blob: 17bc279290733c242ca2fa9fcd3aa56c04154532 [file] [log] [blame]
* 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.
#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/include/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 {
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; } }
// 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_;
} // namespace webrtc