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rtp: opus: add simple payload / depayload test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
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commit
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5 changed files with 115 additions and 16 deletions
1
net/rtp/src/opus/tests/.gitattributes
vendored
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1
net/rtp/src/opus/tests/.gitattributes
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@ -0,0 +1 @@
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*.opus binary
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BIN
net/rtp/src/opus/tests/audiotestsrc-1ch-48kHz-000.opus
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BIN
net/rtp/src/opus/tests/audiotestsrc-1ch-48kHz-000.opus
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net/rtp/src/opus/tests/audiotestsrc-1ch-48kHz-001.opus
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net/rtp/src/opus/tests/audiotestsrc-1ch-48kHz-001.opus
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ø¯êm$t3<0B>H–z ¡YÒ{½#éƒH™:?Q…a€soØòû<C3B2>…{`ãª8ÐÎÂjÁ¨iŒŽšOì·@-¯æPÙŠmZï¾2ÉOª‘‚ʘÏòó´üVÒË8
÷Ç;Ì/Š¨ù‚nâÛv|`œt—ÂîhñI04áþ{êB^ßöog>‰‹GO[Ž{<7B>oyéfß@êè
6^8ËÖ“k%^ G<>E®
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net/rtp/src/opus/tests/audiotestsrc-1ch-48kHz-002.opus
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net/rtp/src/opus/tests/audiotestsrc-1ch-48kHz-002.opus
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@ -20,18 +20,6 @@ fn init() {
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});
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}
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// test_opus_pay_dtx
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//
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// Make sure payloader drops any DTX packets by the encoder (if so requested via the property)
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//
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#[test]
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fn test_opus_pay_dtx() {
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// gst-launch-1.0 audiotestsrc wave=silence
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// ! opusenc dtx=true bitrate-type=vbr
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// ! fakesink silent=false dump=true
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const OPUS_BUFFER_SILENCE: &[u8] = &[0xf8, 0xff, 0xfe];
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const OPUS_BUFFER_SILENCE_DTX: &[u8] = &[0xf8];
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fn make_buffer(
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data: &'static [u8],
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pts: gst::ClockTime,
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@ -48,6 +36,18 @@ fn test_opus_pay_dtx() {
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buf
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}
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// test_opus_pay_dtx
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//
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// Make sure payloader drops any DTX packets by the encoder (if so requested via the property)
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//
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#[test]
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fn test_opus_pay_dtx() {
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// gst-launch-1.0 audiotestsrc wave=silence
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// ! opusenc dtx=true bitrate-type=vbr
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// ! fakesink silent=false dump=true
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const OPUS_BUFFER_SILENCE: &[u8] = &[0xf8, 0xff, 0xfe];
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const OPUS_BUFFER_SILENCE_DTX: &[u8] = &[0xf8];
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init();
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for dtx_prop in [false, true] {
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@ -139,3 +139,100 @@ fn test_opus_pay_dtx() {
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);
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}
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}
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// test_opus_pay_depay
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//
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// Check basic payloading/depayloading
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//
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#[test]
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fn test_opus_pay_depay() {
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// gst-launch-1.0 audiotestsrc ! opusenc ! multifilesink
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const OPUS_BUFFERS: &[&[u8]] = &[
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include_bytes!("audiotestsrc-1ch-48kHz-000.opus").as_slice(),
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include_bytes!("audiotestsrc-1ch-48kHz-001.opus").as_slice(),
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include_bytes!("audiotestsrc-1ch-48kHz-002.opus").as_slice(),
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];
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init();
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let input_caps = gst::Caps::builder("audio/x-opus")
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.field("rate", 48000i32)
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.field("channels", 1i32)
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.field("channel-mapping-family", 0i32)
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.field("stream-count", 1i32)
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.field("coupled-count", 0i32)
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.build();
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let input_buffers = vec![
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make_buffer(
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OPUS_BUFFERS[0],
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gst::ClockTime::ZERO,
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gst::ClockTime::from_mseconds(20), // Note: no ClippingMeta for lead-in here unlike opusenc
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gst::BufferFlags::DISCONT,
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),
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make_buffer(
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OPUS_BUFFERS[1],
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gst::ClockTime::from_mseconds(20),
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gst::ClockTime::from_mseconds(20),
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gst::BufferFlags::empty(),
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),
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make_buffer(
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OPUS_BUFFERS[2],
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gst::ClockTime::from_mseconds(40),
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gst::ClockTime::from_mseconds(20),
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gst::BufferFlags::empty(),
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),
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];
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// TODO: check durations?
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let expected_pay = vec![
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vec![ExpectedPacket::builder()
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.pts(gst::ClockTime::ZERO)
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.flags(gst::BufferFlags::DISCONT | gst::BufferFlags::MARKER)
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.pt(96)
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.rtp_time(0)
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.marker_bit(true)
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.build()],
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vec![ExpectedPacket::builder()
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.pts(gst::ClockTime::from_mseconds(20))
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.flags(gst::BufferFlags::empty())
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.pt(96)
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.rtp_time(960)
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.marker_bit(false)
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.build()],
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vec![ExpectedPacket::builder()
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.pts(gst::ClockTime::from_mseconds(40))
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.flags(gst::BufferFlags::empty())
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.pt(96)
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.rtp_time(960 + 960)
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.marker_bit(false)
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.build()],
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];
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// TODO: check durations?
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let expected_depay = vec![
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vec![ExpectedBuffer::builder()
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.pts(gst::ClockTime::ZERO)
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.size(253)
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.flags(gst::BufferFlags::DISCONT | gst::BufferFlags::RESYNC)
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.build()],
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vec![ExpectedBuffer::builder()
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.pts(gst::ClockTime::from_mseconds(20))
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.size(168)
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.flags(gst::BufferFlags::empty())
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.build()],
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vec![ExpectedBuffer::builder()
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.pts(gst::ClockTime::from_mseconds(40))
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.size(166)
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.flags(gst::BufferFlags::empty())
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.build()],
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];
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run_test_pipeline(
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Source::Buffers(input_caps, input_buffers),
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"rtpopuspay2",
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"rtpopusdepay2",
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expected_pay,
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expected_depay,
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);
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}
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