/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | packet_loss_stats_unittest.cc | 18 PacketLossStats stats_; member in webrtc::PacketLossStatsTest 25 stats_.AddLostPacket(i); in TEST_F() 27 EXPECT_EQ(500, stats_.GetSingleLossCount()); in TEST_F() 28 EXPECT_EQ(0, stats_.GetMultipleLossEventCount()); in TEST_F() 29 EXPECT_EQ(0, stats_.GetMultipleLossPacketCount()); in TEST_F() 36 stats_.AddLostPacket(i & 0xFFFF); in TEST_F() 38 EXPECT_EQ(500, stats_.GetSingleLossCount()); in TEST_F() 39 EXPECT_EQ(0, stats_.GetMultipleLossEventCount()); in TEST_F() 40 EXPECT_EQ(0, stats_.GetMultipleLossPacketCount()); in TEST_F() 48 stats_.AddLostPacket(i & 0xFFFF); in TEST_F() [all …]
|
D | receive_statistics_unittest.cc | 156 : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {} in TEST_F() 162 stats_ = statistics; in TEST_F() 170 RtcpStatistics stats_; in TEST_F() member in webrtc::TEST_F::TestCallback 203 EXPECT_EQ(statistics.cumulative_lost, callback.stats_.cumulative_lost); in TEST_F() 205 callback.stats_.extended_max_sequence_number); in TEST_F() 206 EXPECT_EQ(statistics.fraction_lost, callback.stats_.fraction_lost); in TEST_F() 207 EXPECT_EQ(statistics.jitter, callback.stats_.jitter); in TEST_F() 244 : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {} in RtpTestCallback() 250 stats_ = counters; in DataCountersUpdated() 267 MatchPacketCounter(expected.transmitted, stats_.transmitted); in Matches() [all …]
|
/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
D | rate_statistics_unittest.cc | 20 RateStatisticsTest() : stats_(500, 8000) {} in RateStatisticsTest() 21 RateStatistics stats_; member in __anonab3019290111::RateStatisticsTest 27 EXPECT_EQ(0u, stats_.Rate(now_ms)); in TEST_F() 28 stats_.Update(1500, now_ms); in TEST_F() 30 EXPECT_EQ(24000u, stats_.Rate(now_ms)); in TEST_F() 31 stats_.Reset(); in TEST_F() 33 EXPECT_EQ(0u, stats_.Rate(now_ms)); in TEST_F() 36 stats_.Update(1500, now_ms); in TEST_F() 41 EXPECT_NEAR(1200000u, stats_.Rate(now_ms), 24000u); in TEST_F() 48 EXPECT_EQ(0u, stats_.Rate(now_ms)); in TEST_F() [all …]
|
/external/webrtc/webrtc/modules/video_coding/codecs/test/ |
D | stats_unittest.cc | 25 void SetUp() { stats_ = new Stats(); } in SetUp() 27 void TearDown() { delete stats_; } in TearDown() 29 Stats* stats_; member in webrtc::test::StatsTest 34 EXPECT_EQ(0u, stats_->stats_.size()); in TEST_F() 35 stats_->PrintSummary(); // should not crash in TEST_F() 40 stats_->NewFrame(0u); in TEST_F() 41 FrameStatistic* frameStat = &stats_->stats_[0]; in TEST_F() 49 FrameStatistic& frameStat = stats_->NewFrame(i); in TEST_F() 52 EXPECT_EQ(nbr_of_frames, static_cast<int>(stats_->stats_.size())); in TEST_F() 54 stats_->PrintSummary(); // should not crash in TEST_F()
|
D | stats.cc | 61 stats_.push_back(stat); in NewFrame() 62 return stats_[frame_number]; in NewFrame() 67 if (stats_.size() == 0) { in PrintSummary() 81 for (FrameStatisticsIterator it = stats_.begin(); it != stats_.end(); ++it) { in PrintSummary() 98 frame = std::min_element(stats_.begin(), stats_.end(), LessForEncodeTime); in PrintSummary() 102 frame = std::max_element(stats_.begin(), stats_.end(), LessForEncodeTime); in PrintSummary() 107 static_cast<int>(total_encoding_time_in_us / stats_.size())); in PrintSummary() 114 for (std::vector<FrameStatistic>::iterator it = stats_.begin(); in PrintSummary() 115 it != stats_.end(); ++it) { in PrintSummary() 136 static_cast<int>(stats_.size() - decoded_frames.size())); in PrintSummary() [all …]
|
D | videoprocessor_unittest.cc | 39 Stats stats_; member in webrtc::test::VideoProcessorTest 71 &packet_manipulator_mock_, config_, &stats_); in TEST_F() 83 &packet_manipulator_mock_, config_, &stats_); in TEST_F()
|
D | videoprocessor.cc | 54 stats_(stats), in VideoProcessorImpl() 198 FrameStatistic& stat = stats_->NewFrame(frame_number); in ProcessFrame() 249 FrameStatistic& stat = stats_->stats_[frame_number]; in FrameEncoded() 311 FrameStatistic& stat = stats_->stats_[frame_number]; in FrameDecoded()
|
D | stats.h | 64 std::vector<FrameStatistic> stats_; variable
|
/external/webrtc/webrtc/video/ |
D | receive_statistics_proxy.cc | 30 stats_.ssrc = ssrc; in ReceiveStatisticsProxy() 77 return stats_; in GetStats() 82 stats_.current_payload_type = payload_type; in OnIncomingPayloadType() 88 stats_.decoder_implementation_name = implementation_name; in OnDecoderImplementationName() 93 stats_.network_frame_rate = framerate; in OnIncomingRate() 94 stats_.total_bitrate_bps = bitrate_bps; in OnIncomingRate() 106 stats_.decode_ms = decode_ms; in OnDecoderTiming() 107 stats_.max_decode_ms = max_decode_ms; in OnDecoderTiming() 108 stats_.current_delay_ms = current_delay_ms; in OnDecoderTiming() 109 stats_.target_delay_ms = target_delay_ms; in OnDecoderTiming() [all …]
|
D | send_statistics_proxy.cc | 173 stats_.encoder_implementation_name = implementation_name; in OnEncoderImplementationName() 178 stats_.encode_frame_rate = framerate; in OnOutgoingRate() 179 stats_.media_bitrate_bps = bitrate; in OnOutgoingRate() 185 stats_.encode_usage_percent = metrics.encode_usage_percent; in CpuOveruseMetricsUpdated() 190 stats_.suspended = is_suspended; in OnSuspendChange() 196 stats_.input_frame_rate = in GetStats() 198 return stats_; in GetStats() 204 stats_.substreams.begin(); in PurgeOldStats() 205 it != stats_.substreams.end(); ++it) { in PurgeOldStats() 217 stats_.substreams.find(ssrc); in GetStatsEntry() [all …]
|
/external/webrtc/talk/app/webrtc/test/ |
D | mockpeerconnectionobservers.h | 122 MockStatsObserver() : called_(false), stats_() {} in MockStatsObserver() 128 stats_.Clear(); in OnComplete() 129 stats_.number_of_reports = reports.size(); in OnComplete() 132 stats_.timestamp = r->timestamp(); in OnComplete() 134 &stats_.audio_output_level); in OnComplete() 136 &stats_.audio_input_level); in OnComplete() 138 &stats_.bytes_received); in OnComplete() 140 &stats_.bytes_sent); in OnComplete() 142 stats_.timestamp = r->timestamp(); in OnComplete() 144 &stats_.available_receive_bandwidth); in OnComplete() [all …]
|
/external/webrtc/talk/app/webrtc/ |
D | rtpsender.cc | 69 stats_(stats), in AudioRtpSender() 83 stats_(stats), in AudioRtpSender() 118 if (can_send_track() && stats_) { in SetTrack() 119 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); in SetTrack() 134 if (stats_) { in SetTrack() 135 stats_->AddLocalAudioTrack(track_.get(), ssrc_); in SetTrack() 152 if (stats_) { in SetSsrc() 153 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); in SetSsrc() 159 if (stats_) { in SetSsrc() 160 stats_->AddLocalAudioTrack(track_.get(), ssrc_); in SetSsrc() [all …]
|
/external/webrtc/talk/media/webrtc/ |
D | fakewebrtccall.cc | 51 stats_ = stats; in SetStats() 68 return stats_; in GetStats() 84 stats_ = stats; in SetStats() 92 return stats_; in GetStats() 172 stats_ = stats; in SetStats() 176 return stats_; in GetStats() 229 return stats_; in GetStats() 242 stats_ = stats; in SetStats() 416 stats_ = stats; in SetStats() 428 return stats_; in GetStats()
|
D | fakewebrtccall.h | 80 webrtc::AudioSendStream::Stats stats_; variable 112 webrtc::AudioReceiveStream::Stats stats_; variable 163 webrtc::VideoSendStream::Stats stats_; variable 198 webrtc::VideoReceiveStream::Stats stats_; variable 257 webrtc::Call::Stats stats_; variable
|
/external/v8/src/compiler/ |
D | zone-stats.cc | 15 zone_stats_->stats_.push_back(this); in StatsScope() 26 DCHECK_EQ(zone_stats_->stats_.back(), this); in ~StatsScope() 27 zone_stats_->stats_.pop_back(); in ~StatsScope() 68 DCHECK(stats_.empty()); in ~ZoneStats() 98 for (StatsScope* stat_scope : stats_) { in ReturnZone()
|
/external/libchrome/dbus/ |
D | dbus_statistics.cc | 72 STLDeleteContainerPointers(stats_.begin(), stats_.end()); in ~DBusStatistics() 112 StatSet::iterator found = stats_.find(stat.get()); in GetStat() 113 if (found != stats_.end()) in GetStat() 117 found = stats_.insert(stat.release()).first; in GetStat() 121 StatSet& stats() { return stats_; } in stats() 125 StatSet stats_; member in dbus::__anonbce4a5dd0111::DBusStatistics
|
/external/v8/src/parsing/ |
D | scanner-character-streams.cc | 202 stats_(stats) {} 251 RuntimeCallStats* stats_; member in v8::internal::Utf8ExternalStreamingStream 342 RuntimeCallTimerScope scope(stats_, &RuntimeCallStats::GetMoreDataCallback); in FetchChunk() 532 : source_(source), stats_(stats) {} in OneByteExternalStreamingStream() 541 RuntimeCallStats* stats_; member in v8::internal::OneByteExternalStreamingStream 545 const Chunk& chunk = chunks_[FindChunk(chunks_, source_, position, stats_)]; in FillBuffer() 574 RuntimeCallStats* stats_; member in v8::internal::TwoByteExternalStreamingStream 583 stats_(stats), in TwoByteExternalStreamingStream() 595 size_t chunk_no = FindChunk(chunks_, source_, 2 * position + 1, stats_); in ReadBlock() 681 RuntimeCallStats* stats_; member in v8::internal::TwoByteExternalBufferedStream [all …]
|
/external/compiler-rt/lib/sanitizer_common/ |
D | sanitizer_allocator.h | 217 v += atomic_load(&stats_[i], memory_order_relaxed); in Add() 218 atomic_store(&stats_[i], v, memory_order_relaxed); in Add() 222 v = atomic_load(&stats_[i], memory_order_relaxed) - v; in Sub() 223 atomic_store(&stats_[i], v, memory_order_relaxed); in Sub() 227 atomic_store(&stats_[i], v, memory_order_relaxed); in Set() 231 return atomic_load(&stats_[i], memory_order_relaxed); in Get() 238 atomic_uintptr_t stats_[AllocatorStatCount]; variable 935 stats_.Init(); in Init() 937 s->Register(&stats_); in Init() 943 s->Unregister(&stats_); in Destroy() [all …]
|
/external/libvpx/libvpx/test/ |
D | encode_test_driver.cc | 31 cfg_.rc_twopass_stats_in = stats_->buf(); in InitEncoder() 67 stats_->Append(*pkt); in EncodeFrame() 165 stats_.Reset(); in RunLoop() 181 codec_->CreateEncoder(cfg_, deadline_, init_flags_, &stats_)); in RunLoop()
|
D | encode_test_driver.h | 91 : cfg_(cfg), deadline_(deadline), init_flags_(init_flags), stats_(stats) { in Encoder() 169 TwopassStatsStore *stats_; variable 259 TwopassStatsStore stats_; variable
|
/external/webrtc/webrtc/modules/video_coding/codecs/tools/ |
D | video_quality_measurement.cc | 360 for (unsigned int i = 0; i < stats.stats_.size(); ++i) { in PrintCsvOutput() 361 const webrtc::test::FrameStatistic& f = stats.stats_[i]; in PrintCsvOutput() 438 for (unsigned int i = 0; i < stats.stats_.size(); ++i) { in PrintPythonOutput() 439 const webrtc::test::FrameStatistic& f = stats.stats_[i]; in PrintPythonOutput() 530 assert(frame_number == static_cast<int>(stats.stats_.size())); in main()
|
/external/v8/src/heap/ |
D | object-stats.cc | 246 stats_->RecordObjectStats(map->instance_type(), object_size); in CollectStatistics() 355 return stats_->RecordFixedArraySubTypeStats(array, subtype, array->Size(), in RecordFixedArrayHelper() 400 stats_->RecordFixedArraySubTypeStats(elements, FAST_ELEMENTS_SUB_TYPE, in RecordJSObjectDetails() 413 stats_->RecordFixedArraySubTypeStats(properties, FAST_PROPERTIES_SUB_TYPE, in RecordJSObjectDetails() 512 stats_->RecordCodeSubTypeStats(code->kind(), code->GetAge(), code->Size()); in RecordCodeDetails() 570 stats_->RecordFixedArraySubTypeStats(array, COPY_ON_WRITE_SUB_TYPE, in RecordFixedArrayDetails() 579 stats_->RecordFixedArraySubTypeStats( in RecordFixedArrayDetails()
|
D | object-stats.h | 136 : heap_(heap), stats_(stats) {} in ObjectStatsCollector() 163 ObjectStats* stats_; variable
|
/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | neteq_impl.cc | 332 stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers, in NetworkStatistics() 945 stats_.IncreaseCounter(output_size_samples_, fs_hz_); in GetDecision() 1056 stats_.ResetMcu(); in GetDecision() 1161 stats_.LostSamples(header->timestamp - end_timestamp); in GetDecision() 1458 stats_.ExpandedNoiseSamples(expand_length_correction); in DoMerge() 1461 stats_.ExpandedVoiceSamples(expand_length_correction); in DoMerge() 1485 stats_.ExpandedNoiseSamples(length); in DoExpand() 1488 stats_.ExpandedVoiceSamples(length); in DoExpand() 1532 stats_.AcceleratedSamples(samples_removed); in DoAccelerate() 1612 stats_.PreemptiveExpandedSamples(samples_added); in DoPreemptiveExpand() [all …]
|
/external/compiler-rt/lib/asan/ |
D | asan_thread.h | 125 AsanStats &stats() { return stats_; } in stats() 158 AsanStats stats_; variable
|