| /external/webrtc/media/engine/ |
| D | fake_webrtc_call.h | 39 class FakeAudioSendStream final : public webrtc::AudioSendStream {
49 const webrtc::AudioSendStream::Config& config);
52 const webrtc::AudioSendStream::Config& GetConfig() const override;
53 void SetStats(const webrtc::AudioSendStream::Stats& stats);
60 void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
63 void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override { in SendAudioData()
70 webrtc::AudioSendStream::Stats GetStats() const override;
71 webrtc::AudioSendStream::Stats GetStats(
76 webrtc::AudioSendStream::Config config_;
77 webrtc::AudioSendStream::Stats stats_;
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| D | webrtc_video_engine.h | 40 namespace webrtc {
63 std::map<uint32_t, webrtc::VideoSendStream::StreamStats>
65 const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>& substreams);
84 rtc::VideoSinkInterface<webrtc::VideoFrame>* GetDefaultSink() const;
86 rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
91 rtc::VideoSinkInterface<webrtc::VideoFrame>* default_sink_;
100 std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
101 std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory);
106 webrtc::Call* call,
109 const webrtc::CryptoOptions& crypto_options,
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| D | simulcast.cc | 35 constexpr webrtc::DataRate Interpolate(const webrtc::DataRate& a, in Interpolate()
36 const webrtc::DataRate& b, in Interpolate()
46 constexpr webrtc::DataRate kScreenshareDefaultTl0Bitrate =
47 webrtc::DataRate::KilobitsPerSec(200);
48 constexpr webrtc::DataRate kScreenshareDefaultTl1Bitrate =
49 webrtc::DataRate::KilobitsPerSec(1000);
53 constexpr webrtc::DataRate kScreenshareHighStreamMinBitrate =
54 webrtc::DataRate::KilobitsPerSec(600);
55 constexpr webrtc::DataRate kScreenshareHighStreamMaxBitrate =
56 webrtc::DataRate::KilobitsPerSec(1250);
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| D | fake_webrtc_call.cc | 24 const webrtc::AudioSendStream::Config& config) in FakeAudioSendStream()
28 const webrtc::AudioSendStream::Config& config) { in Reconfigure()
32 const webrtc::AudioSendStream::Config& FakeAudioSendStream::GetConfig() const { in GetConfig()
37 const webrtc::AudioSendStream::Stats& stats) { in SetStats()
61 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { in GetStats()
65 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats( in GetStats()
72 const webrtc::AudioReceiveStream::Config& config) in FakeAudioReceiveStream()
75 const webrtc::AudioReceiveStream::Config& FakeAudioReceiveStream::GetConfig() in GetConfig()
81 const webrtc::AudioReceiveStream::Stats& stats) { in SetStats()
99 const webrtc::AudioReceiveStream::Config& config) { in Reconfigure()
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| D | webrtc_voice_engine.h | 47 webrtc::TaskQueueFactory* task_queue_factory,
48 webrtc::AudioDeviceModule* adm,
49 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
50 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
51 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
52 rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
58 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
60 webrtc::Call* call,
63 const webrtc::CryptoOptions& crypto_options) override;
67 std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
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| D | fake_webrtc_video_engine.cc | 30 const std::vector<webrtc::SdpVideoFormat>& supported_formats, in IsFormatSupported()
31 const webrtc::SdpVideoFormat& format) { in IsFormatSupported()
32 for (const webrtc::SdpVideoFormat& supported_format : supported_formats) { in IsFormatSupported()
54 int32_t FakeWebRtcVideoDecoder::InitDecode(const webrtc::VideoCodec*, int32_t) { in InitDecode()
58 int32_t FakeWebRtcVideoDecoder::Decode(const webrtc::EncodedImage&, in Decode()
66 webrtc::DecodedImageCallback*) { in RegisterDecodeCompleteCallback()
82 std::vector<webrtc::SdpVideoFormat>
84 std::vector<webrtc::SdpVideoFormat> formats; in GetSupportedFormats()
86 for (const webrtc::SdpVideoFormat& format : supported_codec_formats_) { in GetSupportedFormats()
95 std::unique_ptr<webrtc::VideoDecoder>
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| D | fake_webrtc_video_engine.h | 42 class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder {
47 int32_t InitDecode(const webrtc::VideoCodec*, int32_t) override;
48 int32_t Decode(const webrtc::EncodedImage&, bool, int64_t) override;
50 webrtc::DecodedImageCallback*) override;
61 class FakeWebRtcVideoDecoderFactory : public webrtc::VideoDecoderFactory {
65 std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
66 std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
67 const webrtc::SdpVideoFormat& format) override;
75 std::vector<webrtc::SdpVideoFormat> supported_codec_formats_;
81 class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
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| /external/webrtc/ |
| D | WATCHLISTS | 18 # NOTE: if you like this you might like webrtc-reviews@webrtc.org!
105 'root_files': ['peah@webrtc.org',
107 'yujie.mao@webrtc.org'],
108 'build_files': ['mbonadei@webrtc.org'],
109 'common_audio': ['alessiob@webrtc.org',
110 'aluebs@webrtc.org',
112 'minyue@webrtc.org',
113 'peah@webrtc.org',
114 'saza@webrtc.org'],
115 'audio': ['peah@webrtc.org'],
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| D | OWNERS.webrtc | 1 henrika@webrtc.org
2 juberti@webrtc.org
3 kwiberg@webrtc.org
4 mflodman@webrtc.org
5 stefan@webrtc.org
6 tommi@webrtc.org
8 per-file .gn=mbonadei@webrtc.org
9 per-file *.gn=mbonadei@webrtc.org
10 per-file *.gni=mbonadei@webrtc.org
13 per-file pylintrc=phoglund@webrtc.org
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| /external/webrtc/pc/ |
| D | peer_connection_end_to_end_unittest.cc | 41 using webrtc::DataChannelInterface;
42 using webrtc::MediaStreamInterface;
43 using webrtc::PeerConnectionInterface;
44 using webrtc::SdpSemantics;
66 webrtc::PeerConnectionInterface::IceServer ice_server; in PeerConnectionEndToEndBaseTest()
72 webrtc::InitializeAndroidObjects(); in PeerConnectionEndToEndBaseTest()
77 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory1, in CreatePcs()
78 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory1, in CreatePcs()
79 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory2, in CreatePcs()
80 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory2) { in CreatePcs()
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| D | jsep_transport.cc | 30 using webrtc::SdpType;
75 rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport, in JsepTransport()
76 rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport, in JsepTransport()
77 std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport, in JsepTransport()
78 std::unique_ptr<webrtc::SrtpTransport> sdes_transport, in JsepTransport()
79 std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport, in JsepTransport()
80 std::unique_ptr<webrtc::RtpTransportInternal> datagram_rtp_transport, in JsepTransport()
93 rtp_dtls_transport ? new rtc::RefCountedObject<webrtc::DtlsTransport>( in JsepTransport()
98 ? new rtc::RefCountedObject<webrtc::DtlsTransport>( in JsepTransport()
102 sctp_transport ? std::make_unique<webrtc::SctpDataChannelTransport>( in JsepTransport()
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| D | jsep_transport.h | 87 rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport,
88 rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport,
89 std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport,
90 std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
91 std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
92 std::unique_ptr<webrtc::RtpTransportInternal> datagram_rtp_transport,
116 webrtc::RTCError SetLocalJsepTransportDescription(
118 webrtc::SdpType type) RTC_LOCKS_EXCLUDED(accessor_lock_);
122 webrtc::RTCError SetRemoteJsepTransportDescription(
124 webrtc::SdpType type) RTC_LOCKS_EXCLUDED(accessor_lock_);
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| D | data_channel_unittest.cc | 23 using webrtc::DataChannelInterface;
24 using webrtc::SctpDataChannel;
25 using webrtc::SctpSidAllocator;
29 class FakeDataChannelObserver : public webrtc::DataChannelObserver {
42 void OnMessage(const webrtc::DataBuffer& buffer) { ++messages_received_; } in OnMessage()
92 webrtc::InternalDataChannelInit init_;
146 EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, in TEST_F()
152 EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state()); in TEST_F()
156 EXPECT_EQ(webrtc::DataChannelInterface::kClosed, in TEST_F()
170 webrtc::DataBuffer buffer("abcd"); in TEST_F()
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| /external/webrtc/examples/androidnativeapi/jni/ |
| D | android_call_client.cc | 30 class AndroidCallClient::PCObserver : public webrtc::PeerConnectionObserver {
35 webrtc::PeerConnectionInterface::SignalingState new_state) override;
37 rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
40 webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
42 webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
43 void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
51 class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
54 rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
56 void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
57 void OnFailure(webrtc::RTCError error) override;
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| /external/webrtc/sdk/android/ |
| D | BUILD.gn | 12 import("../../webrtc.gni")
78 sources = [ "src/java/org/webrtc/Empty.java" ]
98 sources = [ "api/org/webrtc/Metrics.java" ]
110 "audio_codecs", # TODO(bugs.webrtc.org/8396): Remove.
111 "software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove.
113 public_deps = [ # no-presubmit-check TODO(webrtc:8603)
162 "api/org/webrtc/Predicate.java",
163 "api/org/webrtc/RefCounted.java",
164 "src/java/org/webrtc/CalledByNative.java",
165 "src/java/org/webrtc/CalledByNativeUnchecked.java",
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| /external/webrtc/docs/native-code/ |
| D | index.md | 5 API][webrtc-api] instead.
7 [webrtc-api]: http://dev.w3.org/2011/webrtc/editor/webrtc.html
10 [https://webrtc.googlesource.com/src][webrtc-repo].
12 [webrtc-repo]: https://webrtc.googlesource.com/src/
15 [https://webrtc.googlesource.com/src/+log][webrtc-change-log]
17 [webrtc-change-log]: https://webrtc.googlesource.com/src/+log
19 Please read the [License & Rights][webrtc-license] and [FAQ][webrtc-faq]
22 [webrtc-license]: https://webrtc.org/support/license
23 [webrtc-faq]: https://webrtc.googlesource.com/src/+/refs/heads/master/docs/faq.md
25 The WebRTC [issue tracker][webrtc-issue-tracker] can be used for submitting
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| /external/webrtc/pc/test/ |
| D | peer_connection_test_wrapper.h | 35 : public webrtc::PeerConnectionObserver,
36 public webrtc::CreateSessionDescriptionObserver,
48 const webrtc::PeerConnectionInterface::RTCConfiguration& config,
49 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
50 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory);
52 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory() in pc_factory()
56 webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } in pc()
58 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
60 const webrtc::DataChannelInit& init);
66 webrtc::PeerConnectionInterface::SignalingState new_state) override;
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| D | peer_connection_test_wrapper.cc | 44 using webrtc::FakeVideoTrackRenderer;
45 using webrtc::IceCandidateInterface;
46 using webrtc::MediaStreamInterface;
47 using webrtc::MediaStreamTrackInterface;
48 using webrtc::MockSetSessionDescriptionObserver;
49 using webrtc::PeerConnectionInterface;
50 using webrtc::RtpReceiverInterface;
51 using webrtc::SdpType;
52 using webrtc::SessionDescriptionInterface;
53 using webrtc::VideoTrackInterface;
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| /external/webrtc/rtc_tools/rtc_event_log_visualizer/ |
| D | main.cc | 105 using webrtc::Plot;
243 webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str()); in main()
245 webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions = in main()
246 webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse; in main()
248 header_extensions = webrtc::ParsedRtcEventLog:: in main()
251 webrtc::ParsedRtcEventLog parsed_log(header_extensions, in main()
264 webrtc::AnalyzerConfig config; in main()
277 webrtc::EventLogAnalyzer analyzer(parsed_log, config); in main()
278 webrtc::PlotCollection collection; in main()
282 analyzer.CreatePacketGraph(webrtc::kIncomingPacket, plot); in main()
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| /external/webrtc/system_wrappers/include/ |
| D | metrics.h | 29 namespace webrtc {
46 #define EXPECT_METRIC_EQ(val1, val2) webrtc::metrics_impl::NoOp(val1, val2)
47 #define EXPECT_METRIC_EQ_WAIT(val1, val2, timeout) webrtc::metrics_impl::NoOp(val1, val2, timeout)
48 #define EXPECT_METRIC_GT(val1, val2) webrtc::metrics_impl::NoOp(val1, val2)
49 #define EXPECT_METRIC_LE(val1, val2) webrtc::metrics_impl::NoOp(val1, val2)
50 #define EXPECT_METRIC_TRUE(condition) webrtc::metrics_impl::NoOp(condition || true)
51 #define EXPECT_METRIC_FALSE(condition) webrtc::metrics_impl::NoOp(condition && false)
52 #define EXPECT_METRIC_THAT(value, matcher) webrtc::metrics_impl::NoOp(value, testing::_)
124 webrtc::metrics::HistogramFactoryGetCounts( \
129 webrtc::metrics::HistogramFactoryGetCountsLinear( \
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| /external/webrtc/api/ |
| D | rtp_parameters.cc | 19 namespace webrtc { namespace
139 return uri == webrtc::RtpExtension::kAudioLevelUri || in IsSupportedForAudio()
140 uri == webrtc::RtpExtension::kAbsSendTimeUri || in IsSupportedForAudio()
141 uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri || in IsSupportedForAudio()
142 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || in IsSupportedForAudio()
143 uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri || in IsSupportedForAudio()
144 uri == webrtc::RtpExtension::kMidUri || in IsSupportedForAudio()
145 uri == webrtc::RtpExtension::kRidUri || in IsSupportedForAudio()
146 uri == webrtc::RtpExtension::kRepairedRidUri; in IsSupportedForAudio()
150 return uri == webrtc::RtpExtension::kTimestampOffsetUri || in IsSupportedForVideo()
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| /external/webrtc/examples/objcnativeapi/objc/ |
| D | objc_call_client.mm | 37 class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
39 explicit CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
41 void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
42 void OnFailure(webrtc::RTCError error) override;
45 const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
48 class SetRemoteSessionDescriptionObserver : public webrtc::SetRemoteDescriptionObserverInterface {
50 void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
53 class SetLocalSessionDescriptionObserver : public webrtc::SetSessionDescriptionObserver {
56 void OnFailure(webrtc::RTCError error) override;
71 webrtc::MutexLock lock(&pc_mutex_);
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| /external/webrtc/rtc_base/experiments/ |
| D | OWNERS.webrtc | 1 asapersson@webrtc.org
2 sprang@webrtc.org
3 srte@webrtc.org
5 per-file alr_experiment*=sprang@webrtc.org
6 per-file audio_allocation_settings*=srte@webrtc.org
7 per-file congestion_controller_experiment*=srte@webrtc.org
8 per-file cpu_speed_experiment*=asapersson@webrtc.org
9 per-file field_trial*=srte@webrtc.org
10 per-file jitter_upper_bound_experiment*=sprang@webrtc.org
11 per-file keyframe_interval_settings*=brandtr@webrtc.org
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| D | min_video_bitrate_experiment.cc | 20 namespace webrtc { namespace
34 if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial)) { in GetFallbackMinBpsFromFieldTrial()
39 webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial); in GetFallbackMinBpsFromFieldTrial()
67 if (webrtc::field_trial::IsEnabled(kMinVideoBitrateExperiment)) { in GetExperimentalMinVideoBitrate()
68 webrtc::FieldTrialFlag enabled("Enabled"); in GetExperimentalMinVideoBitrate()
72 webrtc::FieldTrialOptional<webrtc::DataRate> min_video_bitrate("br"); in GetExperimentalMinVideoBitrate()
75 webrtc::FieldTrialOptional<webrtc::DataRate> min_bitrate_vp8("vp8_br"); in GetExperimentalMinVideoBitrate()
76 webrtc::FieldTrialOptional<webrtc::DataRate> min_bitrate_vp9("vp9_br"); in GetExperimentalMinVideoBitrate()
77 webrtc::FieldTrialOptional<webrtc::DataRate> min_bitrate_av1("av1_br"); in GetExperimentalMinVideoBitrate()
78 webrtc::FieldTrialOptional<webrtc::DataRate> min_bitrate_h264("h264_br"); in GetExperimentalMinVideoBitrate()
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| /external/webrtc/examples/unityplugin/ |
| D | simple_peer_connection.cc | 44 static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
52 class CapturerTrackSource : public webrtc::VideoTrackSource {
59 std::unique_ptr<webrtc::test::VcmCapturer> capturer = absl::WrapUnique( in Create()
60 webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, kDeviceIndex)); in Create()
69 std::unique_ptr<webrtc::test::VcmCapturer> capturer) in CapturerTrackSource()
73 rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override { in source()
76 std::unique_ptr<webrtc::test::VcmCapturer> capturer_;
99 : public webrtc::SetSessionDescriptionObserver {
105 virtual void OnFailure(webrtc::RTCError error) { in OnFailure()
130 g_peer_connection_factory = webrtc::CreatePeerConnectionFactory( in InitializePeerConnection()
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