Expose new audio stats on the API
Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL
adds these to the GetStats API.
Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27839}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index d576c60..a751599 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -649,6 +649,12 @@
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.concealment_events);
verifier.TestMemberIsNonNegative<uint64_t>(
+ media_stream_track.inserted_samples_for_deceleration);
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ media_stream_track.removed_samples_for_acceleration);
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ media_stream_track.silent_concealed_samples);
+ verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.delayed_packet_outage_samples);
@@ -722,6 +728,13 @@
verifier.TestMemberIsUndefined(inbound_stream.qp_sum);
}
verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.packets_received);
+ if (inbound_stream.media_type.is_defined() &&
+ *inbound_stream.media_type == "audio") {
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ inbound_stream.fec_packets_received);
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ inbound_stream.fec_packets_discarded);
+ }
verifier.TestMemberIsNonNegative<uint64_t>(inbound_stream.bytes_received);
// packets_lost is defined as signed, but this should never happen in
// this test. See RFC 3550.