| /external/skia/include/private/ |
| GrInstancedPipelineInfo.h | 18 GrInstancedPipelineInfo(const GrRenderTargetProxy* rtp)
19 : fIsMultisampled(rtp->isStencilBufferMultisampled())
20 , fIsMixedSampled(rtp->isMixedSampled())
21 , fIsRenderingToFloat(GrPixelConfigIsFloatingPoint(rtp->desc().fConfig)) {}
|
| /external/webrtc/webrtc/modules/video_coding/ |
| generic_encoder.cc | 25 // Map information from info into rtp. If no relevant information is found
26 // in info, rtp is set to NULL.
27 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
31 rtp->codec = kRtpVideoVp8;
32 rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
33 rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
34 rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference;
35 rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
36 rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
37 rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx
[all...] |
| /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...] |
| send_statistics_proxy_unittest.cc | 40 config.rtp.ssrcs.push_back(17);
41 config.rtp.ssrcs.push_back(42);
42 config.rtp.rtx.ssrcs.push_back(18);
43 config.rtp.rtx.ssrcs.push_back(43);
101 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
102 it != config_.rtp.ssrcs.end();
115 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin();
116 it != config_.rtp.rtx.ssrcs.end();
159 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin();
160 it != config_.rtp.ssrcs.end()
[all...] |
| video_receive_stream.cc | 56 ss << ", rtp: " << rtp.ToString();
71 std::string VideoReceiveStream::Config::Rtp::ToString() const {
157 config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions);
174 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false,
176 RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
179 vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
181 RTC_DCHECK(config_.rtp.remote_ssrc != 0);
183 RTC_DCHECK(config_.rtp.local_ssrc != 0)
[all...] |
| video_send_stream.cc | 49 std::string VideoSendStream::Config::Rtp::Rtx::ToString()
65 std::string VideoSendStream::Config::Rtp::ToString() const {
94 ss << ", rtp: " << rtp.ToString();
133 RTC_DCHECK(!config_.rtp.ssrcs.empty());
137 for (const RtpExtension& extension : config.rtp.extensions) {
145 const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs;
173 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
174 const std::string& extension = config_.rtp.extensions[i].name;
175 int id = config_.rtp.extensions[i].id
[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,
|
| end_to_end_tests.cc | 65 ADD_FAILURE() << "Unexpected RTP sent.";
306 send_config->rtp.nack.rtp_history_ms =
307 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
439 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
440 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
538 // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms;
539 // send_config->rtp.nack.rtp_history_ms = rtp_history_ms;
540 send_config->rtp.fec.red_payload_type = kRedPayloadType;
541 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
543 (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType
[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/webrtc/webrtc/call/ |
| bitrate_estimator_tests.cc | 123 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
133 receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
134 receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
135 receive_config_.rtp.remb = true;
136 receive_config_.rtp.extensions.push_back(
138 receive_config_.rtp.extensions.push_back(
172 test_->video_send_config_.rtp.ssrcs[0]++;
186 receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0]
[all...] |
| rampup_tests.cc | 120 send_config->rtp.extensions.clear();
127 send_config->rtp.extensions.push_back(
132 send_config->rtp.extensions.push_back(RtpExtension(
137 send_config->rtp.extensions.push_back(RtpExtension(
141 send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
142 send_config->rtp.ssrcs = video_ssrcs_;
144 send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
145 send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
148 send_config->rtp.fec.ulpfec_payload_type =
150 send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType
[all...] |
| call_unittest.cc | 47 config.rtp.ssrc = 42;
57 config.rtp.remote_ssrc = 42;
71 config.rtp.ssrc = ssrc;
94 config.rtp.remote_ssrc = ssrc;
|
| rtc_event_log_unittest.cc | 84 << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
131 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
133 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
136 if (config.rtp.rtcp_mode == RtcpMode::kCompound)
143 EXPECT_EQ(config.rtp.remb, receiver_config.remb());
145 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
150 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
152 const VideoReceiveStream::Config::Rtp::Rtx& rtx =
153 config.rtp.rtx.at(rtx_map.payload_type());
160 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size())
[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
262 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/skia/src/gpu/ |
| GrDrawingManager.cpp | 195 GrRenderTargetOpList* GrDrawingManager::newOpList(GrRenderTargetProxy* rtp) {
204 rtp->setLastOpList(fOpLists[0]);
212 GrRenderTargetOpList* opList = new GrRenderTargetOpList(rtp,
296 sk_sp<GrRenderTargetProxy> rtp(sk_ref_sp(sProxy->asRenderTargetProxy()));
304 rtp->isStencilBufferMultisampled()) {
306 sk_sp<GrRenderTarget> rt(sk_ref_sp(rtp->instantiate(fContext->resourceProvider())));
313 fContext, this, std::move(rtp),
319 return sk_sp<GrRenderTargetContext>(new GrRenderTargetContext(fContext, this, std::move(rtp),
|
| GrPathRenderingRenderTargetContext.h | 26 sk_sp<GrRenderTargetProxy> rtp,
30 : INHERITED(ctx, mgr, std::move(rtp), std::move(colorSpace), surfaceProps, at, so) {}
|
| /external/webrtc/webrtc/audio/ |
| audio_receive_stream.cc | 37 if (!config.rtp.transport_cc) {
40 for (const auto& extension : config.rtp.extensions) {
49 std::string AudioReceiveStream::Config::Rtp::ToString() const {
67 ss << "{rtp: " << rtp.ToString();
98 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
99 for (const auto& extension : config.rtp.extensions) {
115 RTC_NOTREACHED() << "Unsupported RTP extension.";
138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
175 // bandwidth estimation. RTP timestamps has different rates for audio an
[all...] |
| audio_send_stream_unittest.cc | 97 stream_config_.rtp.ssrc = kSsrc;
98 stream_config_.rtp.c_name = kCName;
99 stream_config_.rtp.extensions.push_back(
101 stream_config_.rtp.extensions.push_back(
103 stream_config_.rtp.extensions.push_back(RtpExtension(
170 config.rtp.ssrc = kSsrc;
171 config.rtp.extensions.push_back(
173 config.rtp.c_name = kCName;
178 "{rtp: {ssrc: 1234, extensions: [{name: "
179 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}],
[all...] |
| /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_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
|
| /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.
|
| /external/webrtc/webrtc/test/ |
| call_test.cc | 190 video_send_config_.rtp.extensions.push_back(
194 video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
195 video_send_config_.rtp.extensions.push_back(RtpExtension(
202 audio_send_config_.rtp.ssrc = kAudioSendSsrc;
210 RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
212 video_config.rtp.remb = true;
213 video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
214 for (const RtpExtension& extension : video_send_config_.rtp.extensions)
215 video_config.rtp.extensions.push_back(extension);
216 for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i)
[all...] |
