| /cts/tests/tests/net/src/android/net/rtp/cts/ |
| AudioCodecTest.java | 16 package android.net.rtp.cts;
18 import android.net.rtp.AudioCodec;
|
| AudioStreamTest.java | 16 package android.net.rtp.cts;
18 import android.net.rtp.AudioCodec;
19 import android.net.rtp.AudioStream;
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| AudioGroupTest.java | 16 package android.net.rtp.cts;
20 import android.net.rtp.AudioCodec;
21 import android.net.rtp.AudioGroup;
22 import android.net.rtp.AudioStream;
23 import android.net.rtp.RtpStream;
|
| /external/webrtc/webrtc/video/ |
| vie_remb_unittest.cc | 47 MockRtpRtcp rtp; local
48 vie_remb_->AddReceiveChannel(&rtp);
49 vie_remb_->AddRembSender(&rtp);
58 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
63 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, ssrcs))
67 vie_remb_->RemoveReceiveChannel(&rtp);
68 vie_remb_->RemoveRembSender(&rtp);
72 MockRtpRtcp rtp; local
73 vie_remb_->AddReceiveChannel(&rtp);
74 vie_remb_->AddRembSender(&rtp);
200 MockRtpRtcp rtp; local
231 MockRtpRtcp rtp; local
[all...] |
| payload_router_unittest.cc | 37 MockRtpRtcp rtp; local
38 std::list<RtpRtcp*> modules(1, &rtp);
46 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
53 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
60 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
67 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
75 EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
|
| vie_channel.cc | 148 // RTP/RTCP initialization.
178 // Make sure we don't get more callbacks from the RTP module.
236 StreamDataCounters rtp; local
238 GetSendStreamDataCounters(&rtp, &rtx);
239 StreamDataCounters rtp_rtx = rtp;
251 static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
301 StreamDataCounters rtp; local
303 GetReceiveStreamDataCounters(&rtp, &rtx);
304 StreamDataCounters rtp_rtx = rtp;
314 static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000))
[all...] |
| /frameworks/opt/net/voip/src/java/android/net/rtp/ |
| AudioCodec.java | 17 package android.net.rtp;
39 * The RTP payload type of the encoding.
100 * @param type The payload type of the encoding defined in RTP/AVP.
|
| AudioStream.java | 17 package android.net.rtp;
24 * Real-time Transport Protocol (RTP). Two different classes are developed in
130 * Returns the RTP payload type for dual-tone multi-frequency (DTMF) digits,
140 * Sets the RTP payload type for dual-tone multi-frequency (DTMF) digits.
143 * RTP payload type for DTMF is assigned dynamically, so it must be in the
148 * @param type The RTP payload type to be used or {@code -1} to disable it.
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| AudioGroup.java | 17 package android.net.rtp;
|
| RtpStream.java | 17 package android.net.rtp;
26 * packets with media payloads over Real-time Transport Protocol (RTP).
|
| /external/skia/src/gpu/ |
| GrSurfaceProxy.cpp | 220 const GrRenderTargetProxy* rtp = this->asRenderTargetProxy(); local
222 if (rtp) {
223 sampleCount = rtp->numStencilSamples();
235 GrTexturePriv::ComputeScratchKey(this->config(), width, height, SkToBool(rtp), sampleCount,
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| GrContext.cpp | 1387 sk_sp<GrTextureProxy> rtp; local
[all...] |
| /external/skqp/src/gpu/ |
| GrSurfaceProxy.cpp | 192 const GrRenderTargetProxy* rtp = this->asRenderTargetProxy(); local
194 if (rtp) {
195 sampleCount = rtp->numStencilSamples();
207 GrTexturePriv::ComputeScratchKey(this->config(), width, height, SkToBool(rtp), sampleCount,
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| GrContext.cpp | 994 sk_sp<GrTextureProxy> rtp; local
[all...] |
| /external/skqp/src/gpu/glsl/ |
| GrGLSLFragmentShaderBuilder.cpp | 301 const GrRenderTargetPriv& rtp = pipeline.renderTarget()->renderTargetPriv(); local
302 const GrGpu::MultisampleSpecs& specs = rtp.getMultisampleSpecs(pipeline);
|
| /external/webrtc/talk/media/webrtc/ |
| webrtcvoe.h | 102 webrtc::VoERTP_RTCP* rtp,
110 rtp_(rtp),
120 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } function in class:cricket::VoEWrapper
|
| /external/webrtc/webrtc/ |
| audio_receive_stream.h | 66 // Receive-stream specific RTP settings.
67 struct Rtp {
82 // RTP header extensions used for the received stream.
84 } rtp; member in struct:webrtc::AudioReceiveStream::Config
|
| audio_send_stream.h | 60 // Receive-stream specific RTP settings.
61 struct Rtp {
67 // RTP header extensions used for the sent stream.
72 } rtp; member in struct:webrtc::AudioSendStream::Config
|
| video_receive_stream.h | 39 // Received RTP packets with this payload type will be sent to this decoder
86 // Receive-stream specific RTP settings.
87 struct Rtp {
127 // Map from video RTP payload type -> RTX config.
136 // RTP header extensions used for the received stream.
138 } rtp; member in struct:webrtc::VideoReceiveStream::Config
|
| video_send_stream.h | 100 struct Rtp {
108 // Max RTP packet size delivered to send transport from VideoEngine.
111 // RTP header extensions to use for this send stream.
120 // Settings for RTP retransmission payload format, see RFC 4588 for
133 } rtp; member in struct:webrtc::VideoSendStream::Config
|
| /external/curl/lib/ |
| rtsp.c | 71 * Parse and write out any available RTP data.
73 * nread: amount of data left after k->str. will be modified if RTP
75 * readmore: whether or not the RTP parser needs more data right away
143 * The server may send us RTP data at any point, and RTSPREQ_RECEIVE does not
242 infof(data, "Got an RTP Receive with a CSeq of %ld\n", CSeq_recv);
326 /* Treat interleaved RTP as body*/
629 char *rtp; /* moving pointer to rtp data */ local
646 rtp = rtspc->rtp_buf;
651 rtp = k->str
[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/dhcpcd-6.8.2/ |
| ipv4.c | 577 struct rt *rtp, *rtn; local
585 TAILQ_FOREACH(rtp, rt, next) {
586 if (rtp->dest.s_addr != INADDR_ANY)
590 if (rtn == rtp)
593 if (rtn->dest.s_addr == rtp->gate.s_addr)
596 cp = (const char *)&rtp->gate.s_addr;
607 if (rtn != rtp)
618 ifp->name, inet_ntoa(rtp->gate));
620 rtp->gate.s_addr = 0;
629 ifp->name, inet_ntoa(rtp->gate))
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_rtcp_impl_unittest.cc | 414 StreamDataCounters rtp; local
416 rtp.first_packet_time_ms = kStartTimeMs;
417 rtp.transmitted.packets = 1;
418 rtp.transmitted.payload_bytes = 1;
419 rtp.transmitted.header_bytes = 2;
420 rtp.transmitted.padding_bytes = 3;
421 EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes +
422 rtp.transmitted.header_bytes +
423 rtp.transmitted.padding_bytes)
[all...] |
