| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet_unittest.cc | 48 const uint32_t kTimestamp = 47114711;
51 kPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory);
58 EXPECT_EQ(kTimestamp, packet.header().timestamp);
76 const uint32_t kTimestamp = 47114711;
79 kPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory);
89 EXPECT_EQ(kTimestamp, packet.header().timestamp);
141 const uint32_t kTimestamp = 47114711;
144 kRedPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory);
166 EXPECT_EQ(kTimestamp, packet.header().timestamp);
190 EXPECT_EQ(kTimestamp, red_block->timestamp)
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_packet_history_unittest.cc | 37 enum {kTimestamp = 127};
71 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
99 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
112 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
132 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
150 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
168 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
199 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
234 CreateRtpPacket(kSeqNum + i, kSsrc, kPayload, kTimestamp, packet_, &len);
249 CreateRtpPacket(kSeqNum + i, kSsrc, kPayload, kTimestamp, packet_, &len)
[all...] |
| rtp_sender_unittest.cc | 42 const uint32_t kTimestamp = 10;
158 EXPECT_EQ(kTimestamp, rtp_header.timestamp);
204 packet_, kPayload, expect_cvo /* marker_bit */, kTimestamp, 0));
215 EXPECT_EQ(kTimestamp, rtp_header.timestamp);
331 packet_, kPayload, kMarkerBit, kTimestamp, 0));
362 packet_, kPayload, kMarkerBit, kTimestamp, 0));
402 packet_, kPayload, kMarkerBit, kTimestamp, 0));
430 packet_, kPayload, kMarkerBit, kTimestamp, 0));
471 rtp_sender_->BuildRTPheader(packet_, kPayload, true, kTimestamp, 0));
500 rtp_sender_->BuildRTPheader(packet_, kPayload, false, kTimestamp, 0))
[all...] |
| /external/tensorflow/tensorflow/contrib/cloud/kernels/ |
| bigquery_table_accessor.h | 50 kTimestamp,
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| send_time_history_unittest.cc | 208 const int64_t kTimestamp = 2;
209 PacketInfo packets[3] = {{0, kTimestamp, kSeqNo, 0, false},
210 {0, kTimestamp + 1, kSeqNo + 1, 0, false},
211 {0, kTimestamp + 2, kSeqNo + 2, 0, false}};
|
| /external/webrtc/webrtc/modules/video_coding/ |
| jitter_buffer_unittest.cc | 109 const uint32_t kTimestamp = kSsTimestamp1 + kProcessIntervalSec * 90000;
110 map_.RemoveOld(kTimestamp - 1); // Interval not passed.
113 map_.RemoveOld(kTimestamp);
115 EXPECT_TRUE(map_.Find(kTimestamp, &it));
116 EXPECT_EQ(kTimestamp, it->first);
[all...] |
