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