/external/chromium_org/third_party/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],
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_payload_registry.cc | 32 RtpUtility::PayloadTypeMap::iterator it = payload_type_map_.begin(); 72 RtpUtility::PayloadTypeMap::iterator it = 77 RtpUtility::Payload* payload = it->second; 86 RtpUtility::StringCompare( 103 RtpUtility::Payload* payload = NULL; 106 if (RtpUtility::StringCompare(payload_name, "red", 3)) { 108 payload = new RtpUtility::Payload; 112 } else if (RtpUtility::StringCompare(payload_name, "ulpfec", 3)) { 114 payload = new RtpUtility::Payload; 135 RtpUtility::PayloadTypeMap::iterator it [all...] |
rtp_payload_registry_unittest.cc | 39 RtpUtility::Payload* ExpectReturnOfTypicalAudioPayload(uint8_t payload_type, 42 RtpUtility::Payload returned_payload = { 50 RtpUtility::Payload* returned_payload_on_heap = 51 new RtpUtility::Payload(returned_payload); 66 RtpUtility::Payload* returned_payload_on_heap = 76 RtpUtility::Payload* retrieved_payload = NULL; 103 RtpUtility::Payload* retrieved_payload = NULL; 115 RtpUtility::Payload* first_payload_on_heap = 125 RtpUtility::Payload* second_payload_on_heap = 133 RtpUtility::Payload* retrieved_payload = NULL [all...] |
rtp_receiver_video.cc | 117 RtpUtility::AssignUWord16ToBuffer(data_buffer + 2, 119 RtpUtility::AssignUWord32ToBuffer(data_buffer + 4, 121 RtpUtility::AssignUWord32ToBuffer(data_buffer + 8, rtp_header->header.ssrc); 133 RtpUtility::AssignUWord32ToBuffer(ptr, rtp_header->header.arrOfCSRCs[i]);
|
rtp_header_parser.cc | 46 RtpUtility::RtpHeaderParser rtp_parser(packet, length); 53 RtpUtility::RtpHeaderParser rtp_parser(packet, length);
|
rtcp_sender.cc | 626 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); 629 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPsec); 631 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPfrac); 633 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, RTPtime); 637 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, 642 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, 659 RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len); 685 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); 720 RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, SSRC); 751 RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, buffer_length) [all...] |
fec_test_helper.cc | 89 RtpUtility::AssignUWord16ToBuffer(data + 2, header->sequenceNumber); 90 RtpUtility::AssignUWord32ToBuffer(data + 4, header->timestamp); 91 RtpUtility::AssignUWord32ToBuffer(data + 8, header->ssrc);
|
rtp_sender.cc | 127 std::map<int8_t, RtpUtility::Payload*>::iterator it = 230 std::map<int8_t, RtpUtility::Payload*>::iterator it = 235 RtpUtility::Payload* payload = it->second; 239 if (RtpUtility::StringCompare( 256 RtpUtility::Payload* payload = NULL; 274 std::map<int8_t, RtpUtility::Payload*>::iterator it = 280 RtpUtility::Payload* payload = it->second; 387 std::map<int8_t, RtpUtility::Payload*>::iterator it = 394 RtpUtility::Payload* payload = it->second; 477 RtpUtility::RtpHeaderParser rtp_parser(buffer, length) [all...] |
rtp_sender_audio.cc | 92 RtpUtility::Payload*& payload) { 95 if (RtpUtility::StringCompare(payloadName, "cn", 2)) { 113 if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) { 120 payload = new RtpUtility::Payload; 393 RtpUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength, 441 RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); 563 RtpUtility::AssignUWord16ToBuffer(dtmfbuffer + 14, duration);
|
rtp_sender_video.cc | 80 RtpUtility::Payload*& payload) { 84 if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { 86 } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { 88 } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { 93 payload = new RtpUtility::Payload; 212 RtpUtility::AssignUWord32ToBuffer(data + 4, _rtpSender.SSRC());
|
rtp_utility.h | 31 namespace RtpUtility { 118 } // namespace RtpUtility
|
rtp_receiver_audio.h | 86 RtpUtility::PayloadTypeMap* payload_type_map,
|
rtp_sender_audio.h | 34 RtpUtility::Payload*& payload);
|
rtp_sender_video.h | 43 RtpUtility::Payload*& payload);
|
rtp_sender_unittest.cc | 208 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); 239 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); 280 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); 310 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); 348 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); 398 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); 475 webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, 537 webrtc::RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, 750 RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_, [all...] |
forward_error_correction.cc | 233 RtpUtility::AssignUWord16ToBuffer( 435 RtpUtility::AssignUWord16ToBuffer( 540 RtpUtility::BufferToUWord16(&fec_packet->pkt->data[2]); 653 RtpUtility::BufferToUWord16(protection_length)); 661 RtpUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8], fec_packet->ssrc); 670 RtpUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2], 674 RtpUtility::BufferToUWord16(recovered->length_recovery) + kRtpHeaderSize; 689 RtpUtility::AssignUWord16ToBuffer(media_payload_length, [all...] |
rtp_receiver_audio.cc | 162 if (RtpUtility::StringCompare(payload_name, "telephone-event", 15)) { 165 if (RtpUtility::StringCompare(payload_name, "cn", 2)) {
|
rtp_receiver_impl.cc | 25 using RtpUtility::GetCurrentRTP; 26 using RtpUtility::Payload; 27 using RtpUtility::StringCompare;
|
fec_receiver_impl.cc | 174 RtpUtility::BufferToUWord32(&incoming_rtp_packet[8]);
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
rtp_payload_registry.h | 28 virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, 33 virtual void UpdatePayloadRate(RtpUtility::Payload* payload, 36 virtual RtpUtility::Payload* CreatePayloadType( 44 const RtpUtility::Payload& payload) const = 0; 101 RtpUtility::Payload*& payload) const; 151 RtpUtility::PayloadTypeMap payload_type_map_;
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api_video.cc | 86 RtpUtility::AssignUWord16ToBuffer(dataBuffer + 2, sequence_number); 87 RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp); 88 RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, 0x1234); // SSRC.
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testFec/ |
test_fec.cc | 262 RtpUtility::AssignUWord16ToBuffer(&mediaPacket->data[2], seqNum); 263 RtpUtility::AssignUWord32ToBuffer(&mediaPacket->data[4], 265 RtpUtility::AssignUWord32ToBuffer(&mediaPacket->data[8], ssrc); 304 RtpUtility::BufferToUWord16(&mediaPacket->data[2]);
|
/external/chromium_org/third_party/webrtc/test/ |
rtp_file_reader_unittest.cc | 72 RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
audio_coding_module_unittest.cc | 48 class RtpUtility { 50 RtpUtility(int samples_per_packet, uint8_t payload_type) 53 virtual ~RtpUtility() {} 122 : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) { 192 scoped_ptr<RtpUtility> rtp_utility_; [all...] |
audio_coding_module_unittest_oldapi.cc | 47 class RtpUtility { 49 RtpUtility(int samples_per_packet, uint8_t payload_type) 52 virtual ~RtpUtility() {} 122 rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), 187 scoped_ptr<RtpUtility> rtp_utility_; [all...] |