Add histograms for receive statistics:
- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/video_coding/main/source/generic_decoder.cc b/modules/video_coding/main/source/generic_decoder.cc
index cb0faf9..4f8a7ca 100644
--- a/modules/video_coding/main/source/generic_decoder.cc
+++ b/modules/video_coding/main/source/generic_decoder.cc
@@ -68,7 +68,8 @@
_timing.StopDecodeTimer(
decodedImage.timestamp(),
frameInfo->decodeStartTimeMs,
- _clock->TimeInMilliseconds());
+ _clock->TimeInMilliseconds(),
+ frameInfo->renderTimeMs);
if (callback != NULL)
{
diff --git a/modules/video_coding/main/source/timing.cc b/modules/video_coding/main/source/timing.cc
index 2ec149c..0e2fddf 100644
--- a/modules/video_coding/main/source/timing.cc
+++ b/modules/video_coding/main/source/timing.cc
@@ -13,6 +13,7 @@
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
@@ -30,7 +31,11 @@
jitter_delay_ms_(0),
current_delay_ms_(0),
last_decode_ms_(0),
- prev_frame_timestamp_(0) {
+ prev_frame_timestamp_(0),
+ num_decoded_frames_(0),
+ num_delayed_decoded_frames_(0),
+ first_decoded_frame_ms_(-1),
+ sum_missed_render_deadline_ms_(0) {
if (master_timing == NULL) {
master_ = true;
ts_extrapolator_ = new TimestampExtrapolator(clock_->TimeInMilliseconds());
@@ -40,12 +45,34 @@
}
VCMTiming::~VCMTiming() {
+ UpdateHistograms();
if (master_) {
delete ts_extrapolator_;
}
delete crit_sect_;
}
+void VCMTiming::UpdateHistograms() const {
+ CriticalSectionScoped cs(crit_sect_);
+ if (num_decoded_frames_ == 0) {
+ return;
+ }
+ int64_t elapsed_sec =
+ (clock_->TimeInMilliseconds() - first_decoded_frame_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
+ return;
+ }
+ RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond",
+ static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
+ num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
+ if (num_delayed_decoded_frames_ > 0) {
+ RTC_HISTOGRAM_COUNTS_1000(
+ "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
+ sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
+ }
+}
+
void VCMTiming::Reset() {
CriticalSectionScoped cs(crit_sect_);
ts_extrapolator_->Reset(clock_->TimeInMilliseconds());
@@ -139,11 +166,23 @@
int32_t VCMTiming::StopDecodeTimer(uint32_t time_stamp,
int64_t start_time_ms,
- int64_t now_ms) {
+ int64_t now_ms,
+ int64_t render_time_ms) {
CriticalSectionScoped cs(crit_sect_);
int32_t time_diff_ms = codec_timer_.StopTimer(start_time_ms, now_ms);
assert(time_diff_ms >= 0);
last_decode_ms_ = time_diff_ms;
+
+ // Update stats.
+ ++num_decoded_frames_;
+ if (num_decoded_frames_ == 1) {
+ first_decoded_frame_ms_ = now_ms;
+ }
+ int time_until_rendering_ms = render_time_ms - render_delay_ms_ - now_ms;
+ if (time_until_rendering_ms < 0) {
+ sum_missed_render_deadline_ms_ += -time_until_rendering_ms;
+ ++num_delayed_decoded_frames_;
+ }
return 0;
}
diff --git a/modules/video_coding/main/source/timing.h b/modules/video_coding/main/source/timing.h
index bb44a93..61c0273 100644
--- a/modules/video_coding/main/source/timing.h
+++ b/modules/video_coding/main/source/timing.h
@@ -59,7 +59,8 @@
// or when the decoded frame callback is called.
int32_t StopDecodeTimer(uint32_t time_stamp,
int64_t start_time_ms,
- int64_t now_ms);
+ int64_t now_ms,
+ int64_t render_time_ms);
// Used to report that a frame is passed to decoding. Updates the timestamp
// filter which is used to map between timestamps and receiver system time.
@@ -101,6 +102,8 @@
uint32_t TargetDelayInternal() const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
private:
+ void UpdateHistograms() const;
+
CriticalSectionWrapper* crit_sect_;
Clock* const clock_;
bool master_ GUARDED_BY(crit_sect_);
@@ -112,6 +115,12 @@
uint32_t current_delay_ms_ GUARDED_BY(crit_sect_);
int last_decode_ms_ GUARDED_BY(crit_sect_);
uint32_t prev_frame_timestamp_ GUARDED_BY(crit_sect_);
+
+ // Statistics.
+ size_t num_decoded_frames_ GUARDED_BY(crit_sect_);
+ size_t num_delayed_decoded_frames_ GUARDED_BY(crit_sect_);
+ int64_t first_decoded_frame_ms_ GUARDED_BY(crit_sect_);
+ uint64_t sum_missed_render_deadline_ms_ GUARDED_BY(crit_sect_);
};
} // namespace webrtc
diff --git a/modules/video_coding/main/source/timing_unittest.cc b/modules/video_coding/main/source/timing_unittest.cc
index dde26a9..32df31c 100644
--- a/modules/video_coding/main/source/timing_unittest.cc
+++ b/modules/video_coding/main/source/timing_unittest.cc
@@ -86,7 +86,8 @@
int64_t startTimeMs = clock.TimeInMilliseconds();
clock.AdvanceTimeMilliseconds(10);
timing.StopDecodeTimer(timeStamp, startTimeMs,
- clock.TimeInMilliseconds());
+ clock.TimeInMilliseconds(), timing.RenderTimeMs(
+ timeStamp, clock.TimeInMilliseconds()));
timeStamp += 90000 / 25;
clock.AdvanceTimeMilliseconds(1000 / 25 - 10);
timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
