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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 {
RtpSender::RtpSender(base::TickClock* clock,
const AudioSenderConfig* audio_config,
const VideoSenderConfig* video_config,
PacedPacketSender* transport)
: config_(),
transport_(transport),
clock_(clock) {
// Store generic cast config and create packetizer config.
DCHECK(audio_config || video_config) << "Invalid argument";
if (audio_config) {
storage_.reset(new PacketStorage(clock, 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(clock, 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,
const base::TimeTicks& capture_time) {
packetizer_->IncomingEncodedVideoFrame(video_frame, capture_time);
}
void RtpSender::IncomingEncodedAudioFrame(const EncodedAudioFrame* audio_frame,
const base::TimeTicks& recorded_time) {
packetizer_->IncomingEncodedAudioFrame(audio_frame, recorded_time);
}
void RtpSender::ResendPackets(
const MissingFramesAndPacketsMap& missing_frames_and_packets) {
PacketList packets_to_resend =
storage_->GetPackets(missing_frames_and_packets);
PacketList::iterator it = packets_to_resend.begin();
for (; it != packets_to_resend.end(); ++it) {
Packet& packet = *it;
UpdateSequenceNumber(&packet);
}
transport_->ResendPackets(packets_to_resend);
}
void RtpSender::UpdateSequenceNumber(Packet* 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(const base::TimeTicks& now,
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;
ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction);
sender_info->ntp_seconds = ntp_seconds;
sender_info->ntp_fraction = ntp_fraction;
base::TimeTicks time_sent;
uint32 rtp_timestamp;
if (packetizer_->LastSentTimestamp(&time_sent, &rtp_timestamp)) {
base::TimeDelta time_since_last_send = now - time_sent;
sender_info->rtp_timestamp = rtp_timestamp +
time_since_last_send.InMilliseconds() * (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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