| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_header_parser.cc | 46 return rtp_parser.RTCP();
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| rtp_utility.h | 52 bool RTCP() const;
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| rtp_sender_unittest.cc | 212 ASSERT_FALSE(rtp_parser.RTCP());
341 ASSERT_FALSE(rtp_parser.RTCP());
376 ASSERT_FALSE(rtp_parser.RTCP());
416 ASSERT_FALSE(rtp_parser.RTCP());
443 ASSERT_FALSE(rtp_parser.RTCP());
480 ASSERT_FALSE(rtp_parser.RTCP());
508 ASSERT_FALSE(rtp_parser.RTCP());
536 ASSERT_FALSE(rtp_parser.RTCP());
595 ASSERT_FALSE(rtp_parser.RTCP());
[all...] |
| rtp_utility.cc | 83 bool RtpHeaderParser::RTCP() const {
86 // for RTCP 200 SR == marker bit + 72
87 // for RTCP 204 APP == marker bit + 76
89 * RTCP
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| rtp_rtcp_impl.h | 42 // Called when we receive an RTCP packet.
128 // RTCP part.
130 // Get RTCP status.
131 RtcpMode RTCP() const override;
133 // Configure RTCP status i.e on/off.
136 // Set RTCP CName.
161 // Force a send of an RTCP packet.
181 // Get received RTCP report, sender info.
184 // Get received RTCP report, report block.
228 // Called on receipt of RTCP report block from remote side
[all...] |
| rtp_rtcp_impl.cc | 103 // Make sure that RTCP objects are aware of our SSRC.
153 LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
156 "Timeout: No increase in RTCP RR extended highest sequence number.";
225 // Allow receive of non-compound RTCP packets.
230 LOG(LS_WARNING) << "Incoming invalid RTCP packet";
329 // TODO(pbos): Handle media and RTX streams separately (separate RTCP
362 // Sends RTCP BYE when going from true to false
364 LOG(LS_WARNING) << "Failed to send RTCP BYE";
373 // Make sure the RTCP sender has the same timestamp offset.
377 // Make sure that RTCP objects are aware of our SSRC (it could have change
[all...] |
| /external/webrtc/talk/session/media/ |
| channel_unittest.cc | 127 enum Flags { RTCP = 0x1, RTCP_MUX = 0x2, SECURE = 0x4, SSRC_MUX = 0x8,
157 (flags1 & RTCP) != 0));
160 (flags2 & RTCP) != 0));
208 bool rtcp) {
210 thread, engine, ch, transport_controller, cricket::CN_AUDIO, rtcp);
383 // Set SSRC in the rtcp packet copy.
521 // Test that SetLocalContent and SetRemoteContent properly set RTCP
524 CreateChannels(RTCP, RTCP);
529 // Both sides agree on mux. Should no longer be a separate RTCP channel
1746 TransportChannel* rtcp = channel1_->rtcp_transport_channel(); local
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.cc | 142 TEST_F(RtpRtcpAPITest, RTCP) {
143 EXPECT_EQ(RtcpMode::kOff, module_->RTCP());
145 EXPECT_EQ(RtcpMode::kCompound, module_->RTCP());
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| /external/webrtc/webrtc/test/ |
| rtp_file_reader_unittest.cc | 88 if (!rtp_header_parser.RTCP() &&
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| rtp_file_reader.cc | 185 // Use 'len' here because a 'plen' of 0 specifies rtcp.
311 printf("Total RTP/RTCP packets: %" PRIuS "\n", packets_.size());
457 if (rtp_parser.RTCP()) {
462 DEBUG_LOG("Not recognized as RTP/RTCP");
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| /external/webrtc/tools/matlab/ |
| rtpAnalyze.m | 20 %% Filter out RTCP packets.
23 fprintf('Removing %i RTCP packets\n', length(SeqNo) - sum(ix));
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| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| rtp_rtcp.h | 27 namespace rtcp { namespace in namespace:webrtc
36 /* id - Unique identifier of this RTP/RTCP module object
37 * audio - True for a audio version of the RTP/RTCP module
79 * Create a RTP/RTCP module object using the system clock.
81 * configuration - Configuration of the RTP/RTCP module.
324 * RTCP
329 * Get RTCP status
331 virtual RtcpMode RTCP() const = 0;
334 * configure RTCP status i.e on(compound or non- compound)/off
336 * method - RTCP method to us
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 140 MOCK_CONST_METHOD0(RTCP, RtcpMode());
218 MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
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| /external/webrtc/webrtc/voice_engine/ |
| channel.cc | 73 void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
148 // Extend the default RTCP statistics struct with max_jitter, defined as the
149 // maximum jitter value seen in an RTCP report block.
151 ChannelStatistics() : rtcp(), max_jitter(0) {}
153 RtcpStatistics rtcp; member in struct:webrtc::voe::ChannelStatistics
157 // Statistics callback, called at each generation of a new RTCP report block.
171 stats_.rtcp = statistics;
206 // what we get? I.e. do we ever get multiple reports bundled into one RTCP
263 // Store current audio level in the RTP/RTCP module.
269 // Push data from ACM to RTP/RTCP-module to deliver audio frame fo
[all...] |
| /external/webrtc/webrtc/video/ |
| vie_channel.cc | 148 // RTP/RTCP initialization.
514 if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff)
[all...] |
