blob: 6426b00615f329b21821e9a3c6dea95c3daa466e [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 <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/video_encoder.h"
#include "webrtc/system_wrappers/include/atomic32.h"
namespace webrtc {
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 EncodedImageCallback {
// Rtp modules are assumed to be sorted in simulcast index order.
PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
int payload_type);
static size_t DefaultMaxPayloadLength();
// PayloadRouter will only route packets if being active, all packets will be
// dropped otherwise.
void set_active(bool active);
bool active();
// Implements EncodedImageCallback.
// Returns 0 if the packet was routed / sent, -1 otherwise.
EncodedImageCallback::Result OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) override;
// Returns the maximum allowed data payload length, given the configured MTU
// and RTP headers.
size_t MaxPayloadLength() const;
void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_);
rtc::CriticalSection crit_;
bool active_ GUARDED_BY(crit_);
// Rtp modules are assumed to be sorted in simulcast index order. Not owned.
const std::vector<RtpRtcp*> rtp_modules_;
const int payload_type_;
} // namespace webrtc