rtp: gstreamer: Don't count on buffer DTS for capture time

With GStreamer 1.18, the old behaviour of storing the capture time in
DTS is gone (which is reasonable, since the semantics really don't
match). So instead, we get a capture timestamp when the buffer is being
pushed from udpsrc. This should eventually move into udpsrc, and the
timestamp should come from the cmsg instead of the clock.

We still fallback to the DTS if the meta isn't available, as the meta
might be dropped in older versions of rtpL16pay due to a bug.

configure.ac

@@ -1318,7 +1318,7 @@ AC_ARG_ENABLE([gstreamer],
     AS_HELP_STRING([--disable-gstreamer],[Disable optional GStreamer-based RTP support]))
 
 AS_IF([test "x$enable_gstreamer" != "xno"],
-    [PKG_CHECK_MODULES(GSTREAMER, [ gstreamer-1.0 gstreamer-app-1.0 gstreamer-rtp-1.0 gio-2.0 ],
+    [PKG_CHECK_MODULES(GSTREAMER, [ gstreamer-1.0 >= 1.14 gstreamer-app-1.0 gstreamer-rtp-1.0 gio-2.0 ],
                        HAVE_GSTREAMER=1, HAVE_GSTREAMER=0)],
     HAVE_GSTREAMER=0)
 

meson.build

@@ -678,7 +678,7 @@ if webrtc_dep.found()
   cdata.set('HAVE_WEBRTC', 1)
 endif
 
-gst_dep = dependency('gstreamer-1.0', required : get_option('gstreamer'))
+gst_dep = dependency('gstreamer-1.0', version : '>= 1.14', required : get_option('gstreamer'))
 gstapp_dep = dependency('gstreamer-app-1.0', required : get_option('gstreamer'))
 gstrtp_dep = dependency('gstreamer-rtp-1.0', required : get_option('gstreamer'))
 

src/modules/rtp/rtp-gstreamer.c

@@ -52,6 +52,7 @@ struct pa_rtp_context {
     GstElement *pipeline;
     GstElement *appsrc;
     GstElement *appsink;
+    GstCaps *meta_reference;
 
     uint32_t last_timestamp;
 
@@ -70,6 +71,7 @@ static GstCaps* caps_from_sample_spec(const pa_sample_spec *ss) {
             "layout", G_TYPE_STRING, "interleaved",
             NULL);
 }
+
 static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss) {
     GstElement *appsrc = NULL, *pay = NULL, *capsf = NULL, *rtpbin = NULL, *sink = NULL;
     GstCaps *caps;
@@ -354,9 +356,26 @@ static void on_pad_added(GstElement *element, GstPad *pad, gpointer userdata) {
     gst_object_unref(depay);
 }
 
+static GstPadProbeReturn udpsrc_buffer_probe(GstPad *pad, GstPadProbeInfo *info, gpointer userdata) {
+    struct timeval tv;
+    pa_usec_t timestamp;
+    pa_rtp_context *c = (pa_rtp_context *) userdata;
+
+    pa_assert(info->type & GST_PAD_PROBE_TYPE_BUFFER);
+
+    pa_gettimeofday(&tv);
+    timestamp = pa_timeval_load(&tv);
+
+    gst_buffer_add_reference_timestamp_meta(GST_BUFFER(info->data), c->meta_reference, timestamp * GST_USECOND,
+            GST_CLOCK_TIME_NONE);
+
+    return GST_PAD_PROBE_OK;
+}
+
 static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spec *ss) {
     GstElement *udpsrc = NULL, *rtpbin = NULL, *depay = NULL, *appsink = NULL;
     GstCaps *caps;
+    GstPad *pad;
     GSocket *socket;
     GError *error = NULL;
 
@@ -398,6 +417,14 @@ static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spe
 
     g_signal_connect(G_OBJECT(rtpbin), "pad-added", G_CALLBACK(on_pad_added), c);
 
+    /* This logic should go into udpsrc, and we should be populating the
+     * receive timestamp using SCM_TIMESTAMP, but until we have that ... */
+    c->meta_reference = gst_caps_new_empty_simple("timestamp/x-pulseaudio-wallclock");
+
+    pad = gst_element_get_static_pad(udpsrc, "src");
+    gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, udpsrc_buffer_probe, c, NULL);
+    gst_object_unref(pad);
+
     if (gst_element_set_state(c->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
         pa_log("Could not start pipeline");
         goto fail;
@@ -483,6 +510,7 @@ int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_
     GstAdapter *adapter;
     GstBuffer *buf;
     GstMapInfo info;
+    GstClockTime timestamp = GST_CLOCK_TIME_NONE;
     uint8_t *data;
     uint64_t data_len = 0;
 
@@ -499,6 +527,21 @@ int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_
 
         buf = gst_sample_get_buffer(sample);
 
+        /* Get the timestamp from the first buffer */
+        if (timestamp == GST_CLOCK_TIME_NONE) {
+            GstReferenceTimestampMeta *meta = gst_buffer_get_reference_timestamp_meta(buf, c->meta_reference);
+
+            /* Use the meta if we were able to insert it and it came through,
+             * else try to fallback to the DTS, which is only available in
+             * GStreamer 1.16 and earlier. */
+            if (meta)
+                timestamp = meta->timestamp;
+            else if (GST_BUFFER_DTS(buf) != GST_CLOCK_TIME_NONE)
+                timestamp = GST_BUFFER_DTS(buf);
+            else
+                timestamp = 0;
+        }
+
         if (GST_BUFFER_IS_DISCONT(buf))
             pa_log_info("Discontinuity detected, possibly lost some packets");
 
@@ -550,11 +593,10 @@ int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_
 
     /* When buffer-mode = none, the buffer PTS is the RTP timestamp, converted
      * to time units (instead of clock-rate units as is in the header) and
-     * wraparound-corrected, and the DTS is the pipeline clock timestamp from
-     * when the buffer was acquired at the source (this is actually the running
-     * time which is why we need to add base time). */
+     * wraparound-corrected. */
     *rtp_tstamp = gst_util_uint64_scale_int(GST_BUFFER_PTS(gst_buffer_list_get(buf_list, 0)), c->ss.rate, GST_SECOND) & 0xFFFFFFFFU;
-    pa_timeval_rtstore(tstamp, (GST_BUFFER_DTS(gst_buffer_list_get(buf_list, 0)) + gst_element_get_base_time(c->pipeline)) / GST_USECOND, false);
+    if (timestamp != GST_CLOCK_TIME_NONE)
+        pa_timeval_rtstore(tstamp, timestamp / PA_NSEC_PER_USEC, false);
 
     gst_buffer_list_unref(buf_list);
     gst_object_unref(adapter);
@@ -574,6 +616,9 @@ fail:
 void pa_rtp_context_free(pa_rtp_context *c) {
     pa_assert(c);
 
+    if (c->meta_reference)
+        gst_caps_unref(c->meta_reference);
+
     if (c->appsrc) {
         gst_app_src_end_of_stream(GST_APP_SRC(c->appsrc));
         gst_object_unref(c->appsrc);