Add framerate for complete received frames to histogram stats:
"WebRTC.Video.CompleteFramesReceivedPerSecond".
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index 7a866d8..7a95b71 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -133,6 +133,7 @@
num_packets_(0),
num_duplicated_packets_(0),
num_discarded_packets_(0),
+ time_first_packet_ms_(0),
jitter_estimate_(clock),
inter_frame_delay_(clock_->TimeInMilliseconds()),
rtt_ms_(kDefaultRtt),
@@ -169,11 +170,28 @@
}
void VCMJitterBuffer::UpdateHistograms() {
- if (num_packets_ > 0) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
- num_discarded_packets_ * 100 / num_packets_);
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
- num_duplicated_packets_ * 100 / num_packets_);
+ if (num_packets_ <= 0) {
+ return;
+ }
+ int64_t elapsed_sec =
+ (clock_->TimeInMilliseconds() - time_first_packet_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
+ return;
+ }
+
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
+ num_discarded_packets_ * 100 / num_packets_);
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
+ num_duplicated_packets_ * 100 / num_packets_);
+
+ uint32_t total_frames = receive_statistics_[kVideoFrameKey] +
+ receive_statistics_[kVideoFrameDelta];
+ if (total_frames > 0) {
+ RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.CompleteFramesReceivedPerSecond",
+ static_cast<int>((total_frames / elapsed_sec) + 0.5f));
+ RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
+ static_cast<int>((receive_statistics_[kVideoFrameKey] * 1000.0f /
+ total_frames) + 0.5f));
}
}
@@ -191,6 +209,7 @@
num_packets_ = 0;
num_duplicated_packets_ = 0;
num_discarded_packets_ = 0;
+ time_first_packet_ms_ = 0;
// Start in a non-signaled state.
frame_event_->Reset();
@@ -540,6 +559,9 @@
CriticalSectionScoped cs(crit_sect_);
++num_packets_;
+ if (num_packets_ == 1) {
+ time_first_packet_ms_ = clock_->TimeInMilliseconds();
+ }
// Does this packet belong to an old frame?
if (last_decoded_state_.IsOldPacket(&packet)) {
// Account only for media packets.
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h
index 4918aa5..3722912 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h
@@ -17,6 +17,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
@@ -201,6 +202,7 @@
VCMFrameBuffer** frame,
FrameList** frame_list)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
// Returns true if |frame| is continuous in |decoding_state|, not taking
// decodable frames into account.
bool IsContinuousInState(const VCMFrameBuffer& frame,
@@ -281,7 +283,7 @@
uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
- void UpdateHistograms();
+ void UpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Clock* clock_;
// If we are running (have started) or not.
@@ -311,11 +313,13 @@
// Number of packets in a row that have been too old.
int num_consecutive_old_packets_;
// Number of packets received.
- int num_packets_;
+ int num_packets_ GUARDED_BY(crit_sect_);
// Number of duplicated packets received.
- int num_duplicated_packets_;
+ int num_duplicated_packets_ GUARDED_BY(crit_sect_);
// Number of packets discarded by the jitter buffer.
- int num_discarded_packets_;
+ int num_discarded_packets_ GUARDED_BY(crit_sect_);
+ // Time when first packet is received.
+ int64_t time_first_packet_ms_ GUARDED_BY(crit_sect_);
// Jitter estimation.
// Filter for estimating jitter.
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
index 5d5e29a..e5561d5 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
@@ -111,10 +111,10 @@
uint32_t timestamp_;
int size_;
uint8_t data_[1500];
- scoped_ptr<VCMJitterBuffer> jitter_buffer_;
scoped_ptr<VCMPacket> packet_;
scoped_ptr<SimulatedClock> clock_;
NullEventFactory event_factory_;
+ scoped_ptr<VCMJitterBuffer> jitter_buffer_;
};
diff --git a/webrtc/system_wrappers/interface/metrics.h b/webrtc/system_wrappers/interface/metrics.h
index 0390140..36da4cd 100644
--- a/webrtc/system_wrappers/interface/metrics.h
+++ b/webrtc/system_wrappers/interface/metrics.h
@@ -104,6 +104,9 @@
namespace webrtc {
namespace metrics {
+// Time that should have elapsed for stats that are gathered once per call.
+enum { kMinRunTimeInSeconds = 10 };
+
class Histogram;
// Functions for getting pointer to histogram (constructs or finds the named
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 8679c17..e6fbf3c 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -236,10 +236,9 @@
}
void ViEChannel::UpdateHistograms() {
- const float kMinCallLengthInMinutes = 0.5f;
float elapsed_minutes =
(Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
- if (elapsed_minutes < kMinCallLengthInMinutes) {
+ if (elapsed_minutes < metrics::kMinRunTimeInSeconds / 60.0f) {
return;
}
RtcpPacketTypeCounter rtcp_sent;
@@ -272,15 +271,6 @@
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
rtcp_sent.pli_packets / elapsed_minutes);
- webrtc::VCMFrameCount frames;
- if (vcm_->ReceivedFrameCount(frames) == VCM_OK) {
- uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
- if (total_frames > 0) {
- RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
- static_cast<int>((frames.numKeyFrames * 1000.0f / total_frames) +
- 0.5f));
- }
- }
StreamDataCounters data;
StreamDataCounters rtx_data;
GetReceiveStreamDataCounters(&data, &rtx_data);
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 8bd40df..1d6b816 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -241,10 +241,9 @@
}
void ViEEncoder::UpdateHistograms() {
- const float kMinCallLengthInMinutes = 0.5f;
- float elapsed_minutes =
- (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
- if (elapsed_minutes < kMinCallLengthInMinutes) {
+ int64_t elapsed_sec =
+ (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
return;
}
webrtc::VCMFrameCount frames;