From ed54734825c6e820b6d79a1569242ddbb4f5c59c Mon Sep 17 00:00:00 2001
From: Guillaume Desmottes <guillaume.desmottes@onestream.live>
Date: Tue, 2 Apr 2024 15:57:58 +0200
Subject: [PATCH] examples: set perfect-timestamp=true on opusenc
Fix audio streaming on Chrome, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6512>
---
.../gst-examples/webrtc/android/app/src/main/jni/webrtc.c | 2 +-
.../webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c | 2 +-
.../gst-examples/webrtc/sendonly/webrtc-recvonly-h264.c | 2 +-
.../webrtc/sendonly/webrtc-unidirectional-h264.c | 2 +-
.../sendrecv/gst-java/src/main/java/WebrtcSendRecv.java | 2 +-
.../webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs | 2 +-
.../gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c | 2 +-
.../gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py | 6 +++---
.../gst-plugins-bad/tests/examples/webrtc/webrtcswap.c | 4 ++--
9 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/subprojects/gst-examples/webrtc/android/app/src/main/jni/webrtc.c b/subprojects/gst-examples/webrtc/android/app/src/main/jni/webrtc.c
index 2897fa57ba..661cdbdbc6 100644
--- a/subprojects/gst-examples/webrtc/android/app/src/main/jni/webrtc.c
+++ b/subprojects/gst-examples/webrtc/android/app/src/main/jni/webrtc.c
@@ -390,7 +390,7 @@ start_pipeline (WebRTC * webrtc)
"ahcsrc device-facing=front ! video/x-raw,width=[320,1280] ! queue max-size-buffers=1 ! videoconvert ! "
"vp8enc keyframe-max-dist=30 deadline=1 error-resilient=default ! rtpvp8pay picture-id-mode=15-bit mtu=1300 ! "
"queue max-size-time=300000000 ! " RTP_CAPS_VP8 " ! sendrecv.sink_0 "
- "openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! "
+ "openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc perfect-timestamp=true ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS " ! sendrecv.sink_1 ", &error);
if (error) {
diff --git a/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c b/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
index 87dd87be15..21e5c7cb02 100644
--- a/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
+++ b/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
@@ -458,7 +458,7 @@ start_pipeline (void)
* inside the same pipeline. We start by connecting it to a fakesink so that
* we can preroll early. */
pipeline = gst_parse_launch ("tee name=audiotee ! queue ! fakesink "
- "audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay ! "
+ "audiotestsrc is-live=true wave=red-noise ! queue ! opusenc perfect-timestamp=true ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS (96) " ! audiotee. ", &error);
if (error) {
diff --git a/subprojects/gst-examples/webrtc/sendonly/webrtc-recvonly-h264.c b/subprojects/gst-examples/webrtc/sendonly/webrtc-recvonly-h264.c
index 7222ccbb91..835528b058 100644
--- a/subprojects/gst-examples/webrtc/sendonly/webrtc-recvonly-h264.c
+++ b/subprojects/gst-examples/webrtc/sendonly/webrtc-recvonly-h264.c
@@ -329,7 +329,7 @@ create_receiver_entry (SoupWebsocketConnection * connection)
receiver_entry->pipeline =
gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
- "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
+ "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. ", &error);
if (error != NULL) {
g_error ("Could not create WebRTC pipeline: %s\n", error->message);
diff --git a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
index b53719550d..6c785d035b 100644
--- a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
+++ b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
@@ -249,7 +249,7 @@ create_receiver_entry (SoupWebsocketConnection * connection)
"application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE " ! webrtcbin. "
"autoaudiosrc ! queue max-size-buffers=1 leaky=downstream"
- " ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
+ " ! audioconvert ! audioresample ! opusenc perfect-timestamp=true ! rtpopuspay pt="
RTP_AUDIO_PAYLOAD_TYPE " ! application/x-rtp, encoding-name=OPUS !"
" webrtcbin. ", &error);
if (error != NULL) {
diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst-java/src/main/java/WebrtcSendRecv.java b/subprojects/gst-examples/webrtc/sendrecv/gst-java/src/main/java/WebrtcSendRecv.java
index 7aea74a8d3..8b081d72b8 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst-java/src/main/java/WebrtcSendRecv.java
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst-java/src/main/java/WebrtcSendRecv.java
@@ -30,7 +30,7 @@ public class WebrtcSendRecv {
private static final Logger logger = LoggerFactory.getLogger(WebrtcSendRecv.class);
private static final String REMOTE_SERVER_URL = "wss://webrtc.gstreamer.net:8443";
private static final String VIDEO_BIN_DESCRIPTION = "videotestsrc ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! queue ! capsfilter caps=application/x-rtp,media=video,encoding-name=VP8,payload=97";
- private static final String AUDIO_BIN_DESCRIPTION = "audiotestsrc ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! queue ! capsfilter caps=application/x-rtp,media=audio,encoding-name=OPUS,payload=96";
+ private static final String AUDIO_BIN_DESCRIPTION = "audiotestsrc ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay ! queue ! capsfilter caps=application/x-rtp,media=audio,encoding-name=OPUS,payload=96";
private final String serverUrl;
private final String peerId;
diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs b/subprojects/gst-examples/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs
index 7b672ab824..48dc8464c3 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs
@@ -19,7 +19,7 @@ namespace GstWebRTCDemo
const string PIPELINE_DESC = @"webrtcbin name=sendrecv bundle-policy=max-bundle
videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
- audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+ audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.";
readonly int _id;
diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
index 7e4e06f557..35e77135e4 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
@@ -490,7 +490,7 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
audio_desc =
g_strdup_printf
("audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample"
- "! queue ! opusenc ! rtpopuspay name=audiopay pt=%u "
+ "! queue ! opusenc perfect-timestamp=true ! rtpopuspay name=audiopay pt=%u "
"! application/x-rtp, encoding-name=OPUS ! queue", opus_pt);
audio_bin = gst_parse_bin_from_description (audio_desc, TRUE, &audio_error);
g_free (audio_desc);
diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
index 6feef0eb92..bf4abc822e 100755
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
@@ -39,7 +39,7 @@ PIPELINE_DESC_VP8 = WEBRTCBIN + '''
{vsrc} ! videoconvert ! queue !
vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
queue ! application/x-rtp,media=video,encoding-name=VP8,payload={video_pt} ! sendrecv.
- {asrc} ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+ {asrc} ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
'''
PIPELINE_DESC_H264 = WEBRTCBIN + '''
@@ -47,7 +47,7 @@ PIPELINE_DESC_H264 = WEBRTCBIN + '''
x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true !
rtph264pay aggregate-mode=zero-latency config-interval=-1 !
queue ! application/x-rtp,media=video,encoding-name=H264,payload={video_pt} ! sendrecv.
- {asrc} ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+ {asrc} ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
'''
# Force I420 because dav1d bundled with Chrome doesn't support 10-bit choma/luma (I420_10LE)
@@ -55,7 +55,7 @@ PIPELINE_DESC_AV1 = WEBRTCBIN + '''
{vsrc} ! videoconvert ! queue !
video/x-raw,format=I420 ! svtav1enc preset=13 ! av1parse ! rtpav1pay !
queue ! application/x-rtp,media=video,encoding-name=AV1,payload={video_pt} ! sendrecv.
- {asrc} ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+ {asrc} ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
'''
PIPELINE_DESC = {
diff --git a/subprojects/gst-plugins-bad/tests/examples/webrtc/webrtcswap.c b/subprojects/gst-plugins-bad/tests/examples/webrtc/webrtcswap.c
index 02c507dc88..046d3fcee2 100644
--- a/subprojects/gst-plugins-bad/tests/examples/webrtc/webrtcswap.c
+++ b/subprojects/gst-plugins-bad/tests/examples/webrtc/webrtcswap.c
@@ -172,11 +172,11 @@ main (int argc, char *argv[])
gst_parse_launch ("webrtcbin name=smpte webrtcbin name=ball "
"videotestsrc pattern=smpte ! queue ! vp8enc ! rtpvp8pay ! queue ! "
"application/x-rtp,media=video,payload=96,encoding-name=VP8 ! smpte.sink_0 "
- "audiotestsrc ! opusenc ! rtpopuspay ! queue ! "
+ "audiotestsrc ! opusenc perfect-timestamp=true ! rtpopuspay ! queue ! "
"application/x-rtp,media=audio,payload=97,encoding-name=OPUS ! smpte.sink_1 "
"videotestsrc pattern=ball ! queue ! vp8enc ! rtpvp8pay ! queue ! "
"application/x-rtp,media=video,payload=96,encoding-name=VP8 ! ball.sink_1 "
- "audiotestsrc wave=saw ! opusenc ! rtpopuspay ! queue ! "
+ "audiotestsrc wave=saw ! opusenc perfect-timestamp=true ! rtpopuspay ! queue ! "
"application/x-rtp,media=audio,payload=97,encoding-name=OPUS ! ball.sink_0 ",
NULL);
bus1 = gst_pipeline_get_bus (GST_PIPELINE (pipe1));