diff options
Diffstat (limited to 'media/cast/rtp_sender/rtp_sender.cc')
| -rw-r--r-- | media/cast/rtp_sender/rtp_sender.cc | 166 |
1 files changed, 166 insertions, 0 deletions
diff --git a/media/cast/rtp_sender/rtp_sender.cc b/media/cast/rtp_sender/rtp_sender.cc new file mode 100644 index 0000000..c735dd0e --- /dev/null +++ b/media/cast/rtp_sender/rtp_sender.cc @@ -0,0 +1,166 @@ +// Copyright 2013 The Chromium Authors. All rights reserved. +// Use of this source code is governed by a BSD-style license that can be +// found in the LICENSE file. + +#include "media/cast/rtp_sender/rtp_sender.h" + +#include "base/logging.h" +#include "base/rand_util.h" +#include "media/cast/cast_defines.h" +#include "media/cast/pacing/paced_sender.h" +#include "media/cast/rtcp/rtcp_defines.h" + +namespace media { +namespace cast { +namespace { + +// January 1970, in milliseconds. +static const int64 kNtpJan1970 = 2208988800000LL; + +// Magic fractional unit. +static const uint32 kMagicFractionalUnit = 4294967; + +void ConvertTimeToFractions(int64 time_ms, uint32* seconds, + uint32* fractions) { + *seconds = static_cast<uint32>(time_ms / 1000); + *fractions = static_cast<uint32>((time_ms % 1000) * kMagicFractionalUnit); +} + +void ConvertTimeToNtp(int64 time_ms, uint32* ntp_seconds, + uint32* ntp_fractions) { + ConvertTimeToFractions(time_ms + kNtpJan1970, ntp_seconds, ntp_fractions); +} +} // namespace + +RtpSender::RtpSender(const AudioSenderConfig* audio_config, + const VideoSenderConfig* video_config, + PacedPacketSender* transport) + : config_(), + transport_(transport), + default_tick_clock_(new base::DefaultTickClock()), + clock_(default_tick_clock_.get()) { + // Store generic cast config and create packetizer config. + DCHECK(audio_config || video_config) << "Invalid argument"; + if (audio_config) { + storage_.reset(new PacketStorage(audio_config->rtp_history_ms)); + config_.audio = true; + config_.ssrc = audio_config->sender_ssrc; + config_.payload_type = audio_config->rtp_payload_type; + config_.frequency = audio_config->frequency; + config_.audio_codec = audio_config->codec; + } else { + storage_.reset(new PacketStorage(video_config->rtp_history_ms)); + config_.audio = false; + config_.ssrc = video_config->sender_ssrc; + config_.payload_type = video_config->rtp_payload_type; + config_.frequency = kVideoFrequency; + config_.video_codec = video_config->codec; + } + // Randomly set start values. + config_.sequence_number = base::RandInt(0, 65535); + config_.rtp_timestamp = base::RandInt(0, 65535); + config_.rtp_timestamp += base::RandInt(0, 65535) << 16; + packetizer_.reset(new RtpPacketizer(transport, storage_.get(), config_)); +} + +RtpSender::~RtpSender() {} + +void RtpSender::IncomingEncodedVideoFrame(const EncodedVideoFrame& video_frame, + int64 capture_time) { + packetizer_->IncomingEncodedVideoFrame(video_frame, capture_time); +} + +void RtpSender::IncomingEncodedAudioFrame(const EncodedAudioFrame& audio_frame, + int64 recorded_time) { + packetizer_->IncomingEncodedAudioFrame(audio_frame, recorded_time); +} + +void RtpSender::ResendPackets( + const MissingFramesAndPackets& missing_frames_and_packets) { + std::vector<uint8> packet; + // Iterate over all frames in the list. + for (std::map<uint8, std::set<uint16> >::const_iterator it = + missing_frames_and_packets.begin(); + it != missing_frames_and_packets.end(); ++it) { + uint8 frame_id = it->first; + // Iterate over all of the packets in the frame. + const std::set<uint16>& packets = it->second; + if (packets.empty()) { + VLOG(1) << "Missing all packets in frame " << static_cast<int>(frame_id); + + bool success = false; + uint16 packet_id = 0; + do { + // Get packet from storage. + packet.clear(); + success = storage_->GetPacket(frame_id, packet_id, &packet); + + // Resend packet to the network. + if (success) { + VLOG(1) << "Resend " << static_cast<int>(frame_id) << ":" + << packet_id << " size: " << packets.size(); + // Set a unique incremental sequence number for every packet. + UpdateSequenceNumber(&packet); + // Set the size as correspond to each frame. + transport_->ResendPacket(packet, packets.size()); + ++packet_id; + } + } while (success); + + + } else { + for (std::set<uint16>::const_iterator set_it = packets.begin(); + set_it != packets.end(); ++set_it) { + uint16 packet_id = *set_it; + // Get packet from storage. + packet.clear(); + bool success = storage_->GetPacket(frame_id, packet_id, &packet); + // Resend packet to the network. + if (success) { + VLOG(1) << "Resend " << static_cast<int>(frame_id) << ":" + << packet_id << " size: " << packet.size(); + UpdateSequenceNumber(&packet); + // Set the size as correspond to each frame. + transport_->ResendPacket(packet, packets.size()); + } else { + VLOG(1) << "Failed to resend " << static_cast<int>(frame_id) << ":" + << packet_id; + } + } + } + } +} + +void RtpSender::UpdateSequenceNumber(std::vector<uint8>* packet) { + uint16 new_sequence_number = packetizer_->NextSequenceNumber(); + int index = 2; + (*packet)[index] = (static_cast<uint8>(new_sequence_number)); + (*packet)[index + 1] =(static_cast<uint8>(new_sequence_number >> 8)); +} + +void RtpSender::RtpStatistics(int64 now_ms, RtcpSenderInfo* sender_info) { + // The timestamp of this Rtcp packet should be estimated as the timestamp of + // the frame being captured at this moment. We are calculating that + // timestamp as the last frame's timestamp + the time since the last frame + // was captured. + uint32 ntp_seconds = 0; + uint32 ntp_fraction = 0; + ConvertTimeToNtp(now_ms, &ntp_seconds, &ntp_fraction); + // sender_info->ntp_seconds = ntp_seconds; + sender_info->ntp_fraction = ntp_fraction; + + int64 time_sent_ms; + uint32 rtp_timestamp; + if (packetizer_->LastSentTimestamp(&time_sent_ms, &rtp_timestamp)) { + int64 time_since_last_send_ms = now_ms - time_sent_ms; + sender_info->rtp_timestamp = rtp_timestamp + + time_since_last_send_ms * (config_.frequency / 1000); + } else { + sender_info->rtp_timestamp = 0; + } + sender_info->send_packet_count = packetizer_->send_packets_count(); + sender_info->send_octet_count = packetizer_->send_octet_count(); +} + +} // namespace cast +} // namespace media |