e590416722
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out. BUG=4323 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45549004 Cr-Commit-Position: refs/heads/master@{#8864}
86 lines
2.8 KiB
C++
86 lines
2.8 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 WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
|
|
#define WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
|
|
|
|
#include <list>
|
|
#include <vector>
|
|
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/base/scoped_ptr.h"
|
|
#include "webrtc/base/thread_annotations.h"
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/system_wrappers/interface/atomic32.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class CriticalSectionWrapper;
|
|
class RTPFragmentationHeader;
|
|
class RtpRtcp;
|
|
struct RTPVideoHeader;
|
|
|
|
// PayloadRouter routes outgoing data to the correct sending RTP module, based
|
|
// on the simulcast layer in RTPVideoHeader.
|
|
class PayloadRouter {
|
|
public:
|
|
PayloadRouter();
|
|
~PayloadRouter();
|
|
|
|
static size_t DefaultMaxPayloadLength();
|
|
|
|
// Rtp modules are assumed to be sorted in simulcast index order.
|
|
void SetSendingRtpModules(const std::list<RtpRtcp*>& rtp_modules);
|
|
|
|
// PayloadRouter will only route packets if being active, all packets will be
|
|
// dropped otherwise.
|
|
void set_active(bool active);
|
|
bool active();
|
|
|
|
// Input parameters according to the signature of RtpRtcp::SendOutgoingData.
|
|
// Returns true if the packet was routed / sent, false otherwise.
|
|
bool RoutePayload(FrameType frame_type,
|
|
int8_t payload_type,
|
|
uint32_t time_stamp,
|
|
int64_t capture_time_ms,
|
|
const uint8_t* payload_data,
|
|
size_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation,
|
|
const RTPVideoHeader* rtp_video_hdr);
|
|
|
|
// Configures current target bitrate per module. 'stream_bitrates' is assumed
|
|
// to be in the same order as 'SetSendingRtpModules'.
|
|
void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
|
|
|
|
// Returns the maximum allowed data payload length, given the configured MTU
|
|
// and RTP headers.
|
|
size_t MaxPayloadLength() const;
|
|
|
|
void AddRef() { ++ref_count_; }
|
|
void Release() { if (--ref_count_ == 0) { delete this; } }
|
|
|
|
private:
|
|
// TODO(mflodman): When the new video API has launched, remove crit_ and
|
|
// assume rtp_modules_ will never change during a call.
|
|
rtc::scoped_ptr<CriticalSectionWrapper> crit_;
|
|
|
|
// Active sending RTP modules, in layer order.
|
|
std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
|
|
bool active_ GUARDED_BY(crit_.get());
|
|
|
|
Atomic32 ref_count_;
|
|
|
|
DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
|