| /frameworks/opt/net/voip/src/java/android/net/sip/ |
| SipAudioCall.java | 21 import android.net.rtp.AudioCodec;
22 import android.net.rtp.AudioGroup;
23 import android.net.rtp.AudioStream;
24 import android.net.rtp.RtpStream;
740 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
759 && "RTP/AVP".equals(media.getProtocol())) {
770 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
823 "audio", mAudioStream.getLocalPort(), 1, "RTP/AVP");
[all...] |
| /sdk/eclipse/plugins/com.android.ide.eclipse.adt/src/com/android/ide/eclipse/adt/internal/refactorings/core/ |
| AndroidPackageRenameParticipant.java | 228 RenameTypeProcessor rtp = local
230 if (rtp != null) {
231 String pattern = rtp.getFilePatterns();
232 boolean updQualf = rtp.getUpdateQualifiedNames();
|
| AndroidTypeRenameParticipant.java | 206 RenameTypeProcessor rtp = local
208 if (rtp != null) {
209 String pattern = rtp.getFilePatterns();
210 boolean updQualf = rtp.getUpdateQualifiedNames();
|
| /external/libvorbis/doc/ |
| Makefile.am | 68 a2-encapsulation-rtp.tex \
|
| Vorbis_I_spec.tex | 117 \include{a2-encapsulation-rtp}
|
| /external/skia/src/gpu/glsl/ |
| GrGLSLFragmentShaderBuilder.cpp | 302 const GrRenderTargetPriv& rtp = pipeline.getRenderTarget()->renderTargetPriv(); local
303 const GrGpu::MultisampleSpecs& specs = rtp.getMultisampleSpecs(pipeline);
|
| /prebuilts/go/darwin-x86/src/runtime/ |
| defs_freebsd_arm.go | 110 rtp *rtprio
|
| defs_freebsd_386.go | 110 rtp *rtprio
|
| defs_freebsd_amd64.go | 111 rtp *rtprio
|
| /prebuilts/go/linux-x86/src/runtime/ |
| defs_freebsd_arm.go | 110 rtp *rtprio
|
| defs_freebsd_386.go | 110 rtp *rtprio
|
| defs_freebsd_amd64.go | 111 rtp *rtprio
|
| /external/webrtc/talk/media/webrtc/ |
| webrtcvoiceengine.cc | 89 // draft-spittka-payload-rtp-opus-03
431 // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
[all...] |
| webrtcvoiceengine_unittest.cc | 69 engine, // rtp
205 EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
211 EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
216 EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
222 EXPECT_EQ(1u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
223 EXPECT_EQ(ext, GetSendStreamConfig(kSsrc1).rtp.extensions[0].name);
224 EXPECT_EQ(id, GetSendStreamConfig(kSsrc1).rtp.extensions[0].id);
231 EXPECT_EQ(1u, GetSendStreamConfig(kSsrc2).rtp.extensions.size());
232 EXPECT_EQ(ext, GetSendStreamConfig(kSsrc2).rtp.extensions[0].name);
233 EXPECT_EQ(id, GetSendStreamConfig(kSsrc2).rtp.extensions[0].id)
[all...] |
| /external/skia/src/gpu/ |
| GrRenderTargetOpList.cpp | 30 GrRenderTargetOpList::GrRenderTargetOpList(GrRenderTargetProxy* rtp, GrGpu* gpu,
33 : INHERITED(rtp, auditTrail)
|
| GrContext.cpp | 832 sk_sp<GrTextureProxy> rtp = GrSurfaceProxy::MakeDeferred(this->resourceProvider(), local
[all...] |
| GrRenderTargetContext.cpp | 75 sk_sp<GrRenderTargetProxy> rtp,
81 , fRenderTargetProxy(std::move(rtp))
[all...] |
| /external/webrtc/talk/session/media/ |
| srtpfilter.cc | 175 // differently in RTP/RTCP mux and non-mux modes.
177 // - In the non-muxed case, RTP and RTCP are keyed with different
682 crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
685 crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); // rtp is 32,
706 // We want to set this option only for rtp packets.
711 policy.rtp.auth_type = EXTERNAL_HMAC_SHA1;
724 rtp_auth_tag_len_ = policy.rtp.auth_tag_len;
|
| channel_unittest.cc | 372 // Set SSRC in the rtp packet copy.
1745 TransportChannel* rtp = channel1_->transport_channel(); local
1777 TransportChannel* rtp = channel1_->transport_channel(); local
[all...] |
| /external/webrtc/tools/matlab/ |
| rtpAnalyze.m | 2 %RTP_ANALYZE Analyze RTP stream(s) from a txt file
6 % $ out/Debug/rtp_analyze my_file.rtp my_file.txt
21 % These appear as RTP packets having payload types 72 through 76.
|
| /external/linux-kselftest/ |
| .mailmap | 28 Arnaud Patard <arnaud.patard@rtp-net.org>
|
| /external/dhcpcd-6.8.2/ |
| ipv6.c | [all...] |
| /art/compiler/optimizing/ |
| inliner.cc | [all...] |
| /external/skia/src/gpu/instanced/ |
| InstanceProcessor.cpp | 1685 const GrRenderTargetPriv& rtp = pipeline.getRenderTarget()->renderTargetPriv(); local
[all...] |
| /frameworks/opt/telephony/src/java/com/android/internal/telephony/sip/ |
| SipPhone.java | 21 import android.net.rtp.AudioGroup;
[all...] |
