| /external/webrtc/webrtc/modules/audio_processing/ |
| splitting_filter.cc | 35 RTC_DCHECK_EQ(num_bands_, bands->num_bands());
36 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
37 RTC_DCHECK_EQ(data->num_frames(),
48 RTC_DCHECK_EQ(num_bands_, bands->num_bands());
49 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
50 RTC_DCHECK_EQ(data->num_frames(),
61 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
74 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
87 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
97 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels())
[all...] |
| voice_detection_impl.cc | 23 RTC_DCHECK_EQ(0, error);
132 RTC_DCHECK_EQ(0, error);
144 RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported.
|
| noise_suppression_impl.cc | 37 RTC_DCHECK_EQ(0, error);
79 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
95 RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
147 RTC_DCHECK_EQ(0, error);
|
| high_pass_filter_impl.cc | 112 RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
|
| /external/webrtc/webrtc/voice_engine/ |
| channel_proxy.cc | 37 RTC_DCHECK_EQ(0, error);
45 RTC_DCHECK_EQ(0, error);
51 RTC_DCHECK_EQ(0, error);
57 RTC_DCHECK_EQ(0, error);
68 RTC_DCHECK_EQ(0, error);
74 RTC_DCHECK_EQ(0, error);
90 RTC_DCHECK_EQ(0, error);
98 RTC_DCHECK_EQ(0, error);
106 RTC_DCHECK_EQ(0, error);
121 RTC_DCHECK_EQ(0, error)
[all...] |
| /external/webrtc/webrtc/common_audio/ |
| audio_ring_buffer.cc | 33 RTC_DCHECK_EQ(buffers_.size(), channels);
41 RTC_DCHECK_EQ(buffers_.size(), channels);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_decoder_pcm.cc | 28 RTC_DCHECK_EQ(sample_rate_hz, 8000);
52 RTC_DCHECK_EQ(sample_rate_hz, 8000);
|
| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| voice_activity_detector.cc | 40 RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100);
51 RTC_DCHECK_EQ(length, kLength10Ms);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| app.cc | 52 RTC_DCHECK_EQ(0u, data_length % 4) << "Data must be 32 bits aligned.";
74 RTC_DCHECK_EQ(index_end, *index);
|
| dlrr.cc | 38 RTC_DCHECK_EQ(block_length_32bits,
81 RTC_DCHECK_EQ(buffer + BlockLength(), write_at);
|
| extended_jitter_report.cc | 90 RTC_DCHECK_EQ(index_end, *index);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| audio_encoder_ilbc.cc | 104 RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
115 RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
125 RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
|
| audio_decoder_ilbc.cc | 36 RTC_DCHECK_EQ(sample_rate_hz, 8000);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| isac_fix_type.h | 85 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
90 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
95 RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| audio_encoder_copy_red.cc | 71 RTC_DCHECK_EQ(info.redundant.size(), 1u);
76 RTC_DCHECK_EQ(info.redundant.size(), 2u);
81 RTC_DCHECK_EQ(info.speech, info.redundant[0].speech);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_decoder_opus.cc | 33 RTC_DCHECK_EQ(sample_rate_hz, 48000);
54 RTC_DCHECK_EQ(sample_rate_hz, 48000);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_decoder_g722.cc | 38 RTC_DCHECK_EQ(sample_rate_hz, 16000);
77 RTC_DCHECK_EQ(sample_rate_hz, 16000);
|
| /external/webrtc/talk/app/webrtc/ |
| fakemetricsobserver.cc | 58 RTC_DCHECK_EQ(histogram_samples_[type], 0);
|
| /external/webrtc/webrtc/audio/ |
| audio_send_stream.cc | 171 RTC_DCHECK_EQ(0, error);
177 RTC_DCHECK_EQ(0, error);
185 RTC_DCHECK_EQ(0, error);
196 RTC_DCHECK_EQ(0, error);
|
| /external/webrtc/webrtc/base/ |
| array_view.h | 126 void CheckInvariant() const { RTC_DCHECK_EQ(!data_, size_ == 0); }
|
| filerotatingstream.cc | 167 RTC_DCHECK_EQ(current_bytes_written_, max_file_size_);
251 RTC_DCHECK_EQ(current_file_index_, 0u);
279 RTC_DCHECK_EQ(mode_, kWrite);
|
| criticalsection.cc | 153 RTC_DCHECK_EQ(1, old_value) << "Unlock called without calling Lock first";
|
| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| agc.cc | 57 RTC_DCHECK_EQ(rms.size(), probabilities.size());
|
| /external/webrtc/webrtc/modules/video_coding/codecs/h264/ |
| h264_video_toolbox_nalu.cc | 54 RTC_DCHECK_EQ(nalu_header_size, 4);
55 RTC_DCHECK_EQ(param_set_count, 2u);
140 RTC_DCHECK_EQ(bytes_remaining, (size_t)0);
145 RTC_DCHECK_EQ(frag_lengths.size(), frag_offsets.size());
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_track_jni.cc | 242 RTC_DCHECK_EQ(frames_per_buffer_, length / kBytesPerFrame);
253 RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_);
257 RTC_DCHECK_EQ(length, kBytesPerFrame * samples);
|
