Searched refs:rtc (Results 176 - 200 of 1454) sorted by relevance

1234567891011>>

/external/webrtc/talk/app/webrtc/
H A Dlocalaudiosource.cc52 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>());
H A Dmediacontroller.cc44 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,
H A Drtpreceiver.h45 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_;
H A Dpeerconnection.h69 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 Dpeerconnectionendtoend_unittest.cc62 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 DRTCMediaSource.mm37 rtc::scoped_refptr<webrtc::MediaSourceInterface> _mediaSource;
49 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource {
61 - (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource {
H A DRTCPeerConnectionFactory.mm58 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 Dfilevideocapturer.cc58 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 Daudio_receive_stream.h33 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_;
/external/webrtc/webrtc/base/
H A Dgunit.h24 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); \
H A Dsslidentity_unittest.cc18 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 Dautodetectproxy_unittest.cc16 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
H A Dsslfingerprint.cc20 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
/external/webrtc/webrtc/modules/desktop_capture/
H A Ddesktop_capture_options.h42 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>
/external/webrtc/webrtc/modules/utility/source/
H A Dprocess_thread_impl.h35 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_;
/external/webrtc/webrtc/modules/audio_coding/acm2/
H A Drent_a_codec.cc45 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 Dgain_control_impl.cc46 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 Drtpdump.h60 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 Dwebrtcvoiceengine.h67 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 Dvideo_frame_buffer.h31 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 Dincoming_video_stream.h80 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_
/external/webrtc/webrtc/modules/audio_coding/neteq/
H A Dneteq_impl.h84 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 Dchannelmanager.h57 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 Dfaketransportcontroller.h35 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 Daudio_record_jni.h49 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_;

Completed in 422 milliseconds

1234567891011>>