| /external/webrtc/test/fuzzers/ |
| D | rtp_packetizer_av1_fuzzer.cc | 50 RTC_CHECK_LE(rtp_packet.payload_size(), in FuzzOneInput()
56 RTC_CHECK_LE(rtp_packet.payload_size(), in FuzzOneInput()
62 RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len) in FuzzOneInput()
67 RTC_CHECK_LE(rtp_packet.payload_size(), in FuzzOneInput()
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| D | rtp_dependency_descriptor_fuzzer.cc | 57 RTC_CHECK_LE(value_size, raw.size()); in FuzzOneInput()
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| /external/webrtc/api/video/ |
| D | i010_buffer.cc | 216 RTC_CHECK_LE(crop_width, src.width()); in CropAndScaleFrom()
217 RTC_CHECK_LE(crop_height, src.height()); in CropAndScaleFrom()
218 RTC_CHECK_LE(crop_width + offset_x, src.width()); in CropAndScaleFrom()
219 RTC_CHECK_LE(crop_height + offset_y, src.height()); in CropAndScaleFrom()
250 RTC_CHECK_LE(picture.width() + offset_col, width()); in PasteFrom()
251 RTC_CHECK_LE(picture.height() + offset_row, height()); in PasteFrom()
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| D | i420_buffer.cc | 189 RTC_CHECK_LE(crop_width, src.width()); in CropAndScaleFrom()
190 RTC_CHECK_LE(crop_height, src.height()); in CropAndScaleFrom()
191 RTC_CHECK_LE(crop_width + offset_x, src.width()); in CropAndScaleFrom()
192 RTC_CHECK_LE(crop_height + offset_y, src.height()); in CropAndScaleFrom()
233 RTC_CHECK_LE(picture.width() + offset_col, width()); in PasteFrom()
234 RTC_CHECK_LE(picture.height() + offset_row, height()); in PasteFrom()
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| /external/webrtc/audio/utility/ |
| D | channel_mixer.cc | 46 RTC_CHECK_LE(frame->samples_per_channel() * output_channels_, in Transform()
84 RTC_CHECK_LE(index, audio_vector_size_); in Transform()
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| /external/webrtc/video/ |
| D | buffered_frame_decryptor.cc | 74 RTC_CHECK_LE(max_plaintext_byte_size, frame->size()); in DecryptFrame()
102 RTC_CHECK_LE(decrypt_result.bytes_written, max_plaintext_byte_size); in DecryptFrame()
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| /external/webrtc/modules/audio_coding/codecs/opus/test/ |
| D | lapped_transform.h | 51 RTC_CHECK_LE(row, rows_); in Row()
56 RTC_CHECK_LE(row, rows_); in Row()
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| D | blocker.cc | 119 RTC_CHECK_LE(num_output_channels_, num_input_channels_); in Blocker()
120 RTC_CHECK_LE(shift_amount_, block_size_); in Blocker()
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| /external/webrtc/test/pc/e2e/analyzer/video/ |
| D | default_encoded_image_data_injector.cc | 78 RTC_CHECK_LE(insertion_pos + kEncodedImageBufferExpansion, source.size()); in ExtractData()
113 RTC_CHECK_LE(source_pos + kEncodedImageBufferExpansion + info.length, in ExtractData()
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| /external/webrtc/sdk/android/src/jni/pc/ |
| D | data_channel.cc | 120 RTC_CHECK_LE(id, std::numeric_limits<int32_t>::max()) in JNI_DataChannel_Id()
134 RTC_CHECK_LE(buffered_amount, std::numeric_limits<int64_t>::max()) in JNI_DataChannel_BufferedAmount()
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| /external/webrtc/api/audio/ |
| D | audio_frame.cc | 90 RTC_CHECK_LE(length, kMaxDataSizeSamples); in UpdateFrame()
117 RTC_CHECK_LE(length, kMaxDataSizeSamples); in CopyFrom()
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| /external/webrtc/modules/audio_coding/codecs/cng/ |
| D | webrtc_cng.cc | 218 RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
223 RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER); in Reset()
259 RTC_CHECK_LE(num_samples, kCngMaxOutsizeOrder); in Encode()
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| /external/webrtc/test/ |
| D | mock_audio_encoder.cc | 52 RTC_CHECK_LE(info_.encoded_bytes, payload_.size()); in operator ()()
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| D | frame_generator_capturer.cc | 119 RTC_CHECK_LE(crop_width, config.width); in Create()
120 RTC_CHECK_LE(crop_height, config.height); in Create()
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| /external/webrtc/api/ |
| D | rtc_event_log_output_file.cc | 47 RTC_CHECK_LE(max_size_bytes_, kMaxReasonableFileSize); in RtcEventLogOutputFile()
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| /external/webrtc/rtc_base/numerics/ |
| D | histogram_percentile_counter.cc | 53 RTC_CHECK_LE(fraction, 1.0); in GetPercentile()
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| D | samples_stats_counter.cc | 48 RTC_CHECK_LE(percentile, 1); in GetPercentile()
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| D | percentile_filter.h | 64 RTC_CHECK_LE(percentile, 1.0f); in PercentileFilter()
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| /external/webrtc/modules/audio_processing/test/conversational_speech/ |
| D | timing.cc | 34 RTC_CHECK_LE(fields.size(), 4); in LoadTiming()
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| /external/webrtc/rtc_base/ |
| D | checks.h | 404 #define RTC_CHECK_LE(val1, val2) RTC_CHECK_OP(Le, <=, val1, val2) macro
416 #define RTC_DCHECK_LE(v1, v2) RTC_CHECK_LE(v1, v2)
463 #define RTC_CHECK_LE(a, b) RTC_CHECK((a) <= (b)) macro
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtcp_packet.cc | 33 RTC_CHECK_LE(max_length, IP_PACKET_SIZE); in Build()
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| /external/webrtc/test/testsupport/ |
| D | video_frame_writer.cc | 32 RTC_CHECK_LE(std::abs(static_cast<double>(width) / height - in ExtractI420BufferWithSize()
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | send_side_bandwidth_estimation.cc | 81 RTC_CHECK_LE(*low_loss_threshold, 1.0f) in ReadBweLossExperimentParameters()
85 RTC_CHECK_LE(*high_loss_threshold, 1.0f) in ReadBweLossExperimentParameters()
87 RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold) in ReadBweLossExperimentParameters()
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| /external/webrtc/api/test/ |
| D | create_peer_connection_quality_test_frame_generator.cc | 46 RTC_CHECK_LE(screen_share_config.scrolling_params->duration, in ValidateScreenShareConfig()
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| /external/webrtc/modules/audio_processing/agc2/ |
| D | limiter_db_gain_curve.cc | 118 RTC_CHECK_LE(x0, x1); // Valid interval. in GetGainIntegralLinear()
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