/external/webrtc/api/video/ |
D | video_bitrate_allocation.cc | 27 RTC_CHECK_LT(spatial_index, kMaxSpatialLayers); in SetBitrate() 28 RTC_CHECK_LT(temporal_index, kMaxTemporalStreams); in SetBitrate() 47 RTC_CHECK_LT(spatial_index, kMaxSpatialLayers); in HasBitrate() 48 RTC_CHECK_LT(temporal_index, kMaxTemporalStreams); in HasBitrate() 54 RTC_CHECK_LT(spatial_index, kMaxSpatialLayers); in GetBitrate() 55 RTC_CHECK_LT(temporal_index, kMaxTemporalStreams); in GetBitrate() 62 RTC_CHECK_LT(spatial_index, kMaxSpatialLayers); in IsSpatialLayerUsed() 73 RTC_CHECK_LT(spatial_index, kMaxSpatialLayers); in GetSpatialLayerSum() 80 RTC_CHECK_LT(spatial_index, kMaxSpatialLayers); in GetTemporalLayerSum() 81 RTC_CHECK_LT(temporal_index, kMaxTemporalStreams); in GetTemporalLayerSum() [all …]
|
/external/webrtc/test/pc/e2e/analyzer/video/ |
D | multi_head_queue.h | 47 RTC_CHECK_LT(index, queues_.size()); in PopFront() 58 RTC_CHECK_LT(index, queues_.size()); in Front() 68 RTC_CHECK_LT(index, queues_.size()); in IsEmpty() 86 RTC_CHECK_LT(index, queues_.size()); in size()
|
D | default_encoded_image_data_injector.cc | 57 RTC_CHECK_LT(source.size(), 1U << 31) << "High bit is already in use"; in InjectData()
|
D | single_process_encoded_image_data_injector.cc | 138 RTC_CHECK_LT(frame_index, frame_sl_index.size()) in ExtractData()
|
/external/webrtc/test/scenario/ |
D | call_client.cc | 293 RTC_CHECK_LT(next_video_ssrc_index_, kNumSsrcs); in GetNextVideoSsrc() 298 RTC_CHECK_LT(next_video_local_ssrc_index_, kNumSsrcs); in GetNextVideoLocalSsrc() 303 RTC_CHECK_LT(next_audio_ssrc_index_, 1); in GetNextAudioSsrc() 309 RTC_CHECK_LT(next_audio_local_ssrc_index_, 1); in GetNextAudioLocalSsrc() 315 RTC_CHECK_LT(next_rtx_ssrc_index_, kNumSsrcs); in GetNextRtxSsrc()
|
/external/webrtc/modules/audio_processing/agc2/ |
D | compute_interpolated_gain_curve.cc | 40 RTC_CHECK_LT(x0, x1); in ComputeAreaUnderPiecewiseLinearApproximation() 50 RTC_CHECK_LT(x0, x_split); in ComputeAreaUnderPiecewiseLinearApproximation() 51 RTC_CHECK_LT(x_split, x1); in ComputeAreaUnderPiecewiseLinearApproximation()
|
D | limiter_db_gain_curve.cc | 25 RTC_CHECK_LT((compression_ratio - 1.0) * knee_smoothness_db / in ComputeKneeStart()
|
/external/webrtc/modules/audio_coding/audio_network_adaptor/util/ |
D | threshold_curve_unittest.cc | 80 RTC_CHECK_LT((p1.x + p2.x) / 2, p2.x); in TEST() 81 RTC_CHECK_LT((p1.y + p2.y) / 2, p1.y); in TEST() 110 RTC_CHECK_LT(((p1.y + p2.y) / 2) + 1, p1.y); in TEST() 111 RTC_CHECK_LT(p2.y, ((p1.y + p2.y) / 2) - 1); in TEST() 165 RTC_CHECK_LT((p1.x + p2.x) / 2, p2.x); in TEST() 235 RTC_CHECK_LT((p1.y + p2.y) / 2, p1.y); in TEST()
|
/external/webrtc/rtc_base/ |
D | unique_id_generator.cc | 29 RTC_CHECK_LT(known_ids_.size(), std::numeric_limits<uint32_t>::max() - 1); in GenerateId()
|
D | checks.h | 405 #define RTC_CHECK_LT(val1, val2) RTC_CHECK_OP(Lt, <, val1, val2) macro 417 #define RTC_DCHECK_LT(v1, v2) RTC_CHECK_LT(v1, v2) 464 #define RTC_CHECK_LT(a, b) RTC_CHECK((a) < (b)) macro
|
D | unique_id_generator.h | 120 RTC_CHECK_LT(counter_, std::numeric_limits<TIntegral>::max()); in GenerateNumber()
|
/external/webrtc/test/ |
D | explicit_key_value_config.cc | 31 RTC_CHECK_LT(field_start, s.size()) in ExplicitKeyValueConfig()
|
/external/webrtc/video/ |
D | quality_threshold.cc | 35 RTC_CHECK_LT(low_threshold, high_threshold); in QualityThreshold()
|
/external/webrtc/test/fuzzers/ |
D | rtp_dependency_descriptor_fuzzer.cc | 60 RTC_CHECK_LT(value_size, 256); in FuzzOneInput()
|
/external/webrtc/sdk/android/src/jni/ |
D | jvm.cc | 90 RTC_CHECK_LT(snprintf(buf, sizeof(buf), "%ld", in GetThreadId()
|
/external/webrtc/test/pc/e2e/ |
D | peer_configurer.cc | 151 RTC_CHECK_LT(*video_config.simulcast_config->target_spatial_index, in ValidateParams()
|
/external/webrtc/modules/audio_processing/logging/ |
D | apm_data_dumper.h | 67 RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength); in SetOutputDirectory()
|
/external/webrtc/modules/audio_processing/aec3/ |
D | stationarity_estimator.cc | 116 RTC_CHECK_LT(0.f, noise); in EstimateBandStationarity()
|
/external/webrtc/api/test/ |
D | peerconnection_quality_test_fixture.h | 148 RTC_CHECK_LT(target_spatial_index, simulcast_streams_count); in VideoSimulcastConfig()
|
/external/webrtc/rtc_tools/ |
D | video_file_reader.cc | 55 RTC_CHECK_LT(frame_index, frame_positions_.size()); in GetFrame()
|
/external/webrtc/modules/video_coding/utility/ |
D | simulcast_rate_allocator.cc | 54 RTC_CHECK_LT(temporal_id, num_layers); in GetTemporalRateAllocation()
|
/external/webrtc/modules/rtp_rtcp/source/ |
D | forward_error_correction_internal.cc | 503 RTC_CHECK_LT(new_bit_index, 8 * new_mask_bytes); in CopyColumn()
|
/external/webrtc/modules/video_coding/codecs/test/ |
D | videocodec_test_stats_impl.cc | 48 RTC_CHECK_LT(frame_num, layer_stats_[layer_idx].size()); in GetFrame()
|
/external/webrtc/media/engine/ |
D | simulcast.cc | 108 RTC_CHECK_LT(simulcast_id, webrtc::kMaxSimulcastStreams); in DefaultNumberOfTemporalLayers()
|
/external/webrtc/modules/video_coding/codecs/vp8/ |
D | screenshare_layers.cc | 425 RTC_CHECK_LT(num_layers, 3); in GetTemplateStructure()
|