f70446874a
This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd. Reason for revert: Reland by removing the conflict with the broken CL. Original change's description: > Revert "Move allocation and rtp conversion logic out of payload router." > > This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7. > > Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220 > > This causes a merge conflict. So need to revert this first. > > Original change's description: > > Move allocation and rtp conversion logic out of payload router. > > > > Makes it easier to write tests, and allows for moving rtp module > > ownership into the payload router in the future. > > > > The RtpPayloadParams class is split into declaration and definition and > > moved into separate files. > > > > Bug: webrtc:9517 > > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad > > Reviewed-on: https://webrtc-review.googlesource.com/88564 > > Commit-Queue: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#23983} > > TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9517 > Reviewed-on: https://webrtc-review.googlesource.com/88821 > Reviewed-by: JT Teh <jtteh@webrtc.org> > Commit-Queue: JT Teh <jtteh@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23991} TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9517 Reviewed-on: https://webrtc-review.googlesource.com/89020 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23996}
78 lines
2.6 KiB
C++
78 lines
2.6 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 <vector>
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#include "api/video_codecs/video_encoder.h"
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#include "call/rtp_payload_params.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/rtp_rtcp/source/rtp_video_header.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/thread_annotations.h"
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namespace webrtc {
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class RTPFragmentationHeader;
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class RtpRtcp;
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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 EncodedImageCallback {
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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(const std::vector<RtpRtcp*>& rtp_modules,
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const std::vector<uint32_t>& ssrcs,
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int payload_type,
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const std::map<uint32_t, RtpPayloadState>& states);
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~PayloadRouter() 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);
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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);
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bool IsActive();
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std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
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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(const VideoBitrateAllocation& bitrate);
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private:
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void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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rtc::CriticalSection crit_;
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bool active_ RTC_GUARDED_BY(crit_);
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// Rtp modules are assumed to be sorted in simulcast index order. Not owned.
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const std::vector<RtpRtcp*> rtp_modules_;
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const int payload_type_;
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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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