From 35b84d219f8c06c714e0ab9f7e645253b3ea408d Mon Sep 17 00:00:00 2001
From: Guillaume Desmottes <guillaume.desmottes@onestream.live>
Date: Tue, 2 Apr 2024 14:10:36 +0200
Subject: [PATCH] webrtc: webrtcsink: set perfect-timestamp=true on audio
 encoders

Chrome audio decoder doesn't cope well with not perfect ts, generating
noises in the audio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
---
 Cargo.lock                       | 2 ++
 net/webrtc/Cargo.toml            | 1 +
 net/webrtc/README.md             | 2 +-
 net/webrtc/src/webrtcsink/imp.rs | 6 ++++++
 4 files changed, 10 insertions(+), 1 deletion(-)

diff --git a/Cargo.lock b/Cargo.lock
index 99236951..b4cbd654 100644
--- a/Cargo.lock
+++ b/Cargo.lock
@@ -2930,6 +2930,7 @@ dependencies = [
  "gst-plugin-webrtc-signalling-protocol",
  "gstreamer",
  "gstreamer-app",
+ "gstreamer-audio",
  "gstreamer-base",
  "gstreamer-rtp",
  "gstreamer-sdp",
@@ -3071,6 +3072,7 @@ dependencies = [
  "gstreamer-base",
  "libc",
  "once_cell",
+ "serde",
  "smallvec",
 ]
 
diff --git a/net/webrtc/Cargo.toml b/net/webrtc/Cargo.toml
index 9fc73b64..c5ef0379 100644
--- a/net/webrtc/Cargo.toml
+++ b/net/webrtc/Cargo.toml
@@ -11,6 +11,7 @@ rust-version.workspace = true
 [dependencies]
 gst = { workspace = true, features = ["v1_20", "serde"] }
 gst-app = { workspace = true, features = ["v1_20"] }
+gst-audio = { workspace = true, features = ["v1_20", "serde"] }
 gst-video = { workspace = true, features = ["v1_20", "serde"] }
 gst-webrtc = { workspace = true, features = ["v1_20"] }
 gst-sdp = { workspace = true, features = ["v1_20"] }
diff --git a/net/webrtc/README.md b/net/webrtc/README.md
index 175f0808..bc3179d8 100644
--- a/net/webrtc/README.md
+++ b/net/webrtc/README.md
@@ -319,7 +319,7 @@ b. In the second tab start the `simple-whip-client` as shown in the below comman
 
 ``` shell
 ./whip-client --url http://127.0.0.1:8190/whip/endpoint \
-        -A "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay pt=100 ssrc=1 ! queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=100" \
+        -A "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay pt=100 ssrc=1 ! queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=100" \
         -V "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay pt=96 ssrc=2 ! queue ! application/x-rtp,media=video,encoding-name=VP8,payload=96" \
         -S stun://stun.l.google.com:19302 \
         -l 7 \
diff --git a/net/webrtc/src/webrtcsink/imp.rs b/net/webrtc/src/webrtcsink/imp.rs
index cac00ede..15c2e8cb 100644
--- a/net/webrtc/src/webrtcsink/imp.rs
+++ b/net/webrtc/src/webrtcsink/imp.rs
@@ -693,6 +693,12 @@ fn add_nv4l2enc_force_keyunit_workaround(enc: &gst::Element) {
 /// Default configuration for known encoders, can be disabled
 /// by returning True from an encoder-setup handler.
 fn configure_encoder(enc: &gst::Element, start_bitrate: u32) {
+    let audio_encoder = enc.is::<gst_audio::AudioEncoder>();
+    if audio_encoder {
+        // Chrome audio decoder expects perfect timestamps
+        enc.set_property("perfect-timestamp", true);
+    }
+
     if let Some(factory) = enc.factory() {
         match factory.name().as_str() {
             "vp8enc" | "vp9enc" => {