webrtc_m130/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"

#include <assert.h>  // assert
#include <string.h>  // memset

#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"

namespace webrtc {
namespace RTCPHelp {

RTCPPacketInformation::RTCPPacketInformation()
    : rtcpPacketTypeFlags(0),
      remoteSSRC(0),
      nackSequenceNumbers(),
      applicationSubType(0),
      applicationName(0),
      applicationData(),
      applicationLength(0),
      rtt(0),
      interArrivalJitter(0),
      sliPictureId(0),
      rpsiPictureId(0),
      receiverEstimatedMaxBitrate(0),
      ntp_secs(0),
      ntp_frac(0),
      rtp_timestamp(0),
      xr_originator_ssrc(0),
      xr_dlrr_item(false),
      VoIPMetric(nullptr) {}

RTCPPacketInformation::~RTCPPacketInformation()
{
    delete [] applicationData;
}

void
RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
{
    VoIPMetric.reset(new RTCPVoIPMetric());
    memcpy(VoIPMetric.get(), metric, sizeof(RTCPVoIPMetric));
}

void RTCPPacketInformation::AddApplicationData(const uint8_t* data,
                                               const uint16_t size) {
    uint8_t* oldData = applicationData;
    uint16_t oldLength = applicationLength;

    // Don't copy more than kRtcpAppCode_DATA_SIZE bytes.
    uint16_t copySize = size;
    if (size > kRtcpAppCode_DATA_SIZE) {
        copySize = kRtcpAppCode_DATA_SIZE;
    }

    applicationLength += copySize;
    applicationData = new uint8_t[applicationLength];

    if (oldData)
    {
        memcpy(applicationData, oldData, oldLength);
        memcpy(applicationData+oldLength, data, copySize);
        delete [] oldData;
    } else
    {
        memcpy(applicationData, data, copySize);
    }
}

void
RTCPPacketInformation::ResetNACKPacketIdArray()
{
  nackSequenceNumbers.clear();
}

void
RTCPPacketInformation::AddNACKPacket(const uint16_t packetID)
{
  if (nackSequenceNumbers.size() >= kSendSideNackListSizeSanity) {
    return;
  }
  nackSequenceNumbers.push_back(packetID);
}

void
RTCPPacketInformation::AddReportInfo(
    const RTCPReportBlockInformation& report_block_info)
{
  this->rtt = report_block_info.RTT;
  report_blocks.push_back(report_block_info.remoteReceiveBlock);
}

RTCPReportBlockInformation::RTCPReportBlockInformation():
    remoteReceiveBlock(),
    remoteMaxJitter(0),
    RTT(0),
    minRTT(0),
    maxRTT(0),
    avgRTT(0),
    numAverageCalcs(0)
{
    memset(&remoteReceiveBlock,0,sizeof(remoteReceiveBlock));
}

RTCPReportBlockInformation::~RTCPReportBlockInformation()
{
}

RTCPReceiveInformation::RTCPReceiveInformation() = default;
RTCPReceiveInformation::~RTCPReceiveInformation() = default;

void RTCPReceiveInformation::InsertTmmbrItem(uint32_t sender_ssrc,
                                             const rtcp::TmmbItem& tmmbr_item,
                                             int64_t current_time_ms) {
  TimedTmmbrItem* entry = &tmmbr_[sender_ssrc];
  entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc,
                                     tmmbr_item.bitrate_bps(),
                                     tmmbr_item.packet_overhead());
  entry->last_updated_ms = current_time_ms;
}

void RTCPReceiveInformation::GetTmmbrSet(
    int64_t current_time_ms,
    std::vector<rtcp::TmmbItem>* candidates) {
  // Use audio define since we don't know what interval the remote peer use.
  int64_t timeouted_ms = current_time_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
  for (auto it = tmmbr_.begin(); it != tmmbr_.end();) {
    if (it->second.last_updated_ms < timeouted_ms) {
      // Erase timeout entries.
      it = tmmbr_.erase(it);
    } else {
      candidates->push_back(it->second.tmmbr_item);
      ++it;
    }
  }
}

void RTCPReceiveInformation::ClearTmmbr() {
  tmmbr_.clear();
}

}  // namespace RTCPHelp
}  // namespace webrtc