| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| initial_delay_manager.cc | 35 const WebRtcRTPHeader& rtp_info,
45 rtp_info.header.payloadType != audio_payload_type_));
48 const RTPHeader* current_header = &rtp_info.header;
68 audio_payload_type_ = rtp_info.header.payloadType;
72 RecordLastPacket(rtp_info, receive_timestamp, type);
107 RecordLastPacket(rtp_info, receive_timestamp, type);
130 memcpy(&sync_stream->rtp_info, &rtp_info, sizeof(rtp_info));
131 sync_stream->rtp_info.header.payloadType = audio_payload_type_
[all...] |
| initial_delay_manager_unittest.cc | 32 void InitRtpInfo(WebRtcRTPHeader* rtp_info) {
33 memset(rtp_info, 0, sizeof(*rtp_info));
34 rtp_info->header.markerBit = false;
35 rtp_info->header.payloadType = kAudioPayloadType;
36 rtp_info->header.sequenceNumber = 1234;
37 rtp_info->header.timestamp = 0xFFFFFFFD; // Close to wrap around.
38 rtp_info->header.ssrc = 0x87654321; // Arbitrary.
39 rtp_info->header.numCSRCs = 0; // Arbitrary.
40 rtp_info->header.paddingLength = 0
[all...] |
| initial_delay_manager.h | 32 memset(&rtp_info, 0, sizeof(rtp_info));
38 WebRtcRTPHeader rtp_info; member in struct:webrtc::acm2::InitialDelayManager::SyncStream
91 void RecordLastPacket(const WebRtcRTPHeader& rtp_info,
|
| acm_receiver.cc | 852 neteq_->InsertSyncPacket(sync_stream->rtp_info,
854 ++sync_stream->rtp_info.header.sequenceNumber;
855 sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step;
|
| audio_coding_module_impl.h | 157 const WebRtcRTPHeader& rtp_info);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_unittest.cc | 210 WebRtcRTPHeader* rtp_info);
213 WebRtcRTPHeader* rtp_info,
406 WebRtcRTPHeader* rtp_info) {
407 rtp_info->header.sequenceNumber = frame_index;
408 rtp_info->header.timestamp = timestamp;
409 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
410 rtp_info->header.payloadType = 94; // PCM16b WB codec.
411 rtp_info->header.markerBit = 0;
416 WebRtcRTPHeader* rtp_info,
419 rtp_info->header.sequenceNumber = frame_index
460 WebRtcRTPHeader rtp_info; local
579 WebRtcRTPHeader rtp_info; local
620 WebRtcRTPHeader rtp_info; local
653 WebRtcRTPHeader rtp_info; local
684 WebRtcRTPHeader rtp_info; local
727 WebRtcRTPHeader rtp_info; local
753 WebRtcRTPHeader rtp_info; local
800 WebRtcRTPHeader rtp_info; local
816 WebRtcRTPHeader rtp_info; local
926 WebRtcRTPHeader rtp_info; local
937 WebRtcRTPHeader rtp_info; local
1017 WebRtcRTPHeader rtp_info; local
1099 WebRtcRTPHeader rtp_info; local
1178 WebRtcRTPHeader rtp_info; local
1275 WebRtcRTPHeader rtp_info; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| video_coding_robustness_unittest.cc | 63 WebRtcRTPHeader rtp_info; local
64 memset(&rtp_info, 0, sizeof(rtp_info));
65 rtp_info.frameType = frame_type;
66 rtp_info.header.timestamp = timestamp;
67 rtp_info.header.sequenceNumber = seq_no;
68 rtp_info.header.markerBit = marker_bit;
69 rtp_info.header.payloadType = video_codec_.plType;
70 rtp_info.type.Video.codec = kRtpVideoVp8;
71 rtp_info.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8()
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| TestAllCodecs.cc | 59 WebRtcRTPHeader rtp_info; local
62 rtp_info.header.markerBit = false;
63 rtp_info.header.ssrc = 0;
64 rtp_info.header.sequenceNumber = sequence_number_++;
65 rtp_info.header.payloadType = payload_type;
66 rtp_info.header.timestamp = timestamp;
68 rtp_info.type.Audio.isCNG = true;
70 rtp_info.type.Audio.isCNG = false;
78 rtp_info.type.Audio.channel = 1;
81 status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
[all...] |
| TestStereo.cc | 53 WebRtcRTPHeader rtp_info; local
56 rtp_info.header.markerBit = false;
57 rtp_info.header.ssrc = 0;
58 rtp_info.header.sequenceNumber = seq_no_++;
59 rtp_info.header.payloadType = payload_type;
60 rtp_info.header.timestamp = timestamp;
68 rtp_info.type.Audio.isCNG = false;
69 rtp_info.type.Audio.channel = static_cast<int>(codec_mode_);
71 rtp_info.type.Audio.isCNG = true;
72 rtp_info.type.Audio.channel = static_cast<int>(kMono)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/interface/ |
| audio_coding_module.h | 661 // -rtp_info : the relevant information retrieved from RTP
670 const WebRtcRTPHeader& rtp_info) = 0;
[all...] |
