/external/webrtc/talk/app/webrtc/ |
H A D | localaudiosource.cc | 52 rtc::Optional<bool>& value; 76 if (!rtc::FromString(constraint.value, &value)) 81 entry.value = rtc::Optional<bool>(value); 88 rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create( 91 rtc::scoped_refptr<LocalAudioSource> source( 92 new rtc::RefCountedObject<LocalAudioSource>());
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H A D | mediacontroller.cc | 44 MediaController(rtc::Thread* worker_thread, 49 rtc::Bind(&MediaController::Construct_w, this, 53 worker_thread_->Invoke<void>(rtc::Bind(&MediaController::Destruct_w, this)); 81 rtc::Thread* const worker_thread_; 83 rtc::scoped_ptr<webrtc::Call> call_; 92 rtc::Thread* worker_thread,
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H A D | rtpreceiver.h | 45 public rtc::RefCountedObject<RtpReceiverInterface> { 60 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { 72 const rtc::scoped_refptr<AudioTrackInterface> track_; 78 class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> { 87 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { 97 rtc::scoped_refptr<VideoTrackInterface> track_;
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H A D | peerconnection.h | 69 public rtc::MessageHandler, 77 rtc::scoped_ptr<cricket::PortAllocator> allocator, 78 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, 81 rtc::scoped_refptr<StreamCollectionInterface> local_streams() override; 82 rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override; 88 rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( 91 rtc::scoped_refptr<RtpSenderInterface> CreateSender( 95 std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() 97 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() 100 rtc [all...] |
H A D | peerconnectionendtoend_unittest.cc | 62 typedef std::vector<rtc::scoped_refptr<DataChannelInterface> > 66 : caller_(new rtc::RefCountedObject<PeerConnectionTestWrapper>( 68 callee_(new rtc::RefCountedObject<PeerConnectionTestWrapper>( 129 rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc1_observer( 132 rtc::scoped_ptr<webrtc::MockDataChannelObserver> dc2_observer( 170 rtc::scoped_refptr<PeerConnectionTestWrapper> caller_; 171 rtc::scoped_refptr<PeerConnectionTestWrapper> callee_; 202 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); 207 rtc::scoped_refptr<DataChannelInterface> caller_dc( 209 rtc [all...] |
/external/webrtc/talk/app/webrtc/objc/ |
H A D | RTCMediaSource.mm | 37 rtc::scoped_refptr<webrtc::MediaSourceInterface> _mediaSource; 49 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource { 61 - (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource {
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H A D | RTCPeerConnectionFactory.mm | 58 rtc::scoped_ptr<rtc::Thread> _signalingThread; 59 rtc::scoped_ptr<rtc::Thread> _workerThread; 65 BOOL initialized = rtc::InitializeSSL(); 70 BOOL deinitialized = rtc::CleanupSSL(); 76 _signalingThread.reset(new rtc::Thread()); 79 _workerThread.reset(new rtc::Thread()); 87 // rtc::LogMessage::LogToDebug(rtc [all...] |
/external/webrtc/talk/media/devices/ |
H A D | filevideocapturer.cc | 58 if (rtc::SS_CLOSED == video_file_.GetState()) { 71 rtc::ByteBuffer buffer; 84 if (rtc::SR_SUCCESS != video_file_.Write(buffer.Data(), 93 if (rtc::SR_SUCCESS != video_file_.Write(frame.data, 109 : public rtc::Thread, public rtc::MessageHandler { 130 rtc::CritScope cs(&crit_); 135 virtual void OnMessage(rtc::Message* /*pmsg*/) { 146 rtc::CritScope cs(&crit_); 152 mutable rtc [all...] |
/external/webrtc/webrtc/audio/ |
H A D | audio_receive_stream.h | 33 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); 48 void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override; 55 rtc::ThreadChecker thread_checker_; 58 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 59 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; 60 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
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/external/webrtc/webrtc/base/ |
H A D | gunit.h | 24 for (uint32_t start = rtc::Time(); !(ex) && rtc::Time() < start + timeout;) \ 25 rtc::Thread::Current()->ProcessMessages(1); 32 uint32_t start = rtc::Time(); \ 34 while (!res && rtc::Time() < start + timeout) { \ 35 rtc::Thread::Current()->ProcessMessages(1); \
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H A D | sslidentity_unittest.cc | 18 using rtc::SSLIdentity; 73 identity_rsa1_.reset(SSLIdentity::Generate("test1", rtc::KT_RSA)); 74 identity_rsa2_.reset(SSLIdentity::Generate("test2", rtc::KT_RSA)); 75 identity_ecdsa1_.reset(SSLIdentity::Generate("test3", rtc::KT_ECDSA)); 76 identity_ecdsa2_.reset(SSLIdentity::Generate("test4", rtc::KT_ECDSA)); 83 test_cert_.reset(rtc::SSLCertificate::FromPEMString(kTestCertificate)); 92 ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm); 96 ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm); 100 ASSERT_EQ(rtc::DIGEST_SHA_256, digest_algorithm); 104 ASSERT_EQ(rtc [all...] |
H A D | autodetectproxy_unittest.cc | 16 namespace rtc { namespace 89 void OnWorkDone(rtc::SignalThread *thread) { 91 static_cast<rtc::AutoDetectProxy *>(thread); 103 TestCopesWithProxy(rtc::SocketAddress("localhost", 9999)); 107 TestCopesWithProxy(rtc::SocketAddress("invalid", 9999)); 111 TestCopesWithProxy(rtc::SocketAddress("::1", 9999)); 115 TestCopesWithProxy(rtc::SocketAddress("127.0.0.1", 9999)); 131 } // namespace rtc
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H A D | sslfingerprint.cc | 20 namespace rtc { namespace 23 const std::string& algorithm, const rtc::SSLIdentity* identity) { 32 const std::string& algorithm, const rtc::SSLCertificate* cert) { 46 if (algorithm.empty() || !rtc::IsFips180DigestAlgorithm(algorithm)) 53 char value[rtc::MessageDigest::kMaxSize]; 54 value_len = rtc::hex_decode_with_delimiter(value, sizeof(value), 82 rtc::hex_encode_with_delimiter(digest.data<char>(), digest.size(), ':'); 95 } // namespace rtc
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/external/webrtc/webrtc/modules/desktop_capture/ |
H A D | desktop_capture_options.h | 42 void set_x_display(rtc::scoped_refptr<SharedXDisplay> x_display) { 52 rtc::scoped_refptr<DesktopConfigurationMonitor> m) { 60 rtc::scoped_refptr<FullScreenChromeWindowDetector> detector) { 90 rtc::scoped_refptr<SharedXDisplay> x_display_; 94 rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_; 95 rtc::scoped_refptr<FullScreenChromeWindowDetector>
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/external/webrtc/webrtc/modules/utility/source/ |
H A D | process_thread_impl.h | 35 void PostTask(rtc::scoped_ptr<ProcessTask> task) override; 69 rtc::CriticalSection lock_; // Used to guard modules_, tasks_ and stop_. 71 rtc::ThreadChecker thread_checker_; 72 const rtc::scoped_ptr<EventWrapper> wake_up_; 74 rtc::scoped_ptr<rtc::PlatformThread> thread_;
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | rent_a_codec.cc | 45 rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams( 53 rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) { 54 rtc::Optional<int> mi = CodecIndexFromId(codec_id); 55 return mi ? rtc::Optional<CodecInst>(Database()[*mi]) 56 : rtc::Optional<CodecInst>(); 59 rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst( 64 rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name, 67 rtc::Optional<CodecId> codec_id = 70 return rtc::Optional<CodecInst>(); 71 rtc [all...] |
/external/webrtc/webrtc/modules/audio_processing/ |
H A D | gain_control_impl.cc | 46 rtc::CriticalSection* crit_render, 47 rtc::CriticalSection* crit_capture) 70 rtc::CritScope cs(crit_render_); 107 rtc::CritScope cs(crit_capture_); 128 rtc::CritScope cs(crit_capture_); 180 rtc::CritScope cs(crit_capture_); 238 rtc::CritScope cs(crit_capture_); 250 rtc::CritScope cs(crit_capture_); 258 rtc::CritScope cs_render(crit_render_); 259 rtc [all...] |
/external/webrtc/talk/media/base/ |
H A D | rtpdump.h | 60 void WriteToByteBuffer(rtc::ByteBuffer* buf); 106 explicit RtpDumpReader(rtc::StreamInterface* stream) 117 virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet); 120 rtc::StreamResult ReadFileHeader(); 129 rtc::StreamInterface* stream_; 145 explicit RtpDumpLoopReader(rtc::StreamInterface* stream); 146 virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet); 188 explicit RtpDumpWriter(rtc::StreamInterface* stream); 194 rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) { 197 rtc [all...] |
/external/webrtc/talk/media/webrtc/ |
H A D | webrtcvoiceengine.h | 67 bool Init(rtc::Thread* worker_thread); 70 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const; 98 bool StartAecDump(rtc::PlatformFile file); 105 bool StartRtcEventLog(rtc::PlatformFile file); 125 rtc::ThreadChecker signal_thread_checker_; 126 rtc::ThreadChecker worker_thread_checker_; 129 rtc::scoped_ptr<VoEWrapper> voe_wrapper_; 130 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 144 rtc::Optional<bool> extended_filter_aec_; 145 rtc [all...] |
/external/webrtc/webrtc/common_video/include/ |
H A D | video_frame_buffer.h | 31 class VideoFrameBuffer : public rtc::RefCountInterface { 58 virtual rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() = 0; 78 rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override; 89 const rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> data_; 122 const rtc::Callback0<void>& no_longer_used); 131 rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override; 134 friend class rtc::RefCountedObject<WrappedI420Buffer>; 145 rtc::Callback0<void> no_longer_used_cb_; 150 rtc::scoped_refptr<VideoFrameBuffer> ShallowCenterCrop( 151 const rtc [all...] |
H A D | incoming_video_stream.h | 80 const rtc::scoped_ptr<CriticalSectionWrapper> stream_critsect_; 81 const rtc::scoped_ptr<CriticalSectionWrapper> thread_critsect_; 82 const rtc::scoped_ptr<CriticalSectionWrapper> buffer_critsect_; 85 rtc::scoped_ptr<rtc::PlatformThread> incoming_render_thread_ 87 rtc::scoped_ptr<EventTimerWrapper> deliver_buffer_event_; 92 const rtc::scoped_ptr<VideoRenderFrames> render_buffers_
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.h | 84 rtc::ArrayView<const uint8_t> payload, 209 rtc::ArrayView<const uint8_t> payload, 341 const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_; 342 const rtc::scoped_ptr<BufferLevelFilter> buffer_level_filter_ 344 const rtc::scoped_ptr<DecoderDatabase> decoder_database_ 346 const rtc::scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_); 347 const rtc::scoped_ptr<DelayPeakDetector> delay_peak_detector_ 349 const rtc::scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_); 350 const rtc::scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_ 352 const rtc [all...] |
/external/webrtc/talk/session/media/ |
H A D | channelmanager.h | 57 class ChannelManager : public rtc::MessageHandler, 66 rtc::Thread* worker); 69 rtc::Thread* worker); 74 rtc::Thread* worker_thread() const { return worker_thread_; } 75 bool set_worker_thread(rtc::Thread* thread) { 166 bool StartAecDump(rtc::PlatformFile file); 172 bool StartRtcEventLog(rtc::PlatformFile file); 187 rtc::Thread* worker_thread); 216 virtual void OnMessage(rtc::Message *message); 218 rtc [all...] |
/external/webrtc/webrtc/p2p/base/ |
H A D | faketransportcontroller.h | 35 struct PacketMessageData : public rtc::MessageData { 37 rtc::Buffer packet; 46 public rtc::MessageHandler { 62 const rtc::SSLFingerprint& dtls_fingerprint() const { 105 dtls_fingerprint_ = rtc::SSLFingerprint(alg, digest, digest_len); 108 bool SetSslRole(rtc::SSLRole role) override { 112 bool GetSslRole(rtc::SSLRole* role) const override { 197 const rtc::PacketOptions& options, 209 rtc::Thread::Current()->Post(this, 0, packet); 211 rtc [all...] |
/external/webrtc/webrtc/modules/audio_device/android/ |
H A D | audio_record_jni.h | 49 rtc::scoped_ptr<GlobalRef> audio_track); 60 rtc::scoped_ptr<GlobalRef> audio_record_; 109 rtc::ThreadChecker thread_checker_; 113 rtc::ThreadChecker thread_checker_java_; 120 rtc::scoped_ptr<JNIEnvironment> j_environment_; 123 rtc::scoped_ptr<NativeRegistration> j_native_registration_; 126 rtc::scoped_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
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