde_derive = "1" serde_derive = "1"
serde_json = "1" serde_json = "1"

8
subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs
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@ -153,7 +153,7 @@ impl App {
anyhow::Error, anyhow::Error,
> { > {
// Create the GStreamer pipeline // Create the GStreamer pipeline
let pipeline = gst::parse_launch( let pipeline = gst::parse::launch(
&format!( &format!(
"videotestsrc is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay pt=96 picture-id-mode=15-bit ! tee name=video-tee ! \ "videotestsrc is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay pt=96 picture-id-mode=15-bit ! tee name=video-tee ! \
queue ! fakesink sync=true \ queue ! fakesink sync=true \
@ -302,7 +302,7 @@ impl App {
bail!("Peer {peer_id} already called"); bail!("Peer {peer_id} already called");
} }
let peer_bin = gst::parse_bin_from_description( let peer_bin = gst::parse::bin_from_description(
"queue name=video-queue ! webrtcbin. \ "queue name=video-queue ! webrtcbin. \
queue name=audio-queue ! webrtcbin. \ queue name=audio-queue ! webrtcbin. \
webrtcbin name=webrtcbin", webrtcbin name=webrtcbin",
@ -819,14 +819,14 @@ impl Peer {
.ok_or_else(|| anyhow!("no media type in caps {caps:?}"))?; .ok_or_else(|| anyhow!("no media type in caps {caps:?}"))?;
let conv = if media_type == "video" { let conv = if media_type == "video" {
gst::parse_bin_from_description( gst::parse::bin_from_description(
&format!( &format!(
"decodebin name=dbin ! queue ! videoconvert ! videoscale ! capsfilter name=src caps=video/x-raw,width={VIDEO_WIDTH},height={VIDEO_HEIGHT},pixel-aspect-ratio=1/1" "decodebin name=dbin ! queue ! videoconvert ! videoscale ! capsfilter name=src caps=video/x-raw,width={VIDEO_WIDTH},height={VIDEO_HEIGHT},pixel-aspect-ratio=1/1"
), ),
false, false,
)? )?
} else if media_type == "audio" { } else if media_type == "audio" {
gst::parse_bin_from_description( gst::parse::bin_from_description(
"decodebin name=dbin ! queue ! audioconvert ! audioresample name=src", "decodebin name=dbin ! queue ! audioconvert ! audioresample name=src",
false, false,
)? )?

503
subprojects/gst-examples/webrtc/sendrecv/gst-rust/Cargo.lock generated
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10
subprojects/gst-examples/webrtc/sendrecv/gst-rust/Cargo.toml
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@ -10,11 +10,11 @@ async-std = "1"
clap = { version = "4", features = ["derive"] } clap = { version = "4", features = ["derive"] }
anyhow = "1" anyhow = "1"
rand = "0.8" rand = "0.8"
async-tungstenite = { version = "0.24", features = ["async-std-runtime", "async-native-tls"] } async-tungstenite = { version = "0.25", features = ["async-std-runtime", "async-native-tls"] }
gst = { package = "gstreamer", version = "0.21" } gst = { package = "gstreamer", version = "0.22" }
gst-rtp = { package = "gstreamer-rtp", version = "0.21", features = ["v1_20"] } gst-rtp = { package = "gstreamer-rtp", version = "0.22", features = ["v1_20"] }
gst-webrtc = { package = "gstreamer-webrtc", version = "0.21" } gst-webrtc = { package = "gstreamer-webrtc", version = "0.22" }
gst-sdp = { package = "gstreamer-sdp", version = "0.21" } gst-sdp = { package = "gstreamer-sdp", version = "0.22" }
serde = "1" serde = "1"
serde_derive = "1" serde_derive = "1"
serde_json = "1" serde_json = "1"

6
subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
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@ -124,7 +124,7 @@ impl App {
anyhow::Error, anyhow::Error,
> { > {
// Create the GStreamer pipeline // Create the GStreamer pipeline
let pipeline = gst::parse_launch( let pipeline = gst::parse::launch(
"videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay name=vpay pt=96 picture-id-mode=15-bit ! webrtcbin. \ "videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay name=vpay pt=96 picture-id-mode=15-bit ! webrtcbin. \
audiotestsrc is-live=true ! opusenc perfect-timestamp=true ! rtpopuspay name=apay pt=97 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \ audiotestsrc is-live=true ! opusenc perfect-timestamp=true ! rtpopuspay name=apay pt=97 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \
webrtcbin name=webrtcbin" webrtcbin name=webrtcbin"
@ -623,12 +623,12 @@ impl App {
let name = caps.structure(0).unwrap().name(); let name = caps.structure(0).unwrap().name();
let sink = if name.starts_with("video/") { let sink = if name.starts_with("video/") {
gst::parse_bin_from_description( gst::parse::bin_from_description(
"queue ! videoconvert ! videoscale ! autovideosink", "queue ! videoconvert ! videoscale ! autovideosink",
true, true,
)? )?
} else if name.starts_with("audio/") { } else if name.starts_with("audio/") {
gst::parse_bin_from_description( gst::parse::bin_from_description(
"queue ! audioconvert ! audioresample ! autoaudiosink", "queue ! audioconvert ! audioresample ! autoaudiosink",
true, true,
)? )?