| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| mt_test_common.h | 28 SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp,
31 _rtp(rtp),
49 // constructor input: (receive side) rtp module to send encoded data to
63 SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) :
65 _rtp(rtp) {}
74 SharedTransportState(webrtc::RtpRtcp& rtp, TransportCallback& transport):
75 _rtp(rtp),
|
| test_callbacks.h | 92 // RTP module
96 VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) :
99 _RTPModule(rtp) {}
103 // RTP module
155 // Called by the RTP Sender - simulates sending packets through a network to the
156 // RTP receiver. User can set network conditions as: RTT, packet loss,
161 // Constructor input: (receive side) rtp module to send encoded data to
167 // Send Packet to receive side RTP module
169 // Send RTCP Packet to receive side RTP module
210 PacketRequester(RtpRtcp& rtp)
[all...] |
| 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/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...] |
| /external/chromium_org/third_party/webrtc/video/ |
| video_receive_stream.cc | 53 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
55 SetRtcpMode(config_.rtp.rtcp_mode);
57 assert(config_.rtp.remote_ssrc != 0);
59 assert(config_.rtp.local_ssrc != 0);
60 assert(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
62 rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
64 Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
65 if (it != config_.rtp.rtx.end())
[all...] |
| send_statistics_proxy_unittest.cc | 37 config.rtp.ssrcs.push_back(17);
38 config.rtp.ssrcs.push_back(42);
39 config.rtp.rtx.ssrcs.push_back(18);
40 config.rtp.rtx.ssrcs.push_back(43);
92 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
93 it != config_.rtp.ssrcs.end();
106 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
107 it != config_.rtp.rtx.ssrcs.end();
154 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
155 it != config_.rtp.ssrcs.end()
[all...] |
| video_send_stream.cc | 42 std::string VideoSendStream::Config::Rtp::Rtx::ToString()
58 std::string VideoSendStream::Config::Rtp::ToString() const {
95 ss << ", rtp: " << rtp.ToString();
137 assert(config_.rtp.ssrcs.size() > 0);
139 assert(config_.rtp.min_transmit_bitrate_bps >= 0);
141 config_.rtp.min_transmit_bitrate_bps / 1000);
143 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
144 const std::string& extension = config_.rtp.extensions[i].name;
145 int id = config_.rtp.extensions[i].id
[all...] |
| send_statistics_proxy.cc | 55 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) ==
56 config_.rtp.ssrcs.end() &&
57 std::find(config_.rtp.rtx.ssrcs.begin(),
58 config_.rtp.rtx.ssrcs.end(),
59 ssrc) == config_.rtp.rtx.ssrcs.end()) {
|
| bitrate_estimator_tests.cc | 146 send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
159 receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
160 receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
161 receive_config_.rtp.extensions.push_back(
163 receive_config_.rtp.extensions.push_back(
195 test_->send_config_.rtp.ssrcs[0]++;
212 test_->receive_config_.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
213 test_->receive_config_.rtp.local_ssrc++
[all...] |
| loopback.cc | 115 send_config.rtp.ssrcs.push_back(kSendSsrc);
116 send_config.rtp.rtx.ssrcs.push_back(kSendRtxSsrc);
117 send_config.rtp.rtx.payload_type = kRtxPayloadType;
118 send_config.rtp.nack.rtp_history_ms = 1000;
156 receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
157 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
158 receive_config.rtp.nack.rtp_history_ms = 1000;
159 receive_config.rtp.rtx[kRtxPayloadType].ssrc = kSendRtxSsrc;
160 receive_config.rtp.rtx[kRtxPayloadType].payload_type = kRtxPayloadType
[all...] |
| replay.cc | 86 DEFINE_int32(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
95 "RTP extension ID for transmission-offset");
197 receive_config.rtp.remote_ssrc = flags::Ssrc();
198 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
199 receive_config.rtp.fec.ulpfec_payload_type = flags::FecPayloadType();
200 receive_config.rtp.fec.red_payload_type = flags::RedPayloadType();
201 receive_config.rtp.nack.rtp_history_ms = 1000;
203 receive_config.rtp.extensions.push_back(
207 receive_config.rtp.extensions.push_back(
|
| /external/libvorbis/doc/ |
| a2-encapsulation-rtp.tex | 4 \section{Vorbis encapsulation in RTP} \label{vorbis:over:rtp}
6 % TODO: Include draft-rtp.xml somehow?
8 Please consult RFC 5215 \textit{``RTP Payload Format for Vorbis Encoded
9 Audio''} for description of how to embed Vorbis audio in an RTP stream.
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| generic_encoder.cc | 20 // Map information from info into rtp. If no relevant information is found
21 // in info, rtp is set to NULL.
22 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader** rtp) {
24 *rtp = NULL;
29 (*rtp)->codec = kRtpVideoVp8;
30 (*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8();
31 (*rtp)->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
32 (*rtp)->codecHeader.VP8.nonReference =
34 (*rtp)->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
35 (*rtp)->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync
[all...] |
| /external/chromium_org/content/browser/renderer_host/p2p/ |
| socket_host.cc | 70 // Verifies rtp header and message length.
71 bool ValidateRtpHeader(const char* rtp, size_t length, size_t* header_length) {
80 size_t cc_count = rtp[0] & 0x0F;
88 if (!(rtp[0] & 0x10)) {
95 rtp += header_length_without_extension;
103 uint16 extension_length_in_32bits = rtc::GetBE16(rtp + 2);
124 // Absolute send time in RTP streams.
127 // general mechanism for RTP header extensions [RFC5285]. The payload
161 // the RTP packet.
162 void UpdateRtpAuthTag(char* rtp,
[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...] |
| /cts/tests/tests/net/src/android/net/rtp/cts/ |
| 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;
|
| AudioCodecTest.java | 16 package android.net.rtp.cts;
18 import android.net.rtp.AudioCodec;
|
| /external/chromium_org/media/cast/logging/ |
| receiver_time_offset_estimator_impl.cc | 20 uint32 rtp,
24 uint64 key = (static_cast<uint64>(rtp) << 32) | (packet_id << 1) |
31 uint32 rtp,
35 uint64 key = (static_cast<uint64>(rtp) << 32) | (packet_id << 1) |
|
| receiver_time_offset_estimator_impl.h | 68 void SetSent(uint32 rtp,
73 void SetReceived(uint32 rtp,
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvideoengine2_unittest.cc | 53 "urn:ietf:params:rtp-hdrext:unsupported";
685 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
686 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
687 EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name);
694 .rtp.extensions.empty());
696 // Verify that existing RTP header extensions can be removed.
701 EXPECT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
703 // Verify that adding receive RTP header extensions adds them for existing
707 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
708 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id)
[all...] |
| 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
|
| /external/srtp/test/ |
| rtpw.c | 4 * rtp word sender/receiver
9 * This app is a simple RTP application intended only for testing
12 * each USEC_RATE microseconds. Secure RTP protections can be
79 #include "rtp.h"
119 * program_type distinguishes the [s]rtp sender and receiver cases
314 crypto_policy_set_rtp_default(&policy.rtp);
318 crypto_policy_set_aes_cm_128_null_auth(&policy.rtp);
322 crypto_policy_set_null_cipher_hmac_sha1_80(&policy.rtp);
335 policy.rtp.sec_serv = sec_servs;
370 * application is now a vanilla-flavored RTP application
[all...] |
| /frameworks/opt/net/voip/src/java/android/net/rtp/ |
| 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.
|
| /frameworks/av/media/libstagefright/wifi-display/ |
| Android.mk | 8 rtp/RTPSender.cpp \
|
