Remove unused non-standard capture stats.
Removes 'googCaptureJitterMs' and 'googCaptureQueueDelayMsPerS' from
talk/. The overuse-detection method used is based on encoding time,
so these stats aren't useful enough to warrant having them showing up in
GetStats().
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50469004
Cr-Commit-Position: refs/heads/master@{#8874}
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index f76245d..fb15952 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -223,9 +223,6 @@
const IntForAdd ints[] = {
{ StatsReport::kStatsValueNameAdaptationChanges, info.adapt_changes },
{ StatsReport::kStatsValueNameAvgEncodeMs, info.avg_encode_ms },
- { StatsReport::kStatsValueNameCaptureJitterMs, info.capture_jitter_ms },
- { StatsReport::kStatsValueNameCaptureQueueDelayMsPerS,
- info.capture_queue_delay_ms_per_s },
{ StatsReport::kStatsValueNameEncodeUsagePercent,
info.encode_usage_percent },
{ StatsReport::kStatsValueNameFirsReceived, info.firs_rcvd },
diff --git a/talk/app/webrtc/statstypes.cc b/talk/app/webrtc/statstypes.cc
index bca7280..582d203 100644
--- a/talk/app/webrtc/statstypes.cc
+++ b/talk/app/webrtc/statstypes.cc
@@ -422,10 +422,6 @@
return "googBucketDelay";
case kStatsValueNameBandwidthLimitedResolution:
return "googBandwidthLimitedResolution";
- case kStatsValueNameCaptureJitterMs:
- return "googCaptureJitterMs";
- case kStatsValueNameCaptureQueueDelayMsPerS:
- return "googCaptureQueueDelayMsPerS";
// Candidate related attributes. Values are taken from
// http://w3c.github.io/webrtc-stats/#rtcstatstype-enum*.
diff --git a/talk/app/webrtc/statstypes.h b/talk/app/webrtc/statstypes.h
index de4fb5f..9df1de7 100644
--- a/talk/app/webrtc/statstypes.h
+++ b/talk/app/webrtc/statstypes.h
@@ -138,8 +138,6 @@
kStatsValueNameAvgEncodeMs,
kStatsValueNameBandwidthLimitedResolution,
kStatsValueNameBucketDelay,
- kStatsValueNameCaptureJitterMs,
- kStatsValueNameCaptureQueueDelayMsPerS,
kStatsValueNameCaptureStartNtpTimeMs,
kStatsValueNameCandidateIPAddress,
kStatsValueNameCandidateNetworkType,
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 70fd7f0..12bb519 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -832,10 +832,8 @@
preferred_bitrate(0),
adapt_reason(0),
adapt_changes(0),
- capture_jitter_ms(0),
avg_encode_ms(0),
- encode_usage_percent(0),
- capture_queue_delay_ms_per_s(0) {
+ encode_usage_percent(0) {
}
std::vector<SsrcGroup> ssrc_groups;
@@ -853,10 +851,8 @@
int preferred_bitrate;
int adapt_reason;
int adapt_changes;
- int capture_jitter_ms;
int avg_encode_ms;
int encode_usage_percent;
- int capture_queue_delay_ms_per_s;
VariableInfo<int> adapt_frame_drops;
VariableInfo<int> effects_frame_drops;
VariableInfo<double> capturer_frame_time;
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 4bb9bbe..287de72 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -2639,10 +2639,8 @@
webrtc::CpuOveruseMetrics metrics;
engine()->vie()->base()->GetCpuOveruseMetrics(channel_id, &metrics);
- sinfo.capture_jitter_ms = metrics.capture_jitter_ms;
sinfo.avg_encode_ms = metrics.avg_encode_time_ms;
sinfo.encode_usage_percent = metrics.encode_usage_percent;
- sinfo.capture_queue_delay_ms_per_s = metrics.capture_queue_delay_ms_per_s;
webrtc::RtcpPacketTypeCounter rtcp_sent;
webrtc::RtcpPacketTypeCounter rtcp_received;
