Cast:Moving netwrok sender related code to a designated folder

media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.cc

Index: media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.cc
diff --git a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.cc b/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.cc
deleted file mode 100644
index 6c5141d604f64f0f35e75fabc81a76d6dde3211a..0000000000000000000000000000000000000000
--- a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.cc
+++ /dev/null
@@ -1,153 +0,0 @@
-// 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_packetizer/rtp_packetizer.h"
-
-#include "base/logging.h"
-#include "media/cast/cast_defines.h"
-#include "media/cast/pacing/paced_sender.h"
-#include "net/base/big_endian.h"
-
-namespace media {
-namespace cast {
-
-static const uint16 kCommonRtpHeaderLength = 12;
-static const uint16 kCastRtpHeaderLength = 7;
-static const uint8 kCastKeyFrameBitMask = 0x80;
-static const uint8 kCastReferenceFrameIdBitMask = 0x40;
-
-RtpPacketizer::RtpPacketizer(PacedPacketSender* transport,
-                             PacketStorage* packet_storage,
-                             RtpPacketizerConfig rtp_packetizer_config)
-    : config_(rtp_packetizer_config),
-      transport_(transport),
-      packet_storage_(packet_storage),
-      sequence_number_(config_.sequence_number),
-      rtp_timestamp_(config_.rtp_timestamp),
-      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;
-
-  // Timestamp is in 90 KHz for video.
-  rtp_timestamp_ = GetVideoRtpTimestamp(capture_time);
-  time_last_sent_rtp_timestamp_ = capture_time;
-
-  Cast(video_frame->key_frame,
-       video_frame->frame_id,
-       video_frame->last_referenced_frame_id,
-       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;
-
-  rtp_timestamp_ += audio_frame->samples;  // Timestamp is in samples for audio.
-  time_last_sent_rtp_timestamp_ = recorded_time;
-  Cast(true, audio_frame->frame_id, 0, 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;
-
-  // 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);
-    net::BigEndianWriter big_endian_writer(&(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";
-
-  // 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);
-  net::BigEndianWriter big_endian_writer(&((*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 cast
-}  // namespace media