| /external/webrtc/media/engine/ |
| D | webrtc_media_engine_unittest.cc | 20 using webrtc::RtpExtension;
25 std::vector<RtpExtension> MakeUniqueExtensions() { in MakeUniqueExtensions()
26 std::vector<RtpExtension> result; in MakeUniqueExtensions()
29 result.push_back(RtpExtension(name, 1 + i)); in MakeUniqueExtensions()
31 result.push_back(RtpExtension(name, 255 - i)); in MakeUniqueExtensions()
37 std::vector<RtpExtension> MakeRedundantExtensions() { in MakeRedundantExtensions()
38 std::vector<RtpExtension> result; in MakeRedundantExtensions()
41 result.push_back(RtpExtension(name, 1 + i)); in MakeRedundantExtensions()
42 result.push_back(RtpExtension(name, 255 - i)); in MakeRedundantExtensions()
56 bool IsSorted(const std::vector<webrtc::RtpExtension>& extensions) { in IsSorted()
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| D | webrtc_media_engine.cc | 50 std::vector<webrtc::RtpExtension>* extensions, in DiscardRedundantExtensions()
57 [uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; }); in DiscardRedundantExtensions()
69 const std::vector<webrtc::RtpExtension>& extensions) { in ValidateRtpExtensions()
70 bool id_used[1 + webrtc::RtpExtension::kMaxId] = {false}; in ValidateRtpExtensions()
72 if (extension.id < webrtc::RtpExtension::kMinId || in ValidateRtpExtensions()
73 extension.id > webrtc::RtpExtension::kMaxId) { in ValidateRtpExtensions()
87 std::vector<webrtc::RtpExtension> FilterRtpExtensions( in FilterRtpExtensions()
88 const std::vector<webrtc::RtpExtension>& extensions, in FilterRtpExtensions()
93 std::vector<webrtc::RtpExtension> result; in FilterRtpExtensions()
108 absl::c_sort(result, [](const webrtc::RtpExtension& rhs, in FilterRtpExtensions()
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| D | webrtc_media_engine.h | 61 bool ValidateRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
66 std::vector<webrtc::RtpExtension> FilterRtpExtensions(
67 const std::vector<webrtc::RtpExtension>& extensions,
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| /external/webrtc/api/ |
| D | rtp_parameters.cc | 63 RtpExtension::RtpExtension() = default;
64 RtpExtension::RtpExtension(absl::string_view uri, int id) : uri(uri), id(id) {} in RtpExtension() function in webrtc::RtpExtension
65 RtpExtension::RtpExtension(absl::string_view uri, int id, bool encrypt) in RtpExtension() function in webrtc::RtpExtension
67 RtpExtension::~RtpExtension() = default;
102 std::string RtpExtension::ToString() const { in ToString()
114 constexpr char RtpExtension::kEncryptHeaderExtensionsUri[];
115 constexpr char RtpExtension::kAudioLevelUri[];
116 constexpr char RtpExtension::kTimestampOffsetUri[];
117 constexpr char RtpExtension::kAbsSendTimeUri[];
118 constexpr char RtpExtension::kAbsoluteCaptureTimeUri[];
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| D | rtp_parameters_unittest.cc | 17 using webrtc::RtpExtension;
22 static const RtpExtension kExtension1(kExtensionUri1, 1);
23 static const RtpExtension kExtension1Encrypted(kExtensionUri1, 10, true);
24 static const RtpExtension kExtension2(kExtensionUri2, 2);
27 std::vector<RtpExtension> extensions; in TEST()
28 std::vector<RtpExtension> filtered; in TEST()
32 filtered = RtpExtension::FilterDuplicateNonEncrypted(extensions); in TEST()
34 EXPECT_EQ(std::vector<RtpExtension>{kExtension1Encrypted}, filtered); in TEST()
39 filtered = RtpExtension::FilterDuplicateNonEncrypted(extensions); in TEST()
41 EXPECT_EQ(std::vector<RtpExtension>{kExtension1Encrypted}, filtered); in TEST()
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| D | rtp_parameters.h | 248 struct RTC_EXPORT RtpExtension { struct
249 RtpExtension();
250 RtpExtension(absl::string_view uri, int id);
251 RtpExtension(absl::string_view uri, int id, bool encrypt);
252 ~RtpExtension();
255 bool operator==(const RtpExtension& rhs) const {
265 static const RtpExtension* FindHeaderExtensionByUri(
266 const std::vector<RtpExtension>& extensions,
272 static std::vector<RtpExtension> FilterDuplicateNonEncrypted(
273 const std::vector<RtpExtension>& extensions);
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| /external/webrtc/pc/ |
| D | used_ids_unittest.cc | 92 webrtc::RtpExtension extension("", j); in TEST_P()
100 webrtc::RtpExtension id_1("", 1); in TEST_P()
101 webrtc::RtpExtension id_2("", 2); in TEST_P()
102 webrtc::RtpExtension id_2_collision("", 2); in TEST_P()
103 webrtc::RtpExtension id_3("", 3); in TEST_P()
104 webrtc::RtpExtension id_3_collision("", 3); in TEST_P()
122 webrtc::RtpExtension id("", i); in TEST_F()
127 webrtc::RtpExtension id1_collision("", 1); in TEST_F()
128 webrtc::RtpExtension id2_collision("", 2); in TEST_F()
129 webrtc::RtpExtension id3_collision("", 3); in TEST_F()
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| D | used_ids.h | 104 class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> {
115 : UsedIds<webrtc::RtpExtension>( in UsedRtpHeaderExtensionIds()
116 webrtc::RtpExtension::kMinId, in UsedRtpHeaderExtensionIds()
118 ? webrtc::RtpExtension::kMaxId in UsedRtpHeaderExtensionIds()
119 : webrtc::RtpExtension::kOneByteHeaderExtensionMaxId), in UsedRtpHeaderExtensionIds()
121 next_extension_id_(webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) { in UsedRtpHeaderExtensionIds()
132 webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) { in FindUnusedId()
146 webrtc::RtpExtension::kOneByteHeaderExtensionMaxId + 1; in FindUnusedId()
150 webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) { in FindUnusedId()
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| D | peer_connection_header_extension_unittest.cc | 140 ElementsAre(Field(&RtpExtension::uri, "uri2"), in TEST_P()
141 Field(&RtpExtension::uri, "uri3"), in TEST_P()
142 Field(&RtpExtension::uri, "uri4"))); in TEST_P()
164 ElementsAre(Field(&RtpExtension::uri, "uri1"), in TEST_P()
165 Field(&RtpExtension::uri, "uri2"), in TEST_P()
166 Field(&RtpExtension::uri, "uri3"))); in TEST_P()
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| D | media_session_unittest.cc | 102 using webrtc::RtpExtension;
144 static const RtpExtension kAudioRtpExtension1[] = {
145 RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
146 RtpExtension("http://google.com/testing/audio_something", 10),
149 static const RtpExtension kAudioRtpExtensionEncrypted1[] = {
150 RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
151 RtpExtension("http://google.com/testing/audio_something", 10),
152 RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
155 static const RtpExtension kAudioRtpExtension2[] = {
156 RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 2),
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| /external/webrtc/call/ |
| D | bitrate_estimator_tests.cc | 144 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId)); in SetUp()
146 RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId)); in SetUp()
256 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId)); in TEST_F()
267 RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId)); in TEST_F()
280 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId)); in TEST_F()
289 RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId); in TEST_F()
301 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId)); in TEST_F()
311 RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId); in TEST_F()
320 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); in TEST_F()
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| D | rampup_tests.cc | 187 if (extension_type_ == RtpExtension::kAbsSendTimeUri) { in ModifyVideoConfigs()
191 RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); in ModifyVideoConfigs()
192 } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { in ModifyVideoConfigs()
195 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs()
200 send_config->rtp.extensions.push_back(RtpExtension( in ModifyVideoConfigs()
267 EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_) in ModifyAudioConfigs()
269 EXPECT_NE(RtpExtension::kAbsSendTimeUri, extension_type_) in ModifyAudioConfigs()
279 if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { in ModifyAudioConfigs()
281 send_config->rtp.extensions.push_back(RtpExtension( in ModifyAudioConfigs()
301 if (extension_type_ == RtpExtension::kAbsSendTimeUri) { in ModifyFlexfecConfigs()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | flexfec_sender_unittest.cc | 37 const std::vector<RtpExtension> kNoRtpHeaderExtensions;
193 const std::vector<RtpExtension> kRtpHeaderExtensions{}; in TEST()
206 const std::vector<RtpExtension> kRtpHeaderExtensions{ in TEST()
207 {RtpExtension::kAbsSendTimeUri, 1}}; in TEST()
220 const std::vector<RtpExtension> kRtpHeaderExtensions{ in TEST()
221 {RtpExtension::kTimestampOffsetUri, 1}}; in TEST()
234 const std::vector<RtpExtension> kRtpHeaderExtensions{ in TEST()
235 {RtpExtension::kTransportSequenceNumberUri, 1}}; in TEST()
248 const std::vector<RtpExtension> kRtpHeaderExtensions{ in TEST()
249 {RtpExtension::kAbsSendTimeUri, 1}, in TEST()
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| D | rtp_header_extension_map.cc | 73 rtc::ArrayView<const RtpExtension> extensions) in RtpHeaderExtensionMap()
75 for (const RtpExtension& extension : extensions) in RtpHeaderExtensionMap()
97 RTC_DCHECK_GE(id, RtpExtension::kMinId); in GetType()
98 RTC_DCHECK_LE(id, RtpExtension::kMaxId); in GetType()
130 if (id < RtpExtension::kMinId || id > RtpExtension::kMaxId) { in Register()
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| D | rtp_header_extension_size.cc | 31 if (id > RtpExtension::kOneByteHeaderExtensionMaxId || in RtpHeaderExtensionSize()
33 RtpExtension::kOneByteHeaderExtensionMaxValueSize) { in RtpHeaderExtensionSize()
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| /external/webrtc/media/base/ |
| D | media_engine_unittest.cc | 19 using ::webrtc::RtpExtension;
52 ElementsAre(Field(&RtpExtension::uri, StrEq("uri1")), in TEST()
53 Field(&RtpExtension::uri, StrEq("uri2")), in TEST()
54 Field(&RtpExtension::uri, StrEq("uri4")), in TEST()
55 Field(&RtpExtension::uri, StrEq("uri5")))); in TEST()
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| /external/webrtc/video/end_to_end_tests/ |
| D | frame_encryption_tests.cc | 77 RegisterRtpExtension(RtpExtension(RtpExtension::kGenericFrameDescriptorUri00, in TEST_F()
85 RegisterRtpExtension(RtpExtension(RtpExtension::kDependencyDescriptorUri, in TEST_F()
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| D | transport_feedback_tests.cc | 183 RtpExtension(RtpExtension::kTransportSequenceNumberUri, in TEST()
208 RtpExtension(RtpExtension::kTransportSequenceNumberUri, in TEST()
239 RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, in TransportFeedbackEndToEndTest()
305 RtpExtension(RtpExtension::kTransportSequenceNumberUri, in ModifyAudioConfigs()
458 RtpExtension(RtpExtension::kTransportSequenceNumberUri, in TEST_F()
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| D | codec_tests.cc | 41 RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId)); in CodecEndToEndTest()
42 RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri, in CodecEndToEndTest()
229 RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri, in EndToEndTestH264()
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| D | bandwidth_tests.cc | 57 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); in TEST_F()
108 RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); in ModifyVideoConfigs()
112 RtpExtension(RtpExtension::kTransportSequenceNumberUri, in ModifyVideoConfigs()
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| /external/webrtc/test/ |
| D | rtp_header_parser.cc | 31 bool RegisterRtpHeaderExtension(RtpExtension extension) override;
34 bool DeregisterRtpHeaderExtension(RtpExtension extension) override;
80 bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RtpExtension extension) { in RegisterRtpHeaderExtension()
91 bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) { in DeregisterRtpHeaderExtension()
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| /external/webrtc/audio/ |
| D | audio_receive_stream_unittest.cc | 116 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); in ConfigHelper()
117 stream_config_.rtp.extensions.push_back(RtpExtension( in ConfigHelper()
118 RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId)); in ConfigHelper()
229 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); in TEST()
400 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1)); in TEST()
402 RtpExtension(RtpExtension::kTransportSequenceNumberUri, in TEST()
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| /external/webrtc/test/pc/e2e/sdp/ |
| D | sdp_changer.cc | 135 if (extension.uri == RtpExtension::kMidUri) { in FillSimulcastContext()
137 } else if (extension.uri == RtpExtension::kRidUri) { in FillSimulcastContext()
139 } else if (extension.uri == RtpExtension::kRepairedRidUri) { in FillSimulcastContext()
226 std::vector<webrtc::RtpExtension> extensions = in PatchVp8Offer()
229 if (ext_it->uri == RtpExtension::kRidUri) { in PatchVp8Offer()
234 if (ext_it->uri == RtpExtension::kRepairedRidUri) { in PatchVp8Offer()
239 if (ext_it->uri == RtpExtension::kMidUri) { in PatchVp8Offer()
414 std::vector<webrtc::RtpExtension> extensions = in PatchVp8Answer()
418 [](const webrtc::RtpExtension& e) { in PatchVp8Answer()
419 return e.uri == RtpExtension::kMidUri || in PatchVp8Answer()
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| /external/webrtc/sdk/objc/api/peerconnection/ |
| D | RTCRtpHeaderExtension.mm | 25 - (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters { argument
34 - (webrtc::RtpExtension)nativeParameters {
35 webrtc::RtpExtension extension;
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| D | RTCRtpHeaderExtension+Private.h | 21 @property(nonatomic, readonly) webrtc::RtpExtension nativeParameters;
24 - (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters;
|