| /external/chromium_org/media/cast/net/rtcp/ |
| D | rtcp.cc | 57 Rtcp::Rtcp(const RtcpCastMessageCallback& cast_callback, in Rtcp() function in media::cast::Rtcp
78 Rtcp::~Rtcp() {} in ~Rtcp()
80 bool Rtcp::IsRtcpPacket(const uint8* packet, size_t length) { in IsRtcpPacket()
90 uint32 Rtcp::GetSsrcOfSender(const uint8* rtcp_buffer, size_t length) { in GetSsrcOfSender()
101 bool Rtcp::IncomingRtcpPacket(const uint8* data, size_t length) { in IncomingRtcpPacket()
163 bool Rtcp::DedupeReceiverLog(RtcpReceiverLogMessage* receiver_log) { in DedupeReceiverLog()
195 void Rtcp::SendRtcpFromRtpReceiver( in SendRtcpFromRtpReceiver()
239 void Rtcp::SendRtcpFromRtpSender(base::TimeTicks current_time, in SendRtcpFromRtpSender()
262 void Rtcp::OnReceivedNtp(uint32 ntp_seconds, uint32 ntp_fraction) { in OnReceivedNtp()
289 void Rtcp::OnReceivedLipSyncInfo(uint32 rtp_timestamp, uint32 ntp_seconds, in OnReceivedLipSyncInfo()
[all …]
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| D | rtcp.h | 52 class Rtcp {
54 Rtcp(const RtcpCastMessageCallback& cast_callback,
62 virtual ~Rtcp();
177 DISALLOW_COPY_AND_ASSIGN(Rtcp);
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| D | rtcp_unittest.cc | 30 void set_rtcp_destination(Rtcp* rtcp) { rtcp_ = rtcp; } in set_rtcp_destination()
56 Rtcp* rtcp_;
132 Rtcp rtcp_for_sender_;
133 Rtcp rtcp_for_receiver_;
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| D | rtcp.h | 23 class Rtcp {
25 Rtcp() { in Rtcp() function
29 ~Rtcp() {} in ~Rtcp()
54 DISALLOW_COPY_AND_ASSIGN(Rtcp);
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| D | rtcp.cc | 22 void Rtcp::Init(uint16_t start_sequence_number) { in Init()
33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { in Update()
57 void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) { in GetStatistics()
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| D | neteq_impl.h | 365 Rtcp rtcp_ GUARDED_BY(crit_sect_);
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| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
| D | after_initialization_fixture.h | 44 StorePacket(Packet::Rtcp, channel, data, len); in SendRTCPPacket()
50 enum Type { Rtp, Rtcp, } type; enumerator
101 case Packet::Rtcp: in SendPackets()
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| /external/chromium_org/media/cast/net/ |
| D | cast_transport_sender_impl.h | 150 scoped_ptr<Rtcp> audio_rtcp_session_;
151 scoped_ptr<Rtcp> video_rtcp_session_;
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| D | cast_transport_sender_impl.cc | 151 new Rtcp(base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage, in InitializeAudio()
184 new Rtcp(base::Bind(&CastTransportSenderImpl::OnReceivedCastMessage, in InitializeVideo()
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| /external/chromium_org/media/cast/receiver/ |
| D | cast_receiver_impl.cc | 56 if (Rtcp::IsRtcpPacket(data, length)) { in DispatchReceivedPacket()
57 ssrc_of_sender = Rtcp::GetSsrcOfSender(data, length); in DispatchReceivedPacket()
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| D | frame_receiver.h | 159 Rtcp rtcp_;
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| D | frame_receiver.cc | 76 if (Rtcp::IsRtcpPacket(&packet->front(), packet->size())) { in ProcessPacket()
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_unittest_helper.h | 80 RtcpPacket* Rtcp(int64_t time_now_us);
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| D | remote_bitrate_estimator_unittest_helper.cc | 80 RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) { in Rtcp() function in webrtc::testing::RtpStream
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| /external/chromium_org/media/cast/sender/ |
| D | audio_sender_unittest.cc | 30 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { in SendPacket()
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| D | video_sender_unittest.cc | 74 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { in SendPacket()
|