| /external/webrtc/media/base/ |
| D | fake_frame_source.cc | 29 RTC_CHECK_GT(width_, 0); in FakeFrameSource()
30 RTC_CHECK_GT(height_, 0); in FakeFrameSource()
31 RTC_CHECK_GT(interval_us_, 0); in FakeFrameSource()
69 RTC_CHECK_GT(width, 0); in GetFrame()
70 RTC_CHECK_GT(height, 0); in GetFrame()
71 RTC_CHECK_GT(interval_us, 0); in GetFrame()
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| /external/webrtc/modules/video_coding/codecs/test/ |
| D | videocodec_test_stats_impl.cc | 87 RTC_CHECK_GT(num_spatial_layers, 0); in SliceAndCalcLayerVideoStatistic()
88 RTC_CHECK_GT(num_temporal_layers, 0); in SliceAndCalcLayerVideoStatistic()
110 RTC_CHECK_GT(num_spatial_layers, 0); in SliceAndCalcAggregatedVideoStatistic()
111 RTC_CHECK_GT(num_temporal_layers, 0); in SliceAndCalcAggregatedVideoStatistic()
205 RTC_CHECK_GT(target_bitrate_kbps, 0); // We divide by |target_bitrate_kbps|. in SliceAndCalcVideoStatistic()
276 RTC_CHECK_GT(time_since_first_frame_sec, 0); in SliceAndCalcVideoStatistic()
301 RTC_CHECK_GT(timestamp_delta, 0); in SliceAndCalcVideoStatistic()
304 RTC_CHECK_GT(input_framerate_fps, 0); in SliceAndCalcVideoStatistic()
313 RTC_CHECK_GT(duration_sec, 0); in SliceAndCalcVideoStatistic()
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| /external/webrtc/test/ |
| D | explicit_key_value_config.cc | 26 RTC_CHECK_GT(separator_pos, field_start) in ExplicitKeyValueConfig()
36 RTC_CHECK_GT(separator_pos, field_start) in ExplicitKeyValueConfig()
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| D | drifting_clock.cc | 22 RTC_CHECK_GT(speed, 0.0f); in DriftingClock()
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| /external/webrtc/common_audio/ |
| D | window_generator.cc | 40 RTC_CHECK_GT(length, 1); in Hanning()
51 RTC_CHECK_GT(length, 1U); in KaiserBesselDerived()
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| D | real_fourier.cc | 28 RTC_CHECK_GT(length, 0U); in FftOrder()
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| /external/webrtc/api/test/ |
| D | peerconnection_quality_test_fixture.h | 88 RTC_CHECK_GT(duration.ms(), 0); in ScrollingParams()
103 RTC_CHECK_GT(slide_change_interval.ms(), 0); in ScreenShareConfig()
141 RTC_CHECK_GT(simulcast_streams_count, 1); in VideoSimulcastConfig()
146 RTC_CHECK_GT(simulcast_streams_count, 1); in VideoSimulcastConfig()
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| /external/webrtc/video/ |
| D | quality_threshold.cc | 33 RTC_CHECK_GT(fraction, 0.5f); in QualityThreshold()
34 RTC_CHECK_GT(max_measurements, 1); in QualityThreshold()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_format_h264.cc | 146 RTC_CHECK_GT(packet_length, 0); in PacketizeFuA()
187 RTC_CHECK_GT(fragment.size(), 0); in PacketizeStapA()
207 RTC_CHECK_GT(aggregated_fragments, 0); in PacketizeStapA()
230 RTC_CHECK_GT(fragment.size(), 0u); in PacketizeSingleNalu()
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| /external/webrtc/modules/audio_processing/transient/ |
| D | transient_suppression_test.cc | 232 RTC_CHECK_GT(absl::GetFlag(FLAGS_chunk_size_ms), 0); in main()
233 RTC_CHECK_GT(absl::GetFlag(FLAGS_sample_rate_hz), 0); in main()
234 RTC_CHECK_GT(absl::GetFlag(FLAGS_num_channels), 0); in main()
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| /external/webrtc/modules/audio_coding/codecs/opus/test/ |
| D | lapped_transform.cc | 84 RTC_CHECK_GT(block_length_, 0); in LappedTransform()
85 RTC_CHECK_GT(chunk_length_, 0); in LappedTransform()
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| D | lapped_transform.h | 30 RTC_CHECK_GT(alignment, 0); in AlignedArray()
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| /external/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_abs_send_time.h | 58 RTC_CHECK_GT(send_mean_ms, 0.0f); in GetSendBitrateBps()
63 RTC_CHECK_GT(recv_mean_ms, 0.0f); in GetRecvBitrateBps()
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| /external/webrtc/modules/video_coding/utility/ |
| D | ivf_file_writer.cc | 116 RTC_CHECK_GT(width_, 0); in InitFromFirstFrame()
117 RTC_CHECK_GT(height_, 0); in InitFromFirstFrame()
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| /external/webrtc/test/pc/e2e/ |
| D | peer_configurer.cc | 104 RTC_CHECK_GT(run_params.video_encoder_bitrate_multiplier, 0.0); in ValidateParams()
200 RTC_CHECK_GT(media_streams_count, 0) << "No media in the call."; in ValidateParams()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | resample_input_audio_file.cc | 39 RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set."; in Read()
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| D | constant_pcm_packet_source.cc | 39 RTC_CHECK_GT(packet_len_bytes_, kHeaderLenBytes); in NextPacket()
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| D | neteq_quality_test.cc | 220 RTC_CHECK_GT(absl::GetFlag(FLAGS_drift_factor), -0.1) in NetEqQualityTest()
366 RTC_CHECK_GT(loss_events.size(), 0); in SetUp()
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| /external/webrtc/system_wrappers/source/ |
| D | cpu_info.cc | 55 RTC_CHECK_GT(number_of_cores, 0); in DetectNumberOfCores()
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| /external/webrtc/modules/audio_coding/neteq/test/ |
| D | neteq_speed_test.cc | 42 RTC_CHECK_GT(absl::GetFlag(FLAGS_runtime_ms), 0); in main()
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| /external/webrtc/modules/audio_coding/audio_network_adaptor/util/ |
| D | threshold_curve_unittest.cc | 79 RTC_CHECK_GT((p1.x + p2.x) / 2, p1.x); in TEST()
82 RTC_CHECK_GT((p1.y + p2.y) / 2, p2.y); in TEST()
164 RTC_CHECK_GT((p1.x + p2.x) / 2, p1.x); in TEST()
236 RTC_CHECK_GT((p1.y + p2.y) / 2, p2.y); in TEST()
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| /external/webrtc/rtc_base/numerics/ |
| D | moving_median_filter.h | 57 RTC_CHECK_GT(window_size, 0); in MovingMedianFilter()
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| /external/webrtc/rtc_tools/converter/ |
| D | yuv_to_ivf_converter.cc | 262 RTC_CHECK_GT(width, 0) << "width must be greater then 0"; in main()
263 RTC_CHECK_GT(height, 0) << "height must be greater then 0"; in main()
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| /external/webrtc/rtc_base/ |
| D | checks.h | 407 #define RTC_CHECK_GT(val1, val2) RTC_CHECK_OP(Gt, >, val1, val2) macro
419 #define RTC_DCHECK_GT(v1, v2) RTC_CHECK_GT(v1, v2)
466 #define RTC_CHECK_GT(a, b) RTC_CHECK((a) > (b)) macro
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| /external/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_encoder_pcm.cc | 33 RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz"; in AudioEncoderPcm()
|