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Add logging of delay-based bandwidth estimate.

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BUG=webrtc:6423

Review-Url: https://codereview.webrtc.org/2695923004
Cr-Commit-Position: refs/heads/master@{#16663}
This commit is contained in:
terelius
2017-02-17 03:38:28 -08:00
committed by Commit bot
parent 9c997a3b9e
commit 0baf55d23b
22 changed files with 305 additions and 22 deletions
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19
webrtc/logging/rtc_event_log/rtc_event_log.proto
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@ -55,6 +55,9 @@ message Event {
// optional - but required if type == BWE_PACKET_LOSS_EVENT
optional BwePacketLossEvent bwe_packet_loss_event = 6;
// optional - but required if type == BWE_PACKET_DELAY_EVENT
optional BwePacketDelayEvent bwe_packet_delay_event = 7;
// optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
optional VideoReceiveConfig video_receiver_config = 8;
@ -117,6 +120,20 @@ message BwePacketLossEvent {
optional int32 total_packets = 3;
}
message BwePacketDelayEvent {
enum DetectorState {
BWE_NORMAL = 0;
BWE_UNDERUSING = 1;
BWE_OVERUSING = 2;
}
// required - Bandwidth estimate (in bps) after the update.
optional int32 bitrate = 1;
// required - The state of the overuse detector.
optional DetectorState detector_state = 2;
}
// TODO(terelius): Video and audio streams could in principle share SSRC,
// so identifying a stream based only on SSRC might not work.
// It might be better to use a combination of SSRC and media type
@ -251,4 +268,4 @@ message AudioNetworkAdaptation {
// Number of audio channels that each encoded packet consists of.
optional uint32 num_channels = 6;
}
}