| /external/webrtc/audio/voip/ |
| D | audio_channel.cc | 89 if (!rtp_rtcp_->Sending() && rtp_rtcp_->SetSendingStatus(true) != 0) { in StartSend()
101 rtp_rtcp_->SetSendingStatus(false) != 0) { in StopSend()
111 if (!rtp_rtcp_->Sending() && rtp_rtcp_->SetSendingStatus(true) != 0) { in StartPlay()
121 rtp_rtcp_->SetSendingStatus(false) != 0) { in StopPlay()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtcp_sender_unittest.cc | 129 TEST_F(RtcpSenderTest, SetSendingStatus) { in TEST_F() argument
131 EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true)); in TEST_F()
145 rtcp_sender_->SetSendingStatus(feedback_state, true); in TEST_F()
169 rtcp_sender_->SetSendingStatus(feedback_state, true); in TEST_F()
203 rtcp_sender_->SetSendingStatus(feedback_state(), true); in TEST_F()
225 rtcp_sender_->SetSendingStatus(feedback_state(), true); in TEST_F()
312 EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true)); in TEST_F()
313 EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false)); in TEST_F()
496 EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false)); in TEST_F()
509 EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), true)); in TEST_F()
[all …]
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| D | rtp_rtcp_interface.h | 252 virtual int32_t SetSendingStatus(bool sending) = 0;
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| D | rtp_rtcp_impl2.h | 130 int32_t SetSendingStatus(bool sending) override;
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| D | rtp_rtcp_impl2.cc | 274 int32_t ModuleRtpRtcpImpl2::SetSendingStatus(const bool sending) { in SetSendingStatus() function in webrtc::ModuleRtpRtcpImpl2
277 if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { in SetSendingStatus()
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| D | rtp_rtcp_impl.h | 121 int32_t SetSendingStatus(bool sending) override;
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| D | nack_rtx_unittest.cc | 148 EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true)); in SetUp()
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| D | rtp_rtcp_impl.cc | 326 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { in SetSendingStatus() function in webrtc::ModuleRtpRtcpImpl
329 if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { in SetSendingStatus()
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| D | rtp_rtcp_impl_unittest.cc | 176 EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true)); in SetUp()
194 EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false)); in SetUp()
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| D | rtp_rtcp_impl2_unittest.cc | 181 EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true)); in SetUp()
199 EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false)); in SetUp()
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| D | rtcp_sender.h | 74 int32_t SetSendingStatus(const FeedbackState& feedback_state,
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| D | rtcp_sender.cc | 222 int32_t RTCPSender::SetSendingStatus(const FeedbackState& feedback_state, in SetSendingStatus() function in webrtc::RTCPSender
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| /external/webrtc/audio/voip/test/ |
| D | audio_ingress_unittest.cc | 73 rtp_rtcp_->SetSendingStatus(true); in SetUp()
77 rtp_rtcp_->SetSendingStatus(false); in TearDown()
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| D | audio_egress_unittest.cc | 75 rtp_rtcp_->SetSendingStatus(true); in SetUp()
81 rtp_rtcp_->SetSendingStatus(false); in TearDown()
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| /external/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 79 MOCK_METHOD(int32_t, SetSendingStatus, (bool sending), (override));
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| /external/webrtc/audio/ |
| D | channel_send.cc | 536 int ret = rtp_rtcp_->SetSendingStatus(true); in StartSend()
562 if (rtp_rtcp_->SetSendingStatus(false) == -1) { in StopSend()
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| /external/webrtc/call/ |
| D | rtp_video_sender.cc | 261 rtp_rtcp->SetSendingStatus(false); in CreateRtpStreamSenders()
491 rtp_streams_[i].rtp_rtcp->SetSendingStatus(active_modules[i]); in SetActiveModulesLocked()
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