| /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
|
| 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
|
| 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/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
| after_initialization_fixture.h | 45 StorePacket(Packet::Rtp, data, len);
71 enum Type { Rtp, Rtcp, } type;
119 if (p.type == Packet::Rtp) {
128 // Minimum RTP header size.
133 case Packet::Rtp:
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| conference_transport.h | 108 enum Type { Rtp, Rtcp, } type_;
|
| conference_transport.cc | 112 StorePacket(Packet::Rtp, data, len);
141 // This simulates the flow of RTP and RTCP packets. Complications like that
149 case Packet::Rtp: {
|
| /external/webrtc/webrtc/video/ |
| 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...] |
| /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.cc | 31 std::string AudioSendStream::Config::Rtp::ToString() const {
49 ss << "{rtp: " << rtp.ToString();
76 channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
77 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
79 for (const auto& extension : config.rtp.extensions) {
87 RTC_NOTREACHED() << "Registering unsupported RTP extension.";
128 stats.local_ssrc = config_.rtp.ssrc;
|
| /external/webrtc/webrtc/call/ |
| 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...] |
| call.cc | 309 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
311 audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
328 audio_send_stream->config().rtp.ssrc);
342 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
344 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
360 audio_receive_stream->config().rtp.remote_ssrc);
390 for (uint32_t ssrc : config.rtp.ssrcs) {
445 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
447 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
449 VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it
[all...] |
| /external/spirv-llvm/lib/SPIRV/ |
| SPIRVInternal.h | 339 const static char Rtp[] = "rtp";
[all...] |
| /external/webrtc/talk/media/webrtc/ |
| webrtcvideoengine2.cc | 405 !is_screencast && parameters_.config.rtp.ssrcs.size() == 1;
[all...] |
