78 lines
2.5 KiB
C
78 lines
2.5 KiB
C
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/*
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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/config.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 SetSendStreams(const std::vector<VideoStream>& streams);
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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.
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void SetTargetSendBitrate(uint32_t bitrate_bps);
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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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std::vector<VideoStream> streams_ 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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