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// 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
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