From d810049f011a3669ab76e076006001ac25946225 Mon Sep 17 00:00:00 2001
From: Jochen Henneberg <jh@henneberg-systemdesign.com>
Date: Mon, 4 Sep 2023 17:31:57 +0200
Subject: [PATCH] rtpmp4adepay: Enabled header extension aggregation

Because this depayloader may build several output buffers within one process
run we push them all into a GstBufferList and push them out at once to
make sure that each buffer gets notified about each header extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
---
 .../gst/rtp/gstrtpmp4adepay.c                 | 23 ++++++++++++++++---
 1 file changed, 20 insertions(+), 3 deletions(-)

diff --git a/subprojects/gst-plugins-good/gst/rtp/gstrtpmp4adepay.c b/subprojects/gst-plugins-good/gst/rtp/gstrtpmp4adepay.c
index f278fc598c..6ef10d7275 100644
--- a/subprojects/gst-plugins-good/gst/rtp/gstrtpmp4adepay.c
+++ b/subprojects/gst-plugins-good/gst/rtp/gstrtpmp4adepay.c
@@ -111,6 +111,9 @@ gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
 static void
 gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay)
 {
+  gst_rtp_base_depayload_set_aggregate_hdrext_enabled (GST_RTP_BASE_DEPAYLOAD
+      (rtpmp4adepay), TRUE);
+
   rtpmp4adepay->adapter = gst_adapter_new ();
   rtpmp4adepay->framed = FALSE;
 }
@@ -306,6 +309,7 @@ gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
   GstRtpMP4ADepay *rtpmp4adepay;
   GstBuffer *outbuf;
   GstMapInfo map;
+  GstBufferList *outbufs = NULL;
 
   rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
 
@@ -359,6 +363,7 @@ gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
     /* position in data we are at */
     pos = 0;
 
+    outbufs = gst_buffer_list_new_sized (rtpmp4adepay->numSubFrames);
     /* looping through the number of sub-frames in the audio payload */
     for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) {
       /* determine payload length and set buffer data pointer accordingly */
@@ -398,7 +403,7 @@ gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
 
       GST_BUFFER_PTS (tmp) = timestamp;
       gst_rtp_drop_non_audio_meta (depayload, tmp);
-      gst_rtp_base_depayload_push (depayload, tmp);
+      gst_buffer_list_add (outbufs, tmp);
 
       /* shift ts for next buffers */
       if (rtpmp4adepay->frame_len && timestamp != -1
@@ -409,6 +414,9 @@ gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
       }
     }
 
+    /* now push all sub-frames we found */
+    gst_rtp_base_depayload_push_list (depayload, outbufs);
+
     /* just a check that lengths match */
     if (avail) {
       GST_ELEMENT_WARNING (depayload, STREAM, DECODE,
@@ -416,9 +424,9 @@ gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
               "possible wrongly encoded packet."));
     }
 
-    gst_buffer_unmap (outbuf, &map);
-    gst_buffer_unref (outbuf);
+    goto out;
   }
+
   return NULL;
 
   /* ERRORS */
@@ -426,6 +434,15 @@ wrong_size:
   {
     GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
         ("Packet did not validate"), ("wrong packet size"));
+    /* push what we have so far */
+    gst_rtp_base_depayload_push_list (depayload, outbufs);
+  }
+
+out:
+  {
+    /* we may not have sent anything but we consumed all data from the
+       adapter so let's clear the hdrext cache */
+    gst_rtp_base_depayload_flush (depayload, FALSE);
     gst_buffer_unmap (outbuf, &map);
     gst_buffer_unref (outbuf);
     return NULL;