/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...] |