/external/nanopb-c/examples/simple/ |
Makefile | 9 CSRC = simple.c # The main program 10 CSRC += simple.pb.c # The compiled protocol definition 11 CSRC += $(NANOPB_DIR)/pb_encode.c # The nanopb encoder 12 CSRC += $(NANOPB_DIR)/pb_decode.c # The nanopb decoder 15 simple: $(CSRC) 16 $(CC) $(CFLAGS) -osimple $(CSRC)
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/external/chromium_org/third_party/webrtc/video_engine/mock/ |
mock_vie_frame_provider_base.h | 24 const uint32_t CSRC[kRtpCsrcSize]));
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/external/chromium_org/third_party/webrtc/video_engine/ |
vie_frame_provider_base.h | 33 const uint32_t CSRC[kRtpCsrcSize] = NULL) = 0; 80 const uint32_t CSRC[kRtpCsrcSize] = NULL);
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vie_frame_provider_base.cc | 50 const uint32_t CSRC[kRtpCsrcSize]) { 60 frame_callbacks_.front()->DeliverFrame(id_, video_frame, num_csrcs, CSRC); 65 (*it)->DeliverFrame(id_, video_frame, num_csrcs, CSRC); 72 (*it)->DeliverFrame(id_, extra_frame_.get(), num_csrcs, CSRC);
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vie_renderer.h | 107 const uint32_t CSRC[kRtpCsrcSize] = NULL);
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vie_encoder.h | 99 const uint32_t CSRC[kRtpCsrcSize] = NULL) OVERRIDE;
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vie_renderer.cc | 141 const uint32_t CSRC[kRtpCsrcSize]) {
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vie_encoder.cc | 455 const uint32_t CSRC[kRtpCsrcSize]) { 486 // Make sure the CSRC list is correct. 490 if (CSRC[i] == 1) { 493 tempCSRC[i] = CSRC[i];
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vie_channel.h | 145 // Gets the CSRC for the incoming stream. 247 const uint32_t CSRC,
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vie_channel.cc | [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_utility.cc | 356 uint32_t CSRC = *ptr++ << 24; 357 CSRC += *ptr++ << 16; 358 CSRC += *ptr++ << 8; 359 CSRC += *ptr++; 360 header.arrOfCSRCs[i] = CSRC;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
TestSenderReceiver.h | 96 const uint32_t CSRC,
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/external/chromium_org/third_party/webrtc/voice_engine/include/ |
voe_rtp_rtcp.h | 13 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC. 55 int channel, unsigned int CSRC, bool added) = 0;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
rtp_rtcp_defines.h | 271 const uint32_t CSRC, 344 const uint32_t CSRC,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api.cc | 87 TEST_F(RtpRtcpAPITest, CSRC) {
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
rtp_rtcp_test.cc | 25 unsigned int CSRC,
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/external/chromium_org/third_party/webrtc/video_engine/include/ |
vie_rtp_rtcp.h | 12 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC. 63 // This method is called if a field in CSRC changes or if the number of 66 const unsigned int CSRC,
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/external/chromium_org/third_party/webrtc/voice_engine/test/win_test/ |
WinTestDlg.h | 130 int channel, unsigned int CSRC, bool added);
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WinTestDlg.cc | 132 void CWinTestDlg::OnIncomingCSRCChanged(int channel, unsigned int CSRC, bool added) 135 str.Format(_T("OnIncomingCSRCChanged(channel=%d) => CSRC=%u, added=%d"), channel, CSRC, added); [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/ |
channel.h | 394 uint32_t CSRC, bool added);
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channel.cc | 309 uint32_t CSRC, 313 "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)", 314 id, CSRC, added); [all...] |
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
vie_autotest_rtp_rtcp.cc | 38 const unsigned int CSRC, const bool added) [all...] |