Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget.
This is a standardized metric:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
We estimate the target frame size in bytes from the current encoder
target bitrate and encoder framerate.
We would expect that the average bytes produced by the encoder would
over time match the average target, which is calculated by polling
getStats() twice and dividing the delta totalEncodedBytesTarget with
the delta framesEncoded. This is meant to make googTargetEncBitrate
obsolete.
Bug: webrtc:10446
Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28022}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index ebb2f38..438b47b 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -793,6 +793,8 @@
verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
verifier.TestMemberIsNonNegative<double>(
outbound_stream.total_encode_time);
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ outbound_stream.total_encoded_bytes_target);
verifier.TestMemberIsNonNegative<double>(
outbound_stream.total_packet_send_delay);
// The integration test is not set up to test screen share; don't require
@@ -801,6 +803,8 @@
} else {
verifier.TestMemberIsUndefined(outbound_stream.frames_encoded);
verifier.TestMemberIsUndefined(outbound_stream.total_encode_time);
+ verifier.TestMemberIsUndefined(
+ outbound_stream.total_encoded_bytes_target);
// TODO(https://crbug.com/webrtc/10635): Implement for audio as well.
verifier.TestMemberIsUndefined(outbound_stream.total_packet_send_delay);
verifier.TestMemberIsUndefined(outbound_stream.content_type);