| /external/webrtc/modules/video_coding/codecs/vp8/ |
| D | screenshare_layers_unittest.cc | 65 timestamp_(90), in ScreenshareLayerTest()
82 layers_->OnEncodeDone(0, timestamp_, frame_size_, base_sync, kDefaultQp, in EncodeFrame()
88 tl_config_ = NextFrameConfig(0, timestamp_); in ConfigureFrame()
147 timestamp_ += kTimestampDelta5Fps; in RunGracePeriod()
181 layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp, in SkipUntilTlAndSync()
184 timestamp_ += kTimestampDelta5Fps; in SkipUntilTlAndSync()
200 uint32_t timestamp_; member in webrtc::ScreenshareLayerTest
222 timestamp_ += kTimestampDelta5Fps;
241 timestamp_ += kTimestampDelta5Fps;
244 sync_times.push_back(timestamp_);
[all …]
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | constant_pcm_packet_source.cc | 32 timestamp_(0), in ConstantPcmPacketSource()
57 packet_memory[4] = timestamp_ >> 24; in WriteHeader()
58 packet_memory[5] = (timestamp_ >> 16) & 0xFF; in WriteHeader()
59 packet_memory[6] = (timestamp_ >> 8) & 0xFF; in WriteHeader()
60 packet_memory[7] = timestamp_ & 0xFF; in WriteHeader()
66 timestamp_ += static_cast<uint32_t>(payload_len_samples_); in WriteHeader()
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| D | rtp_generator.cc | 26 rtp_header->timestamp = timestamp_; in GetRtpHeader()
27 timestamp_ += static_cast<uint32_t>(payload_length_samples); in GetRtpHeader()
51 if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <= in GetRtpHeader()
53 timestamp_ > jump_from_timestamp_) { in GetRtpHeader()
55 timestamp_ = jump_to_timestamp_; in GetRtpHeader()
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| D | rtp_generator.h | 29 timestamp_(start_timestamp), in seq_number_()
48 uint32_t timestamp_; variable
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| /external/grpc-grpc-java/grpclb/src/generated/main/java/io/grpc/lb/v1/ |
| D | ClientStats.java | 64 if (timestamp_ != null) { in ClientStats()
65 subBuilder = timestamp_.toBuilder(); in ClientStats()
67 … timestamp_ = input.readMessage(com.google.protobuf.Timestamp.parser(), extensionRegistry); in ClientStats()
69 subBuilder.mergeFrom(timestamp_); in ClientStats()
70 timestamp_ = subBuilder.buildPartial(); in ClientStats()
133 private com.google.protobuf.Timestamp timestamp_; field in ClientStats
142 return timestamp_ != null; in hasTimestamp()
152 return timestamp_ == null ? com.google.protobuf.Timestamp.getDefaultInstance() : timestamp_; in getTimestamp()
285 if (timestamp_ != null) { in writeTo()
311 if (timestamp_ != null) { in getSerializedSize()
[all …]
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_send_test.cc | 49 timestamp_(0),
114 input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_); in NextPacket()
134 timestamp_ = timestamp; in SendData()
150 packet_memory[4] = (timestamp_ >> 24) & 0xFF; in CreatePacket()
151 packet_memory[5] = (timestamp_ >> 16) & 0xFF; in CreatePacket()
152 packet_memory[6] = (timestamp_ >> 8) & 0xFF; in CreatePacket()
153 packet_memory[7] = timestamp_ & 0xFF; in CreatePacket()
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| D | acm_receiver_unittest.cc | 36 : timestamp_(0), in AcmReceiverTestOldApi()
38 last_packet_send_timestamp_(timestamp_), in AcmReceiverTestOldApi()
95 last_packet_send_timestamp_ = timestamp_; in InsertOnePacketOfSilence()
98 frame.timestamp_ = timestamp_; in InsertOnePacketOfSilence()
99 timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_); in InsertOnePacketOfSilence()
139 uint32_t timestamp_; member in webrtc::acm2::AcmReceiverTestOldApi
207 EXPECT_EQ(0u, frame.timestamp_); in RunVerifyAudioFrame()
213 EXPECT_EQ(expected_output_ts, frame.timestamp_); in RunVerifyAudioFrame()
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| D | audio_coding_module.cc | 392 input_data->input_timestamp = ptr_frame->timestamp_; in Add10MsDataInternal()
433 expected_in_ts_ = in_frame.timestamp_; in PreprocessToAddData()
434 expected_codec_ts_ = in_frame.timestamp_; in PreprocessToAddData()
436 } else if (in_frame.timestamp_ != expected_in_ts_) { in PreprocessToAddData()
437 RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_ in PreprocessToAddData()
440 (in_frame.timestamp_ - expected_in_ts_) * in PreprocessToAddData()
444 expected_in_ts_ = in_frame.timestamp_; in PreprocessToAddData()
456 preprocess_frame_.timestamp_ = expected_codec_ts_; in PreprocessToAddData()
489 preprocess_frame_.timestamp_ = expected_codec_ts_; in PreprocessToAddData()
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| /external/perfetto/src/traced/probes/ftrace/ |
| D | compact_sched.h | 97 protozero::PackedVarInt& timestamp() { return timestamp_; } in timestamp()
105 return timestamp_.size(); in size()
109 timestamp_.Append(timestamp - last_timestamp_); in AppendTimestamp()
122 protozero::PackedVarInt timestamp_; variable
140 return timestamp_.size(); in size()
144 timestamp_.Append(timestamp - last_timestamp_); in AppendTimestamp()
155 protozero::PackedVarInt timestamp_; variable
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| D | compact_sched.cc | 217 compact_out->set_switch_timestamp(timestamp_); in Write()
226 timestamp_.Reset(); in Reset()
235 compact_out->set_waking_timestamp(timestamp_); in Write()
244 timestamp_.Reset(); in Reset()
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| /external/webrtc/modules/video_coding/ |
| D | jitter_buffer_unittest.cc | 43 timestamp_ = 0; in SetUp()
64 rtp_header.timestamp = timestamp_; in SetUp()
119 uint32_t timestamp_; member in webrtc::TestBasicJitterBuffer
391 packet_->timestamp = timestamp_; in TEST_F()
457 timestamp_ -= 33 * 90; in TEST_F()
462 packet_->timestamp = timestamp_; in TEST_F()
510 packet_->timestamp = timestamp_ + (66 * 90); in TEST_F()
522 packet_->timestamp = timestamp_ + (33 * 90); in TEST_F()
539 rtp_header.timestamp = timestamp_ + (33 * 90); in TEST_F()
562 packet_->timestamp = timestamp_; in TEST_F()
[all …]
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| /external/grpc-grpc-java/services/src/generated/main/java/io/grpc/channelz/v1/ |
| D | ChannelTraceEvent.java | 73 if (timestamp_ != null) { in ChannelTraceEvent()
74 subBuilder = timestamp_.toBuilder(); in ChannelTraceEvent()
76 … timestamp_ = input.readMessage(com.google.protobuf.Timestamp.parser(), extensionRegistry); in ChannelTraceEvent()
78 subBuilder.mergeFrom(timestamp_); in ChannelTraceEvent()
79 timestamp_ = subBuilder.buildPartial(); in ChannelTraceEvent()
361 private com.google.protobuf.Timestamp timestamp_; field in ChannelTraceEvent
370 return timestamp_ != null; in hasTimestamp()
380 return timestamp_ == null ? com.google.protobuf.Timestamp.getDefaultInstance() : timestamp_; in getTimestamp()
463 if (timestamp_ != null) { in writeTo()
487 if (timestamp_ != null) { in getSerializedSize()
[all …]
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| /external/webrtc/api/audio/ |
| D | audio_frame.cc | 29 swap(a.timestamp_, b.timestamp_); in swap()
57 timestamp_ = 0; in ResetWithoutMuting()
78 timestamp_ = timestamp; in UpdateFrame()
103 timestamp_ = src.timestamp_; in CopyFrom()
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| /external/dynamic_depth/internal/dynamic_depth/ |
| D | earth_pose.cc | 41 EarthPose::EarthPose() : timestamp_(-1) {} in EarthPose()
71 earth_pose->timestamp_ = timestamp; in FromData()
102 int64 EarthPose::GetTimestamp() const { return timestamp_; } in GetTimestamp()
138 if (timestamp_ >= 0) { in Serialize()
140 std::to_string(timestamp_)); in Serialize()
186 ×tamp_); in ParseEarthPoseFields()
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| D | pose.cc | 40 Pose::Pose() : timestamp_(-1) {} in Pose()
69 pose->timestamp_ = timestamp; in FromData()
97 int64 Pose::GetTimestamp() const { return timestamp_; } in GetTimestamp()
131 if (timestamp_ >= 0) { in Serialize()
133 std::to_string(timestamp_)); in Serialize()
172 deserializer.ParseLong(prefix, kTimestamp, ×tamp_); in ParsePoseFields()
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| /external/webrtc/modules/audio_coding/codecs/red/ |
| D | audio_encoder_copy_red_unittest.cc | 42 timestamp_(4711), in AudioEncoderCopyRedTest()
62 timestamp_, in Encode()
65 timestamp_ += rtc::checked_cast<uint32_t>(num_audio_samples_10ms); in Encode()
70 uint32_t timestamp_; member in webrtc::AudioEncoderCopyRedTest
204 uint32_t primary_timestamp = timestamp_; in TEST_F()
207 info.encoded_timestamp = timestamp_; in TEST_F()
218 primary_timestamp = timestamp_; in TEST_F()
219 info.encoded_timestamp = timestamp_; in TEST_F()
303 info.encoded_timestamp = timestamp_; in TEST_F()
309 info.encoded_timestamp = timestamp_; // update timestamp. in TEST_F()
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| /external/webrtc/modules/audio_coding/test/ |
| D | PCMFile.cc | 33 timestamp_ = in PCMFile()
46 timestamp_ = timestamp; in PCMFile()
149 audio_frame.timestamp_ = timestamp_; in Read10MsData()
150 timestamp_ += samples_10ms_; in Read10MsData()
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| /external/webrtc/audio/voip/ |
| D | audio_ingress.cc | 77 if (first_rtp_timestamp_ < 0 && audio_frame->timestamp_ != 0) { in GetAudioFrameWithInfo()
78 first_rtp_timestamp_ = audio_frame->timestamp_; in GetAudioFrameWithInfo()
87 timestamp_wrap_handler_.Unwrap(audio_frame->timestamp_); in GetAudioFrameWithInfo()
89 ntp_estimator_.Estimate(audio_frame->timestamp_); in GetAudioFrameWithInfo()
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| /external/grpc-grpc-java/services/src/generated/main/java/io/grpc/binarylog/v1/ |
| D | GrpcLogEntry.java | 63 if (timestamp_ != null) { in GrpcLogEntry()
64 subBuilder = timestamp_.toBuilder(); in GrpcLogEntry()
66 … timestamp_ = input.readMessage(com.google.protobuf.Timestamp.parser(), extensionRegistry); in GrpcLogEntry()
68 subBuilder.mergeFrom(timestamp_); in GrpcLogEntry()
69 timestamp_ = subBuilder.buildPartial(); in GrpcLogEntry()
584 private com.google.protobuf.Timestamp timestamp_; field in GrpcLogEntry
593 return timestamp_ != null; in hasTimestamp()
603 return timestamp_ == null ? com.google.protobuf.Timestamp.getDefaultInstance() : timestamp_; in getTimestamp()
875 if (timestamp_ != null) { in writeTo()
916 if (timestamp_ != null) { in getSerializedSize()
[all …]
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| /external/webrtc/api/audio/test/ |
| D | audio_frame_unittest.cc | 72 EXPECT_EQ(kTimestamp, frame.timestamp_); in TEST()
117 EXPECT_EQ(frame2.timestamp_, frame1.timestamp_); in TEST()
160 EXPECT_EQ(kTimestamp + 1, frame1.timestamp_); in TEST()
173 EXPECT_EQ(kTimestamp, frame2.timestamp_); in TEST()
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| /external/tflite-support/tensorflow_lite_support/cc/task/vision/core/ |
| D | frame_buffer.h | 220 timestamp_(timestamp) {} in FrameBuffer()
233 timestamp_(timestamp) {} in FrameBuffer()
281 const absl::Time timestamp() const { return timestamp_; } in timestamp()
289 absl::Time timestamp_; variable
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| /external/tensorflow/tensorflow/tools/android/test/jni/object_tracking/ |
| D | image_data.h | 39 timestamp_(0), in ImageData()
99 timestamp_ = timestamp; in SetData()
127 inline const uint64_t GetTimestamp() const { return timestamp_; } in GetTimestamp()
237 int64_t timestamp_; variable
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| /external/rust/crates/grpcio-sys/grpc/src/core/lib/channel/ |
| D | channel_trace.cc | 48 timestamp_(grpc_millis_to_timespec(grpc_core::ExecCtx::Get()->Now(), in TraceEvent()
57 timestamp_(grpc_millis_to_timespec(grpc_core::ExecCtx::Get()->Now(), in TraceEvent()
154 {"timestamp", gpr_format_timespec(timestamp_)}, in RenderTraceEvent()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_video_frame_transformer_delegate.cc | 40 timestamp_(rtp_timestamp), in TransformableVideoSenderFrame()
56 uint32_t GetTimestamp() const override { return timestamp_; } in GetTimestamp()
85 const uint32_t timestamp_; member in webrtc::__anon73054be90111::TransformableVideoSenderFrame
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| /external/grpc-grpc/src/core/lib/channel/ |
| D | channel_trace.cc | 48 timestamp_(grpc_millis_to_timespec(grpc_core::ExecCtx::Get()->Now(), in TraceEvent()
56 timestamp_(grpc_millis_to_timespec(grpc_core::ExecCtx::Get()->Now(), in TraceEvent()
146 gpr_format_timespec(timestamp_), in RenderTraceEvent()
|