Files
platform-external-webrtc/video/payload_router.h
Erik Språng 566124a6df Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.

Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.

Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-23 15:31:27 +00:00

85 lines
2.7 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 VIDEO_PAYLOAD_ROUTER_H_
#define VIDEO_PAYLOAD_ROUTER_H_
#include <map>
#include <vector>
#include "api/video_codecs/video_encoder.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class RTPFragmentationHeader;
class RtpRtcp;
struct RTPVideoHeader;
// Currently only VP8/VP9 specific.
struct RtpPayloadState {
int16_t picture_id = -1;
uint8_t tl0_pic_idx = 0;
};
// PayloadRouter routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
class PayloadRouter : public EncodedImageCallback {
public:
// Rtp modules are assumed to be sorted in simulcast index order.
PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
const std::vector<uint32_t>& ssrcs,
int payload_type,
const std::map<uint32_t, RtpPayloadState>& states);
~PayloadRouter();
// PayloadRouter will only route packets if being active, all packets will be
// dropped otherwise.
void SetActive(bool active);
// Sets the sending status of the rtp modules and appropriately sets the
// payload router to active if any rtp modules are active.
void SetActiveModules(const std::vector<bool> active_modules);
bool IsActive();
std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
// 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;
void OnBitrateAllocationUpdated(const VideoBitrateAllocation& bitrate);
private:
class RtpPayloadParams;
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
rtc::CriticalSection crit_;
bool active_ RTC_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_;
std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_);
RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
};
} // namespace webrtc
#endif // VIDEO_PAYLOAD_ROUTER_H_