webrtc/modules/pacing/packet_router.h
Erik Språng 58ee187554 Add support within PacedSender and pacer queue for owning rtp packets.
This CL builds on https://webrtc-review.googlesource.com/c/src/+/142165
It adds the parts within the paced sender that uses those send methods.
A follow-up will add the pre-pacer RTP sender parts. That CL will also
add proper integration testing. Here, I mostly add coverage for the new
send methods. When the old code-path is removed, all tests need to be
converted to exclusively use the owned path.

Bug: webrtc:10633
Change-Id: I870d9a2285f07a7b7b0ef6758aa310808f210f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142179
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28308}
2019-06-18 15:02:19 +00:00

131 lines
5.1 KiB
C++

/*
* Copyright (c) 2015 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.
*/
#ifndef MODULES_PACING_PACKET_ROUTER_H_
#define MODULES_PACING_PACKET_ROUTER_H_
#include <stddef.h>
#include <stdint.h>
#include <list>
#include <memory>
#include <vector>
#include "api/transport/network_types.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class RtpRtcp;
namespace rtcp {
class TransportFeedback;
} // namespace rtcp
// PacketRouter keeps track of rtp send modules to support the pacer.
// In addition, it handles feedback messages, which are sent on a send
// module if possible (sender report), otherwise on receive module
// (receiver report). For the latter case, we also keep track of the
// receive modules.
class PacketRouter : public TransportSequenceNumberAllocator,
public RemoteBitrateObserver,
public TransportFeedbackSenderInterface {
public:
PacketRouter();
~PacketRouter() override;
void AddSendRtpModule(RtpRtcp* rtp_module, bool remb_candidate);
void RemoveSendRtpModule(RtpRtcp* rtp_module);
void AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
bool remb_candidate);
void RemoveReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender);
virtual RtpPacketSendResult TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission,
const PacedPacketInfo& packet_info);
virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info);
virtual size_t TimeToSendPadding(size_t bytes,
const PacedPacketInfo& packet_info);
void SetTransportWideSequenceNumber(uint16_t sequence_number);
uint16_t AllocateSequenceNumber() override;
// Called every time there is a new bitrate estimate for a receive channel
// group. This call will trigger a new RTCP REMB packet if the bitrate
// estimate has decreased or if no RTCP REMB packet has been sent for
// a certain time interval.
// Implements RtpReceiveBitrateUpdate.
void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
uint32_t bitrate_bps) override;
// Ensures remote party notified of the receive bitrate limit no larger than
// |bitrate_bps|.
void SetMaxDesiredReceiveBitrate(int64_t bitrate_bps);
// Send REMB feedback.
bool SendRemb(int64_t bitrate_bps, const std::vector<uint32_t>& ssrcs);
// Send transport feedback packet to send-side.
bool SendTransportFeedback(rtcp::TransportFeedback* packet) override;
private:
void AddRembModuleCandidate(RtcpFeedbackSenderInterface* candidate_module,
bool media_sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
void MaybeRemoveRembModuleCandidate(
RtcpFeedbackSenderInterface* candidate_module,
bool media_sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
void UnsetActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
void DetermineActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
rtc::CriticalSection modules_crit_;
// Rtp and Rtcp modules of the rtp senders.
std::list<RtpRtcp*> rtp_send_modules_ RTC_GUARDED_BY(modules_crit_);
// The last module used to send media.
RtpRtcp* last_send_module_ RTC_GUARDED_BY(modules_crit_);
// Rtcp modules of the rtp receivers.
std::vector<RtcpFeedbackSenderInterface*> rtcp_feedback_senders_
RTC_GUARDED_BY(modules_crit_);
// TODO(eladalon): remb_crit_ only ever held from one function, and it's not
// clear if that function can actually be called from more than one thread.
rtc::CriticalSection remb_crit_;
// The last time a REMB was sent.
int64_t last_remb_time_ms_ RTC_GUARDED_BY(remb_crit_);
int64_t last_send_bitrate_bps_ RTC_GUARDED_BY(remb_crit_);
// The last bitrate update.
int64_t bitrate_bps_ RTC_GUARDED_BY(remb_crit_);
int64_t max_bitrate_bps_ RTC_GUARDED_BY(remb_crit_);
// Candidates for the REMB module can be RTP sender/receiver modules, with
// the sender modules taking precedence.
std::vector<RtcpFeedbackSenderInterface*> sender_remb_candidates_
RTC_GUARDED_BY(modules_crit_);
std::vector<RtcpFeedbackSenderInterface*> receiver_remb_candidates_
RTC_GUARDED_BY(modules_crit_);
RtcpFeedbackSenderInterface* active_remb_module_
RTC_GUARDED_BY(modules_crit_);
volatile int transport_seq_;
RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
};
} // namespace webrtc
#endif // MODULES_PACING_PACKET_ROUTER_H_