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