Searched defs:rtp_packet (Results 1 – 14 of 14) sorted by relevance
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | rtc_event_log_source.cc | 40 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in GetRtpPacket() local
81 const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event); in NextPacket() local
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | producer_fec_unittest.cc | 120 RtpPacket* rtp_packet = generator_->NextPacket(i, 10); in TEST_F() local
162 RtpPacket* rtp_packet = generator_->NextPacket(i * kNumPackets + j, 10); in TEST_F() local
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| D | fec_test_helper.cc | 29 RtpPacket* rtp_packet = new RtpPacket; in NextPacket() local
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| D | rtp_sender.cc | 1444 const uint8_t* rtp_packet, in FindHeaderExtensionPosition()
1482 uint8_t* rtp_packet, in VerifyExtension()
1514 void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet, in UpdateTransmissionTimeOffset()
1539 bool RTPSender::UpdateAudioLevel(uint8_t* rtp_packet, in UpdateAudioLevel()
1565 bool RTPSender::UpdateVideoRotation(uint8_t* rtp_packet, in UpdateVideoRotation()
1590 void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet, in UpdateAbsoluteSendTime()
1618 uint8_t* rtp_packet, in UpdateTransportSequenceNumber()
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| /external/webrtc/webrtc/video/ |
| D | vie_receiver.cc | 222 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, in ReceivedRTPPacket()
250 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet, in OnRecoveredPacket()
261 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet, in InsertRTPPacket()
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| D | vie_channel.cc | 935 int32_t ViEChannel::ReceivedRTPPacket(const void* rtp_packet, in ReceivedRTPPacket()
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| /external/webrtc/webrtc/call/ |
| D | rtc_event_log2rtp_dump.cc | 127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in main() local
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| D | rtc_event_log.proto | 49 optional RtpPacket rtp_packet = 3; field
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| D | rtc_event_log_unittest.cc | 233 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); in VerifyRtpEvent() local
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| /external/webrtc/webrtc/audio/ |
| D | audio_receive_stream_unittest.cc | 244 std::vector<uint8_t> rtp_packet = in TEST() local
270 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( in TEST() local
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| /external/webrtc/talk/media/base/ |
| D | rtpdump_unittest.cc | 45 RtpDumpPacket rtp_packet(rtp_buf.Data(), rtp_buf.Length(), 0, false); in TEST() local
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| D | testutils.cc | 192 RawRtpPacket rtp_packet; in VerifyTestPacketsFromStream() local
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| /external/webrtc/talk/session/media/ |
| D | srtpfilter_unittest.cc | 95 char rtp_packet[sizeof(kPcmuFrame) + 10]; in TestProtectUnprotect() local
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| /external/webrtc/webrtc/voice_engine/ |
| D | channel.cc | 508 bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet, in OnRecoveredPacket()
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