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rtp: opus: add minimal depayload / re-payload test

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
This commit is contained in:
Tim-Philipp Müller 2024-05-14 19:47:53 +01:00 committed by GStreamer Marge Bot
parent 6f871e6ce2
commit 61523baa7b
1 changed files with 40 additions and 0 deletions
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40
net/rtp/src/opus/tests/tests.rs
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@ -9,6 +9,7 @@
// SPDX-License-Identifier: MPL-2.0
use crate::tests::{run_test_pipeline, ExpectedBuffer, ExpectedPacket, Source};
use gst_check::Harness;
fn init() {
use std::sync::Once;
@ -236,3 +237,42 @@ fn test_opus_pay_depay() {
expected_depay,
);
}
// test_opus_depay_pay
//
// Check basic depayloader ! payloader compatibility
//
#[test]
fn test_opus_depay_pay() {
// gst-launch-1.0 audiotestsrc
// ! opusenc bitrate-type=vbr frame-size=2
// ! rtpopuspay ! fakesink dump=true
const OPUS_RTP_BUFFER: &[u8] = &[
0x80, 0xe0, 0x6c, 0xd6, 0x5f, 0x7a, 0xdd, 0xae, 0xa6, 0x79, 0xe0, 0xc9, 0xe0, 0xff, 0xfe,
];
init();
let mut h = Harness::new_parse("rtpopusdepay2 ! rtpopuspay2");
let input_caps = gst::Caps::builder("application/x-rtp")
.field("media", "audio")
.field("encoding-name", "OPUS")
.field("clock-rate", 48000i32)
.field("payload", 96i32)
.build();
h.set_src_caps(input_caps);
let input_buffer = make_buffer(
OPUS_RTP_BUFFER,
gst::ClockTime::ZERO,
gst::ClockTime::from_mseconds(20),
gst::BufferFlags::DISCONT,
);
h.push(input_buffer)
.expect("Got error flow when pushing buffer");
let _output_buffer = h.pull().expect("Didn't get output buffer");
}