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Add receive bitrate UMA stats.

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Review URL: https://codereview.webrtc.org/1440603002

Cr-Commit-Position: refs/heads/master@{#10605}
This commit is contained in:
stefan 2015-11-11 10:13:02 -08:00 committed by Commit bot
parent 4dc941128f
commit 91d926038f
2 changed files with 68 additions and 5 deletions
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64
webrtc/call/call.cc
View File

@ -33,6 +33,7 @@
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/video/video_receive_stream.h"
@ -105,6 +106,10 @@ class Call : public webrtc::Call, public PacketReceiver {
return nullptr;
}
void UpdateHistograms();
const Clock* const clock_;
const int num_cpu_cores_;
const rtc::scoped_ptr<ProcessThread> module_process_thread_;
const rtc::scoped_ptr<CallStats> call_stats_;
@ -135,6 +140,14 @@ class Call : public webrtc::Call, public PacketReceiver {
RtcEventLog* event_log_ = nullptr;
// The RateTrackers are only accessed (exclusively) from DeliverRtp or
// DeliverRtcp, and from the destructor, and therefore doesn't need any
// explicit synchronization.
rtc::RateTracker received_video_bytes_per_sec_;
rtc::RateTracker received_audio_bytes_per_sec_;
rtc::RateTracker received_rtcp_bytes_per_sec_;
int64_t first_rtp_packet_received_ms_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal
@ -146,15 +159,21 @@ Call* Call::Create(const Call::Config& config) {
namespace internal {
Call::Call(const Call::Config& config)
: num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
: clock_(Clock::GetRealTimeClock()),
num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
call_stats_(new CallStats()),
congestion_controller_(new CongestionController(
module_process_thread_.get(), call_stats_.get())),
congestion_controller_(
new CongestionController(module_process_thread_.get(),
call_stats_.get())),
config_(config),
network_enabled_(true),
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()) {
send_crit_(RWLockWrapper::CreateRWLock()),
received_video_bytes_per_sec_(1000, 1),
received_audio_bytes_per_sec_(1000, 1),
received_rtcp_bytes_per_sec_(1000, 1),
first_rtp_packet_received_ms_(-1) {
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
@ -180,6 +199,7 @@ Call::Call(const Call::Config& config)
}
Call::~Call() {
UpdateHistograms();
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
@ -193,6 +213,35 @@ Call::~Call() {
Trace::ReturnTrace();
}
void Call::UpdateHistograms() {
if (first_rtp_packet_received_ms_ == -1)
return;
int64_t elapsed_sec =
(clock_->TimeInMilliseconds() - first_rtp_packet_received_ms_) / 1000;
if (elapsed_sec < metrics::kMinRunTimeInSeconds)
return;
int audio_bitrate_kbps =
received_audio_bytes_per_sec_.ComputeTotalRate() * 8 / 1000;
int video_bitrate_kbps =
received_video_bytes_per_sec_.ComputeTotalRate() * 8 / 1000;
int rtcp_bitrate_bps = received_rtcp_bytes_per_sec_.ComputeTotalRate() * 8;
if (video_bitrate_kbps > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
video_bitrate_kbps);
}
if (audio_bitrate_kbps > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
audio_bitrate_kbps);
}
if (rtcp_bitrate_bps > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
rtcp_bitrate_bps);
}
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.BitrateReceivedInKbps",
audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
}
PacketReceiver* Call::Receiver() {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
@ -527,6 +576,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
// Do NOT broadcast! Also make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
received_rtcp_bytes_per_sec_.AddSamples(length);
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
@ -559,12 +609,15 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
if (length < 12)
return DELIVERY_PACKET_ERROR;
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
if (first_rtp_packet_received_ms_ == -1)
first_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
ReadLockScoped read_lock(*receive_crit_);
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
received_audio_bytes_per_sec_.AddSamples(length);
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
@ -576,6 +629,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
received_video_bytes_per_sec_.AddSamples(length);
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;

9
webrtc/video/end_to_end_tests.cc
View File

@ -1943,7 +1943,16 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {
test::ClearHistograms();
RunBaseTest(&test, FakeNetworkPipe::Config());
// Delete the call for Call stats to be reported.
receiver_call_.reset();
// Verify that stats have been updated once.
EXPECT_EQ(
1, test::NumHistogramSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_EQ(1,
test::NumHistogramSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.BitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(