dbdb3a0079
- Move PayloadRouter to RtpTransportControllerInterface. - Move RetransmissionLimiter inside RtpTransportControllerSend from VideoSendStreamImpl. - Move video RTP specifics into PayloadRouter, in particular ownership of the RTP modules. - PayloadRouter now contains all video specific RTP code, and will be renamed in a follow-up to VideoRtpSender. - Introduce VideoRtpSenderInterface. Bug: webrtc:9517 Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38 Reviewed-on: https://webrtc-review.googlesource.com/88240 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24009}
129 lines
4.8 KiB
C++
129 lines
4.8 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_PAYLOAD_ROUTER_H_
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#define CALL_PAYLOAD_ROUTER_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "api/call/transport.h"
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#include "api/video_codecs/video_encoder.h"
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#include "call/rtp_config.h"
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#include "call/rtp_payload_params.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "call/video_rtp_sender_interface.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/rtp_rtcp/include/flexfec_sender.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/rate_limiter.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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class RTPFragmentationHeader;
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class RtpRtcp;
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class RtpTransportControllerSendInterface;
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// PayloadRouter routes outgoing data to the correct sending RTP module, based
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// on the simulcast layer in RTPVideoHeader.
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class PayloadRouter : public VideoRtpSenderInterface {
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public:
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// Rtp modules are assumed to be sorted in simulcast index order.
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PayloadRouter(
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const std::vector<uint32_t>& ssrcs,
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std::map<uint32_t, RtpState> suspended_ssrcs,
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const std::map<uint32_t, RtpPayloadState>& states,
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const RtpConfig& rtp_config,
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const RtcpConfig& rtcp_config,
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Transport* send_transport,
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const RtpSenderObservers& observers,
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RtpTransportControllerSendInterface* transport,
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RtcEventLog* event_log,
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RateLimiter* retransmission_limiter); // move inside RtpTransport
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~PayloadRouter() override;
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// RegisterProcessThread register |module_process_thread| with those objects
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// that use it. Registration has to happen on the thread were
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// |module_process_thread| was created (libjingle's worker thread).
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// TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
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// maybe |worker_queue|.
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void RegisterProcessThread(ProcessThread* module_process_thread) override;
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void DeRegisterProcessThread() override;
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// PayloadRouter will only route packets if being active, all packets will be
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// dropped otherwise.
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void SetActive(bool active) override;
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// Sets the sending status of the rtp modules and appropriately sets the
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// payload router to active if any rtp modules are active.
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void SetActiveModules(const std::vector<bool> active_modules) override;
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bool IsActive() override;
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void OnNetworkAvailability(bool network_available) override;
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std::map<uint32_t, RtpState> GetRtpStates() const override;
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std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const override;
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bool FecEnabled() const override;
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bool NackEnabled() const override;
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void DeliverRtcp(const uint8_t* packet, size_t length) override;
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void ProtectionRequest(const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params,
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uint32_t* sent_video_rate_bps,
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uint32_t* sent_nack_rate_bps,
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uint32_t* sent_fec_rate_bps) override;
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void SetMaxRtpPacketSize(size_t max_rtp_packet_size) override;
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// Implements EncodedImageCallback.
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// Returns 0 if the packet was routed / sent, -1 otherwise.
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EncodedImageCallback::Result OnEncodedImage(
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const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info,
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const RTPFragmentationHeader* fragmentation) override;
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void OnBitrateAllocationUpdated(
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const VideoBitrateAllocation& bitrate) override;
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private:
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void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void ConfigureProtection(const RtpConfig& rtp_config);
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void ConfigureSsrcs(const RtpConfig& rtp_config);
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rtc::CriticalSection crit_;
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bool active_ RTC_GUARDED_BY(crit_);
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ProcessThread* module_process_thread_;
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rtc::ThreadChecker module_process_thread_checker_;
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std::map<uint32_t, RtpState> suspended_ssrcs_;
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std::unique_ptr<FlexfecSender> flexfec_sender_;
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// Rtp modules are assumed to be sorted in simulcast index order. Not owned.
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const std::vector<std::unique_ptr<RtpRtcp>> rtp_modules_;
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const RtpConfig rtp_config_;
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RtpTransportControllerSendInterface* const transport_;
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std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_);
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RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
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};
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} // namespace webrtc
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#endif // CALL_PAYLOAD_ROUTER_H_
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