/external/webrtc/webrtc/system_wrappers/include/ |
aligned_array.h | 51 RTC_CHECK_LE(row, rows_); 56 RTC_CHECK_LE(row, rows_); 61 RTC_CHECK_LE(col, cols_); 66 RTC_CHECK_LE(col, cols_);
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/external/webrtc/webrtc/common_audio/ |
real_fourier_openmax.cc | 28 RTC_CHECK_LE(order, TWIDDLE_TABLE_ORDER);
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blocker.cc | 121 RTC_CHECK_LE(num_output_channels_, num_input_channels_); 122 RTC_CHECK_LE(shift_amount_, block_size_);
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wav_file.cc | 77 RTC_CHECK_LE(read, num_samples_remaining_);
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
audio_encoder.cc | 36 RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
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/external/webrtc/webrtc/modules/audio_device/ |
fine_audio_buffer.cc | 99 RTC_CHECK_LE(bytes_left, 0); 108 RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
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/external/webrtc/webrtc/base/ |
checks.h | 158 #define RTC_CHECK_LE(val1, val2) RTC_CHECK_OP(LE, <=, val1, val2) 171 #define RTC_DCHECK_LE(v1, v2) RTC_CHECK_LE(v1, v2)
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/external/webrtc/webrtc/common_video/ |
video_frame_buffer.cc | 212 RTC_CHECK_LE(cropped_width, buffer->width()); 213 RTC_CHECK_LE(cropped_height, buffer->height());
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
audio_encoder_isac_t_impl.h | 136 RTC_CHECK_LE(static_cast<size_t>(r), max_encoded_bytes);
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/external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
audio_encoder_cng.cc | 115 RTC_CHECK_LE(static_cast<int>(frames_to_encode * 10), kMaxFrameSizeMs)
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
nack.cc | 201 RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal);
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/external/webrtc/webrtc/video/ |
video_quality_test.cc | 663 RTC_CHECK_LE(params_.ss.selected_sl, params_.ss.num_spatial_layers); 755 RTC_CHECK_LE(params->common.num_temporal_layers, 2); 870 RTC_CHECK_LE(params_.common.width, kWidth); 871 RTC_CHECK_LE(params_.common.height, kHeight); 876 RTC_CHECK_LE(params_.screenshare.scroll_duration, [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
audio_encoder_copy_red_unittest.cc | 93 RTC_CHECK_LE(info_.encoded_bytes, max_encoded_bytes);
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/external/webrtc/webrtc/modules/audio_processing/test/ |
debug_dump_test.cc | 223 RTC_CHECK_LE(out_channels, channels);
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/external/webrtc/talk/app/webrtc/java/jni/ |
peerconnection_jni.cc | [all...] |