| /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/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/video/ |
| video_receive_stream.cc | 53 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
55 switch (config_.rtp.rtcp_mode) {
64 assert(config_.rtp.remote_ssrc != 0);
66 assert(config_.rtp.local_ssrc != 0);
67 assert(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
69 rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
71 Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
72 if (it != config_.rtp.rtx.end())
[all...] |
| bitrate_estimator_tests.cc | 71 send_config_.rtp.ssrcs.push_back(kSendSsrc);
84 receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
85 receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
86 receive_config_.rtp.extensions.push_back(
88 receive_config_.rtp.extensions.push_back(
172 test_->send_config_.rtp.ssrcs[0]++;
189 test_->receive_config_.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
190 test_->receive_config_.rtp.local_ssrc++
[all...] |
| video_send_stream.cc | 41 std::string VideoSendStream::Config::Rtp::Rtx::ToString()
57 std::string VideoSendStream::Config::Rtp::ToString() const {
94 ss << ", rtp: " << rtp.ToString();
134 assert(config_.rtp.ssrcs.size() > 0);
136 assert(config_.rtp.min_transmit_bitrate_bps >= 0);
138 config_.rtp.min_transmit_bitrate_bps / 1000);
140 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
141 const std::string& extension = config_.rtp.extensions[i].name;
142 int id = config_.rtp.extensions[i].id
[all...] |
| send_statistics_proxy_unittest.cc | 38 config.rtp.ssrcs.push_back(17);
39 config.rtp.ssrcs.push_back(42);
102 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
103 it != config_.rtp.ssrcs.end();
151 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
152 it != config_.rtp.ssrcs.end();
170 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
171 it != config_.rtp.ssrcs.end();
192 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
193 it != config_.rtp.ssrcs.end()
[all...] |
| call.cc | 33 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
35 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
185 assert(config.rtp.ssrcs.size() > 0);
201 for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
202 assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
203 send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
232 config.rtp.remb = true;
246 assert(receive_ssrcs_.find(config.rtp.remote_ssrc) == receive_ssrcs_.end());
247 receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
249 VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it
[all...] |
| send_statistics_proxy.cc | 63 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) ==
64 config_.rtp.ssrcs.end())
|
| loopback.cc | 72 send_config.rtp.ssrcs.push_back(kSendSsrc);
103 receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
104 receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
|
| /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 | 59 // Verifies rtp header and message length.
60 bool ValidateRtpHeader(const char* rtp, int length, size_t* header_length) {
64 int cc_count = rtp[0] & 0x0F;
72 if (!(rtp[0] & 0x10)) {
79 rtp += rtp_hdr_len_without_extn;
83 uint16 extn_length = talk_base::GetBE16(rtp + 2) * 4;
97 // Absolute send time in RTP streams.
100 // general mechanism for RTP header extensions [RFC5285]. The payload
130 // the RTP packet.
131 void UpdateRtpAuthTag(char* rtp, int len
[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/media/cast/transport/ |
| cast_transport_sender_impl.cc | 84 LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
85 config.rtp.config.aes_iv_mask.empty())
87 if (!audio_encryptor_.Initialize(config.rtp.config.aes_key,
88 config.rtp.config.aes_iv_mask)) {
94 pacer_.RegisterAudioSsrc(config.rtp.config.ssrc);
104 LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
105 config.rtp.config.aes_iv_mask.empty())
107 if (!video_encryptor_.Initialize(config.rtp.config.aes_key,
108 config.rtp.config.aes_iv_mask)) {
114 pacer_.RegisterVideoSsrc(config.rtp.config.ssrc)
[all...] |
| /external/chromium_org/media/cast/transport/rtp_sender/ |
| rtp_sender.cc | 45 storage_.reset(new PacketStorage(config.rtp.max_outstanding_frames));
50 config_.ssrc = config.rtp.config.ssrc;
51 config_.payload_type = config.rtp.config.payload_type;
59 storage_.reset(new PacketStorage(config.rtp.max_outstanding_frames));
64 config_.ssrc = config.rtp.config.ssrc;
65 config_.payload_type = config.rtp.config.payload_type;
|
| /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/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...] |
| /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/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
|
| webrtcvideoengine2_unittest.cc | 639 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
640 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
641 EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name);
648 .rtp.extensions.empty());
654 EXPECT_FALSE(send_stream->GetConfig().rtp.extensions.empty());
669 ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
670 EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
671 EXPECT_EQ(webrtc_ext, recv_stream->GetConfig().rtp.extensions[0].name);
678 .rtp.extensions.empty());
684 EXPECT_FALSE(recv_stream->GetConfig().rtp.extensions.empty())
[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.
|
