| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| rtp_generator.cc | 20 WebRtcRTPHeader* rtp_header) {
21 assert(rtp_header);
22 if (!rtp_header) {
25 rtp_header->header.sequenceNumber = seq_number_++;
26 rtp_header->header.timestamp = timestamp_;
28 rtp_header->header.payloadType = payload_type;
29 rtp_header->header.markerBit = false;
30 rtp_header->header.ssrc = ssrc_;
31 rtp_header->header.numCSRCs = 0;
32 rtp_header->frameType = kAudioFrameSpeech
[all...] |
| rtp_generator.h | 39 // Writes the next RTP header to |rtp_header|, which will be of type
44 WebRtcRTPHeader* rtp_header);
73 WebRtcRTPHeader* rtp_header) override;
|
| neteq_external_decoder_test.cc | 39 WebRtcRTPHeader rtp_header,
43 neteq_->InsertPacket(rtp_header, payload, receive_timestamp));
|
| neteq_performance_test.cc | 59 WebRtcRTPHeader rtp_header; local
65 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
82 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
87 neteq->InsertPacket(rtp_header, input_payload,
96 &rtp_header);
|
| neteq_external_decoder_test.h | 37 // Inserts a new packet with |rtp_header| and |payload| of
41 virtual void InsertPacket(WebRtcRTPHeader rtp_header,
|
| neteq_rtpplay.cc | 302 WebRtcRTPHeader* rtp_header,
306 if (IsComfortNoise(rtp_header->header.payloadType)) {
318 rtp_header->header.sequenceNumber + 1) {
320 next_packet->header().timestamp - rtp_header->header.timestamp) {
322 next_packet->header().timestamp - rtp_header->header.timestamp;
331 if (CodecTimestampRate(rtp_header->header.payloadType) !=
332 CodecSampleRate(rtp_header->header.payloadType) ||
333 rtp_header->header.payloadType == FLAGS_red ||
334 rtp_header->header.payloadType == FLAGS_avt) {
356 switch (CodecSampleRate(rtp_header->header.payloadType))
546 WebRtcRTPHeader rtp_header; local
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_receiver_video.cc | 52 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
60 "seqnum", rtp_header->header.sequenceNumber, "timestamp",
61 rtp_header->header.timestamp);
62 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
66 payload_length - rtp_header->header.paddingLength;
69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
75 RtpDepacketizer::Create(rtp_header->type.Video.codec));
81 rtp_header->type.Video.isFirstPacket = is_first_packet;
86 rtp_header->frameType = parsed_payload.frame_type
[all...] |
| rtp_sender_unittest.cc | 58 const uint8_t* GetPayloadData(const RTPHeader& rtp_header,
60 return packet + rtp_header.headerLength;
63 size_t GetPayloadDataLength(const RTPHeader& rtp_header,
65 return packet_length - rtp_header.headerLength - rtp_header.paddingLength;
150 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
151 VerifyRTPHeaderCommon(rtp_header, kMarkerBit);
154 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) {
155 EXPECT_EQ(marker_bit, rtp_header.markerBit);
156 EXPECT_EQ(payload_, rtp_header.payloadType)
202 webrtc::RTPHeader rtp_header; local
336 webrtc::RTPHeader rtp_header; local
368 webrtc::RTPHeader rtp_header; local
408 webrtc::RTPHeader rtp_header; local
436 webrtc::RTPHeader rtp_header; local
477 webrtc::RTPHeader rtp_header; local
505 webrtc::RTPHeader rtp_header; local
525 webrtc::RTPHeader rtp_header; local
579 webrtc::RTPHeader rtp_header; local
665 webrtc::RTPHeader rtp_header; local
725 webrtc::RTPHeader rtp_header; local
770 webrtc::RTPHeader rtp_header; local
936 webrtc::RTPHeader rtp_header; local
1219 webrtc::RTPHeader rtp_header; local
1248 webrtc::RTPHeader rtp_header; local
1305 webrtc::RTPHeader rtp_header; local
[all...] |
| rtp_receiver_impl.cc | 161 const RTPHeader& rtp_header,
167 CheckSSRCChanged(rtp_header);
172 if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
184 webrtc_rtp_header.header = rtp_header;
187 size_t payload_data_length = payload_length - rtp_header.paddingLength;
194 last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
195 last_received_timestamp_ != rtp_header.timestamp;
216 if (last_received_timestamp_ != rtp_header.timestamp) {
217 last_received_timestamp_ = rtp_header.timestamp;
220 last_received_sequence_number_ = rtp_header.sequenceNumber
[all...] |
| rtp_receiver_audio.cc | 181 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
189 "seqnum", rtp_header->header.sequenceNumber, "timestamp",
190 rtp_header->header.timestamp);
191 rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs;
192 num_energy_ = rtp_header->type.Audio.numEnergy;
193 if (rtp_header->type.Audio.numEnergy > 0 &&
194 rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) {
196 rtp_header->type.Audio.arrOfEnergy,
197 rtp_header->type.Audio.numEnergy)
[all...] |
| rtp_receiver_impl.h | 44 bool IncomingRtpPacket(const RTPHeader& rtp_header,
70 void CheckSSRCChanged(const RTPHeader& rtp_header);
71 void CheckCSRC(const WebRtcRTPHeader& rtp_header);
72 int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
|
| rtp_sender.cc | 581 RTPHeader rtp_header; local
582 rtp_parser.Parse(&rtp_header);
583 bytes_left -= static_cast<int>(length - rtp_header.headerLength);
672 RTPHeader rtp_header; local
673 rtp_parser.Parse(&rtp_header);
677 padding_packet, length, rtp_header, now_ms - capture_time_ms);
680 UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
685 UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
696 UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
914 RTPHeader rtp_header; local
1033 RTPHeader rtp_header; local
1828 RTPHeader rtp_header; local
[all...] |
| rtp_sender.h | 82 const RTPHeader& rtp_header,
194 const RTPHeader& rtp_header,
201 const RTPHeader& rtp_header,
207 const RTPHeader& rtp_header,
362 const RTPHeader& rtp_header,
367 const RTPHeader& rtp_header,
371 const RTPHeader& rtp_header,
378 const RTPHeader& rtp_header) const;
|
| fec_receiver_impl.h | 31 int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
|
| rtp_receiver_video.h | 29 int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_impl_unittest.cc | 267 WebRtcRTPHeader rtp_header; local
268 rtp_header.header.payloadType = kPayloadType;
269 rtp_header.header.sequenceNumber = kFirstSequenceNumber;
270 rtp_header.header.timestamp = kFirstTimestamp;
271 rtp_header.header.ssrc = kSsrc;
327 .WillOnce(Return(&rtp_header.header));
363 neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime);
366 rtp_header.header.timestamp += 160;
367 rtp_header.header.sequenceNumber += 1;
368 neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155)
380 WebRtcRTPHeader rtp_header; local
421 WebRtcRTPHeader rtp_header; local
515 WebRtcRTPHeader rtp_header; local
609 WebRtcRTPHeader rtp_header; local
676 WebRtcRTPHeader rtp_header; local
813 WebRtcRTPHeader rtp_header; local
908 WebRtcRTPHeader rtp_header; local
944 WebRtcRTPHeader rtp_header; local
1013 WebRtcRTPHeader rtp_header; local
1138 WebRtcRTPHeader rtp_header; local
[all...] |
| rtcp.cc | 33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
36 int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_;
38 if (rtp_header.sequenceNumber < max_seq_no_) {
42 max_seq_no_ = rtp_header.sequenceNumber;
48 int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
54 transit_ = rtp_header.timestamp - receive_timestamp;
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| loudest_filter.cc | 43 bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) {
47 int source_ssrc = rtp_header.ssrc;
48 int audio_level = rtp_header.extension.hasAudioLevel ?
49 rtp_header.extension.audioLevel : kInvalidAudioLevel;
|
| loudest_filter.h | 26 * rtp_header : Header of the RTP packet of interest.
28 bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header);
|
| conference_transport.cc | 150 webrtc::RTPHeader rtp_header; local
151 rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
152 if (rtp_header.ssrc == kLocalSsrc) {
156 if (loudest_filter_.ForwardThisPacket(rtp_header)) {
157 destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
|
| /external/webrtc/webrtc/modules/video_coding/test/ |
| receiver_tests.h | 33 const webrtc::WebRtcRTPHeader* rtp_header) override {
34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
|
| /external/webrtc/webrtc/video/ |
| vie_receiver.cc | 237 const WebRtcRTPHeader* rtp_header) {
238 WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
240 ntp_estimator_->Estimate(rtp_header->header.timestamp);
391 WebRtcRTPHeader rtp_header = {}; local
392 rtp_header.header = header;
393 rtp_header.header.payloadType = last_media_payload_type;
394 rtp_header.header.paddingLength = 0;
401 rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
402 rtp_header.type.Video.rotation = kVideoRotation_0;
404 rtp_header.type.Video.rotation
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.h | 60 const webrtc::WebRtcRTPHeader* rtp_header) override;
64 webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; } function in class:webrtc::TestRtpReceiver
|
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| receive_statistics.h | 56 virtual void IncomingPacket(const RTPHeader& rtp_header,
85 void IncomingPacket(const RTPHeader& rtp_header,
|
| fec_receiver.h | 36 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
|
