| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_unittest.cc | 337 RTPSender* rtp_sender() { in rtp_sender() function in webrtc::RtpSenderTest
375 rtp_sender()->SetSequenceNumber(kSeqNum); in SetUpRtpSender()
376 rtp_sender()->SetTimestampOffset(0); in SetUpRtpSender()
399 auto packet = rtp_sender()->AllocatePacket(); in BuildRtpPacket()
405 EXPECT_TRUE(rtp_sender()->AssignSequenceNumber(packet.get())); in BuildRtpPacket()
418 EXPECT_TRUE(rtp_sender()->SendToNetwork( in SendPacket()
431 rtp_sender()->GeneratePadding(target_size_bytes, true)) { in GenerateAndSendPadding()
448 rtp_sender()->SetRtxPayloadType(kRtxPayload, kPayload); in EnableRtx()
449 rtp_sender()->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads); in EnableRtx()
455 rtp_sender()->RegisterRtpHeaderExtension(kRtpExtensionMid, kMidExtensionId); in EnableMidSending()
[all …]
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| D | rtp_sender_audio.cc | 54 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender) in RTPSenderAudio() argument
56 rtp_sender_(rtp_sender), in RTPSenderAudio()
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| D | rtp_rtcp_impl.h | 289 RTPSender* rtp_sender() { in rtp_sender() function
292 const RTPSender* rtp_sender() const { in rtp_sender() function
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| D | rtp_sender_audio.h | 34 RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
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| D | rtp_sender_video_unittest.cc | 125 RTPSender* rtp_sender, in TestRtpSenderVideo() argument
131 config.rtp_sender = rtp_sender; in TestRtpSenderVideo()
721 config.rtp_sender = rtp_module_->RtpSender(); in TEST_P()
924 config.rtp_sender = rtp_module_->RtpSender(); in CreateSenderWithFrameTransformer()
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| D | rtp_sender_video.h | 74 RTPSender* rtp_sender = nullptr; member
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| D | nack_rtx_unittest.cc | 143 video_config.rtp_sender = rtp_rtcp_module_->RtpSender(); in SetUp()
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| D | rtp_sender_video.cc | 119 : rtp_sender_(config.rtp_sender), in RTPSenderVideo()
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| D | rtp_rtcp_impl_unittest.cc | 184 video_config.rtp_sender = sender_.impl_->RtpSender(); in SetUp()
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| D | rtp_rtcp_impl2_unittest.cc | 189 video_config.rtp_sender = sender_.impl_->RtpSender(); in SetUp()
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| /external/webrtc/pc/scenario_tests/ |
| D | goog_cc_test.cc | 64 for (auto& rtp_sender : caller->pc()->GetSenders()) { in TEST() local
65 auto parameters = rtp_sender->GetParameters(); in TEST()
74 rtp_sender->SetParameters(parameters); in TEST()
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| /external/webrtc/pc/ |
| D | track_media_info_map.cc | 48 for (const auto& rtp_sender : rtp_senders) { in GetAudioAndVideoTrackBySsrc() local
49 cricket::MediaType media_type = rtp_sender->media_type(); in GetAudioAndVideoTrackBySsrc()
50 MediaStreamTrackInterface* track = rtp_sender->track(); in GetAudioAndVideoTrackBySsrc()
55 uint32_t ssrc = rtp_sender->ssrc(); in GetAudioAndVideoTrackBySsrc()
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| D | track_media_info_map_unittest.cc | 113 rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockRtpSender( in AddRtpSenderWithSsrcs() local
118 rtp_senders_.push_back(rtp_sender); in AddRtpSenderWithSsrcs()
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| D | BUILD.gn | 202 "rtp_sender.cc",
203 "rtp_sender.h",
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| D | rtc_stats_collector_unittest.cc | 496 rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender( in CreateMockRtpSendersReceiversAndChannels() local
501 pc_->AddSender(rtp_sender); in CreateMockRtpSendersReceiversAndChannels()
531 rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender( in CreateMockRtpSendersReceiversAndChannels() local
536 pc_->AddSender(rtp_sender); in CreateMockRtpSendersReceiversAndChannels()
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| /external/webrtc/call/ |
| D | BUILD.gn | 132 rtc_library("rtp_sender") {
250 ":rtp_sender",
398 ":rtp_sender",
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| D | rtp_video_sender.cc | 268 video_config.rtp_sender = rtp_rtcp->RtpSender(); in CreateRtpStreamSenders()
924 RTPSender* rtp_sender = it->second->RtpSender(); in OnPacketFeedbackVector() local
926 rtp_sender->ReSendPacket(sequence_number); in OnPacketFeedbackVector()
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| D | rtp_transport_controller_send.cc | 379 for (auto& rtp_sender : video_rtp_senders_) { in OnNetworkAvailability() local
380 rtp_sender->OnNetworkAvailability(network_available); in OnNetworkAvailability()
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| /external/webrtc/test/scenario/ |
| D | BUILD.gn | 106 "../../call:rtp_sender",
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| /external/webrtc/modules/rtp_rtcp/ |
| D | BUILD.gn | 198 "source/rtp_sender.cc",
199 "source/rtp_sender.h",
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| /external/webrtc/video/ |
| D | BUILD.gn | 98 "../call:rtp_sender",
608 "../call:rtp_sender",
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| /external/webrtc/audio/ |
| D | BUILD.gn | 159 "../call:rtp_sender",
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| /external/webrtc/rtc_tools/ |
| D | BUILD.gn | 178 "../call:rtp_sender",
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| /external/webrtc/sdk/android/ |
| D | BUILD.gn | 702 "src/jni/pc/rtp_sender.cc",
703 "src/jni/pc/rtp_sender.h",
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| /external/webrtc/ |
| D | Android.bp | 3931 "modules/rtp_rtcp/source/rtp_sender.cc",
4969 "pc/rtp_sender.cc",
|