blob: 6fd848f27bf47108fcdd2b0974a2d3c76155d63e [file] [log] [blame]
// Copyright 2014 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/cast_transport_sender_impl.h"
#include "base/single_thread_task_runner.h"
#include "media/cast/transport/cast_transport_config.h"
#include "media/cast/transport/cast_transport_defines.h"
#include "net/base/net_util.h"
namespace media {
namespace cast {
namespace transport {
scoped_ptr<CastTransportSender> CastTransportSender::Create(
net::NetLog* net_log,
base::TickClock* clock,
const net::IPEndPoint& remote_end_point,
const CastTransportStatusCallback& status_callback,
const BulkRawEventsCallback& raw_events_callback,
base::TimeDelta raw_events_callback_interval,
const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner) {
return scoped_ptr<CastTransportSender>(
new CastTransportSenderImpl(net_log,
clock,
remote_end_point,
status_callback,
raw_events_callback,
raw_events_callback_interval,
transport_task_runner.get(),
NULL));
}
CastTransportSenderImpl::CastTransportSenderImpl(
net::NetLog* net_log,
base::TickClock* clock,
const net::IPEndPoint& remote_end_point,
const CastTransportStatusCallback& status_callback,
const BulkRawEventsCallback& raw_events_callback,
base::TimeDelta raw_events_callback_interval,
const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
PacketSender* external_transport)
: clock_(clock),
status_callback_(status_callback),
transport_task_runner_(transport_task_runner),
transport_(external_transport ? NULL
: new UdpTransport(net_log,
transport_task_runner,
net::IPEndPoint(),
remote_end_point,
status_callback)),
logging_(),
pacer_(clock,
&logging_,
external_transport ? external_transport : transport_.get(),
transport_task_runner),
rtcp_builder_(&pacer_),
raw_events_callback_(raw_events_callback) {
DCHECK(clock_);
if (!raw_events_callback_.is_null()) {
DCHECK(raw_events_callback_interval > base::TimeDelta());
event_subscriber_.reset(new SimpleEventSubscriber);
logging_.AddRawEventSubscriber(event_subscriber_.get());
raw_events_timer_.Start(FROM_HERE,
raw_events_callback_interval,
this,
&CastTransportSenderImpl::SendRawEvents);
}
if (transport_) {
// The default DSCP value for cast is AF41. Which gives it a higher
// priority over other traffic.
transport_->SetDscp(net::DSCP_AF41);
}
}
CastTransportSenderImpl::~CastTransportSenderImpl() {
if (event_subscriber_.get())
logging_.RemoveRawEventSubscriber(event_subscriber_.get());
}
void CastTransportSenderImpl::InitializeAudio(
const CastTransportAudioConfig& config) {
LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
config.rtp.config.aes_iv_mask.empty())
<< "Unsafe to send audio with encryption DISABLED.";
if (!audio_encryptor_.Initialize(config.rtp.config.aes_key,
config.rtp.config.aes_iv_mask)) {
status_callback_.Run(TRANSPORT_AUDIO_UNINITIALIZED);
return;
}
audio_sender_.reset(new RtpSender(clock_, transport_task_runner_, &pacer_));
if (audio_sender_->InitializeAudio(config)) {
pacer_.RegisterAudioSsrc(config.rtp.config.ssrc);
status_callback_.Run(TRANSPORT_AUDIO_INITIALIZED);
} else {
audio_sender_.reset();
status_callback_.Run(TRANSPORT_AUDIO_UNINITIALIZED);
}
}
void CastTransportSenderImpl::InitializeVideo(
const CastTransportVideoConfig& config) {
LOG_IF(WARNING, config.rtp.config.aes_key.empty() ||
config.rtp.config.aes_iv_mask.empty())
<< "Unsafe to send video with encryption DISABLED.";
if (!video_encryptor_.Initialize(config.rtp.config.aes_key,
config.rtp.config.aes_iv_mask)) {
status_callback_.Run(TRANSPORT_VIDEO_UNINITIALIZED);
return;
}
video_sender_.reset(new RtpSender(clock_, transport_task_runner_, &pacer_));
if (video_sender_->InitializeVideo(config)) {
pacer_.RegisterVideoSsrc(config.rtp.config.ssrc);
status_callback_.Run(TRANSPORT_VIDEO_INITIALIZED);
} else {
video_sender_.reset();
status_callback_.Run(TRANSPORT_VIDEO_UNINITIALIZED);
}
}
void CastTransportSenderImpl::SetPacketReceiver(
const PacketReceiverCallback& packet_receiver) {
transport_->StartReceiving(packet_receiver);
}
namespace {
void EncryptAndSendFrame(const EncodedFrame& frame,
TransportEncryptionHandler* encryptor,
RtpSender* sender) {
if (encryptor->initialized()) {
EncodedFrame encrypted_frame;
frame.CopyMetadataTo(&encrypted_frame);
if (encryptor->Encrypt(frame.frame_id, frame.data, &encrypted_frame.data)) {
sender->SendFrame(encrypted_frame);
} else {
LOG(ERROR) << "Encryption failed. Not sending frame with ID "
<< frame.frame_id;
}
} else {
sender->SendFrame(frame);
}
}
} // namespace
void CastTransportSenderImpl::InsertCodedAudioFrame(
const EncodedFrame& audio_frame) {
DCHECK(audio_sender_) << "Audio sender uninitialized";
EncryptAndSendFrame(audio_frame, &audio_encryptor_, audio_sender_.get());
}
void CastTransportSenderImpl::InsertCodedVideoFrame(
const EncodedFrame& video_frame) {
DCHECK(video_sender_) << "Video sender uninitialized";
EncryptAndSendFrame(video_frame, &video_encryptor_, video_sender_.get());
}
void CastTransportSenderImpl::SendRtcpFromRtpSender(
uint32 packet_type_flags,
uint32 ntp_seconds,
uint32 ntp_fraction,
uint32 rtp_timestamp,
const RtcpDlrrReportBlock& dlrr,
uint32 sending_ssrc,
const std::string& c_name) {
RtcpSenderInfo sender_info;
sender_info.ntp_seconds = ntp_seconds;
sender_info.ntp_fraction = ntp_fraction;
sender_info.rtp_timestamp = rtp_timestamp;
if (audio_sender_ && audio_sender_->ssrc() == sending_ssrc) {
sender_info.send_packet_count = audio_sender_->send_packet_count();
sender_info.send_octet_count = audio_sender_->send_octet_count();
} else if (video_sender_ && video_sender_->ssrc() == sending_ssrc) {
sender_info.send_packet_count = video_sender_->send_packet_count();
sender_info.send_octet_count = video_sender_->send_octet_count();
} else {
LOG(ERROR) << "Sending RTCP with an invalid SSRC.";
return;
}
rtcp_builder_.SendRtcpFromRtpSender(
packet_type_flags, sender_info, dlrr, sending_ssrc, c_name);
}
void CastTransportSenderImpl::ResendPackets(
bool is_audio,
const MissingFramesAndPacketsMap& missing_packets,
bool cancel_rtx_if_not_in_list,
base::TimeDelta dedupe_window) {
if (is_audio) {
DCHECK(audio_sender_) << "Audio sender uninitialized";
audio_sender_->ResendPackets(missing_packets,
cancel_rtx_if_not_in_list,
dedupe_window);
} else {
DCHECK(video_sender_) << "Video sender uninitialized";
video_sender_->ResendPackets(missing_packets,
cancel_rtx_if_not_in_list,
dedupe_window);
}
}
void CastTransportSenderImpl::SendRawEvents() {
DCHECK(event_subscriber_.get());
DCHECK(!raw_events_callback_.is_null());
std::vector<PacketEvent> packet_events;
event_subscriber_->GetPacketEventsAndReset(&packet_events);
raw_events_callback_.Run(packet_events);
}
} // namespace transport
} // namespace cast
} // namespace media