/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | constant_pcm_packet_source.cc | 32 timestamp_(0), 57 packet_memory[4] = timestamp_ >> 24; 58 packet_memory[5] = (timestamp_ >> 16) & 0xFF; 59 packet_memory[6] = (timestamp_ >> 8) & 0xFF; 60 packet_memory[7] = timestamp_ & 0xFF; 66 timestamp_ += static_cast<uint32_t>(payload_len_samples_);
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H A D | rtp_generator.cc | 26 rtp_header->header.timestamp = timestamp_; 27 timestamp_ += static_cast<uint32_t>(payload_length_samples); 52 if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <= 54 timestamp_ > jump_from_timestamp_) { 56 timestamp_ = jump_to_timestamp_;
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H A D | constant_pcm_packet_source.h | 49 uint32_t timestamp_; member in class:webrtc::test::ConstantPcmPacketSource
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H A D | rtp_generator.h | 30 timestamp_(start_timestamp), 50 uint32_t timestamp_; member in class:webrtc::test::RtpGenerator
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | acm_send_test_oldapi.cc | 40 timestamp_(0), 97 input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_); 118 timestamp_ = timestamp; 134 packet_memory[4] = (timestamp_ >> 24) & 0xFF; 135 packet_memory[5] = (timestamp_ >> 16) & 0xFF; 136 packet_memory[6] = (timestamp_ >> 8) & 0xFF; 137 packet_memory[7] = timestamp_ & 0xFF;
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H A D | acm_receiver_unittest_oldapi.cc | 57 : timestamp_(0), 59 last_packet_send_timestamp_(timestamp_), 91 if (timestamp_ == 0) { // This is the first time inserting audio. 107 last_packet_send_timestamp_ = timestamp_; 109 frame.timestamp_ = timestamp_; 110 timestamp_ += frame.samples_per_channel_; 160 uint32_t timestamp_; member in class:webrtc::acm2::AcmReceiverTestOldApi
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H A D | acm_send_test_oldapi.h | 81 uint32_t timestamp_; member in class:webrtc::test::AcmSendTestOldApi
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H A D | rent_a_codec_unittest.cc | 48 encoder_->Encode(timestamp_, kZeroData, kPacketSizeSamples, out); 49 timestamp_ += kDataLengthSamples; 63 uint32_t timestamp_ = 0; member in class:webrtc::acm2::RentACodecTestF 77 uint32_t expected_timestamp = timestamp_;
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/external/webrtc/webrtc/modules/video_coding/ |
H A D | jitter_buffer_unittest.cc | 196 timestamp_ = 0; 214 packet_.reset(new VCMPacket(data_, size_, seq_num_, timestamp_, true)); 273 uint32_t timestamp_; member in class:webrtc::TestBasicJitterBuffer 613 packet_->timestamp = timestamp_; 679 timestamp_ -= 33 * 90; 684 packet_->timestamp = timestamp_; 731 packet_->timestamp = timestamp_ + (66 * 90); 743 packet_->timestamp = timestamp_ + (33 * 90); 757 VCMPacket empty_packet(data_, 0, seq_num_ + 2, timestamp_ + (33 * 90), false); 775 packet_->timestamp = timestamp_; [all...] |
/external/webrtc/webrtc/common_video/ |
H A D | video_frame.cc | 53 timestamp_(timestamp), 72 timestamp_ = 0; 151 timestamp_ = videoFrame.timestamp_; 160 timestamp_ = videoFrame.timestamp_; 168 timestamp_ = 0;
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | fec_test_helper.cc | 19 : num_packets_(0), seq_num_(0), timestamp_(0) {} 23 timestamp_ += 3000; 38 rtp_packet->header.header.timestamp = timestamp_;
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H A D | fec_test_helper.h | 55 uint32_t timestamp_; member in class:webrtc::FrameGenerator
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/external/webrtc/webrtc/ |
H A D | video_frame.h | 106 void set_timestamp(uint32_t timestamp) { timestamp_ = timestamp; } 109 uint32_t timestamp() const { return timestamp_; } 166 uint32_t timestamp_; member in class:webrtc::VideoFrame
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/external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
H A D | default_temporal_layers.h | 87 uint32_t timestamp_; member in class:webrtc::DefaultTemporalLayers
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H A D | realtime_temporal_layers.cc | 92 timestamp_(0), 240 if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) { 241 timestamp_ = timestamp; 259 uint32_t timestamp_; member in class:webrtc::__anon24812::RealTimeTemporalLayers
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H A D | default_temporal_layers.cc | 32 timestamp_(0), 277 if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) { 278 timestamp_ = timestamp;
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/external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
H A D | audio_encoder_copy_red_unittest.cc | 36 : timestamp_(4711), 64 timestamp_, 67 timestamp_ += num_audio_samples_10ms; 72 uint32_t timestamp_; member in class:webrtc::AudioEncoderCopyRedTest 219 helper.info_.encoded_timestamp = timestamp_; 220 uint32_t primary_timestamp = timestamp_; 230 primary_timestamp = timestamp_; 231 helper.info_.encoded_timestamp = timestamp_;
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/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | PCMFile.cc | 33 timestamp_ = (((uint32_t) rand() & 0x0000FFFF) << 16) | 46 timestamp_ = timestamp; 138 audio_frame.timestamp_ = timestamp_; 139 timestamp_ += samples_10ms_;
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H A D | PCMFile.h | 71 uint32_t timestamp_; member in class:webrtc::PCMFile
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/external/libchrome/base/trace_event/ |
H A D | trace_event_impl.h | 129 TimeTicks timestamp() const { return timestamp_; } 157 TimeTicks timestamp_; member in class:base::trace_event::TraceEvent
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/external/webrtc/webrtc/voice_engine/ |
H A D | utility.cc | 30 dst_frame->timestamp_ = src_frame.timestamp_;
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/external/webrtc/webrtc/examples/peerconnection/server/ |
H A D | peer_channel.cc | 59 connected_(true), timestamp_(time(NULL)) { 80 return waiting_socket_ == NULL && (time(NULL) - timestamp_) > 30; 127 timestamp_ = time(NULL); 144 timestamp_ = time(NULL);
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H A D | peer_channel.h | 61 time_t timestamp_; member in class:ChannelMember
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/external/webrtc/webrtc/modules/utility/source/ |
H A D | coder.cc | 85 audioFrame.timestamp_ = _encodeTimestamp;
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/external/webrtc/talk/app/webrtc/ |
H A D | statstypes.h | 350 double timestamp() const { return timestamp_; } 351 void set_timestamp(double t) { timestamp_ = t; } 372 double timestamp_; // Time since 1970-01-01T00:00:00Z in milliseconds. member in class:webrtc::StatsReport
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