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  /external/chromium_org/third_party/webrtc/
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
video_send_stream.h 74 struct Rtp {
75 Rtp()
82 // Max RTP packet size delivered to send transport from VideoEngine.
90 // RTP header extensions to use for this send stream.
99 // Settings for RTP retransmission payload format, see RFC 4588 for
117 } rtp; member in struct:webrtc::VideoSendStream::Config
  /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/
after_initialization_fixture.h 39 StorePacket(Packet::Rtp, channel, data, len);
50 enum Type { Rtp, Rtcp, } type;
97 case Packet::Rtp:
  /external/chromium_org/third_party/webrtc/video/
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...]
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...]
call.cc 36 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
38 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
199 assert(config.rtp.ssrcs.size() > 0);
218 for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
219 assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
220 send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
271 assert(receive_ssrcs_.find(config.rtp.remote_ssrc) == receive_ssrcs_.end());
272 receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
274 VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
275 config.rtp.rtx.begin()
    [all...]

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