| /external/webrtc/webrtc/ |
| config.cc | 24 std::string RtpExtension::ToString() const {
32 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
33 const char* RtpExtension::kAbsSendTime =
35 const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
36 const char* RtpExtension::kAudioLevel =
38 const char* RtpExtension::kTransportSequenceNumber =
41 bool RtpExtension::IsSupportedForAudio(const std::string& name) {
42 return name == webrtc::RtpExtension::kAbsSendTime ||
43 name == webrtc::RtpExtension::kAudioLevel ||
44 name == webrtc::RtpExtension::kTransportSequenceNumber
[all...] |
| config.h | 54 struct RtpExtension {
55 RtpExtension(const std::string& name, int id) : name(name), id(id) {}
57 bool operator==(const RtpExtension& rhs) const {
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| audio_send_stream.h | 68 std::vector<RtpExtension> extensions;
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| audio_receive_stream.h | 83 std::vector<RtpExtension> extensions;
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| video_receive_stream.h | 137 std::vector<RtpExtension> extensions;
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| video_send_stream.h | 112 std::vector<RtpExtension> extensions;
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| /external/webrtc/talk/media/webrtc/ |
| webrtcmediaengine.cc | 77 std::vector<webrtc::RtpExtension>* extensions,
83 [name](const webrtc::RtpExtension& rhs) {
112 std::vector<webrtc::RtpExtension> FilterRtpExtensions(
118 std::vector<webrtc::RtpExtension> result;
132 [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
139 [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
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| webrtcmediaengine.h | 59 // Convert cricket::RtpHeaderExtension:s to webrtc::RtpExtension:s, discarding
63 std::vector<webrtc::RtpExtension> FilterRtpExtensions(
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| webrtcmediaengine_unittest.cc | 66 bool IsSorted(const std::vector<webrtc::RtpExtension>& extensions) {
114 std::vector<webrtc::RtpExtension> filtered =
121 std::vector<webrtc::RtpExtension> filtered =
130 std::vector<webrtc::RtpExtension> filtered =
138 std::vector<webrtc::RtpExtension> filtered =
146 std::vector<webrtc::RtpExtension> filtered =
155 std::vector<webrtc::RtpExtension> filtered =
174 std::vector<webrtc::RtpExtension> filtered =
188 std::vector<webrtc::RtpExtension> filtered =
200 std::vector<webrtc::RtpExtension> filtered
[all...] |
| webrtcvideoengine2.h | 247 const std::vector<webrtc::RtpExtension>& rtp_extensions,
254 const std::vector<webrtc::RtpExtension>& rtp_extensions);
406 void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
522 std::vector<webrtc::RtpExtension> send_rtp_extensions_;
527 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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| webrtcvoiceengine.h | 279 std::vector<webrtc::RtpExtension> send_rtp_extensions_;
283 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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| webrtcvideoengine2_unittest.cc | [all...] |
| webrtcvideoengine2.cc | 246 inline const webrtc::RtpExtension* FindHeaderExtension(
247 const std::vector<webrtc::RtpExtension>& extensions,
[all...] |
| /external/webrtc/webrtc/call/ |
| bitrate_estimator_tests.cc | 137 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
139 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
192 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
269 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
278 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId))
[all...] |
| rampup_tests.cc | 124 if (extension_type_ == RtpExtension::kAbsSendTime) {
128 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
129 } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
132 send_config->rtp.extensions.push_back(RtpExtension(
137 send_config->rtp.extensions.push_back(RtpExtension(
185 EXPECT_NE(RtpExtension::kTOffset, extension_type_)
192 if (extension_type_ == RtpExtension::kAbsSendTime) {
195 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
196 } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
198 send_config->rtp.extensions.push_back(RtpExtension(
[all...] |
| rtc_event_log_unittest.cc | 48 const char* kExtensionNames[] = {RtpExtension::kTOffset,
49 RtpExtension::kAudioLevel,
50 RtpExtension::kAbsSendTime,
51 RtpExtension::kVideoRotation,
52 RtpExtension::kTransportSequenceNumber};
390 RtpExtension(kExtensionNames[i], prng->Rand<int>()));
410 RtpExtension(kExtensionNames[i], prng->Rand<int>()));
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| /external/webrtc/webrtc/audio/ |
| audio_send_stream_unittest.cc | 100 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
102 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
103 stream_config_.rtp.extensions.push_back(RtpExtension(
104 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
172 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
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| audio_receive_stream_unittest.cc | 107 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
109 RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
209 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
264 helper.config().rtp.extensions.push_back(RtpExtension(
265 RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
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| audio_receive_stream.cc | 41 if (extension.name == RtpExtension::kTransportSequenceNumber) {
100 if (extension.name == RtpExtension::kAudioLevel) {
105 } else if (extension.name == RtpExtension::kAbsSendTime) {
110 } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
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| audio_send_stream.cc | 80 if (extension.name == RtpExtension::kAbsSendTime) {
82 } else if (extension.name == RtpExtension::kAudioLevel) {
84 } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
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| /external/webrtc/webrtc/video/ |
| video_receive_stream.cc | 29 static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
31 if (extension.name == RtpExtension::kTransportSequenceNumber)
210 if (extension == RtpExtension::kTOffset) {
212 } else if (extension == RtpExtension::kAbsSendTime) {
214 } else if (extension == RtpExtension::kVideoRotation) {
216 } else if (extension == RtpExtension::kTransportSequenceNumber) {
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| replay.cc | 228 RtpExtension(RtpExtension::kTOffset, flags::TransmissionOffsetId()));
232 RtpExtension(RtpExtension::kAbsSendTime, flags::AbsSendTimeId()));
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| video_send_stream.cc | 137 for (const RtpExtension& extension : config.rtp.extensions) {
138 if (extension.name == RtpExtension::kTransportSequenceNumber) {
179 if (extension == RtpExtension::kTOffset) {
181 } else if (extension == RtpExtension::kAbsSendTime) {
183 } else if (extension == RtpExtension::kVideoRotation) {
185 } else if (extension == RtpExtension::kTransportSequenceNumber) {
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| video_send_stream_tests.cc | 155 send_config->rtp.extensions.push_back(RtpExtension(
156 RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
199 RtpExtension(RtpExtension::kTOffset, test::kTOffsetExtensionId));
243 RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
403 send_config->rtp.extensions.push_back(RtpExtension(
404 RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
406 RtpExtension(RtpExtension::kTransportSequenceNumber
[all...] |
| /external/webrtc/webrtc/test/ |
| call_test.cc | 191 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
195 video_send_config_.rtp.extensions.push_back(RtpExtension(
196 RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
214 for (const RtpExtension& extension : video_send_config_.rtp.extensions)
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