| /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;
|
| 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/chromium_org/third_party/webrtc/video_engine/ |
| vie_remb_unittest.cc | 45 MockRtpRtcp rtp; local
46 vie_remb_->AddReceiveChannel(&rtp);
47 vie_remb_->AddRembSender(&rtp);
56 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
61 EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, 1, _))
65 vie_remb_->RemoveReceiveChannel(&rtp);
66 vie_remb_->RemoveRembSender(&rtp);
70 MockRtpRtcp rtp; local
71 vie_remb_->AddReceiveChannel(&rtp);
72 vie_remb_->AddRembSender(&rtp);
198 MockRtpRtcp rtp; local
229 MockRtpRtcp rtp; local
[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.
|
| 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/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvie.h | 95 webrtc::ViERender* render, webrtc::ViERTP_RTCP* rtp,
104 rtp_(rtp),
116 webrtc::ViERTP_RTCP* rtp() { return rtp_.get(); } function in class:cricket::ViEWrapper
|
| webrtcvoe.h | 112 webrtc::VoERTP_RTCP* rtp,
125 rtp_(rtp),
140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } function in class:cricket::VoEWrapper
|
| /external/chromium_org/third_party/webrtc/ |
| video_send_stream.h | 80 struct Rtp {
81 Rtp()
88 // Max RTP packet size delivered to send transport from VideoEngine.
96 // RTP header extensions to use for this send stream.
105 // Settings for RTP retransmission payload format, see RFC 4588 for
123 } rtp; member in struct:webrtc::VideoSendStream::Config
|
| video_receive_stream.h | 42 // Received RTP packets with this payload type will be sent to this decoder
87 // Receive-stream specific RTP settings.
88 struct Rtp {
89 Rtp()
133 // Map from video RTP payload type -> RTX config.
137 // RTP header extensions used for the received stream.
139 } rtp; member in struct:webrtc::VideoReceiveStream::Config
|
| /external/chromium_org/third_party/libsrtp/srtp/include/ |
| srtp.h | 56 * @defgroup SRTP Secure RTP
58 * @brief libSRTP provides functions for protecting RTP and RTCP. See
84 * the maximum number of octets that will be added to an RTP packet by
216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member in struct:srtp_policy_t
227 * transmissions must have the same RTP
243 * An SRTP session consists of all of the traffic sent to the RTP and
244 * RTCP destination transport addresses, using the RTP/SAVP (Secure
288 * @brief srtp_protect() is the Secure RTP sender-side packet processing
292 * protection to the RTP packet rtp_hdr (which has length *len_ptr) using
298 * The sequence numbers of the RTP packets presented to this functio
[all...] |
| /external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
| voice_engine_jni.cc | 64 rtp(webrtc::VoERTP_RTCP::GetInterface(ve)) {
73 CHECK(rtp != NULL, "Failed to acquire rtp interface");
87 ReleaseSubApi(rtp);
125 webrtc::VoERTP_RTCP* const rtp; member in class:__anon19652::VoiceEngineData::webrtc
406 return voe_data->rtp->StartRTPDump(
414 return voe_data->rtp->StopRTPDump(
|
| video_engine_jni.cc | 159 rtp(webrtc::ViERTP_RTCP::GetInterface(vie)),
167 CHECK(rtp != NULL, "Failed to acquire rtp interface");
183 ReleaseSubApi(rtp);
260 webrtc::ViERTP_RTCP* const rtp; member in class:__anon19651::VideoEngineData::webrtc
580 return vie_data->rtp->SetNACKStatus(channel, enable);
587 return vie_data->rtp->SetKeyFrameRequestMethod(
599 if (vie_data->rtp->GetReceivedRTCPStatistics(channel, fraction_lost,
643 return vie_data->rtp->StartRTPDump(
650 return vie_data->rtp->StopRTPDump
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_rtpplay.cc | 45 DEFINE_int32(pcmu, 0, "RTP payload type for PCM-u");
48 DEFINE_int32(pcma, 8, "RTP payload type for PCM-a");
51 DEFINE_int32(ilbc, 102, "RTP payload type for iLBC");
54 DEFINE_int32(isac, 103, "RTP payload type for iSAC");
57 DEFINE_int32(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
60 DEFINE_int32(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
63 DEFINE_int32(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
66 DEFINE_int32(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
69 DEFINE_int32(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
72 DEFINE_int32(g722, 9, "RTP payload type for G.722")
185 NETEQTEST_RTPpacket* rtp; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| mt_rx_tx_test.cc | 152 new RTPSendCompleteCallback(Clock::GetRealTimeClock(), "dump.rtp");
158 RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration); local
165 // registering codecs for the RTP module
192 TEST(rtp->RegisterSendPayload(video_codec) == 0);
194 // inform RTP Module of error resilience features
195 TEST(rtp->SetGenericFECStatus(fecEnabled, VCM_RED_PAYLOAD_TYPE,
235 PacketRequester packetRequester(*rtp);
238 VCMRTPEncodeCompleteCallback* encodeCompleteCallback = new VCMRTPEncodeCompleteCallback(rtp);
256 // inform RTP Module of error resilience features
259 rtp->SetFecParameters(&delta_params, &key_params)
[all...] |
| /external/srtp/include/ |
| srtp.h | 56 * @defgroup SRTP Secure RTP
58 * @brief libSRTP provides functions for protecting RTP and RTCP. See
84 * the maximum number of octets that will be added to an RTP packet by
216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member in struct:srtp_policy_t
227 * transmissions must have the same RTP
243 * An SRTP session consists of all of the traffic sent to the RTP and
244 * RTCP destination transport addresses, using the RTP/SAVP (Secure
279 * @brief srtp_protect() is the Secure RTP sender-side packet processing
283 * protection to the RTP packet rtp_hdr (which has length *len_ptr) using
289 * The sequence numbers of the RTP packets presented to this functio
[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/chromium_org/media/cast/transport/ |
| cast_transport_config.h | 64 CastTransportRtpConfig rtp; member in struct:media::cast::transport::CastTransportAudioConfig
74 CastTransportRtpConfig rtp; member in struct:media::cast::transport::CastTransportVideoConfig
127 // The stream timestamp, on the timeline of the signal data. For example, RTP
139 // expected to drift with respect to the elapsed time implied by the RTP
180 uint32 media_ssrc; // SSRC of the RTP packet sender.
|
| /external/dhcpcd/ |
| configure.c | 642 struct rt *rtp, *rtl, *rtn; local
645 for (rtp = rt, rtl = NULL; rtp; rtl = rtp, rtp = rtp->next) {
646 if (rtp->dest.s_addr != INADDR_ANY)
649 for (rtn = rt; rtn != rtp; rtn = rtn->next) {
651 if (rtn->dest.s_addr == rtp->gate.s_addr)
654 cp = (const char *)&rtp->gate.s_addr
[all...] |
| /frameworks/av/media/libstagefright/wifi-display/rtp/ |
| RTPSender.cpp | 219 uint8_t *rtp = udpPacket->data(); local
220 rtp[0] = 0x80;
221 rtp[1] = packetType;
223 rtp[2] = (mRTPSeqNo >> 8) & 0xff;
224 rtp[3] = mRTPSeqNo & 0xff;
229 rtp[4] = rtpTime >> 24;
230 rtp[5] = (rtpTime >> 16) & 0xff;
231 rtp[6] = (rtpTime >> 8) & 0xff;
232 rtp[7] = rtpTime & 0xff;
234 rtp[8] = kSourceID >> 24
264 uint8_t *rtp = udpPacket->data(); local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_unittest.cc | 310 void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int* out_len) {
312 while ((sim_clock_ >= rtp->time()) &&
313 (rtp->dataLen() >= 0)) {
314 if (rtp->dataLen() > 0) {
316 rtp->parseHeader(&rtpInfo);
319 rtp->payload(),
320 rtp->payloadLen(),
321 rtp->time() * (output_sample_rate_ / 1000)));
324 ASSERT_NE(-1, rtp->readFromFile(rtp_fp_));
351 NETEQTEST_RTPpacket rtp; local
382 NETEQTEST_RTPpacket rtp; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| channel_unittest.cc | 472 // Set SSRC in the rtp packet copy.
1817 TransportChannel* rtp = channel1_->transport_channel(); local
1849 TransportChannel* rtp = channel1_->transport_channel(); local
[all...] |
