| /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],
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| /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]);
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| rtp_header_parser.cc | 46 RtpUtility::RtpHeaderParser rtp_parser(packet, length);
53 RtpUtility::RtpHeaderParser rtp_parser(packet, length);
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| 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);
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| 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);
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| 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());
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| rtp_utility.h | 31 namespace RtpUtility {
118 } // namespace RtpUtility
|
| rtp_receiver_audio.h | 86 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 | 43 RtpUtility::Payload*& payload);
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| 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;
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| 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...] |
