diff --git a/libavformat/hls.c b/libavformat/hls.c
index 1dd19d659f..80d9b5f016 100644
--- a/libavformat/hls.c
+++ b/libavformat/hls.c
@@ -666,7 +666,7 @@ static int parse_playlist(HLSContext *c, const char *url,
         }
     }
     if (pls)
-        pls->last_load_time = av_gettime();
+        pls->last_load_time = av_gettime_relative();
 
 fail:
     av_free(new_url);
@@ -1035,7 +1035,7 @@ restart:
 
 reload:
         if (!v->finished &&
-            av_gettime() - v->last_load_time >= reload_interval) {
+            av_gettime_relative() - v->last_load_time >= reload_interval) {
             if ((ret = parse_playlist(c, v->url, v, NULL)) < 0) {
                 av_log(v->parent, AV_LOG_WARNING, "Failed to reload playlist %d\n",
                        v->index);
@@ -1055,7 +1055,7 @@ reload:
         if (v->cur_seq_no >= v->start_seq_no + v->n_segments) {
             if (v->finished)
                 return AVERROR_EOF;
-            while (av_gettime() - v->last_load_time < reload_interval) {
+            while (av_gettime_relative() - v->last_load_time < reload_interval) {
                 if (ff_check_interrupt(c->interrupt_callback))
                     return AVERROR_EXIT;
                 av_usleep(100*1000);
@@ -1198,7 +1198,7 @@ static int select_cur_seq_no(HLSContext *c, struct playlist *pls)
     int seq_no;
 
     if (!pls->finished && !c->first_packet &&
-        av_gettime() - pls->last_load_time >= default_reload_interval(pls))
+        av_gettime_relative() - pls->last_load_time >= default_reload_interval(pls))
         /* reload the playlist since it was suspended */
         parse_playlist(c, pls->url, pls, NULL);
 
diff --git a/libavformat/hlsproto.c b/libavformat/hlsproto.c
index e607c10e21..a569e9224e 100644
--- a/libavformat/hlsproto.c
+++ b/libavformat/hlsproto.c
@@ -169,7 +169,7 @@ static int parse_playlist(URLContext *h, const char *url)
             }
         }
     }
-    s->last_load_time = av_gettime();
+    s->last_load_time = av_gettime_relative();
 
 fail:
     avio_close(in);
@@ -273,7 +273,7 @@ start:
                       s->target_duration;
 retry:
     if (!s->finished) {
-        int64_t now = av_gettime();
+        int64_t now = av_gettime_relative();
         if (now - s->last_load_time >= reload_interval) {
             if ((ret = parse_playlist(h, s->playlisturl)) < 0)
                 return ret;
@@ -292,7 +292,7 @@ retry:
     if (s->cur_seq_no - s->start_seq_no >= s->n_segments) {
         if (s->finished)
             return AVERROR_EOF;
-        while (av_gettime() - s->last_load_time < reload_interval) {
+        while (av_gettime_relative() - s->last_load_time < reload_interval) {
             if (ff_check_interrupt(&h->interrupt_callback))
                 return AVERROR_EXIT;
             av_usleep(100*1000);
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 7e68b4c2a6..33205aa6d4 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -143,7 +143,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
                 return AVERROR_INVALIDDATA;
             }
 
-            s->last_rtcp_reception_time = av_gettime();
+            s->last_rtcp_reception_time = av_gettime_relative();
             s->last_rtcp_ntp_time  = AV_RB64(buf + 8);
             s->last_rtcp_timestamp = AV_RB32(buf + 16);
             if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
@@ -323,7 +323,7 @@ int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
         avio_wb32(pb, 0); /* delay since last SR */
     } else {
         uint32_t middle_32_bits   = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
-        uint32_t delay_since_last = av_rescale(av_gettime() - s->last_rtcp_reception_time,
+        uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
                                                65536, AV_TIME_BASE);
 
         avio_wb32(pb, middle_32_bits); /* last SR timestamp */
@@ -448,7 +448,7 @@ int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
     /* Send new feedback if enough time has elapsed since the last
      * feedback packet. */
 
-    now = av_gettime();
+    now = av_gettime_relative();
     if (s->last_feedback_time &&
         (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
         return 0;
@@ -691,7 +691,7 @@ static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
     packet = av_mallocz(sizeof(*packet));
     if (!packet)
         return;
-    packet->recvtime = av_gettime();
+    packet->recvtime = av_gettime_relative();
     packet->seq      = seq;
     packet->len      = len;
     packet->buf      = buf;
@@ -769,7 +769,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
     }
 
     if (s->st) {
-        int64_t received = av_gettime();
+        int64_t received = av_gettime_relative();
         uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
                                            s->st->time_base);
         timestamp = AV_RB32(buf + 4);
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 0a88164d84..43071ea0df 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -1183,7 +1183,7 @@ start:
         }
         ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
 
-        rt->last_cmd_time = av_gettime();
+        rt->last_cmd_time = av_gettime_relative();
         /* Even if the request from the server had data, it is not the data
          * that the caller wants or expects. The memory could also be leaked
          * if the actual following reply has content data. */
@@ -1280,7 +1280,7 @@ static int rtsp_send_cmd_with_content_async(AVFormatContext *s,
         }
         ffurl_write(rt->rtsp_hd_out, send_content, send_content_length);
     }
-    rt->last_cmd_time = av_gettime();
+    rt->last_cmd_time = av_gettime_relative();
 
     return 0;
 }
@@ -1858,7 +1858,7 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
     for (;;) {
         if (ff_check_interrupt(&s->interrupt_callback))
             return AVERROR_EXIT;
-        if (wait_end && wait_end - av_gettime() < 0)
+        if (wait_end && wait_end - av_gettime_relative() < 0)
             return AVERROR(EAGAIN);
         max_p = 0;
         if (rt->rtsp_hd) {
@@ -2057,7 +2057,7 @@ redo:
         break;
     case RTSP_LOWER_TRANSPORT_CUSTOM:
         if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
-            wait_end && wait_end < av_gettime())
+            wait_end && wait_end < av_gettime_relative())
             len = AVERROR(EAGAIN);
         else
             len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);
diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c
index cad8de32c0..fa7383c734 100644
--- a/libavformat/rtspdec.c
+++ b/libavformat/rtspdec.c
@@ -887,7 +887,7 @@ retry:
 
     if (!(rt->rtsp_flags & RTSP_FLAG_LISTEN)) {
         /* send dummy request to keep TCP connection alive */
-        if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2 ||
+        if ((av_gettime_relative() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2 ||
             rt->auth_state.stale) {
             if (rt->server_type == RTSP_SERVER_WMS ||
                 (rt->server_type != RTSP_SERVER_REAL &&