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rtpmp4adepay: Enabled header extension aggregation

Browse source 
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>
This commit is contained in:
Jochen Henneberg 2023-09-04 17:31:57 +02:00 committed by GStreamer Marge Bot
parent 90b5d2eb93
commit d810049f01
1 changed files with 20 additions and 3 deletions
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23
subprojects/gst-plugins-good/gst/rtp/gstrtpmp4adepay.c
View file

@ -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;