New statistic added to VideoReceiveStream to determine latency to first decode.
This change introduces a new measurement into the VideoReceiveStream::Stats
structure to measure the latency between the first frame being received and
the first frame being decoded in WebRTC. The goal here is to measure the latency
difference when a FrameEncryptor is attached and not attached.
Change-Id: I0f0178aff73b66f25dbc6617098033e226da2958
Bug: webrtc:10105
Reviewed-on: https://webrtc-review.googlesource.com/c/113328
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25956}diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index fca6ee4..09e752a 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -558,6 +558,9 @@
// Estimated capture start time in NTP time in ms.
int64_t capture_start_ntp_time_ms = -1;
+ // First frame received to first frame decoded latency.
+ int64_t first_frame_received_to_decoded_ms = -1;
+
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
absl::optional<webrtc::TimingFrameInfo> timing_frame_info;