red: copy audio level from main packet for recovery packet

fill the audio level of the recovery packets from the main packet.
While not exact, this should be close enough. Without this,
the audio level in getStats() will be filled but the audio level
in getSynchronizationSources() will be empty.

In chrome this is easy to test, the audio level graph on
  https://webrtc.github.io/samples/src/content/peerconnection/audio/
will be empty all the time prior to this fix.

BUG=webrtc:11640

Change-Id: Ia1e61fd1852445239021a76d08032120a92d83aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226840
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34635}
diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc
index b7b4520..7438f25 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -136,7 +136,7 @@
             /*ssrc=*/red_packet.packet_info.ssrc(),
             /*csrcs=*/std::vector<uint32_t>(),
             /*rtp_timestamp=*/new_packet.timestamp,
-            /*audio_level=*/absl::nullopt,
+            red_packet.packet_info.audio_level(),
             /*absolute_capture_time=*/absl::nullopt,
             /*receive_time=*/red_packet.packet_info.receive_time());
         new_packets.push_front(std::move(new_packet));