| /external/webrtc/media/base/ |
| D | fake_rtp.cc | 28 RTC_CHECK_GE(packet1_size, 12 + 4); in CompareHeaderExtensions()
29 RTC_CHECK_GE(packet2_size, 12 + 4); in CompareHeaderExtensions()
42 RTC_CHECK_GE(packet1_size, 12 + 4 + extension_words * 4); in CompareHeaderExtensions()
43 RTC_CHECK_GE(packet2_size, 12 + 4 + extension_words * 4); in CompareHeaderExtensions()
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| D | fake_frame_source.cc | 32 RTC_CHECK_GE(next_timestamp_us_, 0); in FakeFrameSource()
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| /external/webrtc/modules/audio_processing/agc2/ |
| D | limiter_db_gain_curve.cc | 83 RTC_CHECK_GE(max_input_level_db_, knee_start_dbfs_ + knee_smoothness_db_); in LimiterDbGainCurve()
111 RTC_CHECK_GE(x, limiter_start_linear_ - 1e-7 * kMaxAbsFloatS16Value); in GetGainFirstDerivativeLinear()
119 RTC_CHECK_GE(x0, limiter_start_linear_); // Beyond-knee region only. in GetGainIntegralLinear()
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| /external/webrtc/modules/audio_processing/vad/ |
| D | voice_activity_detector.cc | 72 RTC_CHECK_GE( in ProcessChunk()
76 RTC_CHECK_GE(pitch_based_vad_.VoicingProbability( in ProcessChunk()
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| /external/webrtc/api/video/ |
| D | i010_buffer.cc | 220 RTC_CHECK_GE(offset_x, 0); in CropAndScaleFrom()
221 RTC_CHECK_GE(offset_y, 0); in CropAndScaleFrom()
252 RTC_CHECK_GE(offset_col, 0); in PasteFrom()
253 RTC_CHECK_GE(offset_row, 0); in PasteFrom()
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| D | i420_buffer.cc | 193 RTC_CHECK_GE(offset_x, 0); in CropAndScaleFrom()
194 RTC_CHECK_GE(offset_y, 0); in CropAndScaleFrom()
235 RTC_CHECK_GE(offset_col, 0); in PasteFrom()
236 RTC_CHECK_GE(offset_row, 0); in PasteFrom()
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| /external/webrtc/api/test/ |
| D | create_peer_connection_quality_test_frame_generator.cc | 48 RTC_CHECK_GE(screen_share_config.scrolling_params->source_width, in ValidateScreenShareConfig()
50 RTC_CHECK_GE(screen_share_config.scrolling_params->source_height, in ValidateScreenShareConfig()
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| /external/webrtc/sdk/android/native_api/base/ |
| D | init.cc | 20 RTC_CHECK_GE(jni::InitGlobalJniVariables(jvm), 0); in InitAndroid()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | fake_decode_from_file.cc | 87 RTC_CHECK_GE(encoded_len, 12); in DecodeInternal()
157 RTC_CHECK_GE(encoded.size(), 12); in PrepareEncoded()
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| D | input_audio_file.cc | 73 RTC_CHECK_GE(new_pos, 0) in Seek()
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| /external/webrtc/rtc_tools/frame_analyzer/ |
| D | video_color_aligner.cc | 44 RTC_CHECK_GE(u_data.size() * 2, y_data.size()); in CalculateYChannel()
45 RTC_CHECK_GE(v_data.size() * 2, y_data.size()); in CalculateYChannel()
142 RTC_CHECK_GE(reference_video->number_of_frames(), in CalculateColorTransformationMatrix()
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| /external/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/ |
| D | fake_polqa.cc | 38 RTC_CHECK_GE(argc, 3) << kErrorMessage; in main()
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| /external/webrtc/test/pc/e2e/ |
| D | peer_configurer.cc | 105 RTC_CHECK_GE(run_params.video_codecs.size(), 1); in ValidateParams()
150 RTC_CHECK_GE(*video_config.simulcast_config->target_spatial_index, 0); in ValidateParams()
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| /external/webrtc/common_audio/ |
| D | real_fourier.cc | 33 RTC_CHECK_GE(order, 0); in FftLength()
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| D | audio_converter.cc | 126 RTC_CHECK_GE(converters_.size(), 2); in CompositionConverter()
216 RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames()); in CheckSizes()
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| D | real_fourier_ooura.cc | 45 RTC_CHECK_GE(fft_order, 1); in RealFourierOoura()
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| /external/webrtc/modules/audio_coding/neteq/test/ |
| D | neteq_speed_test.cc | 43 RTC_CHECK_GE(absl::GetFlag(FLAGS_lossrate), 0); in main()
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| /external/webrtc/modules/audio_processing/test/conversational_speech/ |
| D | simulator.cc | 137 RTC_CHECK_GE(interval_begin, wav_writer->num_samples()); in PadLeftWriteChunk()
153 RTC_CHECK_GE(pad_samples, wav_writer->num_samples()); in PadRightWrite()
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| D | timing.cc | 33 RTC_CHECK_GE(fields.size(), 3); in LoadTiming()
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| /external/webrtc/modules/video_coding/ |
| D | frame_dependencies_calculator.cc | 33 RTC_CHECK_GE(buffer_usage.id, 0); in FromBuffersUsage()
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| /external/webrtc/rtc_base/numerics/ |
| D | histogram_percentile_counter.cc | 54 RTC_CHECK_GE(fraction, 0.0); in GetPercentile()
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| /external/webrtc/common_audio/resampler/ |
| D | push_sinc_resampler.cc | 53 RTC_CHECK_GE(destination_capacity, destination_frames_); in Resample()
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| /external/webrtc/rtc_base/ |
| D | checks.h | 406 #define RTC_CHECK_GE(val1, val2) RTC_CHECK_OP(Ge, >=, val1, val2) macro
418 #define RTC_DCHECK_GE(v1, v2) RTC_CHECK_GE(v1, v2)
465 #define RTC_CHECK_GE(a, b) RTC_CHECK((a) >= (b)) macro
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| /external/webrtc/call/ |
| D | call_factory.cc | 52 RTC_CHECK_GE(queue_length, 0); in ParseDegradationConfig()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_format_h264.cc | 169 RTC_CHECK_GE(payload_size_left, fragment.size()); in PacketizeStapA()
264 RTC_CHECK_GE(payload_capacity, kNalHeaderSize); in NextAggregatePacket()
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