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Remove unused non-standard capture stats.

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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}
This commit is contained in:
Peter Boström 2015-03-27 10:01:02 +01:00
parent 3954e1dfe1
commit 8ed6a4bba4
5 changed files with 1 additions and 16 deletions
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3
talk/app/webrtc/statscollector.cc
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@ -223,9 +223,6 @@ void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
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 },

4
talk/app/webrtc/statstypes.cc
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@ -422,10 +422,6 @@ const char* StatsReport::Value::display_name() const {
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*.

2
talk/app/webrtc/statstypes.h
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@ -138,8 +138,6 @@ class StatsReport {
kStatsValueNameAvgEncodeMs,
kStatsValueNameBandwidthLimitedResolution,
kStatsValueNameBucketDelay,
kStatsValueNameCaptureJitterMs,
kStatsValueNameCaptureQueueDelayMsPerS,
kStatsValueNameCaptureStartNtpTimeMs,
kStatsValueNameCandidateIPAddress,
kStatsValueNameCandidateNetworkType,

6
talk/media/base/mediachannel.h
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@ -832,10 +832,8 @@ struct VideoSenderInfo : public MediaSenderInfo {
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 @@ struct VideoSenderInfo : public MediaSenderInfo {
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;

2
talk/media/webrtc/webrtcvideoengine.cc
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@ -2639,10 +2639,8 @@ bool WebRtcVideoMediaChannel::GetStats(VideoMediaInfo* info) {
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;