| /external/webrtc/webrtc/modules/audio_processing/logging/ |
| D | aec_logging_file_handling.cc | 37 RTC_DCHECK_GE(written, 0); in WebRtcAec_ReopenWav()
50 RTC_DCHECK_GE(written, 0); in WebRtcAec_RawFileOpen()
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| /external/webrtc/webrtc/video/ |
| D | video_send_stream.cc | 177 RTC_DCHECK_GE(id, 1); in VideoSendStream()
218 RTC_DCHECK_GE(config.encoder_settings.payload_type, 0); in VideoSendStream()
303 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size()); in ReconfigureVideoEncoder()
398 RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0); in ReconfigureVideoEncoder()
399 RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps); in ReconfigureVideoEncoder()
400 RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps); in ReconfigureVideoEncoder()
401 RTC_DCHECK_GE(streams[i].max_qp, 0); in ReconfigureVideoEncoder()
442 RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0); in ReconfigureVideoEncoder()
494 RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0); in ConfigureSsrcs()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_decoder_pcm.h | 22 RTC_DCHECK_GE(num_channels, 1u); in AudioDecoderPcmU()
43 RTC_DCHECK_GE(num_channels, 1u); in AudioDecoderPcmA()
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| /external/webrtc/webrtc/common_video/ |
| D | video_frame_buffer.cc | 47 RTC_DCHECK_GE(stride_y, width); in I420Buffer()
48 RTC_DCHECK_GE(stride_u, (width + 1) / 2); in I420Buffer()
49 RTC_DCHECK_GE(stride_v, (width + 1) / 2); in I420Buffer()
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| D | video_frame.cc | 67 RTC_DCHECK_GE(stride_y, width); in CreateEmptyFrame()
68 RTC_DCHECK_GE(stride_u, half_width); in CreateEmptyFrame()
69 RTC_DCHECK_GE(stride_v, half_width); in CreateEmptyFrame()
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| /external/webrtc/webrtc/test/ |
| D | frame_generator.cc | 151 RTC_DCHECK_GE(source_height, target_height); in ScrollingImageFrameGenerator()
152 RTC_DCHECK_GE(source_width, target_width); in ScrollingImageFrameGenerator()
153 RTC_DCHECK_GE(scroll_time_ms, 0); in ScrollingImageFrameGenerator()
154 RTC_DCHECK_GE(pause_time_ms, 0); in ScrollingImageFrameGenerator()
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | noise_suppression_impl.cc | 78 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); in AnalyzeCaptureAudio()
94 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); in ProcessCaptureAudio()
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| D | high_pass_filter_impl.cc | 111 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); in ProcessCaptureAudio()
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| D | voice_detection_impl.cc | 65 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); in ProcessCaptureAudio()
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| /external/webrtc/talk/app/webrtc/java/jni/ |
| D | jni_onload.cc | 40 RTC_DCHECK_GE(ret, 0); in JNI_OnLoad()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| D | audio_decoder_pcm16b.cc | 20 RTC_DCHECK_GE(num_channels, 1u); in AudioDecoderPcm16B()
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| /external/webrtc/webrtc/base/ |
| D | checks.h | 173 #define RTC_DCHECK_GE(v1, v2) RTC_CHECK_GE(v1, v2) macro
182 #define RTC_DCHECK_GE(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) >= (v2))
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| D | macifaddrs_converter.cc | 194 RTC_DCHECK_GE(ioctl_socket_, 0); in IPv6AttributesGetter()
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| D | timeutils.cc | 139 RTC_DCHECK_GE(elapsed, 0); in TimeAfter()
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| D | filerotatingstream.cc | 363 RTC_DCHECK_GE(max_total_log_size, 4u); in CallSessionFileRotatingStream()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_encoder_opus.cc | 44 RTC_DCHECK_GE(new_loss_rate, 0.0); in OptimizePacketLossRate()
46 RTC_DCHECK_GE(old_loss_rate, 0.0); in OptimizePacketLossRate()
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| /external/webrtc/webrtc/call/ |
| D | call.cc | 205 RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); in Call()
206 RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, in Call()
209 RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps, in Call()
526 RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0); in SetBitrateConfig()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_receiver_video.cc | 64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength); in ParseRtpPacket()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| D | audio_encoder_ilbc.cc | 97 RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes()); in EncodeInternal()
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| /external/webrtc/webrtc/common_audio/ |
| D | channel_buffer.h | 95 RTC_DCHECK_GE(channel, 0u); in bands()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | nack.cc | 220 RTC_DCHECK_GE(round_trip_time_ms, 0); in GetNackList()
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| D | statistics_calculator.cc | 49 RTC_DCHECK_GE(timer_, 0); in AdvanceClock()
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| /external/webrtc/webrtc/modules/bitrate_controller/ |
| D | send_side_bandwidth_estimation.cc | 79 RTC_DCHECK_GE(min_bitrate, 0); in SetMinMaxBitrate()
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| /external/webrtc/webrtc/modules/video_coding/ |
| D | codec_database.cc | 200 RTC_DCHECK_GE(number_of_cores, 1); in SetSendCodec()
201 RTC_DCHECK_GE(send_codec->plType, 1); in SetSendCodec()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| D | transport_feedback.cc | 523 RTC_DCHECK_GE(first_symbol_cardinality_, symbol_vec_.size()); in EmitRunLengthChunk()
743 RTC_DCHECK_GE(index, end_index - 3); in ParseFrom()
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