root/media/cast/transport/rtp_sender/rtp_packetizer/rtp_packetizer.cc

DEFINITIONS

1. channels
2. send_octet_count_
3. IncomingEncodedVideoFrame
4. IncomingEncodedAudioFrame
5. NextSequenceNumber
6. LastSentTimestamp
7. Cast
8. BuildCommonRTPheader

// 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/transport/rtp_sender/rtp_packetizer/rtp_packetizer.h"

#include "base/big_endian.h"
#include "base/logging.h"
#include "media/cast/transport/pacing/paced_sender.h"

namespace media {
namespace cast {
namespace transport {

static const uint16 kCommonRtpHeaderLength = 12;
static const uint16 kCastRtpHeaderLength = 7;
static const uint8 kCastKeyFrameBitMask = 0x80;
static const uint8 kCastReferenceFrameIdBitMask = 0x40;
static const uint8 kRtpMarkerBitMask = 0x80;

RtpPacketizerConfig::RtpPacketizerConfig()
    : audio(false),
      payload_type(-1),
      max_payload_length(kMaxIpPacketSize - 28),  // Default is IP-v4/UDP.
      sequence_number(0),
      frequency(8000),
      ssrc(0),
      channels(0) {}

RtpPacketizerConfig::~RtpPacketizerConfig() {}

RtpPacketizer::RtpPacketizer(PacedSender* const transport,
                             PacketStorage* packet_storage,
                             RtpPacketizerConfig rtp_packetizer_config,
                             base::TickClock* clock,
                             LoggingImpl* logging)
    : config_(rtp_packetizer_config),
      transport_(transport),
      packet_storage_(packet_storage),
      clock_(clock),
      logging_(logging),
      sequence_number_(config_.sequence_number),
      rtp_timestamp_(0),
      packet_id_(0),
      send_packets_count_(0),
      send_octet_count_(0) {
  DCHECK(transport) << "Invalid argument";
}

RtpPacketizer::~RtpPacketizer() {}

void RtpPacketizer::IncomingEncodedVideoFrame(
    const EncodedVideoFrame* video_frame,
    const base::TimeTicks& capture_time) {
  DCHECK(!config_.audio) << "Invalid state";
  if (config_.audio)
    return;

  time_last_sent_rtp_timestamp_ = capture_time;
  Cast(video_frame->key_frame,
       video_frame->frame_id,
       video_frame->last_referenced_frame_id,
       video_frame->rtp_timestamp,
       video_frame->data);
}

void RtpPacketizer::IncomingEncodedAudioFrame(
    const EncodedAudioFrame* audio_frame,
    const base::TimeTicks& recorded_time) {
  DCHECK(config_.audio) << "Invalid state";
  if (!config_.audio)
    return;

  time_last_sent_rtp_timestamp_ = recorded_time;
  Cast(true,
       audio_frame->frame_id,
       0,
       audio_frame->rtp_timestamp,
       audio_frame->data);
}

uint16 RtpPacketizer::NextSequenceNumber() {
  ++sequence_number_;
  return sequence_number_ - 1;
}

bool RtpPacketizer::LastSentTimestamp(base::TimeTicks* time_sent,
                                      uint32* rtp_timestamp) const {
  if (time_last_sent_rtp_timestamp_.is_null())
    return false;

  *time_sent = time_last_sent_rtp_timestamp_;
  *rtp_timestamp = rtp_timestamp_;
  return true;
}

// TODO(mikhal): Switch to pass data with a const_ref.
void RtpPacketizer::Cast(bool is_key,
                         uint32 frame_id,
                         uint32 reference_frame_id,
                         uint32 timestamp,
                         const std::string& data) {
  uint16 rtp_header_length = kCommonRtpHeaderLength + kCastRtpHeaderLength;
  uint16 max_length = config_.max_payload_length - rtp_header_length - 1;
  rtp_timestamp_ = timestamp;

  // Split the payload evenly (round number up).
  size_t num_packets = (data.size() + max_length) / max_length;
  size_t payload_length = (data.size() + num_packets) / num_packets;
  DCHECK_LE(payload_length, max_length) << "Invalid argument";

  PacketList packets;

  size_t remaining_size = data.size();
  std::string::const_iterator data_iter = data.begin();
  while (remaining_size > 0) {
    Packet packet;

    if (remaining_size < payload_length) {
      payload_length = remaining_size;
    }
    remaining_size -= payload_length;
    BuildCommonRTPheader(&packet, remaining_size == 0, timestamp);

    // Build Cast header.
    packet.push_back((is_key ? kCastKeyFrameBitMask : 0) |
                     kCastReferenceFrameIdBitMask);
    packet.push_back(frame_id);
    size_t start_size = packet.size();
    packet.resize(start_size + 4);
    base::BigEndianWriter big_endian_writer(
        reinterpret_cast<char*>(&(packet[start_size])), 4);
    big_endian_writer.WriteU16(packet_id_);
    big_endian_writer.WriteU16(static_cast<uint16>(num_packets - 1));
    packet.push_back(static_cast<uint8>(reference_frame_id));

    // Copy payload data.
    packet.insert(packet.end(), data_iter, data_iter + payload_length);

    // Store packet.
    packet_storage_->StorePacket(frame_id, packet_id_, &packet);
    ++packet_id_;
    data_iter += payload_length;

    // Update stats.
    ++send_packets_count_;
    send_octet_count_ += payload_length;
    packets.push_back(packet);
  }
  DCHECK(packet_id_ == num_packets) << "Invalid state";

  logging_->InsertPacketListEvent(
      clock_->NowTicks(),
      config_.audio ? kAudioPacketSentToPacer : kVideoPacketSentToPacer,
      packets);

  // Send to network.
  transport_->SendPackets(packets);

  // Prepare for next frame.
  packet_id_ = 0;
}

void RtpPacketizer::BuildCommonRTPheader(Packet* packet,
                                         bool marker_bit,
                                         uint32 time_stamp) {
  packet->push_back(0x80);
  packet->push_back(static_cast<uint8>(config_.payload_type) |
                    (marker_bit ? kRtpMarkerBitMask : 0));
  size_t start_size = packet->size();
  packet->resize(start_size + 10);
  base::BigEndianWriter big_endian_writer(
      reinterpret_cast<char*>(&((*packet)[start_size])), 10);
  big_endian_writer.WriteU16(sequence_number_);
  big_endian_writer.WriteU32(time_stamp);
  big_endian_writer.WriteU32(config_.ssrc);
  ++sequence_number_;
}

}  // namespace transport
}  // namespace cast
}  // namespace media