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  /external/protobuf/src/google/protobuf/util/internal/
type_info.cc 71 google::protobuf::scoped_ptr<google::protobuf::Type> type(new google::protobuf::Type());
96 google::protobuf::scoped_ptr<google::protobuf::Enum> enum_type(
  /external/v8/testing/gtest/include/gtest/internal/
gtest-port.h 221 // scoped_ptr - as in TR2.
1114 class scoped_ptr { class in namespace:testing::internal
1118 explicit scoped_ptr(T* p = NULL) : ptr_(p) {} function in class:testing::internal::scoped_ptr
    [all...]
  /external/webrtc/talk/app/webrtc/
peerconnectionfactory.cc 252 rtc::scoped_ptr<cricket::PortAllocator> allocator,
253 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
peerconnectioninterface.h 531 rtc::scoped_ptr<cricket::PortAllocator> allocator,
532 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
  /external/webrtc/talk/media/devices/
gdivideorenderer.cc 36 #include "webrtc/base/scoped_ptr.h"
103 rtc::scoped_ptr<uint8_t[]> image_;
104 rtc::scoped_ptr<WindowThread> window_thread_;
  /external/webrtc/talk/media/webrtc/
fakewebrtccall.h 109 void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
114 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
  /external/webrtc/webrtc/base/
buffer.h 20 #include "webrtc/base/scoped_ptr.h"
167 scoped_ptr<uint8_t[]> new_data(new uint8_t[capacity]);
225 scoped_ptr<uint8_t[]> data_;
criticalsection_unittest.cc 17 #include "webrtc/base/scoped_ptr.h"
226 scoped_ptr<Foo> a(new Foo());
227 scoped_ptr<Foo> b(new Foo());
  /external/webrtc/webrtc/call/
bitrate_allocator_unittest.cc 44 rtc::scoped_ptr<BitrateAllocator> allocator_;
111 rtc::scoped_ptr<BitrateAllocator> allocator_;
congestion_controller.cc 136 rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
137 rtc::scoped_ptr<RemoteBitrateEstimator> rbe_;
  /external/webrtc/webrtc/common_audio/
audio_converter.cc 139 rtc::scoped_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
143 rtc::scoped_ptr<AudioConverter> sp;
audio_converter_unittest.cc 18 #include "webrtc/base/scoped_ptr.h"
25 typedef rtc::scoped_ptr<ChannelBuffer<float>> ScopedBuffer;
135 rtc::scoped_ptr<AudioConverter> converter = AudioConverter::Create(
  /external/webrtc/webrtc/libjingle/xmpp/
chatroommodule_unittest.cc 164 rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
168 rtc::scoped_ptr<XmppChatroomModule> chatroom(XmppChatroomModule::Create());
pubsubstateclient.h 23 #include "webrtc/base/scoped_ptr.h"
259 rtc::scoped_ptr<PubSubStateKeySerializer> key_serializer_;
260 rtc::scoped_ptr<PubSubStateSerializer<C> > state_serializer_;
pubsubtasks_unittest.cc 83 rtc::scoped_ptr<rtc::FakeTaskRunner> runner;
86 rtc::scoped_ptr<TestPubSubTasksListener> listener;
xmppengine_unittest.cc 80 rtc::scoped_ptr<XmppEngine> engine_;
81 rtc::scoped_ptr<XmppTestHandler> handler_;
  /external/webrtc/webrtc/modules/audio_coding/codecs/cng/
audio_encoder_cng.cc 22 rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
26 rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst;
  /external/webrtc/webrtc/modules/audio_coding/test/
delay_test.cc 18 #include "webrtc/base/scoped_ptr.h"
226 rtc::scoped_ptr<AudioCodingModule> acm_a_;
227 rtc::scoped_ptr<AudioCodingModule> acm_b_;
insert_packet_with_timing.cc 15 #include "webrtc/base/scoped_ptr.h"
244 rtc::scoped_ptr<AudioCodingModule> send_acm_;
245 rtc::scoped_ptr<AudioCodingModule> receive_acm_;
  /external/webrtc/webrtc/modules/audio_device/android/
audio_device_unittest.cc 23 #include "webrtc/base/scoped_ptr.h"
147 rtc::scoped_ptr<int16_t[]> file_;
242 rtc::scoped_ptr<AudioBufferList> fifo_;
494 rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
691 rtc::scoped_ptr<EventWrapper> test_is_done_;
695 rtc::scoped_ptr<BuildInfo> build_info_;
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  /external/webrtc/webrtc/modules/audio_device/dummy/
file_audio_device.h 185 rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
186 rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
  /external/webrtc/webrtc/modules/audio_device/linux/
audio_device_alsa_linux.h 190 rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
191 rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
  /external/webrtc/webrtc/modules/desktop_capture/win/
cursor.cc 15 #include "webrtc/base/scoped_ptr.h"
138 rtc::scoped_ptr<uint32_t[]> mask_data(new uint32_t[width * height]);
165 rtc::scoped_ptr<DesktopFrame> image(
screen_capturer_win_gdi.cc 162 rtc::scoped_ptr<Desktop> input_desktop(Desktop::GetInputDesktop());
244 rtc::scoped_ptr<DesktopFrame> buffer(
  /external/webrtc/webrtc/modules/rtp_rtcp/source/
fec_receiver_impl.cc 16 #include "webrtc/base/scoped_ptr.h"
92 rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
140 rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket>

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