Fix unit for inbound RTP stat lastPacketReceivedTimestamp (s -> ms)
Both inbound RTP stats `estimatedPlayoutTimestamp` and `lastPacketReceivedTimestamp` are surfaced to JS land as `DOMHighResTimeStamp` - i.e., time values in milliseconds. This CL fixes `lastPacketReceivedTimestamp` which is incorrectly surfaced as time value in seconds. Bug: webrtc:12605 Change-Id: I290103071cca3331d2a3066b6b6b9fcb4f4fd0af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212742 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33530}
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@ -347,10 +347,8 @@ void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
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// |fir_count|, |pli_count| and |sli_count| are only valid for video and are
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// purposefully left undefined for audio.
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if (voice_receiver_info.last_packet_received_timestamp_ms) {
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inbound_audio->last_packet_received_timestamp =
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static_cast<double>(
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*voice_receiver_info.last_packet_received_timestamp_ms) /
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rtc::kNumMillisecsPerSec;
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inbound_audio->last_packet_received_timestamp = static_cast<double>(
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*voice_receiver_info.last_packet_received_timestamp_ms);
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}
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if (voice_receiver_info.estimated_playout_ntp_timestamp_ms) {
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inbound_audio->estimated_playout_timestamp = static_cast<double>(
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@ -1864,7 +1864,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
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// Set previously undefined values and "GetStats" again.
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voice_media_info.receivers[0].last_packet_received_timestamp_ms = 3000;
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expected_audio.last_packet_received_timestamp = 3.0;
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expected_audio.last_packet_received_timestamp = 3000.0;
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voice_media_info.receivers[0].estimated_playout_ntp_timestamp_ms = 4567;
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expected_audio.estimated_playout_timestamp = 4567;
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voice_media_channel->SetStats(voice_media_info);
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