webrtc_m130/pc/rtc_stats_integrationtest.cc

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

1260 lines
58 KiB
C++
Raw Normal View History

/*
* Copyright 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdint.h>
#include <string.h>
#include <algorithm>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/data_channel_interface.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
#include "pc/rtc_stats_traversal.h"
#include "pc/test/peer_connection_test_wrapper.h"
#include "pc/test/rtc_stats_obtainer.h"
#include "rtc_base/checks.h"
#include "rtc_base/event_tracer.h"
#include "rtc_base/gunit.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/virtual_socket_server.h"
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::Contains;
namespace webrtc {
namespace {
const int64_t kGetStatsTimeoutMs = 10000;
const unsigned char* GetCategoryEnabledHandler(const char* name) {
if (strcmp("webrtc_stats", name) != 0) {
return reinterpret_cast<const unsigned char*>("");
}
return reinterpret_cast<const unsigned char*>(name);
}
class RTCStatsReportTraceListener {
public:
static void SetUp() {
if (!traced_report_)
traced_report_ = new RTCStatsReportTraceListener();
traced_report_->last_trace_ = "";
SetupEventTracer(&GetCategoryEnabledHandler,
&RTCStatsReportTraceListener::AddTraceEventHandler);
}
static const std::string& last_trace() {
RTC_DCHECK(traced_report_);
return traced_report_->last_trace_;
}
private:
static void AddTraceEventHandler(
char phase,
const unsigned char* category_enabled,
const char* name,
unsigned long long id, // NOLINT(runtime/int)
int num_args,
const char** arg_names,
const unsigned char* arg_types,
const unsigned long long* arg_values, // NOLINT(runtime/int)
unsigned char flags) {
RTC_DCHECK(traced_report_);
EXPECT_STREQ("webrtc_stats",
reinterpret_cast<const char*>(category_enabled));
EXPECT_STREQ("webrtc_stats", name);
EXPECT_EQ(1, num_args);
EXPECT_STREQ("report", arg_names[0]);
EXPECT_EQ(TRACE_VALUE_TYPE_COPY_STRING, arg_types[0]);
traced_report_->last_trace_ = reinterpret_cast<const char*>(arg_values[0]);
}
static RTCStatsReportTraceListener* traced_report_;
std::string last_trace_;
};
RTCStatsReportTraceListener* RTCStatsReportTraceListener::traced_report_ =
nullptr;
class RTCStatsIntegrationTest : public ::testing::Test {
public:
RTCStatsIntegrationTest()
: network_thread_(new rtc::Thread(&virtual_socket_server_)),
worker_thread_(rtc::Thread::Create()) {
RTCStatsReportTraceListener::SetUp();
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"caller", network_thread_.get(), worker_thread_.get());
callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"callee", network_thread_.get(), worker_thread_.get());
}
void StartCall() {
// Create PeerConnections and "connect" sigslots
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:1.1.1.1:3478";
config.servers.push_back(ice_server);
EXPECT_TRUE(caller_->CreatePc(config, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory()));
EXPECT_TRUE(callee_->CreatePc(config, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory()));
PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
// Get user media for audio and video
caller_->GetAndAddUserMedia(true, cricket::AudioOptions(), true);
callee_->GetAndAddUserMedia(true, cricket::AudioOptions(), true);
// Create data channels
DataChannelInit init;
caller_->CreateDataChannel("data", init);
callee_->CreateDataChannel("data", init);
// Negotiate and wait for call to establish
caller_->CreateOffer(PeerConnectionInterface::RTCOfferAnswerOptions());
caller_->WaitForCallEstablished();
callee_->WaitForCallEstablished();
}
rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCaller() {
return GetStats(caller_->pc());
}
2018-03-20 13:24:20 +01:00
rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCaller(
rtc::scoped_refptr<RtpSenderInterface> selector) {
return GetStats(caller_->pc(), selector);
}
rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCaller(
rtc::scoped_refptr<RtpReceiverInterface> selector) {
return GetStats(caller_->pc(), selector);
}
rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCallee() {
return GetStats(callee_->pc());
}
2018-03-20 13:24:20 +01:00
rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCallee(
rtc::scoped_refptr<RtpSenderInterface> selector) {
return GetStats(callee_->pc(), selector);
}
rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCallee(
rtc::scoped_refptr<RtpReceiverInterface> selector) {
return GetStats(callee_->pc(), selector);
}
protected:
static rtc::scoped_refptr<const RTCStatsReport> GetStats(
PeerConnectionInterface* pc) {
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
pc->GetStats(stats_obtainer);
EXPECT_TRUE_WAIT(stats_obtainer->report(), kGetStatsTimeoutMs);
return stats_obtainer->report();
}
2018-03-20 13:24:20 +01:00
template <typename T>
static rtc::scoped_refptr<const RTCStatsReport> GetStats(
PeerConnectionInterface* pc,
rtc::scoped_refptr<T> selector) {
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
pc->GetStats(selector, stats_obtainer);
EXPECT_TRUE_WAIT(stats_obtainer->report(), kGetStatsTimeoutMs);
return stats_obtainer->report();
}
// |network_thread_| uses |virtual_socket_server_| so they must be
// constructed/destructed in the correct order.
rtc::VirtualSocketServer virtual_socket_server_;
std::unique_ptr<rtc::Thread> network_thread_;
std::unique_ptr<rtc::Thread> worker_thread_;
rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
};
class RTCStatsVerifier {
public:
RTCStatsVerifier(const RTCStatsReport* report, const RTCStats* stats)
: report_(report), stats_(stats), all_tests_successful_(true) {
RTC_CHECK(report_);
RTC_CHECK(stats_);
for (const RTCStatsMemberInterface* member : stats_->Members()) {
untested_members_.insert(member);
}
}
void MarkMemberTested(const RTCStatsMemberInterface& member,
bool test_successful) {
untested_members_.erase(&member);
all_tests_successful_ &= test_successful;
}
void TestMemberIsDefined(const RTCStatsMemberInterface& member) {
EXPECT_TRUE(member.is_defined())
<< stats_->type() << "." << member.name() << "[" << stats_->id()
<< "] was undefined.";
MarkMemberTested(member, member.is_defined());
}
void TestMemberIsUndefined(const RTCStatsMemberInterface& member) {
EXPECT_FALSE(member.is_defined())
<< stats_->type() << "." << member.name() << "[" << stats_->id()
<< "] was defined (" << member.ValueToString() << ").";
MarkMemberTested(member, !member.is_defined());
}
template <typename T>
void TestMemberIsPositive(const RTCStatsMemberInterface& member) {
EXPECT_TRUE(member.is_defined())
<< stats_->type() << "." << member.name() << "[" << stats_->id()
<< "] was undefined.";
if (!member.is_defined()) {
MarkMemberTested(member, false);
return;
}
bool is_positive = *member.cast_to<RTCStatsMember<T>>() > T(0);
EXPECT_TRUE(is_positive)
<< stats_->type() << "." << member.name() << "[" << stats_->id()
<< "] was not positive (" << member.ValueToString() << ").";
MarkMemberTested(member, is_positive);
}
template <typename T>
void TestMemberIsNonNegative(const RTCStatsMemberInterface& member) {
EXPECT_TRUE(member.is_defined())
<< stats_->type() << "." << member.name() << "[" << stats_->id()
<< "] was undefined.";
if (!member.is_defined()) {
MarkMemberTested(member, false);
return;
}
bool is_non_negative = *member.cast_to<RTCStatsMember<T>>() >= T(0);
EXPECT_TRUE(is_non_negative)
<< stats_->type() << "." << member.name() << "[" << stats_->id()
<< "] was not non-negative (" << member.ValueToString() << ").";
MarkMemberTested(member, is_non_negative);
}
void TestMemberIsIDReference(const RTCStatsMemberInterface& member,
const char* expected_type) {
TestMemberIsIDReference(member, expected_type, false);
}
void TestMemberIsOptionalIDReference(const RTCStatsMemberInterface& member,
const char* expected_type) {
TestMemberIsIDReference(member, expected_type, true);
}
bool ExpectAllMembersSuccessfullyTested() {
if (untested_members_.empty())
return all_tests_successful_;
for (const RTCStatsMemberInterface* member : untested_members_) {
EXPECT_TRUE(false) << stats_->type() << "." << member->name() << "["
<< stats_->id() << "] was not tested.";
}
return false;
}
private:
void TestMemberIsIDReference(const RTCStatsMemberInterface& member,
const char* expected_type,
bool optional) {
if (optional && !member.is_defined()) {
MarkMemberTested(member, true);
return;
}
bool valid_reference = false;
if (member.is_defined()) {
if (member.type() == RTCStatsMemberInterface::kString) {
// A single ID.
const RTCStatsMember<std::string>& id =
member.cast_to<RTCStatsMember<std::string>>();
const RTCStats* referenced_stats = report_->Get(*id);
valid_reference =
referenced_stats && referenced_stats->type() == expected_type;
} else if (member.type() == RTCStatsMemberInterface::kSequenceString) {
// A vector of IDs.
valid_reference = true;
const RTCStatsMember<std::vector<std::string>>& ids =
member.cast_to<RTCStatsMember<std::vector<std::string>>>();
for (const std::string& id : *ids) {
const RTCStats* referenced_stats = report_->Get(id);
if (!referenced_stats || referenced_stats->type() != expected_type) {
valid_reference = false;
break;
}
}
}
}
EXPECT_TRUE(valid_reference)
<< stats_->type() << "." << member.name()
<< " is not a reference to an "
"existing dictionary of type "
<< expected_type << " (value: "
<< (member.is_defined() ? member.ValueToString() : "null") << ").";
MarkMemberTested(member, valid_reference);
}
rtc::scoped_refptr<const RTCStatsReport> report_;
const RTCStats* stats_;
std::set<const RTCStatsMemberInterface*> untested_members_;
bool all_tests_successful_;
};
class RTCStatsReportVerifier {
public:
static std::set<const char*> StatsTypes() {
std::set<const char*> stats_types;
stats_types.insert(RTCCertificateStats::kType);
stats_types.insert(RTCCodecStats::kType);
stats_types.insert(RTCDataChannelStats::kType);
stats_types.insert(RTCIceCandidatePairStats::kType);
stats_types.insert(RTCLocalIceCandidateStats::kType);
stats_types.insert(RTCRemoteIceCandidateStats::kType);
stats_types.insert(RTCMediaStreamStats::kType);
stats_types.insert(RTCMediaStreamTrackStats::kType);
stats_types.insert(RTCPeerConnectionStats::kType);
stats_types.insert(RTCInboundRTPStreamStats::kType);
stats_types.insert(RTCOutboundRTPStreamStats::kType);
stats_types.insert(RTCTransportStats::kType);
return stats_types;
}
explicit RTCStatsReportVerifier(const RTCStatsReport* report)
: report_(report) {}
void VerifyReport(std::vector<const char*> allowed_missing_stats) {
std::set<const char*> missing_stats = StatsTypes();
bool verify_successful = true;
std::vector<const RTCTransportStats*> transport_stats =
report_->GetStatsOfType<RTCTransportStats>();
EXPECT_EQ(transport_stats.size(), 1U);
std::string selected_candidate_pair_id =
*transport_stats[0]->selected_candidate_pair_id;
for (const RTCStats& stats : *report_) {
missing_stats.erase(stats.type());
if (stats.type() == RTCCertificateStats::kType) {
verify_successful &=
VerifyRTCCertificateStats(stats.cast_to<RTCCertificateStats>());
} else if (stats.type() == RTCCodecStats::kType) {
verify_successful &=
VerifyRTCCodecStats(stats.cast_to<RTCCodecStats>());
} else if (stats.type() == RTCDataChannelStats::kType) {
verify_successful &=
VerifyRTCDataChannelStats(stats.cast_to<RTCDataChannelStats>());
} else if (stats.type() == RTCIceCandidatePairStats::kType) {
verify_successful &= VerifyRTCIceCandidatePairStats(
stats.cast_to<RTCIceCandidatePairStats>(),
stats.id() == selected_candidate_pair_id);
} else if (stats.type() == RTCLocalIceCandidateStats::kType) {
verify_successful &= VerifyRTCLocalIceCandidateStats(
stats.cast_to<RTCLocalIceCandidateStats>());
} else if (stats.type() == RTCRemoteIceCandidateStats::kType) {
verify_successful &= VerifyRTCRemoteIceCandidateStats(
stats.cast_to<RTCRemoteIceCandidateStats>());
} else if (stats.type() == RTCMediaStreamStats::kType) {
verify_successful &=
VerifyRTCMediaStreamStats(stats.cast_to<RTCMediaStreamStats>());
} else if (stats.type() == RTCMediaStreamTrackStats::kType) {
verify_successful &= VerifyRTCMediaStreamTrackStats(
stats.cast_to<RTCMediaStreamTrackStats>());
} else if (stats.type() == RTCPeerConnectionStats::kType) {
verify_successful &= VerifyRTCPeerConnectionStats(
stats.cast_to<RTCPeerConnectionStats>());
} else if (stats.type() == RTCInboundRTPStreamStats::kType) {
verify_successful &= VerifyRTCInboundRTPStreamStats(
stats.cast_to<RTCInboundRTPStreamStats>());
} else if (stats.type() == RTCOutboundRTPStreamStats::kType) {
verify_successful &= VerifyRTCOutboundRTPStreamStats(
stats.cast_to<RTCOutboundRTPStreamStats>());
} else if (stats.type() == RTCRemoteInboundRtpStreamStats::kType) {
verify_successful &= VerifyRTCRemoteInboundRtpStreamStats(
stats.cast_to<RTCRemoteInboundRtpStreamStats>());
} else if (stats.type() == RTCAudioSourceStats::kType) {
// RTCAudioSourceStats::kType and RTCVideoSourceStats::kType both have
// the value "media-source", but they are distinguishable with pointer
// equality (==). In JavaScript they would be distinguished with |kind|.
verify_successful &=
VerifyRTCAudioSourceStats(stats.cast_to<RTCAudioSourceStats>());
} else if (stats.type() == RTCVideoSourceStats::kType) {
// RTCAudioSourceStats::kType and RTCVideoSourceStats::kType both have
// the value "media-source", but they are distinguishable with pointer
// equality (==). In JavaScript they would be distinguished with |kind|.
verify_successful &=
VerifyRTCVideoSourceStats(stats.cast_to<RTCVideoSourceStats>());
} else if (stats.type() == RTCTransportStats::kType) {
verify_successful &=
VerifyRTCTransportStats(stats.cast_to<RTCTransportStats>());
} else {
EXPECT_TRUE(false) << "Unrecognized stats type: " << stats.type();
verify_successful = false;
}
}
2018-03-20 13:24:20 +01:00
for (const char* missing : missing_stats) {
if (!absl::c_linear_search(allowed_missing_stats, missing)) {
2018-03-20 13:24:20 +01:00
verify_successful = false;
EXPECT_TRUE(false) << "Missing expected stats type: " << missing;
}
}
EXPECT_TRUE(verify_successful)
<< "One or more problems with the stats. This is the report:\n"
<< report_->ToJson();
}
bool VerifyRTCCertificateStats(const RTCCertificateStats& certificate) {
RTCStatsVerifier verifier(report_, &certificate);
verifier.TestMemberIsDefined(certificate.fingerprint);
verifier.TestMemberIsDefined(certificate.fingerprint_algorithm);
verifier.TestMemberIsDefined(certificate.base64_certificate);
verifier.TestMemberIsOptionalIDReference(certificate.issuer_certificate_id,
RTCCertificateStats::kType);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCCodecStats(const RTCCodecStats& codec) {
RTCStatsVerifier verifier(report_, &codec);
verifier.TestMemberIsDefined(codec.payload_type);
verifier.TestMemberIsDefined(codec.mime_type);
verifier.TestMemberIsPositive<uint32_t>(codec.clock_rate);
if (codec.mime_type->rfind("audio", 0) == 0)
verifier.TestMemberIsPositive<uint32_t>(codec.channels);
else
verifier.TestMemberIsUndefined(codec.channels);
// sdp_fmtp_line is an optional field.
verifier.MarkMemberTested(codec.sdp_fmtp_line, true);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCDataChannelStats(const RTCDataChannelStats& data_channel) {
RTCStatsVerifier verifier(report_, &data_channel);
verifier.TestMemberIsDefined(data_channel.label);
verifier.TestMemberIsDefined(data_channel.protocol);
verifier.TestMemberIsDefined(data_channel.data_channel_identifier);
verifier.TestMemberIsDefined(data_channel.state);
verifier.TestMemberIsNonNegative<uint32_t>(data_channel.messages_sent);
verifier.TestMemberIsNonNegative<uint64_t>(data_channel.bytes_sent);
verifier.TestMemberIsNonNegative<uint32_t>(data_channel.messages_received);
verifier.TestMemberIsNonNegative<uint64_t>(data_channel.bytes_received);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCIceCandidatePairStats(
const RTCIceCandidatePairStats& candidate_pair,
bool is_selected_pair) {
RTCStatsVerifier verifier(report_, &candidate_pair);
verifier.TestMemberIsIDReference(candidate_pair.transport_id,
RTCTransportStats::kType);
verifier.TestMemberIsIDReference(candidate_pair.local_candidate_id,
RTCLocalIceCandidateStats::kType);
verifier.TestMemberIsIDReference(candidate_pair.remote_candidate_id,
RTCRemoteIceCandidateStats::kType);
verifier.TestMemberIsDefined(candidate_pair.state);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.priority);
verifier.TestMemberIsDefined(candidate_pair.nominated);
verifier.TestMemberIsDefined(candidate_pair.writable);
verifier.TestMemberIsUndefined(candidate_pair.readable);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.bytes_received);
verifier.TestMemberIsNonNegative<double>(
candidate_pair.total_round_trip_time);
verifier.TestMemberIsNonNegative<double>(
candidate_pair.current_round_trip_time);
if (is_selected_pair) {
verifier.TestMemberIsNonNegative<double>(
candidate_pair.available_outgoing_bitrate);
// A pair should be nominated in order to be selected.
EXPECT_TRUE(*candidate_pair.nominated);
} else {
verifier.TestMemberIsUndefined(candidate_pair.available_outgoing_bitrate);
}
verifier.TestMemberIsUndefined(candidate_pair.available_incoming_bitrate);
verifier.TestMemberIsNonNegative<uint64_t>(
candidate_pair.requests_received);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.requests_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
candidate_pair.responses_received);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.responses_sent);
verifier.TestMemberIsUndefined(candidate_pair.retransmissions_received);
verifier.TestMemberIsUndefined(candidate_pair.retransmissions_sent);
verifier.TestMemberIsUndefined(candidate_pair.consent_requests_received);
verifier.TestMemberIsNonNegative<uint64_t>(
candidate_pair.consent_requests_sent);
verifier.TestMemberIsUndefined(candidate_pair.consent_responses_received);
verifier.TestMemberIsUndefined(candidate_pair.consent_responses_sent);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCIceCandidateStats(const RTCIceCandidateStats& candidate) {
RTCStatsVerifier verifier(report_, &candidate);
verifier.TestMemberIsIDReference(candidate.transport_id,
RTCTransportStats::kType);
verifier.TestMemberIsDefined(candidate.is_remote);
if (*candidate.is_remote) {
verifier.TestMemberIsUndefined(candidate.network_type);
} else {
verifier.TestMemberIsDefined(candidate.network_type);
}
verifier.TestMemberIsDefined(candidate.ip);
verifier.TestMemberIsNonNegative<int32_t>(candidate.port);
verifier.TestMemberIsDefined(candidate.protocol);
verifier.TestMemberIsDefined(candidate.candidate_type);
verifier.TestMemberIsNonNegative<int32_t>(candidate.priority);
verifier.TestMemberIsUndefined(candidate.url);
verifier.TestMemberIsDefined(candidate.deleted);
verifier.TestMemberIsUndefined(candidate.relay_protocol);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCLocalIceCandidateStats(
const RTCLocalIceCandidateStats& local_candidate) {
return VerifyRTCIceCandidateStats(local_candidate);
}
bool VerifyRTCRemoteIceCandidateStats(
const RTCRemoteIceCandidateStats& remote_candidate) {
return VerifyRTCIceCandidateStats(remote_candidate);
}
bool VerifyRTCMediaStreamStats(const RTCMediaStreamStats& media_stream) {
RTCStatsVerifier verifier(report_, &media_stream);
verifier.TestMemberIsDefined(media_stream.stream_identifier);
verifier.TestMemberIsIDReference(media_stream.track_ids,
RTCMediaStreamTrackStats::kType);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCMediaStreamTrackStats(
const RTCMediaStreamTrackStats& media_stream_track) {
RTCStatsVerifier verifier(report_, &media_stream_track);
verifier.TestMemberIsDefined(media_stream_track.track_identifier);
verifier.TestMemberIsDefined(media_stream_track.remote_source);
verifier.TestMemberIsDefined(media_stream_track.ended);
verifier.TestMemberIsDefined(media_stream_track.detached);
verifier.TestMemberIsDefined(media_stream_track.kind);
RTC_DCHECK(media_stream_track.remote_source.is_defined());
// Video or audio media stream track?
if (*media_stream_track.kind == RTCMediaStreamTrackKind::kVideo) {
// The type of the referenced media source depends on kind.
if (*media_stream_track.remote_source) {
verifier.TestMemberIsUndefined(media_stream_track.media_source_id);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_emitted_count);
verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frames_received);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frames_decoded);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frames_dropped);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.freeze_count);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.pause_count);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.total_freezes_duration);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.total_pauses_duration);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.total_frames_duration);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.sum_squared_frame_durations);
} else {
verifier.TestMemberIsIDReference(media_stream_track.media_source_id,
RTCVideoSourceStats::kType);
// Local tracks have no jitter buffer.
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_emitted_count);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frames_sent);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.huge_frames_sent);
verifier.TestMemberIsUndefined(media_stream_track.frames_received);
verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
verifier.TestMemberIsUndefined(media_stream_track.freeze_count);
verifier.TestMemberIsUndefined(media_stream_track.pause_count);
verifier.TestMemberIsUndefined(
media_stream_track.total_freezes_duration);
verifier.TestMemberIsUndefined(
media_stream_track.total_pauses_duration);
verifier.TestMemberIsUndefined(
media_stream_track.total_frames_duration);
verifier.TestMemberIsUndefined(
media_stream_track.sum_squared_frame_durations);
}
// Video-only members
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frame_width);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frame_height);
verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
verifier.TestMemberIsUndefined(media_stream_track.frames_corrupted);
verifier.TestMemberIsUndefined(media_stream_track.partial_frames_lost);
verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
// Audio-only members should be undefined
verifier.TestMemberIsUndefined(media_stream_track.audio_level);
verifier.TestMemberIsUndefined(media_stream_track.echo_return_loss);
verifier.TestMemberIsUndefined(
media_stream_track.echo_return_loss_enhancement);
verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
verifier.TestMemberIsUndefined(media_stream_track.total_samples_duration);
verifier.TestMemberIsUndefined(media_stream_track.total_samples_received);
verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
verifier.TestMemberIsUndefined(
media_stream_track.silent_concealed_samples);
verifier.TestMemberIsUndefined(media_stream_track.concealment_events);
verifier.TestMemberIsUndefined(
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsUndefined(
media_stream_track.removed_samples_for_acceleration);
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsUndefined(
media_stream_track.delayed_packet_outage_samples);
verifier.TestMemberIsUndefined(
media_stream_track.relative_packet_arrival_delay);
verifier.TestMemberIsUndefined(media_stream_track.interruption_count);
verifier.TestMemberIsUndefined(
media_stream_track.total_interruption_duration);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_target_delay);
} else {
RTC_DCHECK_EQ(*media_stream_track.kind, RTCMediaStreamTrackKind::kAudio);
// The type of the referenced media source depends on kind.
if (*media_stream_track.remote_source) {
// Remote tracks don't have media source stats.
verifier.TestMemberIsUndefined(media_stream_track.media_source_id);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_emitted_count);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.jitter_buffer_target_delay);
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
verifier.TestMemberIsPositive<double>(media_stream_track.audio_level);
verifier.TestMemberIsPositive<double>(
media_stream_track.total_audio_energy);
verifier.TestMemberIsPositive<uint64_t>(
media_stream_track.total_samples_received);
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
verifier.TestMemberIsPositive<double>(
media_stream_track.total_samples_duration);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.silent_concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.concealment_events);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.removed_samples_for_acceleration);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.delayed_packet_outage_samples);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.relative_packet_arrival_delay);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.interruption_count);
verifier.TestMemberIsNonNegative<double>(
media_stream_track.total_interruption_duration);
} else {
verifier.TestMemberIsIDReference(media_stream_track.media_source_id,
RTCAudioSourceStats::kType);
// Local audio tracks have no jitter buffer.
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_emitted_count);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_target_delay);
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
verifier.TestMemberIsUndefined(media_stream_track.audio_level);
verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
verifier.TestMemberIsUndefined(
media_stream_track.total_samples_received);
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
verifier.TestMemberIsUndefined(
media_stream_track.total_samples_duration);
verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
verifier.TestMemberIsUndefined(
media_stream_track.silent_concealed_samples);
verifier.TestMemberIsUndefined(media_stream_track.concealment_events);
verifier.TestMemberIsUndefined(
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsUndefined(
media_stream_track.removed_samples_for_acceleration);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsUndefined(
media_stream_track.delayed_packet_outage_samples);
verifier.TestMemberIsUndefined(
media_stream_track.relative_packet_arrival_delay);
verifier.TestMemberIsUndefined(media_stream_track.interruption_count);
verifier.TestMemberIsUndefined(
media_stream_track.total_interruption_duration);
}
// Video-only members should be undefined
verifier.TestMemberIsUndefined(media_stream_track.frame_width);
verifier.TestMemberIsUndefined(media_stream_track.frame_height);
verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
verifier.TestMemberIsUndefined(media_stream_track.frames_received);
verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
verifier.TestMemberIsUndefined(media_stream_track.frames_corrupted);
verifier.TestMemberIsUndefined(media_stream_track.partial_frames_lost);
verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
verifier.TestMemberIsUndefined(media_stream_track.freeze_count);
verifier.TestMemberIsUndefined(media_stream_track.pause_count);
verifier.TestMemberIsUndefined(media_stream_track.total_freezes_duration);
verifier.TestMemberIsUndefined(media_stream_track.total_pauses_duration);
verifier.TestMemberIsUndefined(media_stream_track.total_frames_duration);
verifier.TestMemberIsUndefined(
media_stream_track.sum_squared_frame_durations);
// Audio-only members
// TODO(hbos): |echo_return_loss| and |echo_return_loss_enhancement| are
// flaky on msan bot (sometimes defined, sometimes undefined). Should the
// test run until available or is there a way to have it always be
// defined? crbug.com/627816
verifier.MarkMemberTested(media_stream_track.echo_return_loss, true);
verifier.MarkMemberTested(media_stream_track.echo_return_loss_enhancement,
true);
}
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCPeerConnectionStats(
const RTCPeerConnectionStats& peer_connection) {
RTCStatsVerifier verifier(report_, &peer_connection);
verifier.TestMemberIsNonNegative<uint32_t>(
peer_connection.data_channels_opened);
verifier.TestMemberIsNonNegative<uint32_t>(
peer_connection.data_channels_closed);
return verifier.ExpectAllMembersSuccessfullyTested();
}
void VerifyRTCRTPStreamStats(const RTCRTPStreamStats& stream,
RTCStatsVerifier* verifier) {
verifier->TestMemberIsDefined(stream.ssrc);
verifier->TestMemberIsDefined(stream.is_remote);
verifier->TestMemberIsDefined(stream.media_type);
verifier->TestMemberIsDefined(stream.kind);
verifier->TestMemberIsIDReference(stream.track_id,
RTCMediaStreamTrackStats::kType);
verifier->TestMemberIsIDReference(stream.transport_id,
RTCTransportStats::kType);
verifier->TestMemberIsIDReference(stream.codec_id, RTCCodecStats::kType);
if (stream.media_type.is_defined() && *stream.media_type == "video") {
verifier->TestMemberIsNonNegative<uint32_t>(stream.fir_count);
verifier->TestMemberIsNonNegative<uint32_t>(stream.pli_count);
verifier->TestMemberIsNonNegative<uint32_t>(stream.nack_count);
} else {
verifier->TestMemberIsUndefined(stream.fir_count);
verifier->TestMemberIsUndefined(stream.pli_count);
verifier->TestMemberIsUndefined(stream.nack_count);
}
verifier->TestMemberIsUndefined(stream.sli_count);
}
bool VerifyRTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& inbound_stream) {
RTCStatsVerifier verifier(report_, &inbound_stream);
VerifyRTCRTPStreamStats(inbound_stream, &verifier);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "video") {
verifier.TestMemberIsNonNegative<uint64_t>(inbound_stream.qp_sum);
2019-08-16 13:09:51 +02:00
verifier.TestMemberIsDefined(inbound_stream.decoder_implementation);
} else {
verifier.TestMemberIsUndefined(inbound_stream.qp_sum);
2019-08-16 13:09:51 +02:00
verifier.TestMemberIsUndefined(inbound_stream.decoder_implementation);
}
verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.packets_received);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "audio") {
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.fec_packets_received);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.fec_packets_discarded);
} else {
verifier.TestMemberIsUndefined(inbound_stream.fec_packets_received);
verifier.TestMemberIsUndefined(inbound_stream.fec_packets_discarded);
}
verifier.TestMemberIsNonNegative<uint64_t>(inbound_stream.bytes_received);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.header_bytes_received);
// packets_lost is defined as signed, but this should never happen in
// this test. See RFC 3550.
verifier.TestMemberIsNonNegative<int32_t>(inbound_stream.packets_lost);
verifier.TestMemberIsDefined(inbound_stream.last_packet_received_timestamp);
if (inbound_stream.frames_received.ValueOrDefault(0) > 0) {
verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.frame_width);
verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.frame_height);
} else {
verifier.TestMemberIsUndefined(inbound_stream.frame_width);
verifier.TestMemberIsUndefined(inbound_stream.frame_height);
}
if (inbound_stream.frames_per_second.is_defined()) {
verifier.TestMemberIsNonNegative<double>(
inbound_stream.frames_per_second);
} else {
verifier.TestMemberIsUndefined(inbound_stream.frames_per_second);
}
verifier.TestMemberIsUndefined(inbound_stream.frame_bit_depth);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "video") {
verifier.TestMemberIsUndefined(inbound_stream.jitter);
verifier.TestMemberIsUndefined(inbound_stream.jitter_buffer_delay);
verifier.TestMemberIsUndefined(
inbound_stream.jitter_buffer_emitted_count);
verifier.TestMemberIsUndefined(inbound_stream.total_samples_received);
verifier.TestMemberIsUndefined(inbound_stream.concealed_samples);
verifier.TestMemberIsUndefined(inbound_stream.silent_concealed_samples);
verifier.TestMemberIsUndefined(inbound_stream.concealment_events);
verifier.TestMemberIsUndefined(
inbound_stream.inserted_samples_for_deceleration);
verifier.TestMemberIsUndefined(
inbound_stream.removed_samples_for_acceleration);
verifier.TestMemberIsUndefined(inbound_stream.audio_level);
verifier.TestMemberIsUndefined(inbound_stream.total_audio_energy);
verifier.TestMemberIsUndefined(inbound_stream.total_samples_duration);
verifier.TestMemberIsNonNegative<int32_t>(inbound_stream.frames_received);
} else {
verifier.TestMemberIsNonNegative<double>(inbound_stream.jitter);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.jitter_buffer_emitted_count);
verifier.TestMemberIsPositive<uint64_t>(
inbound_stream.total_samples_received);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.silent_concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.concealment_events);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.inserted_samples_for_deceleration);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.removed_samples_for_acceleration);
verifier.TestMemberIsPositive<double>(inbound_stream.audio_level);
verifier.TestMemberIsPositive<double>(inbound_stream.total_audio_energy);
verifier.TestMemberIsPositive<double>(
inbound_stream.total_samples_duration);
verifier.TestMemberIsUndefined(inbound_stream.frames_received);
}
verifier.TestMemberIsUndefined(inbound_stream.round_trip_time);
verifier.TestMemberIsUndefined(inbound_stream.packets_discarded);
verifier.TestMemberIsUndefined(inbound_stream.packets_repaired);
verifier.TestMemberIsUndefined(inbound_stream.burst_packets_lost);
verifier.TestMemberIsUndefined(inbound_stream.burst_packets_discarded);
verifier.TestMemberIsUndefined(inbound_stream.burst_loss_count);
verifier.TestMemberIsUndefined(inbound_stream.burst_discard_count);
verifier.TestMemberIsUndefined(inbound_stream.burst_loss_rate);
verifier.TestMemberIsUndefined(inbound_stream.burst_discard_rate);
verifier.TestMemberIsUndefined(inbound_stream.gap_loss_rate);
verifier.TestMemberIsUndefined(inbound_stream.gap_discard_rate);
// Test runtime too short to get an estimate (at least two RTCP sender
// reports need to be received).
verifier.MarkMemberTested(inbound_stream.estimated_playout_timestamp, true);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "video") {
verifier.TestMemberIsDefined(inbound_stream.frames_decoded);
verifier.TestMemberIsDefined(inbound_stream.key_frames_decoded);
verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.frames_dropped);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.total_decode_time);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.total_inter_frame_delay);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.total_squared_inter_frame_delay);
// The integration test is not set up to test screen share; don't require
// this to be present.
verifier.MarkMemberTested(inbound_stream.content_type, true);
} else {
verifier.TestMemberIsUndefined(inbound_stream.frames_decoded);
verifier.TestMemberIsUndefined(inbound_stream.key_frames_decoded);
verifier.TestMemberIsUndefined(inbound_stream.frames_dropped);
verifier.TestMemberIsUndefined(inbound_stream.total_decode_time);
verifier.TestMemberIsUndefined(inbound_stream.total_inter_frame_delay);
verifier.TestMemberIsUndefined(
inbound_stream.total_squared_inter_frame_delay);
verifier.TestMemberIsUndefined(inbound_stream.content_type);
}
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCOutboundRTPStreamStats(
const RTCOutboundRTPStreamStats& outbound_stream) {
RTCStatsVerifier verifier(report_, &outbound_stream);
VerifyRTCRTPStreamStats(outbound_stream, &verifier);
if (outbound_stream.media_type.is_defined() &&
*outbound_stream.media_type == "video") {
verifier.TestMemberIsIDReference(outbound_stream.media_source_id,
RTCVideoSourceStats::kType);
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
if (*outbound_stream.frames_encoded > 0) {
verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.qp_sum);
} else {
verifier.TestMemberIsUndefined(outbound_stream.qp_sum);
}
} else {
verifier.TestMemberIsIDReference(outbound_stream.media_source_id,
RTCAudioSourceStats::kType);
verifier.TestMemberIsUndefined(outbound_stream.qp_sum);
}
verifier.TestMemberIsOptionalIDReference(
outbound_stream.remote_id, RTCRemoteInboundRtpStreamStats::kType);
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.retransmitted_packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.header_bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.retransmitted_bytes_sent);
verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
if (outbound_stream.media_type.is_defined() &&
*outbound_stream.media_type == "video") {
verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
verifier.TestMemberIsDefined(outbound_stream.key_frames_encoded);
verifier.TestMemberIsNonNegative<double>(
outbound_stream.total_encode_time);
verifier.TestMemberIsNonNegative<uint64_t>(
outbound_stream.total_encoded_bytes_target);
verifier.TestMemberIsNonNegative<double>(
outbound_stream.total_packet_send_delay);
verifier.TestMemberIsDefined(outbound_stream.quality_limitation_reason);
verifier.TestMemberIsNonNegative<uint32_t>(
outbound_stream.quality_limitation_resolution_changes);
// The integration test is not set up to test screen share; don't require
// this to be present.
verifier.MarkMemberTested(outbound_stream.content_type, true);
2019-08-16 13:09:51 +02:00
verifier.TestMemberIsDefined(outbound_stream.encoder_implementation);
// Unless an implementation-specific amount of time has passed and at
// least one frame has been encoded, undefined is reported. Because it
// is hard to tell what is the case here, we treat FPS as optional.
// TODO(hbos): Update the tests to run until all implemented metrics
// should be populated.
if (outbound_stream.frames_per_second.is_defined()) {
verifier.TestMemberIsNonNegative<double>(
outbound_stream.frames_per_second);
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
} else {
verifier.TestMemberIsUndefined(outbound_stream.frames_per_second);
}
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.frame_height);
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.frame_width);
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.frames_sent);
verifier.TestMemberIsNonNegative<uint32_t>(
outbound_stream.huge_frames_sent);
verifier.MarkMemberTested(outbound_stream.rid, true);
} else {
verifier.TestMemberIsUndefined(outbound_stream.frames_encoded);
verifier.TestMemberIsUndefined(outbound_stream.key_frames_encoded);
verifier.TestMemberIsUndefined(outbound_stream.total_encode_time);
verifier.TestMemberIsUndefined(
outbound_stream.total_encoded_bytes_target);
// TODO(https://crbug.com/webrtc/10635): Implement for audio as well.
verifier.TestMemberIsUndefined(outbound_stream.total_packet_send_delay);
verifier.TestMemberIsUndefined(outbound_stream.quality_limitation_reason);
verifier.TestMemberIsUndefined(
outbound_stream.quality_limitation_resolution_changes);
verifier.TestMemberIsUndefined(outbound_stream.content_type);
2019-08-16 13:09:51 +02:00
// TODO(hbos): Implement for audio as well.
verifier.TestMemberIsUndefined(outbound_stream.encoder_implementation);
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
verifier.TestMemberIsUndefined(outbound_stream.rid);
verifier.TestMemberIsUndefined(outbound_stream.frames_per_second);
verifier.TestMemberIsUndefined(outbound_stream.frame_height);
verifier.TestMemberIsUndefined(outbound_stream.frame_width);
verifier.TestMemberIsUndefined(outbound_stream.frames_sent);
verifier.TestMemberIsUndefined(outbound_stream.huge_frames_sent);
}
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& remote_inbound_stream) {
RTCStatsVerifier verifier(report_, &remote_inbound_stream);
verifier.TestMemberIsDefined(remote_inbound_stream.ssrc);
verifier.TestMemberIsDefined(remote_inbound_stream.kind);
verifier.TestMemberIsIDReference(remote_inbound_stream.transport_id,
RTCTransportStats::kType);
verifier.TestMemberIsIDReference(remote_inbound_stream.codec_id,
RTCCodecStats::kType);
verifier.TestMemberIsDefined(remote_inbound_stream.packets_lost);
// Note that the existance of RTCCodecStats is needed for |codec_id| and
// |jitter| to be present.
verifier.TestMemberIsNonNegative<double>(remote_inbound_stream.jitter);
verifier.TestMemberIsIDReference(remote_inbound_stream.local_id,
RTCOutboundRTPStreamStats::kType);
verifier.TestMemberIsNonNegative<double>(
remote_inbound_stream.round_trip_time);
return verifier.ExpectAllMembersSuccessfullyTested();
}
void VerifyRTCMediaSourceStats(const RTCMediaSourceStats& media_source,
RTCStatsVerifier* verifier) {
verifier->TestMemberIsDefined(media_source.track_identifier);
verifier->TestMemberIsDefined(media_source.kind);
if (media_source.kind.is_defined()) {
EXPECT_TRUE((*media_source.kind == "audio" &&
media_source.type() == RTCAudioSourceStats::kType) ||
(*media_source.kind == "video" &&
media_source.type() == RTCVideoSourceStats::kType));
}
}
bool VerifyRTCAudioSourceStats(const RTCAudioSourceStats& audio_source) {
RTCStatsVerifier verifier(report_, &audio_source);
VerifyRTCMediaSourceStats(audio_source, &verifier);
// Audio level, unlike audio energy, only gets updated at a certain
// frequency, so we don't require that one to be positive to avoid a race
// (https://crbug.com/webrtc/10962).
verifier.TestMemberIsNonNegative<double>(audio_source.audio_level);
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
verifier.TestMemberIsPositive<double>(audio_source.total_audio_energy);
verifier.TestMemberIsPositive<double>(audio_source.total_samples_duration);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCVideoSourceStats(const RTCVideoSourceStats& video_source) {
RTCStatsVerifier verifier(report_, &video_source);
VerifyRTCMediaSourceStats(video_source, &verifier);
// TODO(hbos): This integration test uses fakes that doesn't support
// VideoTrackSourceInterface::Stats. When this is fixed we should
// TestMemberIsNonNegative<uint32_t>() for |width| and |height| instead to
// reflect real code.
verifier.TestMemberIsUndefined(video_source.width);
verifier.TestMemberIsUndefined(video_source.height);
// TODO(hbos): When |frames| is implemented test that this member should be
// expected to be non-negative.
verifier.TestMemberIsUndefined(video_source.frames);
verifier.TestMemberIsNonNegative<uint32_t>(video_source.frames_per_second);
return verifier.ExpectAllMembersSuccessfullyTested();
}
bool VerifyRTCTransportStats(const RTCTransportStats& transport) {
RTCStatsVerifier verifier(report_, &transport);
verifier.TestMemberIsNonNegative<uint64_t>(transport.bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(transport.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(transport.bytes_received);
verifier.TestMemberIsNonNegative<uint64_t>(transport.packets_received);
verifier.TestMemberIsOptionalIDReference(transport.rtcp_transport_stats_id,
RTCTransportStats::kType);
verifier.TestMemberIsDefined(transport.dtls_state);
verifier.TestMemberIsIDReference(transport.selected_candidate_pair_id,
RTCIceCandidatePairStats::kType);
verifier.TestMemberIsIDReference(transport.local_certificate_id,
RTCCertificateStats::kType);
verifier.TestMemberIsIDReference(transport.remote_certificate_id,
RTCCertificateStats::kType);
verifier.TestMemberIsDefined(transport.tls_version);
verifier.TestMemberIsDefined(transport.dtls_cipher);
verifier.TestMemberIsDefined(transport.srtp_cipher);
verifier.TestMemberIsPositive<uint32_t>(
transport.selected_candidate_pair_changes);
return verifier.ExpectAllMembersSuccessfullyTested();
}
private:
rtc::scoped_refptr<const RTCStatsReport> report_;
};
#ifdef HAVE_SCTP
TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) {
StartCall();
rtc::scoped_refptr<const RTCStatsReport> report = GetStatsFromCaller();
RTCStatsReportVerifier(report.get()).VerifyReport({});
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
#if RTC_TRACE_EVENTS_ENABLED
EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
#endif
}
TEST_F(RTCStatsIntegrationTest, GetStatsFromCallee) {
StartCall();
rtc::scoped_refptr<const RTCStatsReport> report = GetStatsFromCallee();
RTCStatsReportVerifier(report.get()).VerifyReport({});
Reland "Improve outbound-rtp statistics for simulcast" This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 15:54:46 +02:00
#if RTC_TRACE_EVENTS_ENABLED
EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
#endif
}
2018-03-20 13:24:20 +01:00
// These tests exercise the integration of the stats selection algorithm inside
// of PeerConnection. See rtcstatstraveral_unittest.cc for more detailed stats
// traversal tests on particular stats graphs.
TEST_F(RTCStatsIntegrationTest, GetStatsWithSenderSelector) {
StartCall();
ASSERT_FALSE(caller_->pc()->GetSenders().empty());
rtc::scoped_refptr<const RTCStatsReport> report =
GetStatsFromCaller(caller_->pc()->GetSenders()[0]);
std::vector<const char*> allowed_missing_stats = {
// TODO(hbos): Include RTC[Audio/Video]ReceiverStats when implemented.
// TODO(hbos): Include RTCRemoteOutboundRtpStreamStats when implemented.
// TODO(hbos): Include RTCRtpContributingSourceStats when implemented.
RTCInboundRTPStreamStats::kType,
RTCPeerConnectionStats::kType,
RTCMediaStreamStats::kType,
RTCDataChannelStats::kType,
};
RTCStatsReportVerifier(report.get()).VerifyReport(allowed_missing_stats);
2018-03-20 13:24:20 +01:00
EXPECT_TRUE(report->size());
}
TEST_F(RTCStatsIntegrationTest, GetStatsWithReceiverSelector) {
StartCall();
ASSERT_FALSE(caller_->pc()->GetReceivers().empty());
rtc::scoped_refptr<const RTCStatsReport> report =
GetStatsFromCaller(caller_->pc()->GetReceivers()[0]);
std::vector<const char*> allowed_missing_stats = {
// TODO(hbos): Include RTC[Audio/Video]SenderStats when implemented.
// TODO(hbos): Include RTCRemoteInboundRtpStreamStats when implemented.
// TODO(hbos): Include RTCRtpContributingSourceStats when implemented.
RTCOutboundRTPStreamStats::kType,
RTCPeerConnectionStats::kType,
RTCMediaStreamStats::kType,
RTCDataChannelStats::kType,
};
RTCStatsReportVerifier(report.get()).VerifyReport(allowed_missing_stats);
2018-03-20 13:24:20 +01:00
EXPECT_TRUE(report->size());
}
TEST_F(RTCStatsIntegrationTest, GetStatsWithInvalidSenderSelector) {
StartCall();
ASSERT_FALSE(callee_->pc()->GetSenders().empty());
// The selector is invalid for the caller because it belongs to the callee.
auto invalid_selector = callee_->pc()->GetSenders()[0];
rtc::scoped_refptr<const RTCStatsReport> report =
GetStatsFromCaller(invalid_selector);
EXPECT_FALSE(report->size());
}
TEST_F(RTCStatsIntegrationTest, GetStatsWithInvalidReceiverSelector) {
StartCall();
ASSERT_FALSE(callee_->pc()->GetReceivers().empty());
// The selector is invalid for the caller because it belongs to the callee.
auto invalid_selector = callee_->pc()->GetReceivers()[0];
rtc::scoped_refptr<const RTCStatsReport> report =
GetStatsFromCaller(invalid_selector);
EXPECT_FALSE(report->size());
}
// TODO(bugs.webrtc.org/10041) For now this is equivalent to the following
// test GetsStatsWhileClosingPeerConnection, because pc() is closed by
// PeerConnectionTestWrapper. See: bugs.webrtc.org/9847
TEST_F(RTCStatsIntegrationTest,
DISABLED_GetStatsWhileDestroyingPeerConnection) {
StartCall();
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
caller_->pc()->GetStats(stats_obtainer);
// This will destroy the peer connection.
caller_ = nullptr;
// Any pending stats requests should have completed in the act of destroying
// the peer connection.
ASSERT_TRUE(stats_obtainer->report());
#if RTC_TRACE_EVENTS_ENABLED
EXPECT_EQ(stats_obtainer->report()->ToJson(),
RTCStatsReportTraceListener::last_trace());
#endif
}
TEST_F(RTCStatsIntegrationTest, GetsStatsWhileClosingPeerConnection) {
StartCall();
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
caller_->pc()->GetStats(stats_obtainer);
caller_->pc()->Close();
ASSERT_TRUE(stats_obtainer->report());
#if RTC_TRACE_EVENTS_ENABLED
EXPECT_EQ(stats_obtainer->report()->ToJson(),
RTCStatsReportTraceListener::last_trace());
#endif
}
// GetStatsReferencedIds() is optimized to recognize what is or isn't a
// referenced ID based on dictionary type information and knowing what members
// are used as references, as opposed to iterating all members to find the ones
// with the "Id" or "Ids" suffix. As such, GetStatsReferencedIds() is tested as
// an integration test instead of a unit test in order to guard against adding
// new references and forgetting to update GetStatsReferencedIds().
TEST_F(RTCStatsIntegrationTest, GetStatsReferencedIds) {
StartCall();
rtc::scoped_refptr<const RTCStatsReport> report = GetStatsFromCallee();
for (const RTCStats& stats : *report) {
// Find all references by looking at all string members with the "Id" or
// "Ids" suffix.
std::set<const std::string*> expected_ids;
for (const auto* member : stats.Members()) {
if (!member->is_defined())
continue;
if (member->type() == RTCStatsMemberInterface::kString) {
if (absl::EndsWith(member->name(), "Id")) {
const auto& id = member->cast_to<const RTCStatsMember<std::string>>();
expected_ids.insert(&(*id));
}
} else if (member->type() == RTCStatsMemberInterface::kSequenceString) {
if (absl::EndsWith(member->name(), "Ids")) {
const auto& ids =
member->cast_to<const RTCStatsMember<std::vector<std::string>>>();
for (const std::string& id : *ids)
expected_ids.insert(&id);
}
}
}
std::vector<const std::string*> neighbor_ids = GetStatsReferencedIds(stats);
EXPECT_EQ(neighbor_ids.size(), expected_ids.size());
for (const std::string* neighbor_id : neighbor_ids) {
EXPECT_THAT(expected_ids, Contains(neighbor_id));
}
for (const std::string* expected_id : expected_ids) {
EXPECT_THAT(neighbor_ids, Contains(expected_id));
}
}
}
#endif // HAVE_SCTP
} // namespace
} // namespace webrtc