| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.h | 31 rtp_receiver_(NULL),
40 rtp_receiver_ = receiver;
64 if (!rtp_receiver_->IncomingRtpPacket(header,
82 RtpReceiver* rtp_receiver_; member in class:webrtc::LoopBackTransport
|
| test_api_video.cc | 50 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
55 rtp_receiver_->SetNACKStatus(kNackRtcp);
60 rtp_receiver_.get(), receive_statistics_.get());
68 EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
132 scoped_ptr<RtpReceiver> rtp_receiver_; member in class:webrtc::RtpRtcpVideoTest
163 EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(codec.plName,
181 EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, payload,
|
| test_api.cc | 38 rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
48 scoped_ptr<RtpReceiver> rtp_receiver_; member in class:RtpRtcpAPITest
112 EXPECT_EQ(kNackOff, rtp_receiver_->NACK());
113 rtp_receiver_->SetNACKStatus(kNackRtcp);
114 EXPECT_EQ(kNackRtcp, rtp_receiver_->NACK());
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| nack_rtx_unittest.cc | 77 rtp_receiver_(NULL),
85 rtp_receiver_ = receiver;
125 &restored_packet_ptr, ptr, &packet_length, rtp_receiver_->SSRC(),
138 if (!rtp_receiver_->IncomingRtpPacket(header, restored_packet_ptr,
159 RtpReceiver* rtp_receiver_; member in class:RtxLoopBackTransport
187 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
193 rtp_receiver_->SetNACKStatus(kNackRtcp);
200 rtp_receiver_.get());
208 EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(video_codec.plName,
286 scoped_ptr<RtpReceiver> rtp_receiver_; member in class:RtpRtcpRtxNackTest
[all...] |
| rtcp_sender_unittest.cc | 300 rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
326 scoped_ptr<RtpReceiver> rtp_receiver_; member in class:webrtc::RtcpSenderTest
366 EXPECT_EQ(0, rtp_receiver_->RegisterReceivePayload(codec_inst.plName,
380 EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, packet_, packet_length,
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_receiver.cc | 41 rtp_receiver_(RtpReceiver::CreateVideoReceiver(
80 return rtp_receiver_->RegisterReceivePayload(video_codec.plName,
96 rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
108 return rtp_receiver_->SSRC();
112 return rtp_receiver_->CSRCs(csrcs);
120 return rtp_receiver_.get();
268 return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
302 &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
340 ntp_estimator_->UpdateRtcpTimestamp(rtp_receiver_->SSRC(), rtp_rtcp_);
416 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL)
[all...] |
| vie_receiver.h | 110 scoped_ptr<RtpReceiver> rtp_receiver_; member in class:webrtc::ViEReceiver
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| media_opt_test.h | 56 webrtc::RtpReceiver* rtp_receiver_; member in class:MediaOptTest
|
| mt_test_common.cc | 105 if (!rtp_receiver_->IncomingRtpPacket(header, packet->data,
|
| test_callbacks.cc | 213 rtp_receiver_(NULL),
309 if (!rtp_receiver_->IncomingRtpPacket(header, packet->data,
|
| media_opt_test.cc | 217 rtp_receiver_->RegisterReceivePayload(video_codec.plName,
226 rtp_receiver_->RegisterReceivePayload(video_codec.plName,
235 rtp_receiver_->RegisterReceivePayload(video_codec.plName,
|
| test_callbacks.h | 194 RtpReceiver* rtp_receiver_; member in class:webrtc::RTPSendCompleteCallback
|
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| channel.cc | 549 _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time,
853 rtp_receiver_(RtpReceiver::CreateAudioReceiver(
856 telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
[all...] |
| channel.h | 522 scoped_ptr<RtpReceiver> rtp_receiver_; member in class:webrtc::voe::Channel
|
