| /external/webrtc/modules/video_coding/ |
| D | video_codec_initializer.cc | 56 RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0); in VideoEncoderConfigToVideoCodec()
101 RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0); in VideoEncoderConfigToVideoCodec()
102 RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps); in VideoEncoderConfigToVideoCodec()
103 RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps); in VideoEncoderConfigToVideoCodec()
104 RTC_DCHECK_GE(streams[i].max_qp, 0); in VideoEncoderConfigToVideoCodec()
154 RTC_DCHECK_GE(video_codec.VP8()->numberOfTemporalLayers, 1); in VideoEncoderConfigToVideoCodec()
171 RTC_DCHECK_GE(video_codec.VP9()->numberOfTemporalLayers, 1); in VideoEncoderConfigToVideoCodec()
235 RTC_DCHECK_GE(video_codec.VP9()->numberOfSpatialLayers, 1); in VideoEncoderConfigToVideoCodec()
241 RTC_DCHECK_GE(video_codec.VP9()->numberOfTemporalLayers, 1); in VideoEncoderConfigToVideoCodec()
253 RTC_DCHECK_GE(video_codec.H264()->numberOfTemporalLayers, 1); in VideoEncoderConfigToVideoCodec()
|
| /external/webrtc/audio/ |
| D | audio_transport_impl.cc | 108 RTC_DCHECK_GE(number_of_channels, 1); in RecordedDataIsAvailable()
111 RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz); in RecordedDataIsAvailable()
182 RTC_DCHECK_GE(nChannels, 1); in NeedMorePlayData()
184 RTC_DCHECK_GE( in NeedMorePlayData()
219 RTC_DCHECK_GE(number_of_channels, 1); in PullRenderData()
220 RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz); in PullRenderData()
|
| /external/webrtc/modules/desktop_capture/win/ |
| D | dxgi_adapter_duplicator.cc | 146 RTC_DCHECK_GE(monitor_id, 0); in DuplicateMonitor()
154 RTC_DCHECK_GE(id, 0); in ScreenRect()
160 RTC_DCHECK_GE(id, 0); in GetDeviceName()
180 RTC_DCHECK_GE(desktop_rect_.left(), 0); in TranslateRect()
181 RTC_DCHECK_GE(desktop_rect_.top(), 0); in TranslateRect()
|
| /external/webrtc/call/ |
| D | rtp_bitrate_configurator.cc | 37 RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0); in RtpBitrateConfigurator()
38 RTC_DCHECK_GE(bitrate_config.start_bitrate_bps, in RtpBitrateConfigurator()
41 RTC_DCHECK_GE(bitrate_config.max_bitrate_bps, in RtpBitrateConfigurator()
55 RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0); in UpdateWithSdpParameters()
|
| /external/webrtc/audio/voip/ |
| D | audio_egress.cc | 43 RTC_DCHECK_GE(payload_type, 0); in SetEncoder()
136 RTC_DCHECK_GE(rtp_payload_type, 0); in RegisterTelephoneEventType()
145 RTC_DCHECK_GE(dtmf_event, 0); in SendTelephoneEvent()
147 RTC_DCHECK_GE(duration_ms, 0); in SendTelephoneEvent()
|
| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_generic_frame_descriptor.cc | 34 RTC_DCHECK_GE(temporal_layer, 0); in SetTemporalLayer()
63 RTC_DCHECK_GE(width, 0); in SetResolution()
65 RTC_DCHECK_GE(height, 0); in SetResolution()
|
| D | rtp_format_vp8.cc | 39 RTC_DCHECK_GE(hdr_info.pictureId, 0); in ValidateHeader()
43 RTC_DCHECK_GE(hdr_info.tl0PicIdx, 0); in ValidateHeader()
47 RTC_DCHECK_GE(hdr_info.temporalIdx, 0); in ValidateHeader()
53 RTC_DCHECK_GE(hdr_info.keyIdx, 0); in ValidateHeader()
|
| D | rtp_dependency_descriptor_writer.cc | 222 RTC_DCHECK_GE(structure_.structure_id, 0); in WriteTemplateDependencyStructure()
270 RTC_DCHECK_GE(fdiff - 1, 0); in WriteTemplateFdiffs()
280 RTC_DCHECK_GE(structure_.num_chains, 0); in WriteTemplateChains()
290 RTC_DCHECK_GE(protected_by, 0); in WriteTemplateChains()
297 RTC_DCHECK_GE(chain_diff, 0); in WriteTemplateChains()
381 RTC_DCHECK_GE(chain_diff, 0); in WriteFrameChains()
|
| /external/webrtc/modules/audio_mixer/ |
| D | audio_frame_manipulator.cc | 37 RTC_DCHECK_GE(start_gain, 0.0f); in Ramp()
38 RTC_DCHECK_GE(target_gain, 0.0f); in Ramp()
59 RTC_DCHECK_GE(target_number_of_channels, 1); in RemixFrame()
|
| /external/webrtc/api/video_codecs/ |
| D | video_encoder.cc | 226 RTC_DCHECK_GE(bitrate_limits[i].min_bitrate_bps, 0); in GetEncoderBitrateLimitsForResolution()
227 RTC_DCHECK_GE(bitrate_limits[i].min_start_bitrate_bps, 0); in GetEncoderBitrateLimitsForResolution()
228 RTC_DCHECK_GE(bitrate_limits[i].max_bitrate_bps, in GetEncoderBitrateLimitsForResolution()
232 RTC_DCHECK_GE(bitrate_limits[i].min_bitrate_bps, in GetEncoderBitrateLimitsForResolution()
234 RTC_DCHECK_GE(bitrate_limits[i].min_start_bitrate_bps, in GetEncoderBitrateLimitsForResolution()
236 RTC_DCHECK_GE(bitrate_limits[i].max_bitrate_bps, in GetEncoderBitrateLimitsForResolution()
|
| D | vp8_temporal_layers_factory.cc | 32 RTC_DCHECK_GE(num_streams, 1); in Create()
38 RTC_DCHECK_GE(num_temporal_layers, 1); in Create()
|
| /external/webrtc/modules/audio_processing/agc2/ |
| D | adaptive_digital_gain_applier.cc | 111 RTC_DCHECK_GE(signal_with_levels.input_level_dbfs, -150.f); in Process()
112 RTC_DCHECK_GE(signal_with_levels.float_frame.num_channels(), 1); in Process()
113 RTC_DCHECK_GE(signal_with_levels.float_frame.samples_per_channel(), 1); in Process()
|
| /external/webrtc/modules/audio_processing/vad/ |
| D | vad_audio_proc.cc | 124 RTC_DCHECK_GE(length_corr, kLpcOrder + 1); in SubframeCorrelation()
140 RTC_DCHECK_GE(length_lpc, kNum10msSubframes * (kLpcOrder + 1)); in GetLpcPolynomials()
179 RTC_DCHECK_GE(length_f_peak, kNum10msSubframes); in FindFirstSpectralPeaks()
235 RTC_DCHECK_GE(length, kNum10msSubframes); in PitchAnalysis()
265 RTC_DCHECK_GE(length_rms, kNum10msSubframes); in Rms()
|
| /external/webrtc/common_video/libyuv/ |
| D | webrtc_libyuv.cc | 24 RTC_DCHECK_GE(width, 0); in CalcBufferSize()
25 RTC_DCHECK_GE(height, 0); in CalcBufferSize()
239 RTC_DCHECK_GE(ref_buffer.width(), test_buffer.width()); in I420APSNR()
240 RTC_DCHECK_GE(ref_buffer.height(), test_buffer.height()); in I420APSNR()
285 RTC_DCHECK_GE(ref_buffer.width(), test_buffer.width()); in I420PSNR()
286 RTC_DCHECK_GE(ref_buffer.height(), test_buffer.height()); in I420PSNR()
316 RTC_DCHECK_GE(ref_buffer.width(), test_buffer.width()); in I420ASSIM()
317 RTC_DCHECK_GE(ref_buffer.height(), test_buffer.height()); in I420ASSIM()
351 RTC_DCHECK_GE(ref_buffer.width(), test_buffer.width()); in I420SSIM()
352 RTC_DCHECK_GE(ref_buffer.height(), test_buffer.height()); in I420SSIM()
|
| /external/webrtc/api/video/ |
| D | encoded_image.cc | 92 RTC_DCHECK_GE(spatial_index, 0); in SpatialLayerFrameSize()
105 RTC_DCHECK_GE(spatial_index, 0); in SetSpatialLayerFrameSize()
107 RTC_DCHECK_GE(size_bytes, 0); in SetSpatialLayerFrameSize()
|
| D | video_adaptation_counters.h | 27 RTC_DCHECK_GE(resolution_adaptations, 0); in VideoAdaptationCounters()
28 RTC_DCHECK_GE(fps_adaptations, 0); in VideoAdaptationCounters()
|
| /external/webrtc/rtc_base/ |
| D | rate_statistics.cc | 58 RTC_DCHECK_GE(count, 0); in Update()
132 RTC_DCHECK_GE(accumulated_count_, oldest_bucket.sum); in EraseOld()
133 RTC_DCHECK_GE(num_samples_, oldest_bucket.num_samples); in EraseOld()
|
| D | bounded_inline_vector.h | 116 RTC_DCHECK_GE(new_size, 0); in resize()
132 RTC_DCHECK_GE(index, 0);
137 RTC_DCHECK_GE(index, 0);
|
| /external/webrtc/rtc_base/numerics/ |
| D | divide_round.h | 25 RTC_DCHECK_GE(dividend, 0); in DivideRoundUp()
37 RTC_DCHECK_GE(dividend, 0); in DivideRoundToNearest()
|
| /external/webrtc/test/logging/ |
| D | log_writer.h | 31 RTC_DCHECK_GE(predicted_length, 0); in LogWriteFormat()
37 RTC_DCHECK_GE(actual_length, 0); in LogWriteFormat()
|
| /external/webrtc/modules/audio_coding/neteq/ |
| D | expand_uma_logger.cc | 58 RTC_DCHECK_GE(last_value_, *last_logged_value_); in UpdateSampleCounter()
64 RTC_DCHECK_GE(rate, 0); in UpdateSampleCounter()
|
| /external/webrtc/modules/audio_processing/aec3/ |
| D | spectrum_buffer.h | 41 RTC_DCHECK_GE(size, offset); in OffsetIndex()
43 RTC_DCHECK_GE(size + index + offset, 0); in OffsetIndex()
|
| D | alignment_mixer.cc | 76 RTC_DCHECK_GE(x.size(), ch); in ProduceOutput()
83 RTC_DCHECK_GE(num_channels_, 2); in Downmix()
98 RTC_DCHECK_GE(num_channels_, 2); in SelectChannel()
|
| /external/webrtc/modules/audio_processing/ |
| D | echo_control_mobile_impl.cc | 132 RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, in PackRenderAudioBuffer()
163 RTC_DCHECK_GE(160, audio->num_frames_per_band()); in ProcessCaptureAudio()
165 RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_reverse_channels * in ProcessCaptureAudio()
179 RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, in ProcessCaptureAudio()
|
| /external/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_decoder_pcm.h | 29 RTC_DCHECK_GE(num_channels, 1); in AudioDecoderPcmU()
53 RTC_DCHECK_GE(num_channels, 1); in AudioDecoderPcmA()
|