| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_payload_registry.cc | 33 RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.begin(); in ~RTPPayloadRegistry()
73 RtpUtility::PayloadTypeMap::iterator it = in RegisterReceivePayload()
78 RtpUtility::Payload* payload = it->second; in RegisterReceivePayload()
87 RtpUtility::StringCompare( in RegisterReceivePayload()
105 RtpUtility::Payload* payload = rtp_payload_strategy_->CreatePayloadType( in RegisterReceivePayload()
111 if (RtpUtility::StringCompare(payload_name, "red", 3)) { in RegisterReceivePayload()
113 } else if (RtpUtility::StringCompare(payload_name, "ulpfec", 6)) { in RegisterReceivePayload()
127 RtpUtility::PayloadTypeMap::iterator it = in DeRegisterReceivePayload()
144 RtpUtility::PayloadTypeMap::iterator iterator = payload_type_map_.begin(); in DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType()
146 RtpUtility::Payload* payload = iterator->second; in DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType()
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| D | rtp_payload_registry_unittest.cc | 41 RtpUtility::Payload* ExpectReturnOfTypicalAudioPayload(uint8_t payload_type, in ExpectReturnOfTypicalAudioPayload()
44 RtpUtility::Payload returned_payload = { in ExpectReturnOfTypicalAudioPayload()
52 RtpUtility::Payload* returned_payload_on_heap = in ExpectReturnOfTypicalAudioPayload()
53 new RtpUtility::Payload(returned_payload); in ExpectReturnOfTypicalAudioPayload()
67 RtpUtility::Payload* returned_payload_on_heap = in TEST_F()
77 const RtpUtility::Payload* retrieved_payload = in TEST_F()
108 const RtpUtility::Payload* retrieved_payload = in TEST_F()
124 RtpUtility::Payload* first_payload_on_heap = in TEST_F()
135 RtpUtility::Payload* second_payload_on_heap = in TEST_F()
143 const RtpUtility::Payload* retrieved_payload = in TEST_F()
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| D | rtp_sender_video.cc | 73 RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload( in CreateVideoPayload()
78 if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { in CreateVideoPayload()
80 } else if (RtpUtility::StringCompare(payloadName, "VP9", 3)) { in CreateVideoPayload()
82 } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { in CreateVideoPayload()
84 } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { in CreateVideoPayload()
89 RtpUtility::Payload* payload = new RtpUtility::Payload(); in CreateVideoPayload()
305 RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); in SendVideo()
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| D | rtp_sender_unittest.cc | 200 webrtc::RtpUtility::RtpHeaderParser rtp_parser(data, len); in VerifyCVOPacket()
242 EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + in TEST_F()
255 RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength), in TEST_F()
267 EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + in TEST_F()
273 EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + in TEST_F()
279 EXPECT_EQ(RtpUtility::Word32Align( in TEST_F()
286 EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + in TEST_F()
295 EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + in TEST_F()
301 EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength + in TEST_F()
307 RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength), in TEST_F()
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| D | rtp_sender_audio.cc | 71 RtpUtility::Payload** payload) { in RegisterAudioPayload()
72 if (RtpUtility::StringCompare(payloadName, "cn", 2)) { in RegisterAudioPayload()
91 } else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) { in RegisterAudioPayload()
99 *payload = new RtpUtility::Payload; in RegisterAudioPayload()
351 RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); in SendAudio()
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| D | rtp_header_parser.cc | 45 RtpUtility::RtpHeaderParser rtp_parser(packet, length); in IsRtcp()
52 RtpUtility::RtpHeaderParser rtp_parser(packet, length); in Parse()
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| D | rtp_sender.cc | 201 std::map<int8_t, RtpUtility::Payload*>::iterator it = in ~RTPSender()
304 std::map<int8_t, RtpUtility::Payload*>::iterator it = in RegisterPayload()
309 RtpUtility::Payload* payload = it->second; in RegisterPayload()
313 if (RtpUtility::StringCompare( in RegisterPayload()
330 RtpUtility::Payload* payload = nullptr; in RegisterPayload()
347 std::map<int8_t, RtpUtility::Payload*>::iterator it = in DeRegisterSendPayload()
353 RtpUtility::Payload* payload = it->second; in DeRegisterSendPayload()
473 std::map<int8_t, RtpUtility::Payload*>::iterator it = in CheckPayloadType()
481 RtpUtility::Payload* payload = it->second; in CheckPayloadType()
580 RtpUtility::RtpHeaderParser rtp_parser(buffer, length); in TrySendRedundantPayloads()
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| D | rtp_utility.h | 32 namespace RtpUtility {
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| D | rtp_receiver_audio.cc | 160 if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) { in OnNewPayloadTypeCreated()
163 if (RtpUtility::StringCompare(payload_name, "cn", 2)) { in OnNewPayloadTypeCreated()
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| D | rtp_receiver_audio.h | 81 RtpUtility::PayloadTypeMap* payload_type_map,
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| D | rtp_sender_audio.h | 34 RtpUtility::Payload** payload);
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| D | rtp_sender_video.h | 42 static RtpUtility::Payload* CreateVideoPayload(
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| D | rtp_receiver_impl.cc | 25 using RtpUtility::Payload;
26 using RtpUtility::StringCompare;
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| D | rtp_header_extension.cc | 149 length = RtpUtility::Word32Align(length); in GetTotalLengthInBytes()
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| D | rtp_utility.cc | 40 namespace RtpUtility { namespace
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| D | rtp_sender.h | 411 std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
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| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| D | rtp_payload_registry.h | 30 virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
35 virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
38 virtual RtpUtility::Payload* CreatePayloadType(
46 const RtpUtility::Payload& payload) const = 0;
117 RtpUtility::Payload*& payload) const { // NOLINT in PayloadTypeToPayload()
119 const_cast<RtpUtility::Payload*>(PayloadTypeToPayload(payload_type)); in PayloadTypeToPayload()
122 const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
182 RtpUtility::PayloadTypeMap payload_type_map_;
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/mock/ |
| D | mock_rtp_payload_strategy.h | 24 bool(const RtpUtility::Payload& payload,
29 void(RtpUtility::Payload* payload, const uint32_t rate));
31 int(const RtpUtility::Payload& payload));
34 RtpUtility::Payload*(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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| /external/webrtc/webrtc/test/ |
| D | layer_filtering_transport.cc | 48 RtpUtility::RtpHeaderParser parser(packet, length); in SendRtp()
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| D | rtp_file_reader_unittest.cc | 86 RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length); in CountRtpPacketsPerSsrc()
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| D | rtp_file_reader.cc | 456 RtpUtility::RtpHeaderParser rtp_parser(read_buffer_, marker.payload_length); in ReadPacket()
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| D | audio_coding_module_unittest_oldapi.cc | 60 class RtpUtility { class
62 RtpUtility(int samples_per_packet, uint8_t payload_type) in RtpUtility() function in webrtc::RtpUtility
65 virtual ~RtpUtility() {} in ~RtpUtility()
158 rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), in AudioCodingModuleTestOldApi()
229 rtc::scoped_ptr<RtpUtility> rtp_utility_;
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| /external/webrtc/webrtc/video/ |
| D | video_quality_test.cc | 117 RtpUtility::RtpHeaderParser parser(packet, length); in DeliverPacket()
153 RtpUtility::RtpHeaderParser parser(packet, length); in SendRtp()
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