02b3d275a0
Reason for revert: A fix is being prepared downstream so this can now go in. Original issue's description: > Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ ) > > Reason for revert: > API changes broke downstream. > > Original issue's description: > > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. > > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport. > > EncodedImageCallback can of course be cleaned up in the future. > > > > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter. > > > > BUG=webrtc::5687 > > > > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb > > Cr-Commit-Position: refs/heads/master@{#12436} > > TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5687 > > Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4 > Cr-Commit-Position: refs/heads/master@{#12441} TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5687 Review URL: https://codereview.webrtc.org/1905583002 Cr-Commit-Position: refs/heads/master@{#12442}
77 lines
2.5 KiB
C++
77 lines
2.5 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 WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
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#define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
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#include <vector>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common_types.h"
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#include "webrtc/video_encoder.h"
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#include "webrtc/system_wrappers/include/atomic32.h"
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namespace webrtc {
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class RTPFragmentationHeader;
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class RtpRtcp;
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struct RTPVideoHeader;
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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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explicit PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
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int payload_type);
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~PayloadRouter();
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static size_t DefaultMaxPayloadLength();
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void SetSendingRtpModules(size_t num_sending_modules);
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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 set_active(bool active);
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bool active();
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// Implements EncodedImageCallback.
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// Returns 0 if the packet was routed / sent, -1 otherwise.
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int32_t Encoded(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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// Configures current target bitrate per module. 'stream_bitrates' is assumed
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// to be in the same order as 'SetSendingRtpModules'.
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void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
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// Returns the maximum allowed data payload length, given the configured MTU
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// and RTP headers.
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size_t MaxPayloadLength() const;
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private:
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void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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rtc::CriticalSection crit_;
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bool active_ GUARDED_BY(crit_);
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size_t num_sending_modules_ 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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RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
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