Searched refs:rtcp (Results 1 - 25 of 57) sorted by relevance

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/external/chromium_org/third_party/webrtc/system_wrappers/source/
H A Drtp_to_ntp_unittest.cc37 RtcpList rtcp; local
43 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
46 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
52 EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, &timestamp_in_ms));
56 RtcpList rtcp; local
62 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
65 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
67 EXPECT_TRUE(RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp, &timestamp_in_ms));
76 RtcpList rtcp; local
96 RtcpList rtcp; local
113 RtcpList rtcp; local
133 RtcpList rtcp; local
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H A Drtp_to_ntp.cc92 // pairs in |rtcp|. The converted timestamp is returned in
96 const RtcpList& rtcp,
98 assert(rtcp.size() == 2);
99 int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs,
100 rtcp.front().ntp_frac);
101 int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs,
102 rtcp.back().ntp_frac);
103 int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp;
104 int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp;
95 RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp, int64_t* rtp_timestamp_in_ms) argument
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtcp_receiver_unittest.cc153 rtcp::SenderReport sr;
155 rtcp::RawPacket p = sr.Build();
167 rtcp::SenderReport sr;
169 rtcp::RawPacket p = sr.Build();
177 rtcp::ReceiverReport rr;
179 rtcp::RawPacket p = rr.Build();
193 rtcp::ReportBlock rb;
195 rtcp::ReceiverReport rr;
198 rtcp::RawPacket p = rr.Build();
212 rtcp
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H A Drtcp_packet_unittest.cc18 using webrtc::rtcp::App;
19 using webrtc::rtcp::Bye;
20 using webrtc::rtcp::Dlrr;
21 using webrtc::rtcp::Empty;
22 using webrtc::rtcp::Fir;
23 using webrtc::rtcp::Ij;
24 using webrtc::rtcp::Nack;
25 using webrtc::rtcp::Pli;
26 using webrtc::rtcp::Sdes;
27 using webrtc::rtcp
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/external/chromium_org/third_party/webrtc/video_engine/
H A Dstream_synchronization.h26 Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
27 RtcpList rtcp; member in struct:webrtc::StreamSynchronization::Measurements
H A Dstream_synchronization.cc62 if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) {
68 audio_measurement.rtcp,
74 video_measurement.rtcp,
H A Dstream_synchronization_unittest.cc37 RtcpMeasurement rtcp; local
38 NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac);
39 rtcp.rtp_timestamp = NowRtp(frequency, offset);
40 return rtcp;
104 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
108 video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
112 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
116 video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency,
H A Dvie_sync_module.cc47 ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
/external/chromium_org/third_party/libjingle/source/talk/session/media/
H A Dmediasink.h42 virtual void OnPacket(const void* data, size_t size, bool rtcp) = 0;
H A Dbundlefilter.h46 // For rtp packets, this is decided based on the payload type. For rtcp packets,
54 bool DemuxPacket(const char* data, size_t len, bool rtcp);
H A Dbundlefilter.cc43 bool BundleFilter::DemuxPacket(const char* data, size_t len, bool rtcp) { argument
45 // For rtcp packets, we check whether the ssrc can be found or is the special
47 // |streams_| is empty, we will allow all rtcp packets pass through provided
48 // that they are valid rtcp packets in case that they are for early media.
49 if (!rtcp) {
78 // Pass through if |streams_| is empty to allow early rtcp packets in.
H A Dchannelmanager.h113 BaseSession* session, const std::string& content_name, bool rtcp);
119 BaseSession* session, const std::string& content_name, bool rtcp,
125 bool rtcp, DataChannelType data_channel_type);
262 BaseSession* session, const std::string& content_name, bool rtcp);
265 BaseSession* session, const std::string& content_name, bool rtcp,
270 bool rtcp, DataChannelType data_channel_type);
H A Dchannel.cc128 static const char* PacketType(bool rtcp) { argument
129 return (!rtcp) ? "RTP" : "RTCP";
132 static bool ValidPacket(bool rtcp, const rtc::Buffer* packet) { argument
135 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
157 const std::string& content_name, bool rtcp)
163 rtcp_(rtcp),
202 if (rtcp() && rtcp_transport_channel == NULL) {
374 bool rtcp = PacketIsRtcp(channel, data, len); local
376 HandlePacket(rtcp, &packet, packet_time);
393 // Notify the MediaChannel when either rtp or rtcp channe
154 BaseChannel(rtc::Thread* thread, MediaEngineInterface* media_engine, MediaChannel* media_channel, BaseSession* session, const std::string& content_name, bool rtcp) argument
409 SendPacket(bool rtcp, rtc::Buffer* packet, rtc::DiffServCodePoint dscp) argument
539 WantsPacket(bool rtcp, rtc::Buffer* packet) argument
552 HandlePacket(bool rtcp, rtc::Buffer* packet, const rtc::PacketTime& packet_time) argument
748 SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp) argument
1252 VoiceChannel(rtc::Thread* thread, MediaEngineInterface* media_engine, VoiceMediaChannel* media_channel, BaseSession* session, const std::string& content_name, bool rtcp) argument
1651 VideoChannel(rtc::Thread* thread, MediaEngineInterface* media_engine, VideoMediaChannel* media_channel, BaseSession* session, const std::string& content_name, bool rtcp, VoiceChannel* voice_channel) argument
2100 DataChannel(rtc::Thread* thread, DataMediaChannel* media_channel, BaseSession* session, const std::string& content_name, bool rtcp) argument
2152 WantsPacket(bool rtcp, rtc::Buffer* packet) argument
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H A Dmediarecorder.h63 virtual void OnPacket(const void* data, size_t size, bool rtcp);
H A Dchannelmanager.cc319 BaseSession* session, const std::string& content_name, bool rtcp) {
322 session, content_name, rtcp));
326 BaseSession* session, const std::string& content_name, bool rtcp) {
335 session, content_name, rtcp);
365 BaseSession* session, const std::string& content_name, bool rtcp,
369 content_name, rtcp, voice_channel));
373 BaseSession* session, const std::string& content_name, bool rtcp,
386 session, content_name, rtcp, voice_channel);
417 bool rtcp, DataChannelType channel_type) {
420 rtcp, channel_typ
318 CreateVoiceChannel( BaseSession* session, const std::string& content_name, bool rtcp) argument
325 CreateVoiceChannel_w( BaseSession* session, const std::string& content_name, bool rtcp) argument
364 CreateVideoChannel( BaseSession* session, const std::string& content_name, bool rtcp, VoiceChannel* voice_channel) argument
372 CreateVideoChannel_w( BaseSession* session, const std::string& content_name, bool rtcp, VoiceChannel* voice_channel) argument
415 CreateDataChannel( BaseSession* session, const std::string& content_name, bool rtcp, DataChannelType channel_type) argument
423 CreateDataChannel_w( BaseSession* session, const std::string& content_name, bool rtcp, DataChannelType data_channel_type) argument
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H A Dchannel.h80 const std::string& content_name, bool rtcp);
248 bool rtcp() const { return rtcp_; } function in class:cricket::BaseChannel
270 bool SendPacket(bool rtcp, rtc::Buffer* packet,
272 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
273 void HandlePacket(bool rtcp, rtc::Buffer* packet,
295 bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
403 const std::string& content_name, bool rtcp);
512 const std::string& content_name, bool rtcp,
607 bool rtcp);
695 virtual bool WantsPacket(bool rtcp, rt
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/external/chromium_org/third_party/webrtc/system_wrappers/interface/
H A Drtp_to_ntp.h41 bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp,
/external/srtp/test/
H A Drtpw.c315 crypto_policy_set_rtcp_default(&policy.rtcp);
319 crypto_policy_set_rtcp_default(&policy.rtcp);
323 crypto_policy_set_rtcp_default(&policy.rtcp);
336 policy.rtcp.sec_serv = sec_serv_none; /* we don't do RTCP anyway */
381 policy.rtcp.cipher_type = NULL_CIPHER;
382 policy.rtcp.cipher_key_len = 0;
383 policy.rtcp.auth_type = NULL_AUTH;
384 policy.rtcp.auth_key_len = 0;
385 policy.rtcp.auth_tag_len = 0;
386 policy.rtcp
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/external/chromium_org/third_party/libjingle/source/talk/media/base/
H A Drtpdump.h74 RtpDumpPacket(const void* d, size_t s, uint32 elapsed, bool rtcp) argument
76 original_data_len((rtcp) ? 0 : s) {
218 uint32 elapsed, bool rtcp);
219 size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
H A Drtpdump.cc353 const void* data, size_t data_len, uint32 elapsed, bool rtcp) {
367 size_t write_len = FilterPacket(data, data_len, rtcp);
376 buf.WriteUInt16(static_cast<uint16>(rtcp ? 0 : data_len));
388 bool rtcp) {
390 if (!rtcp) {
352 WritePacket( const void* data, size_t data_len, uint32 elapsed, bool rtcp) argument
387 FilterPacket(const void* data, size_t data_len, bool rtcp) argument
/external/chromium_org/third_party/libsrtp/srtp/test/
H A Drtpw.c335 crypto_policy_set_rtcp_default(&policy.rtcp);
339 crypto_policy_set_rtcp_default(&policy.rtcp);
343 crypto_policy_set_rtcp_default(&policy.rtcp);
357 policy.rtcp.sec_serv = sec_serv_none; /* we don't do RTCP anyway */
402 policy.rtcp.cipher_type = NULL_CIPHER;
403 policy.rtcp.cipher_key_len = 0;
404 policy.rtcp.auth_type = NULL_AUTH;
405 policy.rtcp.auth_key_len = 0;
406 policy.rtcp.auth_tag_len = 0;
407 policy.rtcp
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/
H A Dremote_bitrate_estimator_unittest_helper.cc84 RtcpPacket* rtcp = new RtcpPacket; local
86 rtcp->timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
88 rtcp->ntp_secs = send_time_us / 1000000;
89 rtcp->ntp_frac = static_cast<int64_t>((send_time_us % 1000000) *
91 rtcp->ssrc = ssrc_;
93 return rtcp;
/external/chromium_org/third_party/libsrtp/srtp/include/
H A Dsrtp.h217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member in struct:srtp_policy_t
/external/srtp/include/
H A Dsrtp.h217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member in struct:srtp_policy_t
/external/chromium_org/media/cast/net/rtcp/
H A Drtcp_unittest.cc11 #include "media/cast/net/rtcp/rtcp.h"
30 void set_rtcp_destination(Rtcp* rtcp) { rtcp_ = rtcp; } argument

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