| /external/webrtc/webrtc/modules/audio_processing/ |
| D | splitting_filter.cc | 35 RTC_DCHECK_EQ(num_bands_, bands->num_bands()); in Analysis()
36 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels()); in Analysis()
37 RTC_DCHECK_EQ(data->num_frames(), in Analysis()
48 RTC_DCHECK_EQ(num_bands_, bands->num_bands()); in Synthesis()
49 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels()); in Synthesis()
50 RTC_DCHECK_EQ(data->num_frames(), in Synthesis()
61 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels()); in TwoBandsAnalysis()
74 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels()); in TwoBandsSynthesis()
87 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels()); in ThreeBandsAnalysis()
97 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels()); in ThreeBandsSynthesis()
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| D | noise_suppression_impl.cc | 37 RTC_DCHECK_EQ(0, error); in Suppressor()
79 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels()); in AnalyzeCaptureAudio()
95 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels()); in ProcessCaptureAudio()
147 RTC_DCHECK_EQ(0, error); in set_level()
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| D | voice_detection_impl.cc | 23 RTC_DCHECK_EQ(0, error); in Vad()
132 RTC_DCHECK_EQ(0, error); in set_likelihood()
144 RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported. in set_frame_size_ms()
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| /external/webrtc/webrtc/voice_engine/ |
| D | channel_proxy.cc | 37 RTC_DCHECK_EQ(0, error); in SetLocalSSRC()
45 RTC_DCHECK_EQ(0, error); in SetRTCP_CNAME()
51 RTC_DCHECK_EQ(0, error); in SetSendAbsoluteSenderTimeStatus()
57 RTC_DCHECK_EQ(0, error); in SetSendAudioLevelIndicationStatus()
68 RTC_DCHECK_EQ(0, error); in SetReceiveAbsoluteSenderTimeStatus()
74 RTC_DCHECK_EQ(0, error); in SetReceiveAudioLevelIndicationStatus()
90 RTC_DCHECK_EQ(0, error); in GetRTCPStatistics()
98 RTC_DCHECK_EQ(0, error); in GetRemoteRTCPReportBlocks()
106 RTC_DCHECK_EQ(0, error); in GetNetworkStatistics()
121 RTC_DCHECK_EQ(0, error); in GetSpeechOutputLevelFullRange()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| D | audio_encoder_ilbc.cc | 104 RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size()); in EncodeInternal()
115 RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); in EncodeInternal()
125 RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes()); in EncodeInternal()
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| D | audio_decoder_ilbc.cc | 36 RTC_DCHECK_EQ(sample_rate_hz, 8000); in DecodeInternal()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| D | audio_encoder_copy_red.cc | 71 RTC_DCHECK_EQ(info.redundant.size(), 1u); in EncodeInternal()
76 RTC_DCHECK_EQ(info.redundant.size(), 2u); in EncodeInternal()
81 RTC_DCHECK_EQ(info.speech, info.redundant[0].speech); in EncodeInternal()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| D | isac_fix_type.h | 85 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); in SetDecSampRate()
90 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); in SetEncSampRate()
95 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); in SetEncSampRateInDecoder()
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| /external/webrtc/webrtc/audio/ |
| D | audio_send_stream.cc | 171 RTC_DCHECK_EQ(0, error); in GetStats()
177 RTC_DCHECK_EQ(0, error); in GetStats()
185 RTC_DCHECK_EQ(0, error); in GetStats()
196 RTC_DCHECK_EQ(0, error); in GetStats()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_decoder_pcm.cc | 28 RTC_DCHECK_EQ(sample_rate_hz, 8000); in DecodeInternal()
52 RTC_DCHECK_EQ(sample_rate_hz, 8000); in DecodeInternal()
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| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| D | voice_activity_detector.cc | 40 RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100); in ProcessChunk()
51 RTC_DCHECK_EQ(length, kLength10Ms); in ProcessChunk()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| D | app.cc | 52 RTC_DCHECK_EQ(0u, data_length % 4) << "Data must be 32 bits aligned."; in WithData()
74 RTC_DCHECK_EQ(index_end, *index); in Create()
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| D | dlrr.cc | 38 RTC_DCHECK_EQ(block_length_32bits, in Parse()
81 RTC_DCHECK_EQ(buffer + BlockLength(), write_at); in Create()
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| D | extended_jitter_report.cc | 90 RTC_DCHECK_EQ(index_end, *index); in Create()
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| /external/webrtc/webrtc/common_audio/ |
| D | audio_ring_buffer.cc | 33 RTC_DCHECK_EQ(buffers_.size(), channels); in Write()
41 RTC_DCHECK_EQ(buffers_.size(), channels); in Read()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_decoder_opus.cc | 33 RTC_DCHECK_EQ(sample_rate_hz, 48000); in DecodeInternal()
54 RTC_DCHECK_EQ(sample_rate_hz, 48000); in DecodeRedundantInternal()
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| /external/webrtc/webrtc/modules/video_coding/codecs/h264/ |
| D | h264_video_toolbox_nalu.cc | 54 RTC_DCHECK_EQ(nalu_header_size, 4); in H264CMSampleBufferToAnnexBBuffer()
55 RTC_DCHECK_EQ(param_set_count, 2u); in H264CMSampleBufferToAnnexBBuffer()
140 RTC_DCHECK_EQ(bytes_remaining, (size_t)0); in H264CMSampleBufferToAnnexBBuffer()
145 RTC_DCHECK_EQ(frag_lengths.size(), frag_offsets.size()); in H264CMSampleBufferToAnnexBBuffer()
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_track_jni.cc | 242 RTC_DCHECK_EQ(frames_per_buffer_, length / kBytesPerFrame); in OnGetPlayoutData()
253 RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_); in OnGetPlayoutData()
257 RTC_DCHECK_EQ(length, kBytesPerFrame * samples); in OnGetPlayoutData()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| D | audio_decoder_g722.cc | 38 RTC_DCHECK_EQ(sample_rate_hz, 16000); in DecodeInternal()
77 RTC_DCHECK_EQ(sample_rate_hz, 16000); in DecodeInternal()
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| /external/webrtc/webrtc/base/ |
| D | filerotatingstream.cc | 167 RTC_DCHECK_EQ(current_bytes_written_, max_file_size_); in Write()
251 RTC_DCHECK_EQ(current_file_index_, 0u); in OpenCurrentFile()
279 RTC_DCHECK_EQ(mode_, kWrite); in RotateFiles()
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| D | checks.h | 169 #define RTC_DCHECK_EQ(v1, v2) RTC_CHECK_EQ(v1, v2) macro
178 #define RTC_DCHECK_EQ(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) == (v2))
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| D | array_view.h | 126 void CheckInvariant() const { RTC_DCHECK_EQ(!data_, size_ == 0); } in CheckInvariant()
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| /external/webrtc/talk/app/webrtc/ |
| D | fakemetricsobserver.cc | 58 RTC_DCHECK_EQ(histogram_samples_[type], 0); in AddHistogramSample()
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| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| D | agc.cc | 57 RTC_DCHECK_EQ(rms.size(), probabilities.size()); in Process()
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| /external/webrtc/webrtc/video/ |
| D | video_send_stream.cc | 354 RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers, 1); in ReconfigureVideoEncoder()
355 RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2); in ReconfigureVideoEncoder()
397 RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate); in ReconfigureVideoEncoder()
484 RTC_DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size()); in ConfigureSsrcs()
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