| Index: media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer.cc
|
| diff --git a/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer.cc b/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer.cc
|
| deleted file mode 100644
|
| index 8a50f8a8aad702d24dc12e0fa9090cffbcaa92f5..0000000000000000000000000000000000000000
|
| --- a/media/cast/net/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/net/rtp_sender/rtp_packetizer/rtp_packetizer.h"
|
| -
|
| -#include "base/logging.h"
|
| -#include "media/cast/cast_defines.h"
|
| -#include "media/cast/net/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
|
|
|