| /external/webrtc/talk/app/webrtc/ |
| dtmfsender_unittest.cc | 234 rtc::scoped_ptr<FakeDtmfObserver> observer_;
235 rtc::scoped_ptr<FakeDtmfProvider> provider_;
|
| /external/webrtc/talk/media/webrtc/ |
| webrtcvideoframe_unittest.cc | 75 rtc::scoped_ptr<uint8_t[]> captured_frame_buffer(
329 rtc::scoped_ptr<rtc::MemoryStream> ms(CreateYuvSample(kWidth, kHeight, 12));
|
| webrtcvideoengine2_unittest.cc | 157 rtc::scoped_ptr<webrtc::Call> call_;
290 rtc::scoped_ptr<VideoMediaChannel> channel(
320 rtc::scoped_ptr<VideoMediaChannel> channel(
344 rtc::scoped_ptr<VideoMediaChannel> channel(
357 rtc::scoped_ptr<VideoMediaChannel> channel(
370 rtc::scoped_ptr<VideoMediaChannel> channel(
407 rtc::scoped_ptr<VideoMediaChannel> channel(
422 rtc::scoped_ptr<VideoMediaChannel> channel(
482 rtc::scoped_ptr<VideoMediaChannel> channel(
499 rtc::scoped_ptr<char[]> data(new char[frame.data_size])
[all...] |
| /external/webrtc/webrtc/base/ |
| diskcache.cc | 126 scoped_ptr<FileStream> file(new FileStream);
164 scoped_ptr<FileStream> file(new FileStream);
|
| network.cc | 42 #include "webrtc/base/scoped_ptr.h"
452 scoped_ptr<Network> network(new Network(cursor->ifa_name,
478 rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(CreateIfAddrsConverter());
534 scoped_ptr<char[]> adapter_info(new char[buffer_size]);
570 scoped_ptr<Network> network;
609 scoped_ptr<Network> network(new Network(name, description, prefix,
771 scoped_ptr<AsyncSocket> socket(
|
| httpclient.cc | 20 #include "webrtc/base/scoped_ptr.h"
469 scoped_ptr<StreamInterface> stream(cache_->WriteResource(id, kCacheBody));
488 scoped_ptr<StreamInterface> stream(cache_->WriteResource(id, kCacheHeader));
566 scoped_ptr<StreamInterface> stream(cache_->ReadResource(id, kCacheHeader));
589 scoped_ptr<StreamInterface> stream(cache_->ReadResource(id, kCacheBody));
602 scoped_ptr<char[]> buffer(new char[array_size]);
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| EncodeDecodeTest.cc | 18 #include "webrtc/base/scoped_ptr.h"
278 rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
334 rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
|
| /external/webrtc/webrtc/modules/audio_device/linux/ |
| audio_device_pulse_linux.h | 287 // TODO(pbos): Remove scoped_ptr and use directly without resetting.
288 rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
289 rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
|
| /external/webrtc/webrtc/modules/audio_device/mac/ |
| audio_device_mac.h | 14 #include "webrtc/base/scoped_ptr.h"
293 rtc::scoped_ptr<rtc::PlatformThread> capture_worker_thread_;
296 rtc::scoped_ptr<rtc::PlatformThread> render_worker_thread_;
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| inter_arrival_unittest.cc | 13 #include "webrtc/base/scoped_ptr.h"
202 rtc::scoped_ptr<InterArrival> inter_arrival_rtp_;
203 rtc::scoped_ptr<InterArrival> inter_arrival_ast_;
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_video.cc | 125 rtc::scoped_ptr<RedPacket> red_packet;
238 rtc::scoped_ptr<RtpPacketizer> packetizer(
|
| rtp_format_vp9_unittest.cc | 79 rtc::scoped_ptr<RtpDepacketizer> depacketizer(new RtpDepacketizerVp9());
130 rtc::scoped_ptr<uint8_t[]> packet_;
131 rtc::scoped_ptr<uint8_t[]> payload_;
135 rtc::scoped_ptr<RtpPacketizerVp9> packetizer_;
472 rtc::scoped_ptr<RtpDepacketizer> depacketizer_;
|
| /external/webrtc/webrtc/modules/utility/source/ |
| helpers_ios.mm | 20 #include "webrtc/base/scoped_ptr.h"
89 rtc::scoped_ptr<char[]> machine;
|
| /external/webrtc/webrtc/modules/video_coding/ |
| video_coding_impl.cc | 68 rtc::scoped_ptr<CriticalSectionWrapper> cs_;
291 rtc::scoped_ptr<EventFactory> own_event_factory_;
|
| /prebuilts/tools/darwin-x86_64/protoc/include/google/protobuf/util/ |
| message_differencer.h | [all...] |
| /external/webrtc/webrtc/call/ |
| call.cc | 22 #include "webrtc/base/scoped_ptr.h"
123 const rtc::scoped_ptr<ProcessThread> module_process_thread_;
124 const rtc::scoped_ptr<CallStats> call_stats_;
125 const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
131 rtc::scoped_ptr<RWLockWrapper> receive_crit_;
142 rtc::scoped_ptr<RWLockWrapper> send_crit_;
169 const rtc::scoped_ptr<CongestionController> congestion_controller_;
|
| rtc_event_log_unittest.cc | 21 #include "webrtc/base/scoped_ptr.h"
351 rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(Random* prng) {
428 std::vector<rtc::scoped_ptr<rtcp::RawPacket> > rtcp_packets;
474 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
605 rtc::scoped_ptr<rtcp::RawPacket> old_rtcp_packet;
606 rtc::scoped_ptr<rtcp::RawPacket> recent_rtcp_packet;
640 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
|
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 24 #include "webrtc/base/scoped_ptr.h"
148 rtc::scoped_ptr<int16_t[]> file_;
236 rtc::scoped_ptr<AudioBufferList> fifo_;
596 rtc::scoped_ptr<EventWrapper> test_is_done_;
764 rtc::scoped_ptr<FileAudioStream> file_audio_stream(
798 rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream(
827 rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream
[all...] |
| /external/google-breakpad/src/processor/ |
| minidump.cc | 62 #include "common/scoped_ptr.h"
234 scoped_ptr<string> out(new string());
338 scoped_ptr<string> temp(UTF16ToUTF8(utf16_vector, swap));
450 scoped_ptr<MDRawContextAMD64> context_amd64(new MDRawContextAMD64());
561 scoped_ptr<MDRawContextPPC64> context_ppc64(new MDRawContextPPC64());
657 scoped_ptr<MDRawContextARM64> context_arm64(new MDRawContextARM64());
771 scoped_ptr<MDRawContextX86> context_x86(new MDRawContextX86());
844 scoped_ptr<MDRawContextPPC> context_ppc(new MDRawContextPPC());
920 scoped_ptr<MDRawContextSPARC> context_sparc(new MDRawContextSPARC());
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| process_test.cc | 21 #include "webrtc/base/scoped_ptr.h"
149 rtc::scoped_ptr<AudioProcessing> apm(AudioProcessing::Create());
494 rtc::scoped_ptr<WavWriter> output_wav_file;
495 rtc::scoped_ptr<RawFile> output_raw_file;
537 rtc::scoped_ptr<char[]> echo_path(new char[path_size]);
579 rtc::scoped_ptr<ChannelBuffer<float> > reverse_cb;
580 rtc::scoped_ptr<ChannelBuffer<float> > primary_cb;
[all...] |
| /external/googletest/googletest/include/gtest/ |
| gtest.h | 335 internal::scoped_ptr< ::std::string> message_;
455 const internal::scoped_ptr< GTEST_FLAG_SAVER_ > gtest_flag_saver_;
755 const internal::scoped_ptr<const ::std::string> type_param_;
758 const internal::scoped_ptr<const ::std::string> value_param_;
[all...] |
| /external/libvpx/libvpx/third_party/googletest/src/include/gtest/ |
| gtest.h | 335 internal::scoped_ptr< ::std::string> message_;
455 const internal::scoped_ptr< GTEST_FLAG_SAVER_ > gtest_flag_saver_;
755 const internal::scoped_ptr<const ::std::string> type_param_;
758 const internal::scoped_ptr<const ::std::string> value_param_;
[all...] |
| /external/v8/testing/gtest/include/gtest/ |
| gtest.h | 335 internal::scoped_ptr< ::std::string> message_;
455 const internal::scoped_ptr< GTEST_FLAG_SAVER_ > gtest_flag_saver_;
755 const internal::scoped_ptr<const ::std::string> type_param_;
758 const internal::scoped_ptr<const ::std::string> value_param_;
[all...] |
| /external/webrtc/webrtc/p2p/base/ |
| transportcontroller_unittest.cc | 16 #include "webrtc/base/scoped_ptr.h"
180 rtc::scoped_ptr<rtc::Thread> worker_thread_; // Not used for most tests.
181 rtc::scoped_ptr<TransportControllerForTest> transport_controller_;
272 rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
275 rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
306 rtc::scoped_ptr<rtc::SSLCertificate> returned_certificate;
|
| turnport_unittest.cc | 29 #include "webrtc/base/scoped_ptr.h"
510 rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
511 rtc::scoped_ptr<TurnPortTestVirtualSocketServer> ss_;
515 rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
517 rtc::scoped_ptr<TurnPort> turn_port_;
518 rtc::scoped_ptr<UDPPort> udp_port_;
[all...] |
