Searched refs:recv_stream (Results 1 – 9 of 9) sorted by relevance
/external/webrtc/webrtc/audio/ |
D | audio_receive_stream_unittest.cc | 222 internal::AudioReceiveStream recv_stream( in TEST() local 241 internal::AudioReceiveStream recv_stream( in TEST() local 257 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); in TEST() 267 internal::AudioReceiveStream recv_stream( in TEST() local 283 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); in TEST() 288 internal::AudioReceiveStream recv_stream( in TEST() local 291 AudioReceiveStream::Stats stats = recv_stream.GetStats(); in TEST()
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/external/tensorflow/tensorflow/core/common_runtime/gpu/ |
D | gpu_stream_util.cc | 43 (opts.recv_stream >= opts.max_streams) || in AssignStreams() 95 if (opts.recv_stream >= 0) stream_id = opts.recv_stream; in AssignStreams()
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D | gpu_util.cc | 218 auto recv_stream = in DeviceToDeviceCopy() local 220 if (recv_stream == nullptr) { in DeviceToDeviceCopy() 229 send_device_to_device_stream->ThenWaitFor(recv_stream); in DeviceToDeviceCopy() 306 se::Stream* recv_stream = nullptr; in CopyCPUTensorToGPU() local 308 &dev_info, &recv_stream); in CopyCPUTensorToGPU() 322 recv_host_to_device_stream->ThenWaitFor(recv_stream); in CopyCPUTensorToGPU()
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D | gpu_stream_util.h | 32 int32 recv_stream = -1; member
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D | gpu_stream_util_test.cc | 123 opts.recv_stream = 92; in TEST_F()
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/external/grpc-grpc/test/cpp/microbenchmarks/ |
D | bm_chttp2_transport.cc | 532 grpc_core::OrphanablePtr<grpc_core::ByteStream> recv_stream; in BM_TransportStreamRecv() local 571 op.payload->recv_message.recv_message = &recv_stream; in BM_TransportStreamRecv() 578 if (recv_stream == nullptr) { in BM_TransportStreamRecv() 587 if (received == recv_stream->length()) { in BM_TransportStreamRecv() 588 recv_stream.reset(); in BM_TransportStreamRecv() 592 } while (recv_stream->Next(recv_stream->length() - received, in BM_TransportStreamRecv() 594 GRPC_ERROR_NONE == recv_stream->Pull(&recv_slice) && in BM_TransportStreamRecv() 600 recv_stream->Pull(&recv_slice); in BM_TransportStreamRecv()
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvideoengine2_unittest.cc | 1011 FakeVideoReceiveStream* recv_stream = in TestSetRecvRtpHeaderExtensions() local 1015 ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size()); in TestSetRecvRtpHeaderExtensions() 1016 EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id); in TestSetRecvRtpHeaderExtensions() 1017 EXPECT_EQ(webrtc_ext, recv_stream->GetConfig().rtp.extensions[0].name); in TestSetRecvRtpHeaderExtensions() 1030 recv_stream = fake_call_->GetVideoReceiveStreams()[0]; in TestSetRecvRtpHeaderExtensions() 1031 EXPECT_TRUE(recv_stream->GetConfig().rtp.extensions.empty()); in TestSetRecvRtpHeaderExtensions() 1036 recv_stream = fake_call_->GetVideoReceiveStreams()[0]; in TestSetRecvRtpHeaderExtensions() 1037 ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size()); in TestSetRecvRtpHeaderExtensions() 1038 EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id); in TestSetRecvRtpHeaderExtensions() 1039 EXPECT_EQ(webrtc_ext, recv_stream->GetConfig().rtp.extensions[0].name); in TestSetRecvRtpHeaderExtensions() [all …]
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D | webrtcvoiceengine_unittest.cc | 135 const auto* recv_stream = call_.GetAudioReceiveStream(ssrc); in GetRecvStreamConfig() local 136 EXPECT_TRUE(recv_stream); in GetRecvStreamConfig() 137 return recv_stream->GetConfig(); in GetRecvStreamConfig()
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/external/grpc-grpc/src/core/ext/transport/inproc/ |
D | inproc_transport.cc | 103 grpc_core::ManualConstructor<grpc_core::SliceBufferByteStream> recv_stream; member 556 receiver->recv_stream.Init(&receiver->recv_message, 0); in message_transfer_locked() 558 receiver->recv_stream.get()); in message_transfer_locked()
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