| /external/chromium_org/third_party/webrtc/system_wrappers/source/ |
| rtp_to_ntp_unittest.cc | 37 RtcpList rtcp; local
43 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
46 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp));
50 // This expected to fail since it's highly unlikely that the older RTCP
52 EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, ×tamp_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, ×tamp_in_ms))
76 RtcpList rtcp; local
96 RtcpList rtcp; local
113 RtcpList rtcp; local
133 RtcpList rtcp; local
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| stream_synchronization.h | 26 Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {}
27 RtcpList rtcp; member in struct:webrtc::StreamSynchronization::Measurements
|
| stream_synchronization_unittest.cc | 37 RtcpMeasurement rtcp; local
38 NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac);
39 rtcp.rtp_timestamp = NowRtp(frequency, offset);
40 return rtcp;
84 // Generates the necessary RTCP measurements and RTP timestamps and computes
103 // Generate NTP/RTP timestamp pair for both streams corresponding to 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
[all...] |
| /external/chromium_org/third_party/libsrtp/srtp/include/ |
| srtp.h | 58 * @brief libSRTP provides functions for protecting RTP and RTCP. See
217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member in struct:srtp_policy_t
244 * RTCP destination transport addresses, using the RTP/SAVP (Secure
466 * structure to the SRTP default policy for RTCP protection.
471 * crypto_policy_t at location p to the SRTP default policy for RTCP
731 * structure to the appropriate value for RTCP based on an srtp_profile_t
736 * sets the crypto_policy_t at location policy to the policy for RTCP
792 * @defgroup SRTCP Secure RTCP
795 * @brief Secure RTCP functions are used to protect RTCP traffic
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_packet.cc | 51 namespace rtcp { namespace in namespace:webrtc
694 } // namespace rtcp
|
| rtcp_packet.h | 24 namespace rtcp { namespace in namespace:webrtc
31 // Class for building RTCP packets.
52 // // the built rtcp packet.
54 // rr.Append(&fir) // Builds a compound RTCP packet with
95 // RTCP report block (RFC 3550).
150 // RTCP sender report (RFC 3550).
218 // RTCP receiver report (RFC 3550).
274 // If present, this RTCP packet must be placed after a receiver report
275 // (inside a compound RTCP packet), and MUST have the same value for RC
695 // Class holding a RTCP packet
[all...] |
| /external/srtp/include/ |
| srtp.h | 58 * @brief libSRTP provides functions for protecting RTP and RTCP. See
217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member in struct:srtp_policy_t
244 * RTCP destination transport addresses, using the RTP/SAVP (Secure
457 * structure to the SRTP default policy for RTCP protection.
462 * crypto_policy_t at location p to the SRTP default policy for RTCP
658 * structure to the appropriate value for RTCP based on an srtp_profile_t
663 * sets the crypto_policy_t at location policy to the policy for RTCP
719 * @defgroup SRTCP Secure RTCP
722 * @brief Secure RTCP functions are used to protect RTCP traffic
[all...] |
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.cc | 79 // Generates an RTCP packet.
80 RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) {
84 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;
206 0)); // RTCP receive time.
456 0)); // RTCP receive time
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| channel.h | 81 const std::string& content_name, bool rtcp);
249 bool rtcp() const { return rtcp_; } function in class:cricket::BaseChannel
271 bool SendPacket(bool rtcp, talk_base::Buffer* packet,
273 virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
274 void HandlePacket(bool rtcp, talk_base::Buffer* packet,
293 // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
296 bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
397 const std::string& content_name, bool rtcp);
514 const std::string& content_name, bool rtcp,
614 bool rtcp);
[all...] |
| call.cc | 316 bool rtcp = false; local
319 session, data_offer->name, rtcp, data_channel_type);
[all...] |
| channel.cc | 143 static const char* PacketType(bool rtcp) {
144 return (!rtcp) ? "RTP" : "RTCP";
147 static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) {
150 packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
172 const std::string& content_name, bool rtcp)
178 rtcp_(rtcp),
200 FlushRtcpMessages(); // Send any outstanding RTCP packets.
217 if (rtcp() && rtcp_transport_channel == NULL) {
239 // Both RTP and RTCP channels are set, we can call SetInterface o
389 bool rtcp = PacketIsRtcp(channel, data, len); local
[all...] |
| channel_unittest.cc | 169 enum Flags { RTCP = 0x1, RTCP_MUX = 0x2, SECURE = 0x4, SSRC_MUX = 0x8,
224 (flags1 & RTCP) != 0));
226 (flags2 & RTCP) != 0));
279 (flags & RTCP) != 0));
281 (flags & RTCP) != 0));
311 bool rtcp) {
313 thread, engine, ch, session, cricket::CN_AUDIO, rtcp);
482 // Set SSRC in the rtcp packet copy.
628 // Test that SetLocalContent and SetRemoteContent properly set RTCP
631 CreateChannels(RTCP, RTCP)
1818 TransportChannel* rtcp = channel1_->rtcp_transport_channel(); local
[all...] |
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| channel.cc | 45 // Extend the default RTCP statistics struct with max_jitter, defined as the
46 // maximum jitter value seen in an RTCP report block.
48 ChannelStatistics() : rtcp(), max_jitter(0) {}
50 RtcpStatistics rtcp; member in struct:webrtc::voe::ChannelStatistics
54 // Statistics callback, called at each generation of a new RTCP report block.
68 stats_.rtcp = statistics;
126 // Store current audio level in the RTP/RTCP module.
132 // Push data from ACM to RTP/RTCP-module to deliver audio frame for
134 // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
148 "Channel::SendData() failed to send data to RTP/RTCP module")
[all...] |
