| /external/webrtc/talk/media/base/ |
| rtputils.h | 59 bool GetRtpSsrc(const void* data, size_t len, uint32_t* value);
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| rtputils.cc | 121 bool GetRtpSsrc(const void* data, size_t len, uint32_t* value) {
150 GetRtpSsrc(data, len, &(header->ssrc)));
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| rtpdump_unittest.cc | 61 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc));
133 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc));
149 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc));
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| fakenetworkinterface.h | 136 if (!GetRtpSsrc(packet->data(), packet->size(), &cur_ssrc)) {
210 if (!GetRtpSsrc(rtp_packets_[i].data(), rtp_packets_[i].size(),
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| rtputils_unittest.cc | 96 EXPECT_TRUE(GetRtpSsrc(kPcmuFrame, sizeof(kPcmuFrame), &ssrc));
109 EXPECT_FALSE(GetRtpSsrc(kInvalidPacket, sizeof(kInvalidPacket), &ssrc));
|
| rtpdump.h | 90 bool GetRtpSsrc(uint32_t* ssrc) const;
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| rtpdump.cc | 91 bool RtpDumpPacket::GetRtpSsrc(uint32_t* ssrc) const {
93 cricket::GetRtpSsrc(&data[0], data.size(), ssrc);
|
| /external/webrtc/talk/media/webrtc/ |
| fakewebrtccall.cc | 393 if (!GetRtpSsrc(packet, length, &ssrc))
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| webrtcvideoengine2.cc | [all...] |
| webrtcvoiceengine.cc | [all...] |
| /external/webrtc/talk/session/media/ |
| srtpfilter.cc | 540 if (GetRtpSsrc(p, in_len, &ssrc)) {
598 if (GetRtpSsrc(p, in_len, &ssrc)) {
|
| channel.cc | 633 GetRtpSsrc(data, len, &ssrc);
717 GetRtpSsrc(data, len, &ssrc);
[all...] |
