blob: cbfbe55297ee7ed101e3918335fd5e00dbe9e9ec [file] [log] [blame]
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "call/flexfec_receive_stream_impl.h"
#include <stddef.h>
#include <cstdint>
#include <string>
#include <utility>
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/rtp_parameters.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/rtp_rtcp/include/flexfec_receiver.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
std::string FlexfecReceiveStream::Config::ToString() const {
char buf[1024];
rtc::SimpleStringBuilder ss(buf);
ss << "{payload_type: " << payload_type;
ss << ", remote_ssrc: " << rtp.remote_ssrc;
ss << ", local_ssrc: " << rtp.local_ssrc;
ss << ", protected_media_ssrcs: [";
size_t i = 0;
for (; i + 1 < protected_media_ssrcs.size(); ++i)
ss << protected_media_ssrcs[i] << ", ";
if (!protected_media_ssrcs.empty())
ss << protected_media_ssrcs[i];
ss << "}";
return ss.str();
}
bool FlexfecReceiveStream::Config::IsCompleteAndEnabled() const {
// Check if FlexFEC is enabled.
if (payload_type < 0)
return false;
// Do we have the necessary SSRC information?
if (rtp.remote_ssrc == 0)
return false;
// TODO(brandtr): Update this check when we support multistream protection.
if (protected_media_ssrcs.size() != 1u)
return false;
return true;
}
namespace {
// TODO(brandtr): Update this function when we support multistream protection.
std::unique_ptr<FlexfecReceiver> MaybeCreateFlexfecReceiver(
Clock* clock,
const FlexfecReceiveStream::Config& config,
RecoveredPacketReceiver* recovered_packet_receiver) {
if (config.payload_type < 0) {
RTC_LOG(LS_WARNING)
<< "Invalid FlexFEC payload type given. "
"This FlexfecReceiveStream will therefore be useless.";
return nullptr;
}
RTC_DCHECK_GE(config.payload_type, 0);
RTC_DCHECK_LE(config.payload_type, 127);
if (config.rtp.remote_ssrc == 0) {
RTC_LOG(LS_WARNING)
<< "Invalid FlexFEC SSRC given. "
"This FlexfecReceiveStream will therefore be useless.";
return nullptr;
}
if (config.protected_media_ssrcs.empty()) {
RTC_LOG(LS_WARNING)
<< "No protected media SSRC supplied. "
"This FlexfecReceiveStream will therefore be useless.";
return nullptr;
}
if (config.protected_media_ssrcs.size() > 1) {
RTC_LOG(LS_WARNING)
<< "The supplied FlexfecConfig contained multiple protected "
"media streams, but our implementation currently only "
"supports protecting a single media stream. "
"To avoid confusion, disabling FlexFEC completely.";
return nullptr;
}
RTC_DCHECK_EQ(1U, config.protected_media_ssrcs.size());
return std::unique_ptr<FlexfecReceiver>(new FlexfecReceiver(
clock, config.rtp.remote_ssrc, config.protected_media_ssrcs[0],
recovered_packet_receiver));
}
std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
Clock* clock,
ReceiveStatistics* receive_statistics,
const FlexfecReceiveStreamImpl::Config& config,
RtcpRttStats* rtt_stats) {
RtpRtcpInterface::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = true;
configuration.clock = clock;
configuration.receive_statistics = receive_statistics;
configuration.outgoing_transport = config.rtcp_send_transport;
configuration.rtt_stats = rtt_stats;
configuration.local_media_ssrc = config.rtp.local_ssrc;
return ModuleRtpRtcpImpl2::Create(configuration);
}
} // namespace
FlexfecReceiveStreamImpl::FlexfecReceiveStreamImpl(
Clock* clock,
Config config,
RecoveredPacketReceiver* recovered_packet_receiver,
RtcpRttStats* rtt_stats)
: remote_ssrc_(config.rtp.remote_ssrc),
payload_type_(config.payload_type),
receiver_(
MaybeCreateFlexfecReceiver(clock, config, recovered_packet_receiver)),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
rtp_rtcp_(CreateRtpRtcpModule(clock,
rtp_receive_statistics_.get(),
config,
rtt_stats)) {
RTC_LOG(LS_INFO) << "FlexfecReceiveStreamImpl: " << config.ToString();
RTC_DCHECK_GE(payload_type_, -1);
packet_sequence_checker_.Detach();
// RTCP reporting.
rtp_rtcp_->SetRTCPStatus(config.rtcp_mode);
}
FlexfecReceiveStreamImpl::~FlexfecReceiveStreamImpl() {
RTC_DLOG(LS_INFO) << "~FlexfecReceiveStreamImpl: ssrc: " << remote_ssrc_;
}
void FlexfecReceiveStreamImpl::RegisterWithTransport(
RtpStreamReceiverControllerInterface* receiver_controller) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(!rtp_stream_receiver_);
if (!receiver_)
return;
// TODO(nisse): OnRtpPacket in this class delegates all real work to
// `receiver_`. So maybe we don't need to implement RtpPacketSinkInterface
// here at all, we'd then delete the OnRtpPacket method and instead register
// `receiver_` as the RtpPacketSinkInterface for this stream.
rtp_stream_receiver_ =
receiver_controller->CreateReceiver(remote_ssrc(), this);
}
void FlexfecReceiveStreamImpl::UnregisterFromTransport() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_stream_receiver_.reset();
}
void FlexfecReceiveStreamImpl::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (!receiver_)
return;
receiver_->OnRtpPacket(packet);
// Do not report media packets in the RTCP RRs generated by `rtp_rtcp_`.
if (packet.Ssrc() == remote_ssrc()) {
rtp_receive_statistics_->OnRtpPacket(packet);
}
}
void FlexfecReceiveStreamImpl::SetPayloadType(int payload_type) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_GE(payload_type, -1);
payload_type_ = payload_type;
}
int FlexfecReceiveStreamImpl::payload_type() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return payload_type_;
}
void FlexfecReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (local_ssrc == rtp_rtcp_->local_media_ssrc())
return;
rtp_rtcp_->SetLocalSsrc(local_ssrc);
}
} // namespace webrtc