| /* |
| * 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; |
| |
| // 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); |
| 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 BitrateAllocation& 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_; |
| |
| const bool forced_fallback_enabled_; |
| std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_); |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // VIDEO_PAYLOAD_ROUTER_H_ |