rtp: opus: add simple payload / depayload test

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
This commit is contained in:
Tim-Philipp Müller 2024-05-14 18:47:06 +01:00 committed by GStreamer Marge Bot
parent 92c0cf1285
commit 6f871e6ce2
5 changed files with 115 additions and 16 deletions

1
net/rtp/src/opus/tests/.gitattributes vendored Normal file
View file

@ -0,0 +1 @@
*.opus binary

Binary file not shown.

View file

@ -0,0 +1 @@
ø¯êm$t3 <0B>Hz ¡YÒ{½#éƒH™:?Q…a€soØòû<C3B2>…{`ãª8ÐÎÂjÁ¨iŒŽšOì·@-¯æPÙŠmZï¾2ÉOªÊ˜Ïòó´üVÒË8 ÷Ç;Ì/Š¨ùnâÛv|`œt—ÂîhñI04áþ{êB^ßöog>‰GO[Ž{<7B>oyéfß@êè 6^8ËÖ“k%^ G<>

Binary file not shown.

View file

@ -20,18 +20,6 @@ fn init() {
});
}
// test_opus_pay_dtx
//
// Make sure payloader drops any DTX packets by the encoder (if so requested via the property)
//
#[test]
fn test_opus_pay_dtx() {
// gst-launch-1.0 audiotestsrc wave=silence
// ! opusenc dtx=true bitrate-type=vbr
// ! fakesink silent=false dump=true
const OPUS_BUFFER_SILENCE: &[u8] = &[0xf8, 0xff, 0xfe];
const OPUS_BUFFER_SILENCE_DTX: &[u8] = &[0xf8];
fn make_buffer(
data: &'static [u8],
pts: gst::ClockTime,
@ -48,6 +36,18 @@ fn test_opus_pay_dtx() {
buf
}
// test_opus_pay_dtx
//
// Make sure payloader drops any DTX packets by the encoder (if so requested via the property)
//
#[test]
fn test_opus_pay_dtx() {
// gst-launch-1.0 audiotestsrc wave=silence
// ! opusenc dtx=true bitrate-type=vbr
// ! fakesink silent=false dump=true
const OPUS_BUFFER_SILENCE: &[u8] = &[0xf8, 0xff, 0xfe];
const OPUS_BUFFER_SILENCE_DTX: &[u8] = &[0xf8];
init();
for dtx_prop in [false, true] {
@ -139,3 +139,100 @@ fn test_opus_pay_dtx() {
);
}
}
// test_opus_pay_depay
//
// Check basic payloading/depayloading
//
#[test]
fn test_opus_pay_depay() {
// gst-launch-1.0 audiotestsrc ! opusenc ! multifilesink
const OPUS_BUFFERS: &[&[u8]] = &[
include_bytes!("audiotestsrc-1ch-48kHz-000.opus").as_slice(),
include_bytes!("audiotestsrc-1ch-48kHz-001.opus").as_slice(),
include_bytes!("audiotestsrc-1ch-48kHz-002.opus").as_slice(),
];
init();
let input_caps = gst::Caps::builder("audio/x-opus")
.field("rate", 48000i32)
.field("channels", 1i32)
.field("channel-mapping-family", 0i32)
.field("stream-count", 1i32)
.field("coupled-count", 0i32)
.build();
let input_buffers = vec![
make_buffer(
OPUS_BUFFERS[0],
gst::ClockTime::ZERO,
gst::ClockTime::from_mseconds(20), // Note: no ClippingMeta for lead-in here unlike opusenc
gst::BufferFlags::DISCONT,
),
make_buffer(
OPUS_BUFFERS[1],
gst::ClockTime::from_mseconds(20),
gst::ClockTime::from_mseconds(20),
gst::BufferFlags::empty(),
),
make_buffer(
OPUS_BUFFERS[2],
gst::ClockTime::from_mseconds(40),
gst::ClockTime::from_mseconds(20),
gst::BufferFlags::empty(),
),
];
// TODO: check durations?
let expected_pay = vec![
vec![ExpectedPacket::builder()
.pts(gst::ClockTime::ZERO)
.flags(gst::BufferFlags::DISCONT | gst::BufferFlags::MARKER)
.pt(96)
.rtp_time(0)
.marker_bit(true)
.build()],
vec![ExpectedPacket::builder()
.pts(gst::ClockTime::from_mseconds(20))
.flags(gst::BufferFlags::empty())
.pt(96)
.rtp_time(960)
.marker_bit(false)
.build()],
vec![ExpectedPacket::builder()
.pts(gst::ClockTime::from_mseconds(40))
.flags(gst::BufferFlags::empty())
.pt(96)
.rtp_time(960 + 960)
.marker_bit(false)
.build()],
];
// TODO: check durations?
let expected_depay = vec![
vec![ExpectedBuffer::builder()
.pts(gst::ClockTime::ZERO)
.size(253)
.flags(gst::BufferFlags::DISCONT | gst::BufferFlags::RESYNC)
.build()],
vec![ExpectedBuffer::builder()
.pts(gst::ClockTime::from_mseconds(20))
.size(168)
.flags(gst::BufferFlags::empty())
.build()],
vec![ExpectedBuffer::builder()
.pts(gst::ClockTime::from_mseconds(40))
.size(166)
.flags(gst::BufferFlags::empty())
.build()],
];
run_test_pipeline(
Source::Buffers(input_caps, input_buffers),
"rtpopuspay2",
"rtpopusdepay2",
expected_pay,
expected_depay,
);
}