Collapse most stdout spammy output of webrtc_perf_tests with PrintResultList
BUG=webrtc:5032
NOTRY=true
Review-Url: https://codereview.webrtc.org/2030073002
Cr-Commit-Position: refs/heads/master@{#13037}
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 11861ac..50e1a62 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -11,7 +11,6 @@
#include <algorithm>
#include <limits>
#include <memory>
-#include <sstream>
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
@@ -100,14 +99,7 @@
int64_t now_ms = clock_->TimeInMilliseconds();
- std::stringstream ss;
- ss << stats.sync_offset_ms;
- webrtc::test::PrintResult("stream_offset",
- "",
- "synchronization",
- ss.str(),
- "ms",
- false);
+ sync_offset_ms_list_.push_back(stats.sync_offset_ms);
int64_t time_since_creation = now_ms - creation_time_ms_;
// During the first couple of seconds audio and video can falsely be
// estimated as being synchronized. We don't want to trigger on those.
@@ -133,12 +125,19 @@
receive_stream_ = receive_stream;
}
+ void PrintResults() {
+ test::PrintResultList("stream_offset", "", "synchronization",
+ test::ValuesToString(sync_offset_ms_list_), "ms",
+ false);
+ }
+
private:
Clock* const clock_;
const int64_t creation_time_ms_;
int64_t first_time_in_sync_;
rtc::CriticalSection crit_;
VideoReceiveStream* receive_stream_ GUARDED_BY(crit_);
+ std::vector<int> sync_offset_ms_list_;
};
void CallPerfTest::TestAudioVideoSync(FecMode fec,
@@ -300,6 +299,7 @@
VoiceEngine::Delete(voice_engine);
+ observer.PrintResults();
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
}
@@ -387,11 +387,8 @@
uint32_t real_capture_timestamp = iter->second;
int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
time_offset_ms = time_offset_ms / 90;
- std::stringstream ss;
- ss << time_offset_ms;
+ time_offset_ms_list_.push_back(time_offset_ms);
- webrtc::test::PrintResult(
- "capture_ntp_time", "", "real - estimated", ss.str(), "ms", true);
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
}
@@ -434,6 +431,9 @@
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
"estimated capture NTP time to be "
"within bounds.";
+ test::PrintResultList("capture_ntp_time", "", "real - estimated",
+ test::ValuesToString(time_offset_ms_list_), "ms",
+ true);
}
rtc::CriticalSection crit_;
@@ -448,6 +448,7 @@
uint32_t rtp_start_timestamp_;
typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
+ std::vector<int> time_offset_ms_list_;
} test(net_config, threshold_ms, start_time_ms, run_time_ms);
RunBaseTest(&test);
@@ -543,14 +544,7 @@
int bitrate_kbps =
stats.substreams.begin()->second.total_bitrate_bps / 1000;
if (bitrate_kbps > 0) {
- test::PrintResult(
- "bitrate_stats_",
- (pad_to_min_bitrate_ ? "min_transmit_bitrate"
- : "without_min_transmit_bitrate"),
- "bitrate_kbps",
- static_cast<size_t>(bitrate_kbps),
- "kbps",
- false);
+ bitrate_kbps_list.push_back(bitrate_kbps);
if (pad_to_min_bitrate_) {
if (bitrate_kbps > kMinAcceptableTransmitBitrate &&
bitrate_kbps < kMaxAcceptableTransmitBitrate) {
@@ -592,11 +586,18 @@
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
+ test::PrintResultList(
+ "bitrate_stats_",
+ (pad_to_min_bitrate_ ? "min_transmit_bitrate"
+ : "without_min_transmit_bitrate"),
+ "bitrate_kbps", test::ValuesToString(bitrate_kbps_list), "kbps",
+ false);
}
VideoSendStream* send_stream_;
const bool pad_to_min_bitrate_;
int num_bitrate_observations_in_range_;
+ std::vector<size_t> bitrate_kbps_list;
} test(pad_to_min_bitrate);
fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
