Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
Partial implementation: currently only populated when a/v sync is enabled.
Bug: webrtc:7065
Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29581}
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 5f71ea3..fb79dc2 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -105,7 +105,12 @@
// Audio+Video Sync.
virtual uint32_t GetDelayEstimate() const = 0;
virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
- virtual uint32_t GetPlayoutTimestamp() const = 0;
+ virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+ int64_t* time_ms) const = 0;
+ virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+ int64_t time_ms) = 0;
+ virtual absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+ int64_t now_ms) const = 0;
// Audio quality.
// Base minimum delay sets lower bound on minimum delay value which