| /external/webrtc/webrtc/common_audio/ |
| window_generator.cc | 41 RTC_CHECK_GT(length, 1);
51 RTC_CHECK_GT(length, 1U);
|
| real_fourier.cc | 33 RTC_CHECK_GT(length, 0U);
|
| lapped_transform.cc | 87 RTC_CHECK_GT(block_length_, 0u);
88 RTC_CHECK_GT(chunk_length_, 0u);
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| resample_input_audio_file.cc | 38 RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
|
| constant_pcm_packet_source.cc | 39 RTC_CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
|
| /external/webrtc/webrtc/system_wrappers/include/ |
| aligned_array.h | 26 RTC_CHECK_GT(alignment, 0u);
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_abs_send_time.h | 49 RTC_CHECK_GT(send_mean_ms, 0.0f);
54 RTC_CHECK_GT(recv_mean_ms, 0.0f);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.cc | 47 RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
|
| /external/webrtc/webrtc/base/ |
| checks.h | 161 #define RTC_CHECK_GT(val1, val2) RTC_CHECK_OP(GT, > , val1, val2)
174 #define RTC_DCHECK_GT(v1, v2) RTC_CHECK_GT(v1, v2)
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| array_util.cc | 26 RTC_CHECK_GT(array_geometry.size(), 1u);
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| test_utils.cc | 135 RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng.cc | 251 RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| nack.cc | 198 RTC_CHECK_GT(max_nack_list_size, 0u);
|
| /external/webrtc/webrtc/video/ |
| video_quality_test.cc | 768 RTC_CHECK_GT(v[2], 0);
872 RTC_CHECK_GT(params_.screenshare.slide_change_interval, 0);
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| RTPencode.cc | [all...] |
