| /external/webrtc/talk/app/webrtc/test/ |
| androidtestinitializer.cc | 61 RTC_CHECK_GE(webrtc_jni::InitGlobalJniVariables(jvm), 0);
|
| /external/webrtc/webrtc/common_audio/ |
| sparse_fir_filter.cc | 25 RTC_CHECK_GE(num_nonzero_coeffs, 1u);
26 RTC_CHECK_GE(sparsity, 1u);
|
| real_fourier.cc | 38 RTC_CHECK_GE(order, 0);
|
| real_fourier_openmax.cc | 26 RTC_CHECK_GE(order, 1);
|
| real_fourier_ooura.cc | 45 RTC_CHECK_GE(fft_order, 1);
|
| audio_converter.cc | 110 RTC_CHECK_GE(converters_.size(), 2u);
198 RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
|
| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| voice_activity_detector.cc | 73 RTC_CHECK_GE(
77 RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
|
| /external/webrtc/webrtc/common_audio/resampler/ |
| push_sinc_resampler.cc | 54 RTC_CHECK_GE(destination_capacity, destination_frames_);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng.cc | 103 RTC_CHECK_GE(max_encoded_bytes,
127 RTC_CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
210 RTC_CHECK_GE(max_encoded_bytes, frames_to_encode * samples_per_10ms_frame);
218 RTC_CHECK_GE(WebRtcCng_Encode(cng_inst_.get(),
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| input_audio_file.cc | 55 RTC_CHECK_GE(new_pos, 0)
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.cc | 93 RTC_CHECK_GE(max_encoded_bytes, full_frame_samples_);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| audio_encoder_ilbc.cc | 122 RTC_CHECK_GE(output_len, 0);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| audio_encoder_copy_red.cc | 62 RTC_CHECK_GE(max_encoded_bytes,
|
| /external/webrtc/webrtc/modules/audio_device/ |
| fine_audio_buffer.cc | 96 RTC_CHECK_GE(bytes_left, 0);
|
| /external/webrtc/webrtc/base/ |
| checks.h | 160 #define RTC_CHECK_GE(val1, val2) RTC_CHECK_OP(GE, >=, val1, val2)
173 #define RTC_DCHECK_GE(v1, v2) RTC_CHECK_GE(v1, v2)
|
| random_unittest.cc | 25 RTC_CHECK_GE(n, static_cast<T>(0));
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_send_test_oldapi.cc | 96 RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_encoder_g722.cc | 99 RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| audio_encoder_isac_t_impl.h | 131 RTC_CHECK_GE(r, 0) << "Encode failed (error code "
|
| /external/webrtc/webrtc/video/ |
| video_quality_test.cc | 646 RTC_CHECK_GE(params_.common.max_bitrate_bps,
648 RTC_CHECK_GE(params_.common.target_bitrate_bps,
653 RTC_CHECK_GE(stream.min_bitrate_bps, 0);
654 RTC_CHECK_GE(stream.target_bitrate_bps, stream.min_bitrate_bps);
655 RTC_CHECK_GE(stream.max_bitrate_bps, stream.target_bitrate_bps);
662 RTC_CHECK_GE(params_.ss.num_spatial_layers, 1);
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.cc | 152 RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_packet_history.cc | 180 RTC_CHECK_GE(*packet_length, static_cast<size_t>(IP_PACKET_SIZE));
|
| h264_bitstream_parser.cc | 547 RTC_CHECK_GE(length, 4u);
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| gain_control_impl.cc | 371 RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;
|
| /external/webrtc/webrtc/call/ |
| rtc_event_log_unittest.cc | 304 RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
|
