| /external/webrtc/webrtc/modules/video_coding/ |
| H A D | jitter_buffer_unittest.cc | 34 Vp9SsMapTest() : packet_(data_, 1400, 1234, 1, true) {}
37 packet_.isFirstPacket = true;
38 packet_.markerBit = true;
39 packet_.frameType = kVideoFrameKey;
40 packet_.codec = kVideoCodecVP9;
41 packet_.codecSpecificHeader.codec = kRtpVideoVp9;
42 packet_.codecSpecificHeader.codecHeader.VP9.flexible_mode = false;
43 packet_.codecSpecificHeader.codecHeader.VP9.gof_idx = 0;
44 packet_.codecSpecificHeader.codecHeader.VP9.temporal_idx = kNoTemporalIdx;
45 packet_
53 VCMPacket packet_; member in class:webrtc::Vp9SsMapTest
276 rtc::scoped_ptr<VCMPacket> packet_; member in class:webrtc::TestBasicJitterBuffer
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| H A D | session_info_unittest.cc | 26 packet_.Reset();
27 packet_.frameType = kVideoFrameDelta;
28 packet_.sizeBytes = packet_buffer_size();
29 packet_.dataPtr = packet_buffer_;
30 packet_.seqNum = 0;
31 packet_.timestamp = 0;
60 VCMPacket packet_; member in class:webrtc::TestSessionInfo
101 packet_.codec = kVideoCodecVP8;
150 packet_.isFirstPacket = true;
151 packet_
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_packet_history_unittest.cc | 39 uint8_t packet_[kMaxPacketLength]; member in class:webrtc::RtpPacketHistoryTest
71 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
72 EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
77 EXPECT_FALSE(hist_->GetPacketAndSetSendTime(kSeqNum, 0, false, packet_, &len,
84 EXPECT_EQ(-1, hist_->PutRTPPacket(packet_, kMaxPacketLength + 1,
92 EXPECT_FALSE(hist_->GetPacketAndSetSendTime(0, 0, false, packet_, &len,
99 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
103 EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
112 CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
113 EXPECT_EQ(0, hist_->PutRTPPacket(packet_, le
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| H A D | rtp_sender_unittest.cc | 148 uint8_t packet_[kMaxPacketLength]; member in class:webrtc::RtpSenderTest
167 packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
172 packet_, payload_length, rtp_length, capture_time_ms,
204 packet_, kPayload, expect_cvo /* marker_bit */, kTimestamp, 0));
331 packet_, kPayload, kMarkerBit, kTimestamp, 0));
335 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
362 packet_, kPayload, kMarkerBit, kTimestamp, 0));
367 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
402 packet_, kPayload, kMarkerBit, kTimestamp, 0));
407 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, lengt
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| H A D | rtp_format_vp9_unittest.cc | 130 rtc::scoped_ptr<uint8_t[]> packet_; member in class:webrtc::RtpPacketizerVp9Test
146 packet_.reset(new uint8_t[payload_size_ + kMaxPayloadDescriptorLength]);
166 EXPECT_FALSE(packetizer_->NextPacket(packet_.get(), &length, &last));
170 EXPECT_TRUE(packetizer_->NextPacket(packet_.get(), &length, &last));
175 ParseAndCheckPacket(packet_.get(), hdr, expected_hdr_sizes[i], length);
176 CheckPayload(packet_.get(), expected_hdr_sizes[i], length, last);
|
| H A D | rtcp_packet.cc | 309 : called_(false), packet_(packet) {}
314 packet_->SetLength(length);
319 RawPacket* const packet_; member in class:webrtc::rtcp::PacketVerifier
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| H A D | neteq_unittest.cc | 324 rtc::scoped_ptr<test::Packet> packet_; member in class:webrtc::NetEqDecodingTest
411 while (packet_ && sim_clock_ >= packet_->time_ms()) {
412 if (packet_->payload_length_bytes() > 0) {
414 packet_->ConvertHeader(&rtp_header);
418 packet_->payload(), packet_->payload_length_bytes()),
419 static_cast<uint32_t>(packet_->time_ms() *
423 packet_.reset(rtp_source_->NextPacket());
466 packet_
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