| /external/chromium_org/media/cast/rtcp/ |
| D | rtcp.cc | 26 explicit LocalRtcpRttFeedback(Rtcp* rtcp) : rtcp_(rtcp) {} in LocalRtcpRttFeedback()
36 Rtcp* rtcp_;
41 LocalRtcpReceiverFeedback(Rtcp* rtcp, in LocalRtcpReceiverFeedback()
72 Rtcp* rtcp_;
76 Rtcp::Rtcp(scoped_refptr<CastEnvironment> cast_environment, in Rtcp() function in media::cast::Rtcp
109 Rtcp::~Rtcp() {} in ~Rtcp()
112 bool Rtcp::IsRtcpPacket(const uint8* packet, size_t length) { in IsRtcpPacket()
125 uint32 Rtcp::GetSsrcOfSender(const uint8* rtcp_buffer, size_t length) { in GetSsrcOfSender()
135 base::TimeTicks Rtcp::TimeToSendNextRtcpReport() { in TimeToSendNextRtcpReport()
142 void Rtcp::IncomingRtcpPacket(const uint8* rtcp_buffer, size_t length) { in IncomingRtcpPacket()
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| D | rtcp_unittest.cc | 41 void set_rtcp_receiver(Rtcp* rtcp) { rtcp_receiver_ = rtcp; } in set_rtcp_receiver()
66 Rtcp* rtcp_receiver_;
80 void set_rtcp_receiver(Rtcp* rtcp) { rtcp_ = rtcp; } in set_rtcp_receiver()
119 Rtcp* rtcp_;
126 class RtcpPeer : public Rtcp {
138 : Rtcp(cast_environment, in RtcpPeer()
150 using Rtcp::OnReceivedNtp;
151 using Rtcp::OnReceivedLipSyncInfo;
213 Rtcp rtcp(cast_environment_, in TEST_F()
235 Rtcp rtcp(cast_environment_, in TEST_F()
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| D | rtcp.h | 56 class Rtcp {
61 Rtcp(scoped_refptr<CastEnvironment> cast_environment,
73 virtual ~Rtcp();
200 DISALLOW_COPY_AND_ASSIGN(Rtcp);
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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 | 370 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/ |
| D | cast_sender_impl.cc | 173 if (!Rtcp::IsRtcpPacket(data, length)) { in ReceivedPacket()
178 uint32 ssrc_of_sender = Rtcp::GetSsrcOfSender(data, length); in ReceivedPacket()
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| /external/chromium_org/media/cast/audio_sender/ |
| D | audio_sender.h | 114 Rtcp rtcp_;
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| D | audio_sender_unittest.cc | 31 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { in SendPacket()
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| /external/chromium_org/media/cast/video_sender/ |
| D | video_sender.h | 126 Rtcp rtcp_;
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| D | video_sender_unittest.cc | 69 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { in SendPacket()
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| /external/chromium_org/media/cast/receiver/ |
| D | frame_receiver.h | 152 Rtcp rtcp_;
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| D | cast_receiver_impl.cc | 54 if (Rtcp::IsRtcpPacket(data, length)) { in DispatchReceivedPacket()
55 ssrc_of_sender = Rtcp::GetSsrcOfSender(data, length); in DispatchReceivedPacket()
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| D | frame_receiver.cc | 80 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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