mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-09-08 21:28:32 +00:00
169 lines
5.7 KiB
Rust
169 lines
5.7 KiB
Rust
// Copyright (C) 2020 François Laignel <fengalin@free.fr>
|
|
//
|
|
// This library is free software; you can redistribute it and/or
|
|
// modify it under the terms of the GNU Library General Public
|
|
// License as published by the Free Software Foundation; either
|
|
// version 2 of the License, or (at your option) any later version.
|
|
//
|
|
// This library is distributed in the hope that it will be useful,
|
|
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
// Library General Public License for more details.
|
|
//
|
|
// You should have received a copy of the GNU Library General Public
|
|
// License along with this library; if not, write to the
|
|
// Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
|
|
// Boston, MA 02110-1335, USA.
|
|
|
|
use gst::gst_debug;
|
|
use gst::prelude::*;
|
|
|
|
use lazy_static::lazy_static;
|
|
|
|
use std::sync::mpsc;
|
|
|
|
lazy_static! {
|
|
static ref CAT: gst::DebugCategory = gst::DebugCategory::new(
|
|
"ts-test",
|
|
gst::DebugColorFlags::empty(),
|
|
Some("Thread-sharing test"),
|
|
);
|
|
}
|
|
|
|
fn init() {
|
|
use std::sync::Once;
|
|
static INIT: Once = Once::new();
|
|
|
|
INIT.call_once(|| {
|
|
gst::init().unwrap();
|
|
gstthreadshare::plugin_register_static().expect("gstthreadshare jitterbuffer test");
|
|
});
|
|
}
|
|
|
|
#[test]
|
|
fn jb_pipeline() {
|
|
init();
|
|
|
|
const CONTEXT_WAIT: u32 = 20;
|
|
const LATENCY: u32 = 20;
|
|
const BUFFER_NB: i32 = 3;
|
|
|
|
let pipeline = gst::Pipeline::new(None);
|
|
|
|
let src = gst::ElementFactory::make("audiotestsrc", Some("audiotestsrc")).unwrap();
|
|
src.set_property("is-live", &true).unwrap();
|
|
src.set_property("num-buffers", &BUFFER_NB).unwrap();
|
|
|
|
let enc = gst::ElementFactory::make("alawenc", Some("alawenc")).unwrap();
|
|
let pay = gst::ElementFactory::make("rtppcmapay", Some("rtppcmapay")).unwrap();
|
|
|
|
let jb = gst::ElementFactory::make("ts-jitterbuffer", Some("ts-jitterbuffer")).unwrap();
|
|
jb.set_property("context", &"jb_pipeline").unwrap();
|
|
jb.set_property("context-wait", &CONTEXT_WAIT).unwrap();
|
|
jb.set_property("latency", &LATENCY).unwrap();
|
|
|
|
let depay = gst::ElementFactory::make("rtppcmadepay", Some("rtppcmadepay")).unwrap();
|
|
let dec = gst::ElementFactory::make("alawdec", Some("alawdec")).unwrap();
|
|
|
|
let sink = gst::ElementFactory::make("appsink", Some("appsink")).unwrap();
|
|
sink.set_property("sync", &false).unwrap();
|
|
sink.set_property("async", &false).unwrap();
|
|
sink.set_property("emit-signals", &true).unwrap();
|
|
|
|
pipeline
|
|
.add_many(&[&src, &enc, &pay, &jb, &depay, &dec, &sink])
|
|
.unwrap();
|
|
gst::Element::link_many(&[&src, &enc, &pay, &jb, &depay, &dec, &sink]).unwrap();
|
|
|
|
let appsink = sink.dynamic_cast::<gst_app::AppSink>().unwrap();
|
|
let (sender, receiver) = mpsc::channel();
|
|
appsink.connect_new_sample(move |appsink| {
|
|
let _sample = appsink
|
|
.emit("pull-sample", &[])
|
|
.unwrap()
|
|
.unwrap()
|
|
.get::<gst::Sample>()
|
|
.unwrap()
|
|
.unwrap();
|
|
|
|
sender.send(()).unwrap();
|
|
Ok(gst::FlowSuccess::Ok)
|
|
});
|
|
|
|
pipeline.set_state(gst::State::Playing).unwrap();
|
|
|
|
gst_debug!(CAT, "jb_pipeline: waiting for {} buffers", BUFFER_NB);
|
|
for idx in 0..BUFFER_NB {
|
|
receiver.recv().unwrap();
|
|
gst_debug!(CAT, "jb_pipeline: received buffer #{}", idx);
|
|
}
|
|
|
|
pipeline.set_state(gst::State::Null).unwrap();
|
|
}
|
|
|
|
#[test]
|
|
fn jb_ts_pipeline() {
|
|
init();
|
|
|
|
const CONTEXT_WAIT: u32 = 20;
|
|
const LATENCY: u32 = 20;
|
|
const BUFFER_NB: i32 = 3;
|
|
|
|
let pipeline = gst::Pipeline::new(None);
|
|
|
|
let src = gst::ElementFactory::make("audiotestsrc", Some("audiotestsrc")).unwrap();
|
|
src.set_property("is-live", &true).unwrap();
|
|
src.set_property("num-buffers", &BUFFER_NB).unwrap();
|
|
|
|
let queue = gst::ElementFactory::make("ts-queue", Some("ts-queue")).unwrap();
|
|
queue
|
|
.set_property("context", &"jb_ts_pipeline_queue")
|
|
.unwrap();
|
|
queue.set_property("context-wait", &CONTEXT_WAIT).unwrap();
|
|
|
|
let enc = gst::ElementFactory::make("alawenc", Some("alawenc")).unwrap();
|
|
let pay = gst::ElementFactory::make("rtppcmapay", Some("rtppcmapay")).unwrap();
|
|
|
|
let jb = gst::ElementFactory::make("ts-jitterbuffer", Some("ts-jitterbuffer")).unwrap();
|
|
jb.set_property("context", &"jb_ts_pipeline").unwrap();
|
|
jb.set_property("context-wait", &CONTEXT_WAIT).unwrap();
|
|
jb.set_property("latency", &LATENCY).unwrap();
|
|
|
|
let depay = gst::ElementFactory::make("rtppcmadepay", Some("rtppcmadepay")).unwrap();
|
|
let dec = gst::ElementFactory::make("alawdec", Some("alawdec")).unwrap();
|
|
|
|
let sink = gst::ElementFactory::make("appsink", Some("appsink")).unwrap();
|
|
sink.set_property("sync", &false).unwrap();
|
|
sink.set_property("async", &false).unwrap();
|
|
sink.set_property("emit-signals", &true).unwrap();
|
|
|
|
pipeline
|
|
.add_many(&[&src, &queue, &enc, &pay, &jb, &depay, &dec, &sink])
|
|
.unwrap();
|
|
gst::Element::link_many(&[&src, &queue, &enc, &pay, &jb, &depay, &dec, &sink]).unwrap();
|
|
|
|
let appsink = sink.dynamic_cast::<gst_app::AppSink>().unwrap();
|
|
let (sender, receiver) = mpsc::channel();
|
|
appsink.connect_new_sample(move |appsink| {
|
|
let _sample = appsink
|
|
.emit("pull-sample", &[])
|
|
.unwrap()
|
|
.unwrap()
|
|
.get::<gst::Sample>()
|
|
.unwrap()
|
|
.unwrap();
|
|
|
|
sender.send(()).unwrap();
|
|
Ok(gst::FlowSuccess::Ok)
|
|
});
|
|
|
|
pipeline.set_state(gst::State::Playing).unwrap();
|
|
|
|
gst_debug!(CAT, "jb_ts_pipeline: waiting for {} buffers", BUFFER_NB);
|
|
for idx in 0..BUFFER_NB {
|
|
receiver.recv().unwrap();
|
|
gst_debug!(CAT, "jb_ts_pipeline: received buffer #{}", idx);
|
|
}
|
|
|
|
pipeline.set_state(gst::State::Null).unwrap();
|
|
}
|