/external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
D | simulcast_encoder_adapter.h | 72 send_stream(true) {} in StreamInfo() 77 bool send_stream) in StreamInfo() 83 send_stream(send_stream) {} in StreamInfo() 90 bool send_stream; member 100 bool* send_stream) const; 108 bool* send_stream);
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D | simulcast_encoder_adapter.cc | 201 bool send_stream = true; in InitEncode() local 209 &send_stream); in InitEncode() 226 stream_codec.height, send_stream)); in InitEncode() 255 streaminfos_[stream_idx].send_stream) { in Encode() 265 if (!streaminfos_[stream_idx].send_stream) in Encode() 345 bool send_stream = true; in SetRates() local 349 new_bitrate_kbit, &send_stream); in SetRates() 351 if (send_stream && !streaminfos_[stream_idx].send_stream) { in SetRates() 354 streaminfos_[stream_idx].send_stream = send_stream; in SetRates() 394 bool* send_stream) const { in GetStreamBitrate() [all …]
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D | vp8_impl.h | 90 void SetStreamState(bool send_stream, int stream_idx);
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D | vp8_impl.cc | 305 void VP8EncoderImpl::SetStreamState(bool send_stream, in SetStreamState() argument 307 if (send_stream && !send_stream_[stream_idx]) { in SetStreamState() 311 send_stream_[stream_idx] = send_stream; in SetStreamState()
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/external/webrtc/webrtc/call/ |
D | call.cc | 63 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; 73 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; 303 AudioSendStream* send_stream = new AudioSendStream( in CreateAudioSendStream() local 306 send_stream->SignalNetworkState(kNetworkDown); in CreateAudioSendStream() 311 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; in CreateAudioSendStream() 313 return send_stream; in CreateAudioSendStream() 316 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { in DestroyAudioSendStream() argument 319 RTC_DCHECK(send_stream != nullptr); in DestroyAudioSendStream() 321 send_stream->Stop(); in DestroyAudioSendStream() 324 static_cast<webrtc::internal::AudioSendStream*>(send_stream); in DestroyAudioSendStream() [all …]
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D | call_perf_tests.cc | 634 VideoSendStream* send_stream, in TestMinTransmitBitrate() argument 636 send_stream_ = send_stream; in TestMinTransmitBitrate() 732 VideoSendStream* send_stream, in TEST_F() argument 734 send_stream_ = send_stream; in TEST_F()
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D | rampup_tests.h | 76 VideoSendStream* send_stream,
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D | rampup_tests.cc | 79 VideoSendStream* send_stream, in OnVideoStreamsCreated() argument 81 send_stream_ = send_stream; in OnVideoStreamsCreated()
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/external/tensorflow/tensorflow/core/common_runtime/gpu/ |
D | gpu_util.cc | 121 se::Stream* send_stream = nullptr; in SetProtoFromGPU() local 123 &send_stream); in SetProtoFromGPU() 137 send_device_to_host_stream->ThenWaitFor(send_stream); in SetProtoFromGPU() 194 se::Stream* send_stream = nullptr; in DeviceToDeviceCopy() local 196 &send_stream); in DeviceToDeviceCopy() 210 send_device_to_device_stream->ThenWaitFor(send_stream); in DeviceToDeviceCopy() 261 se::Stream* send_stream = nullptr; in CopyGPUTensorToCPU() local 263 &dev_info, &send_stream); in CopyGPUTensorToCPU() 277 send_device_to_host_stream->ThenWaitFor(send_stream); in CopyGPUTensorToCPU() 429 se::Stream* send_stream = nullptr; in CopyGPUTensorToSameGPU() local [all …]
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D | gpu_stream_util.cc | 42 if ((opts.max_streams < 1) || (opts.send_stream >= opts.max_streams) || in AssignStreams() 93 if (opts.send_stream >= 0) stream_id = opts.send_stream; in AssignStreams()
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D | gpu_stream_util.h | 31 int32 send_stream = -1; member
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D | gpu_stream_util_test.cc | 122 opts.send_stream = 91; in TEST_F()
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/external/webrtc/webrtc/audio/ |
D | audio_send_stream_unittest.cc | 187 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST() local 193 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST() local 196 EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, in TEST() 202 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST() local 205 AudioSendStream::Stats stats = send_stream.GetStats(); in TEST() 230 internal::AudioSendStream send_stream(helper.config(), helper.audio_state(), in TEST() local 233 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); in TEST() 240 EXPECT_TRUE(send_stream.GetStats().typing_noise_detected); in TEST() 242 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); in TEST()
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvideoengine2_unittest.cc | 972 FakeVideoSendStream* send_stream = in TestSetSendRtpHeaderExtensions() local 976 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); in TestSetSendRtpHeaderExtensions() 977 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id); in TestSetSendRtpHeaderExtensions() 978 EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name); in TestSetSendRtpHeaderExtensions() 990 send_stream = fake_call_->GetVideoSendStreams()[0]; in TestSetSendRtpHeaderExtensions() 991 EXPECT_TRUE(send_stream->GetConfig().rtp.extensions.empty()); in TestSetSendRtpHeaderExtensions() 996 send_stream = fake_call_->GetVideoSendStreams()[0]; in TestSetSendRtpHeaderExtensions() 997 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); in TestSetSendRtpHeaderExtensions() 998 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id); in TestSetSendRtpHeaderExtensions() 999 EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name); in TestSetSendRtpHeaderExtensions() [all …]
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D | fakewebrtccall.cc | 308 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { in DestroyAudioSendStream() argument 311 static_cast<FakeAudioSendStream*>(send_stream)); in DestroyAudioSendStream() 350 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { in DestroyVideoSendStream() argument 353 static_cast<FakeVideoSendStream*>(send_stream)); in DestroyVideoSendStream()
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D | fakewebrtccall.h | 224 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; 234 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
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D | webrtcvoiceengine_unittest.cc | 123 const auto* send_stream = call_.GetAudioSendStream(ssrc); in GetSendStream() local 124 EXPECT_TRUE(send_stream); in GetSendStream() 125 return *send_stream; in GetSendStream() 129 const auto* send_stream = call_.GetAudioSendStream(ssrc); in GetSendStreamConfig() local 130 EXPECT_TRUE(send_stream); in GetSendStreamConfig() 131 return send_stream->GetConfig(); in GetSendStreamConfig()
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/external/webrtc/webrtc/ |
D | call.h | 103 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; 113 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
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/external/webrtc/webrtc/test/ |
D | call_test.h | 163 VideoSendStream* send_stream, 170 AudioSendStream* send_stream,
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D | call_test.cc | 414 VideoSendStream* send_stream, in OnVideoStreamsCreated() argument 422 AudioSendStream* send_stream, in OnAudioStreamsCreated() argument
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/external/webrtc/webrtc/video/ |
D | video_send_stream_tests.cc | 832 VideoSendStream* send_stream, in TEST_F() argument 834 stream_ = send_stream; in TEST_F() 1024 VideoSendStream* send_stream, in TEST_F() argument 1026 stream_ = send_stream; in TEST_F() 1325 VideoSendStream* send_stream, in TEST_F() argument 1330 stream_ = send_stream; in TEST_F() 1389 VideoSendStream* send_stream, in TEST_F() argument 1391 stream_ = send_stream; in TEST_F() 1459 VideoSendStream* send_stream, in OnVideoStreamsCreated() argument 1461 stream_ = send_stream; in OnVideoStreamsCreated() [all …]
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D | end_to_end_tests.cc | 1891 VideoSendStream* send_stream, in TEST_F() argument 1893 send_stream_ = send_stream; in TEST_F() 2296 VideoSendStream* send_stream, in TestSendsSetSsrcs() argument 2298 send_stream_ = send_stream; in TestSendsSetSsrcs() 2342 VideoSendStream* send_stream, in TEST_F() argument 2344 send_stream_ = send_stream; in TEST_F() 2601 VideoSendStream* send_stream, in TEST_F() argument 2603 send_stream_ = send_stream; in TEST_F() 2684 VideoSendStream* send_stream, in TEST_F() argument
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/external/grpc-grpc/test/cpp/microbenchmarks/ |
D | bm_chttp2_transport.cc | 408 grpc_core::ManualConstructor<grpc_core::SliceBufferByteStream> send_stream; in BM_TransportStreamSend() local 431 send_stream.Init(&send_buffer, 0); in BM_TransportStreamSend() 439 op.payload->send_message.send_message.reset(send_stream.get()); in BM_TransportStreamSend()
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