| /external/webrtc/test/pc/e2e/analyzer/audio/ |
| D | default_audio_quality_analyzer.cc | 64 sample.jitter_buffer_emitted_count = in OnStatsReports() 65 stat->jitter_buffer_emitted_count.ValueOrDefault(0ul); in OnStatsReports() 104 sample.jitter_buffer_emitted_count - in OnStatsReports() 105 prev_sample.jitter_buffer_emitted_count; in OnStatsReports()
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| D | default_audio_quality_analyzer.h | 61 uint64_t jitter_buffer_emitted_count = 0; member
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| /external/webrtc/audio/voip/test/ |
| D | audio_channel_unittest.cc | 173 EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 160ULL); in TEST_F() 196 EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 160ULL); in TEST_F() 215 EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 320ULL); in TEST_F()
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| /external/webrtc/stats/ |
| D | rtcstats_objects.cc | 353 &jitter_buffer_emitted_count, 393 jitter_buffer_emitted_count("jitterBufferEmittedCount"), in DEPRECATED_RTCMediaStreamTrackStats() 529 &jitter_buffer_emitted_count, 593 jitter_buffer_emitted_count("jitterBufferEmittedCount"), in RTCInboundRTPStreamStats()
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| /external/webrtc/call/ |
| D | audio_receive_stream.h | 60 uint64_t jitter_buffer_emitted_count = 0; member
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| D | video_receive_stream.h | 100 uint64_t jitter_buffer_emitted_count = 0; member
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| D | video_receive_stream.cc | 75 ss << "jb_emitted_count: " << jitter_buffer_emitted_count << ", "; in ToString()
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| /external/webrtc/audio/voip/ |
| D | audio_channel.cc | 137 ingress_stats.neteq_stats.jitter_buffer_emitted_count = in GetIngressStatistics()
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| /external/webrtc/api/neteq/ |
| D | neteq.h | 68 uint64_t jitter_buffer_emitted_count = 0; member
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| /external/webrtc/pc/ |
| D | rtc_stats_integrationtest.cc | 591 media_stream_track.jitter_buffer_emitted_count); in VerLegacyifyRTCMediaStreamTrackStats() 606 media_stream_track.jitter_buffer_emitted_count); in VerLegacyifyRTCMediaStreamTrackStats() 645 media_stream_track.jitter_buffer_emitted_count); in VerLegacyifyRTCMediaStreamTrackStats() 669 media_stream_track.jitter_buffer_emitted_count); in VerLegacyifyRTCMediaStreamTrackStats() 796 inbound_stream.jitter_buffer_emitted_count); in VerifyRTCInboundRTPStreamStats()
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| D | rtc_stats_collector.cc | 430 inbound_stats->jitter_buffer_emitted_count = in SetInboundRTPStreamStatsFromMediaReceiverInfo() 431 media_receiver_info.jitter_buffer_emitted_count; in SetInboundRTPStreamStatsFromMediaReceiverInfo() 1049 audio_track_stats->jitter_buffer_emitted_count = in ProduceMediaStreamTrackStatsFromVoiceReceiverInfo() 1050 voice_receiver_info.jitter_buffer_emitted_count; in ProduceMediaStreamTrackStatsFromVoiceReceiverInfo() 1123 video_track_stats->jitter_buffer_emitted_count = in ProduceMediaStreamTrackStatsFromVideoReceiverInfo() 1124 video_receiver_info.jitter_buffer_emitted_count; in ProduceMediaStreamTrackStatsFromVideoReceiverInfo()
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| D | rtc_stats_collector_unittest.cc | 2246 voice_receiver_info.jitter_buffer_emitted_count = 13; in TEST_F() 2286 expected_remote_audio_track.jitter_buffer_emitted_count = 13; in TEST_F() 2377 video_receiver_info_ssrc3.jitter_buffer_emitted_count = 25; in TEST_F() 2423 expected_remote_video_track_ssrc3.jitter_buffer_emitted_count = 25; in TEST_F() 2454 voice_media_info.receivers[0].jitter_buffer_emitted_count = 2; in TEST_F() 2515 expected_audio.jitter_buffer_emitted_count = 2; in TEST_F() 2591 video_media_info.receivers[0].jitter_buffer_emitted_count = 13; in TEST_F() 2659 expected_video.jitter_buffer_emitted_count = 13; in TEST_F()
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_receiver.cc | 293 neteq_lifetime_stat.jitter_buffer_emitted_count; in GetNetworkStatistics()
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| /external/webrtc/modules/audio_coding/neteq/ |
| D | statistics_calculator.cc | 275 lifetime_stats_.jitter_buffer_emitted_count += num_samples; in JitterBufferDelay()
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| D | neteq_unittest.cc | 931 EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count); in TestJitterBufferDelay() 975 EXPECT_EQ(kSamples * 3, stats.jitter_buffer_emitted_count); in TEST_F()
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| /external/webrtc/api/stats/ |
| D | rtcstats_objects.h | 331 RTCStatsMember<uint64_t> jitter_buffer_emitted_count; variable 451 RTCStatsMember<uint64_t> jitter_buffer_emitted_count; variable
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| /external/webrtc/audio/ |
| D | audio_receive_stream_unittest.cc | 281 stats.jitter_buffer_emitted_count); in TEST()
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| D | audio_receive_stream.cc | 338 stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount; in GetStats()
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| /external/webrtc/media/base/ |
| D | media_channel.h | 474 uint64_t jitter_buffer_emitted_count = 0; member
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| /external/webrtc/video/ |
| D | receive_statistics_proxy2.cc | 636 stats_.jitter_buffer_emitted_count = current_delay_counter_.NumSamples(); in GetStats()
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| /external/webrtc/video/g3doc/ |
| D | stats.md | 133 * `jitter_buffer_emitted_count` - total number of frames that have come out from the jitter buffe…
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| /external/webrtc/media/engine/ |
| D | webrtc_voice_engine.cc | 2360 rinfo.jitter_buffer_emitted_count = stats.jitter_buffer_emitted_count; in GetStats()
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| D | webrtc_voice_engine_unittest.cc | 684 stats.jitter_buffer_emitted_count = 77; in GetAudioReceiveStreamStats() 731 EXPECT_EQ(info.jitter_buffer_emitted_count, in VerifyVoiceReceiverInfo() 732 stats.jitter_buffer_emitted_count); in VerifyVoiceReceiverInfo()
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| D | webrtc_video_engine.cc | 3292 info.jitter_buffer_emitted_count = stats.jitter_buffer_emitted_count; in GetVideoReceiverInfo()
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| D | webrtc_video_engine_unittest.cc | 6238 stats.jitter_buffer_emitted_count = 6; in TEST_F() 6265 EXPECT_EQ(stats.jitter_buffer_emitted_count, in TEST_F() 6266 info.receivers[0].jitter_buffer_emitted_count); in TEST_F()
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