Comments

src/media/event.rs

@@ -112,12 +112,6 @@ pub struct MediaData {
     /// For audio this flag most likely doesn't matter.
     pub contiguous: bool,
 
-    /// First packet of a talkspurt, that is the first packet after a silence period during
-    /// which packets have not been transmitted contiguously.
-    ///
-    /// For audio only when dtx or silence suppression is enabled.
-    pub audio_start_of_talk_spurt: bool,
-
     /// The actual packet data a.k.a Sample.
     ///
     /// Bigger samples don't fit in one UDP packet, thus WebRTC RTP is chopping up codec
@@ -137,6 +131,12 @@ pub struct MediaData {
     ///
     /// If no Sender Report(SR) has been received this is [`None`].
     pub last_sender_info: Option<SenderInfo>,
+
+    /// First packet of a talkspurt, that is the first packet after a silence period during
+    /// which packets have not been transmitted contiguously.
+    ///
+    /// For audio only when dtx or silence suppression is enabled.
+    pub audio_start_of_talk_spurt: bool,
 }
 
 /// Details for an incoming a keyframe request (PLI or FIR).

src/media/mod.rs

@@ -265,10 +265,14 @@ impl Media {
                     network_time: dep.first_network_time(),
                     seq_range: dep.seq_range(),
                     contiguous: dep.contiguous,
-                    audio_start_of_talk_spurt: dep.start_of_talkspurt(),
                     ext_vals: dep.ext_vals().clone(),
                     codec_extra: dep.codec_extra,
                     last_sender_info: dep.first_sender_info(),
+                    audio_start_of_talk_spurt: if codec.spec().codec.is_audio() {
+                        dep.start_of_talkspurt()
+                    } else {
+                        false
+                    },
                     data: dep.data,
                 }));
             }