| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| bye.cc | 108 RTC_DCHECK_EQ(index_end, *index);
|
| transport_feedback.cc | 270 RTC_DCHECK_EQ(0, buffer[0] & 0x80);
317 RTC_DCHECK_EQ(-1, base_seq_);
717 RTC_DCHECK_EQ(num_packets, symbols.size());
|
| /external/webrtc/webrtc/base/ |
| checks.h | 21 // RTC_DCHECK_EQ(foo, bar) << "I'm printed when foo != bar.";
49 // - RTC_CHECK_EQ, _NE, _GT, ..., and RTC_DCHECK_EQ, _NE, _GT, ... are
169 #define RTC_DCHECK_EQ(v1, v2) RTC_CHECK_EQ(v1, v2)
178 #define RTC_DCHECK_EQ(v1, v2) RTC_EAT_STREAM_PARAMETERS((v1) == (v2))
|
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_ios.mm | 706 RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(),
[all...] |
| /external/webrtc/webrtc/video/ |
| video_send_stream.cc | 354 RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers, 1);
355 RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2);
397 RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
484 RTC_DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size());
|
| vie_channel.cc | 484 RTC_DCHECK_EQ(payload_type_red, -1);
485 RTC_DCHECK_EQ(payload_type_fec, -1);
[all...] |
| /external/webrtc/webrtc/modules/audio_device/android/ |
| opensles_player.cc | 142 RTC_DCHECK_EQ(0u, buffer_queue_state.count);
143 RTC_DCHECK_EQ(0u, buffer_queue_state.index);
|
| audio_device_template.h | 84 RTC_DCHECK_EQ(err, 0);
|
| /external/webrtc/webrtc/modules/pacing/ |
| paced_sender.cc | 127 RTC_DCHECK_EQ(packet_list_.size(), prio_queue_.size());
129 RTC_DCHECK_EQ(0u, queue_time_sum_);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng.cc | 109 RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.cc | 140 RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| PCMFile.cc | 196 RTC_DCHECK_EQ(error, 0);
|
| /external/webrtc/webrtc/call/ |
| call_perf_tests.cc | 599 RTC_DCHECK_EQ(1u, stats.substreams.size());
646 RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
|
| bitrate_estimator_tests.cc | 176 RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
|
| rtc_event_log.cc | 480 RTC_DCHECK_EQ(stream_.stream_size(), 1);
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| nonlinear_beamformer.cc | 371 RTC_DCHECK_EQ(input.num_channels(), num_input_channels_);
372 RTC_DCHECK_EQ(input.num_frames_per_band(), chunk_length_);
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| echo_control_mobile_impl.cc | 133 RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
|
| gain_control_impl.cc | 98 RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
|
| echo_cancellation_impl.cc | 129 RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_video.cc | 145 RTC_DCHECK_EQ(num_fec_packets, fec_packets.size());
|
| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| intelligibility_enhancer.cc | 61 RTC_DCHECK_EQ(parent_->freqs_, frames);
|
| /external/webrtc/webrtc/modules/video_coding/codecs/h264/ |
| h264_video_toolbox_encoder.cc | 221 RTC_DCHECK_EQ(codec_settings->codecType, kVideoCodecH264);
|
| /external/webrtc/webrtc/modules/video_coding/codecs/vp9/ |
| vp9_impl.cc | 492 RTC_DCHECK_EQ(input_image.width(), static_cast<int>(raw_->d_w));
493 RTC_DCHECK_EQ(input_image.height(), static_cast<int>(raw_->d_h));
[all...] |
| /external/webrtc/talk/app/webrtc/ |
| statstypes.cc | 286 RTC_DCHECK_EQ(type_, other.type_);
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| audio_coding_module_impl.cc | 511 RTC_DCHECK_EQ(enable_dtx, enable_vad);
|
