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Revert "Record audio/video bytes sent in analyzer stream stats."

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This reverts commit d978cb43c238ca24b2320acd7b656f446b906101.

Reason for revert: It breaks perf tests: https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Android32%20(L%20Nexus4)/1561

Original change's description:
> Record audio/video bytes sent in analyzer stream stats.
> 
> For each SSRC report, record the number of bytes sent for that stream
> and expose them in analyzer stats.  These numbers can be used to
> determine useful metrics such as total media throughput (by adding the
> bytes sent for all streams) and overhead (by subtracting that amount
> from the total bytes sent to the network).
> 
> Bug: webrtc:9719
> Change-Id: I977bbd40acdd0a1ec64763ddd55a642b9a50f309
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28637}

TBR=mbonadei@webrtc.org,mellem@webrtc.org,titovartem@webrtc.org

Change-Id: I3e46307dd6ef121b9377b93fc8d9fa788245ea5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146605
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28646}
This commit is contained in:
Mirko Bonadei 2019-07-23 13:24:26 +00:00 committed by Commit Bot
parent 61689ab063
commit cfefa0aef3
4 changed files with 3 additions and 47 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

21
test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
View File

@ -45,24 +45,10 @@ void DefaultAudioQualityAnalyzer::OnStatsReports(
if (strcmp(media_type->static_string_val(), kStatsAudioMediaType) != 0) {
continue;
}
if (stats_report->FindValue(
webrtc::StatsReport::kStatsValueNameBytesSent)) {
// If kStatsValueNameBytesSent is present, it means it's a send stream.
// All we need from a send stream is bytes sent.
const webrtc::StatsReport::Value* bytes_sent = stats_report->FindValue(
StatsReport::StatsValueName::kStatsValueNameBytesSent);
const webrtc::StatsReport::Value* report_track_id =
stats_report->FindValue(
StatsReport::StatsValueName::kStatsValueNameTrackId);
rtc::CritScope crit(&lock_);
// Note: outgoing streams have their "stream label" directly in the
// report's track id field. There is no need to look it up using
// GetStreamLabelFromStatsReport(), and in fact doing so will crash.
AudioStreamStats& audio_stream_stats =
streams_stats_[report_track_id->string_val()];
audio_stream_stats.bytes_sent = bytes_sent->int64_val();
// If kStatsValueNameBytesSent is present, it means it's a send stream,
// but we need audio metrics for receive stream, so skip it.
continue;
}
@ -126,9 +112,6 @@ void DefaultAudioQualityAnalyzer::Stop() {
item.second.speech_expand_rate, "unitless");
ReportResult("preferred_buffer_size_ms", item.first,
item.second.preferred_buffer_size_ms, "ms");
test::PrintResult("bytes_sent", "", GetTestCaseName(item.first),
item.second.bytes_sent, "sizeInBytes",
/*important=*/false);
}
}

1
test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h
View File

@ -29,7 +29,6 @@ struct AudioStreamStats {
SamplesStatsCounter preemptive_rate;
SamplesStatsCounter speech_expand_rate;
SamplesStatsCounter preferred_buffer_size_ms;
int64_t bytes_sent;
};
// TODO(bugs.webrtc.org/10430): Migrate to the new GetStats as soon as

26
test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
View File

@ -27,7 +27,6 @@ constexpr int kMaxActiveComparisons = 10;
constexpr int kFreezeThresholdMs = 150;
constexpr int kMicrosPerSecond = 1000000;
constexpr int kBitsInByte = 8;
constexpr char kStatsVideoMediaType[] = "video";
void LogFrameCounters(const std::string& name, const FrameCounters& counters) {
RTC_LOG(INFO) << "[" << name << "] Captured : " << counters.captured;
@ -374,28 +373,7 @@ void DefaultVideoQualityAnalyzer::OnStatsReports(
const std::string& pc_label,
const StatsReports& stats_reports) {
for (const StatsReport* stats_report : stats_reports) {
// Record the number of video bytes sent from outgoing SSRC reports.
if (stats_report->type() == StatsReport::StatsType::kStatsReportTypeSsrc &&
strcmp(stats_report
->FindValue(
StatsReport::StatsValueName::kStatsValueNameMediaType)
->static_string_val(),
kStatsVideoMediaType) == 0 &&
stats_report->FindValue(StatsReport::kStatsValueNameBytesSent)) {
const webrtc::StatsReport::Value* bytes_sent = stats_report->FindValue(
StatsReport::StatsValueName::kStatsValueNameBytesSent);
const webrtc::StatsReport::Value* track_id = stats_report->FindValue(
StatsReport::StatsValueName::kStatsValueNameTrackId);
rtc::CritScope crit(&comparison_lock_);
// Note: outgoing streams have their "stream label" directly in the
// report's track id field. There is no need to look it up using
// GetStreamLabelFromStatsReport(), and in fact doing so will crash.
StreamStats& stream_stats = stream_stats_[track_id->string_val()];
stream_stats.bytes_sent = bytes_sent->int64_val();
}
// The only other stats collected by this analyzer are present in
// The only stats collected by this analyzer are present in
// kStatsReportTypeBwe reports, so all other reports are just ignored.
if (stats_report->type() != StatsReport::StatsType::kStatsReportTypeBwe) {
continue;
@ -651,8 +629,6 @@ void DefaultVideoQualityAnalyzer::ReportResults(
/*important=*/false);
ReportResult("max_skipped", test_case_name, stats.skipped_between_rendered,
"unitless");
test::PrintResult("bytes_sent", "", test_case_name, stats.bytes_sent,
"sizeInBytes", /*important=*/false);
}
void DefaultVideoQualityAnalyzer::ReportResult(

2
test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
View File

@ -96,8 +96,6 @@ struct StreamStats {
int64_t dropped_by_encoder = 0;
int64_t dropped_before_encoder = 0;
int64_t bytes_sent = 0;
};
struct AnalyzerStats {