1 // Copyright 2013 The Chromium Authors. All rights reserved. 2 // Use of this source code is governed by a BSD-style license that can be 3 // found in the LICENSE file. 4 5 #include "media/cast/net/rtp_sender/rtp_sender.h" 6 7 #include "base/logging.h" 8 #include "base/rand_util.h" 9 #include "media/cast/cast_defines.h" 10 #include "media/cast/net/pacing/paced_sender.h" 11 #include "media/cast/rtcp/rtcp_defines.h" 12 #include "net/base/big_endian.h" 13 14 namespace media { 15 namespace cast { 16 17 RtpSender::RtpSender(scoped_refptr<CastEnvironment> cast_environment, 18 const AudioSenderConfig* audio_config, 19 const VideoSenderConfig* video_config, 20 PacedPacketSender* transport) 21 : cast_environment_(cast_environment), 22 config_(), 23 transport_(transport) { 24 // Store generic cast config and create packetizer config. 25 DCHECK(audio_config || video_config) << "Invalid argument"; 26 if (audio_config) { 27 storage_.reset(new PacketStorage(cast_environment->Clock(), 28 audio_config->rtp_history_ms)); 29 config_.audio = true; 30 config_.ssrc = audio_config->sender_ssrc; 31 config_.payload_type = audio_config->rtp_payload_type; 32 config_.frequency = audio_config->frequency; 33 config_.audio_codec = audio_config->codec; 34 } else { 35 storage_.reset(new PacketStorage(cast_environment->Clock(), 36 video_config->rtp_history_ms)); 37 config_.audio = false; 38 config_.ssrc = video_config->sender_ssrc; 39 config_.payload_type = video_config->rtp_payload_type; 40 config_.frequency = kVideoFrequency; 41 config_.video_codec = video_config->codec; 42 } 43 // Randomly set start values. 44 config_.sequence_number = base::RandInt(0, 65535); 45 config_.rtp_timestamp = base::RandInt(0, 65535); 46 config_.rtp_timestamp += base::RandInt(0, 65535) << 16; 47 packetizer_.reset(new RtpPacketizer(transport, storage_.get(), config_)); 48 } 49 50 RtpSender::~RtpSender() {} 51 52 void RtpSender::IncomingEncodedVideoFrame(const EncodedVideoFrame* video_frame, 53 const base::TimeTicks& capture_time) { 54 packetizer_->IncomingEncodedVideoFrame(video_frame, capture_time); 55 } 56 57 void RtpSender::IncomingEncodedAudioFrame(const EncodedAudioFrame* audio_frame, 58 const base::TimeTicks& recorded_time) { 59 packetizer_->IncomingEncodedAudioFrame(audio_frame, recorded_time); 60 } 61 62 void RtpSender::ResendPackets( 63 const MissingFramesAndPacketsMap& missing_frames_and_packets) { 64 // Iterate over all frames in the list. 65 for (MissingFramesAndPacketsMap::const_iterator it = 66 missing_frames_and_packets.begin(); 67 it != missing_frames_and_packets.end(); ++it) { 68 PacketList packets_to_resend; 69 uint8 frame_id = it->first; 70 const PacketIdSet& packets_set = it->second; 71 bool success = false; 72 73 if (packets_set.empty()) { 74 VLOG(1) << "Missing all packets in frame " << static_cast<int>(frame_id); 75 76 uint16 packet_id = 0; 77 do { 78 // Get packet from storage. 79 success = storage_->GetPacket(frame_id, packet_id, &packets_to_resend); 80 81 // Resend packet to the network. 82 if (success) { 83 VLOG(1) << "Resend " << static_cast<int>(frame_id) 84 << ":" << packet_id; 85 // Set a unique incremental sequence number for every packet. 86 Packet& packet = packets_to_resend.back(); 87 UpdateSequenceNumber(&packet); 88 // Set the size as correspond to each frame. 89 ++packet_id; 90 } 91 } while (success); 92 } else { 93 // Iterate over all of the packets in the frame. 94 for (PacketIdSet::const_iterator set_it = packets_set.begin(); 95 set_it != packets_set.end(); ++set_it) { 96 uint16 packet_id = *set_it; 97 success = storage_->GetPacket(frame_id, packet_id, &packets_to_resend); 98 99 // Resend packet to the network. 100 if (success) { 101 VLOG(1) << "Resend " << static_cast<int>(frame_id) 102 << ":" << packet_id; 103 Packet& packet = packets_to_resend.back(); 104 UpdateSequenceNumber(&packet); 105 } 106 } 107 } 108 transport_->ResendPackets(packets_to_resend); 109 } 110 } 111 112 void RtpSender::UpdateSequenceNumber(Packet* packet) { 113 uint16 new_sequence_number = packetizer_->NextSequenceNumber(); 114 int index = 2; 115 (*packet)[index] = (static_cast<uint8>(new_sequence_number)); 116 (*packet)[index + 1] =(static_cast<uint8>(new_sequence_number >> 8)); 117 } 118 119 void RtpSender::RtpStatistics(const base::TimeTicks& now, 120 RtcpSenderInfo* sender_info) { 121 // The timestamp of this Rtcp packet should be estimated as the timestamp of 122 // the frame being captured at this moment. We are calculating that 123 // timestamp as the last frame's timestamp + the time since the last frame 124 // was captured. 125 uint32 ntp_seconds = 0; 126 uint32 ntp_fraction = 0; 127 ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction); 128 sender_info->ntp_seconds = ntp_seconds; 129 sender_info->ntp_fraction = ntp_fraction; 130 131 base::TimeTicks time_sent; 132 uint32 rtp_timestamp; 133 if (packetizer_->LastSentTimestamp(&time_sent, &rtp_timestamp)) { 134 base::TimeDelta time_since_last_send = now - time_sent; 135 sender_info->rtp_timestamp = rtp_timestamp + 136 time_since_last_send.InMilliseconds() * (config_.frequency / 1000); 137 } else { 138 sender_info->rtp_timestamp = 0; 139 } 140 sender_info->send_packet_count = packetizer_->send_packets_count(); 141 sender_info->send_octet_count = packetizer_->send_octet_count(); 142 } 143 144 } // namespace cast 145 } // namespace media 146