| Index: media/cast/net/rtp_sender/rtp_sender.cc
|
| diff --git a/media/cast/net/rtp_sender/rtp_sender.cc b/media/cast/net/rtp_sender/rtp_sender.cc
|
| deleted file mode 100644
|
| index 2b017bc178442cc1e832ed3a0ea6e0e41b73b30f..0000000000000000000000000000000000000000
|
| --- a/media/cast/net/rtp_sender/rtp_sender.cc
|
| +++ /dev/null
|
| @@ -1,145 +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_sender.h"
|
| -
|
| -#include "base/logging.h"
|
| -#include "base/rand_util.h"
|
| -#include "media/cast/cast_defines.h"
|
| -#include "media/cast/net/pacing/paced_sender.h"
|
| -#include "media/cast/rtcp/rtcp_defines.h"
|
| -#include "net/base/big_endian.h"
|
| -
|
| -namespace media {
|
| -namespace cast {
|
| -
|
| -RtpSender::RtpSender(scoped_refptr<CastEnvironment> cast_environment,
|
| - const AudioSenderConfig* audio_config,
|
| - const VideoSenderConfig* video_config,
|
| - PacedPacketSender* transport)
|
| - : cast_environment_(cast_environment),
|
| - config_(),
|
| - transport_(transport) {
|
| - // Store generic cast config and create packetizer config.
|
| - DCHECK(audio_config || video_config) << "Invalid argument";
|
| - if (audio_config) {
|
| - storage_.reset(new PacketStorage(cast_environment->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(cast_environment->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) {
|
| - // Iterate over all frames in the list.
|
| - for (MissingFramesAndPacketsMap::const_iterator it =
|
| - missing_frames_and_packets.begin();
|
| - it != missing_frames_and_packets.end(); ++it) {
|
| - PacketList packets_to_resend;
|
| - uint8 frame_id = it->first;
|
| - const PacketIdSet& packets_set = it->second;
|
| - bool success = false;
|
| -
|
| - if (packets_set.empty()) {
|
| - VLOG(1) << "Missing all packets in frame " << static_cast<int>(frame_id);
|
| -
|
| - uint16 packet_id = 0;
|
| - do {
|
| - // Get packet from storage.
|
| - success = storage_->GetPacket(frame_id, packet_id, &packets_to_resend);
|
| -
|
| - // Resend packet to the network.
|
| - if (success) {
|
| - VLOG(1) << "Resend " << static_cast<int>(frame_id)
|
| - << ":" << packet_id;
|
| - // Set a unique incremental sequence number for every packet.
|
| - Packet& packet = packets_to_resend.back();
|
| - UpdateSequenceNumber(&packet);
|
| - // Set the size as correspond to each frame.
|
| - ++packet_id;
|
| - }
|
| - } while (success);
|
| - } else {
|
| - // Iterate over all of the packets in the frame.
|
| - for (PacketIdSet::const_iterator set_it = packets_set.begin();
|
| - set_it != packets_set.end(); ++set_it) {
|
| - uint16 packet_id = *set_it;
|
| - success = storage_->GetPacket(frame_id, packet_id, &packets_to_resend);
|
| -
|
| - // Resend packet to the network.
|
| - if (success) {
|
| - VLOG(1) << "Resend " << static_cast<int>(frame_id)
|
| - << ":" << packet_id;
|
| - Packet& packet = packets_to_resend.back();
|
| - 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
|
|
|