/external/webrtc/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->header.timestamp = timestamp_; in GetRtpHeader() 27 timestamp_ += static_cast<uint32_t>(payload_length_samples); in GetRtpHeader() 52 if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <= in GetRtpHeader() 54 timestamp_ > jump_from_timestamp_) { in GetRtpHeader() 56 timestamp_ = jump_to_timestamp_; in GetRtpHeader()
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D | rtp_generator.h | 30 timestamp_(start_timestamp), in seq_number_() 50 uint32_t timestamp_; variable
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/external/webrtc/webrtc/modules/video_coding/ |
D | jitter_buffer_unittest.cc | 196 timestamp_ = 0; in SetUp() 214 packet_.reset(new VCMPacket(data_, size_, seq_num_, timestamp_, true)); in SetUp() 273 uint32_t timestamp_; member in webrtc::TestBasicJitterBuffer 613 packet_->timestamp = timestamp_; in TEST_F() 679 timestamp_ -= 33 * 90; in TEST_F() 684 packet_->timestamp = timestamp_; in TEST_F() 731 packet_->timestamp = timestamp_ + (66 * 90); in TEST_F() 743 packet_->timestamp = timestamp_ + (33 * 90); in TEST_F() 757 VCMPacket empty_packet(data_, 0, seq_num_ + 2, timestamp_ + (33 * 90), false); in TEST_F() 775 packet_->timestamp = timestamp_; in TEST_F() [all …]
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
D | acm_send_test_oldapi.cc | 40 timestamp_(0), in AcmSendTestOldApi() 97 input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_); in NextPacket() 118 timestamp_ = timestamp; in SendData() 134 packet_memory[4] = (timestamp_ >> 24) & 0xFF; in CreatePacket() 135 packet_memory[5] = (timestamp_ >> 16) & 0xFF; in CreatePacket() 136 packet_memory[6] = (timestamp_ >> 8) & 0xFF; in CreatePacket() 137 packet_memory[7] = timestamp_ & 0xFF; in CreatePacket()
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D | acm_receiver_unittest_oldapi.cc | 57 : timestamp_(0), in AcmReceiverTestOldApi() 59 last_packet_send_timestamp_(timestamp_), in AcmReceiverTestOldApi() 91 if (timestamp_ == 0) { // This is the first time inserting audio. in InsertOnePacketOfSilence() 107 last_packet_send_timestamp_ = timestamp_; in InsertOnePacketOfSilence() 109 frame.timestamp_ = timestamp_; in InsertOnePacketOfSilence() 110 timestamp_ += frame.samples_per_channel_; in InsertOnePacketOfSilence() 160 uint32_t timestamp_; member in webrtc::acm2::AcmReceiverTestOldApi
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D | rent_a_codec_unittest.cc | 48 encoder_->Encode(timestamp_, kZeroData, kPacketSizeSamples, out); in EncodeAndVerify() 49 timestamp_ += kDataLengthSamples; in EncodeAndVerify() 63 uint32_t timestamp_ = 0; member in webrtc::acm2::RentACodecTestF 77 uint32_t expected_timestamp = timestamp_; in TEST_F()
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D | audio_coding_module_impl.cc | 350 input_data->input_timestamp = ptr_frame->timestamp_; in Add10MsDataInternal() 375 expected_in_ts_ = in_frame.timestamp_; in PreprocessToAddData() 376 expected_codec_ts_ = in_frame.timestamp_; in PreprocessToAddData() 378 } else if (in_frame.timestamp_ != expected_in_ts_) { in PreprocessToAddData() 381 (in_frame.timestamp_ - expected_in_ts_) * in PreprocessToAddData() 384 expected_in_ts_ = in_frame.timestamp_; in PreprocessToAddData() 413 preprocess_frame_.timestamp_ = expected_codec_ts_; in PreprocessToAddData()
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/external/webrtc/webrtc/common_video/ |
D | video_frame.cc | 53 timestamp_(timestamp), in VideoFrame() 72 timestamp_ = 0; in CreateEmptyFrame() 151 timestamp_ = videoFrame.timestamp_; in CopyFrame() 160 timestamp_ = videoFrame.timestamp_; in ShallowCopy() 168 timestamp_ = 0; in Reset()
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/external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
D | audio_encoder_copy_red_unittest.cc | 36 : timestamp_(4711), in AudioEncoderCopyRedTest() 64 timestamp_, in Encode() 67 timestamp_ += num_audio_samples_10ms; in Encode() 72 uint32_t timestamp_; member in webrtc::AudioEncoderCopyRedTest 219 helper.info_.encoded_timestamp = timestamp_; in TEST_F() 220 uint32_t primary_timestamp = timestamp_; in TEST_F() 230 primary_timestamp = timestamp_; in TEST_F() 231 helper.info_.encoded_timestamp = timestamp_; in TEST_F()
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/external/webrtc/webrtc/modules/audio_coding/test/ |
D | PCMFile.cc | 33 timestamp_ = (((uint32_t) rand() & 0x0000FFFF) << 16) | in PCMFile() 46 timestamp_ = timestamp; in PCMFile() 138 audio_frame.timestamp_ = timestamp_; in Read10MsData() 139 timestamp_ += samples_10ms_; in Read10MsData()
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/external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
D | realtime_temporal_layers.cc | 92 timestamp_(0), in RealTimeTemporalLayers() 240 if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) { in PopulateCodecSpecific() 241 timestamp_ = timestamp; in PopulateCodecSpecific() 259 uint32_t timestamp_; member in webrtc::__anon4357a9a10111::RealTimeTemporalLayers
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D | default_temporal_layers.cc | 32 timestamp_(0), in DefaultTemporalLayers() 277 if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) { in PopulateCodecSpecific() 278 timestamp_ = timestamp; in PopulateCodecSpecific()
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/external/webrtc/webrtc/ |
D | video_frame.h | 106 void set_timestamp(uint32_t timestamp) { timestamp_ = timestamp; } in set_timestamp() 109 uint32_t timestamp() const { return timestamp_; } in timestamp() 166 uint32_t timestamp_; variable
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/external/tensorflow/tensorflow/examples/android/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/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | fec_test_helper.cc | 19 : num_packets_(0), seq_num_(0), timestamp_(0) {} in FrameGenerator() 23 timestamp_ += 3000; in NewFrame() 38 rtp_packet->header.header.timestamp = timestamp_; in NextPacket()
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/external/nos/host/generic/nugget/include/ |
D | signed_header.h | 96 printf("hdr.timestamp : %016" PRIx64 ", %s", timestamp_, in print() 97 asctime(localtime(reinterpret_cast<const time_t*>(×tamp_)))); in print() 144 uint64_t timestamp_; // time of signing member
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | neteq_impl.cc | 89 timestamp_(0), in NetEqImpl() 523 sync_buffer_->IncreaseEndTimestamp(main_header.timestamp - timestamp_); in InsertPacketInternal() 526 timestamp_ = main_header.timestamp; in InsertPacketInternal() 727 if ((int32_t) (main_header.timestamp - timestamp_) >= 0 && in InsertPacketInternal() 1025 timestamp_ = dtmf_event->timestamp; in GetDecision() 1031 timestamp_ = header->timestamp; in GetDecision() 1050 sync_buffer_->IncreaseEndTimestamp(timestamp_ - end_timestamp); in GetDecision() 1051 end_timestamp = timestamp_; in GetDecision() 1066 timestamp_ = end_timestamp; in GetDecision() 1076 timestamp_ = end_timestamp; in GetDecision() [all …]
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/external/libchrome/base/trace_event/ |
D | trace_event_impl.cc | 62 timestamp_ = other->timestamp_; in MoveFrom() 101 timestamp_ = timestamp; in Initialize() 191 duration_ = now - timestamp_; in UpdateDuration() 281 int64_t time_int64 = timestamp_.ToInternalValue(); in AppendAsJSON()
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D | trace_event_impl.h | 124 TimeTicks timestamp() const { return timestamp_; } in timestamp() 152 TimeTicks timestamp_;
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/external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
D | audio_encoder_cng_unittest.cc | 38 timestamp_(4711), in AudioEncoderCngTest() 79 timestamp_, in Encode() 82 timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_); in Encode() 191 uint32_t timestamp_; member in webrtc::AudioEncoderCngTest 263 uint32_t expected_timestamp = timestamp_; in TEST_F()
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/external/webrtc/webrtc/examples/peerconnection/server/ |
D | peer_channel.cc | 59 connected_(true), timestamp_(time(NULL)) { in ChannelMember() 80 return waiting_socket_ == NULL && (time(NULL) - timestamp_) > 30; in TimedOut() 127 timestamp_ = time(NULL); in OnClosing() 144 timestamp_ = time(NULL); in QueueResponse()
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/external/webrtc/talk/app/webrtc/ |
D | statstypes.h | 350 double timestamp() const { return timestamp_; } in timestamp() 351 void set_timestamp(double t) { timestamp_ = t; } in set_timestamp() 372 double timestamp_; // Time since 1970-01-01T00:00:00Z in milliseconds. variable
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/external/webrtc/webrtc/voice_engine/ |
D | utility.cc | 30 dst_frame->timestamp_ = src_frame.timestamp_; in RemixAndResample()
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/external/webrtc/webrtc/modules/include/ |
D | module_common_types.h | 525 uint32_t timestamp_; variable 558 timestamp_ = 0; in Reset() 580 timestamp_ = timestamp; in UpdateFrame() 601 timestamp_ = src.timestamp_; in CopyFrom()
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