cc7125f240
When a simulcast stream is enabled or disabled, we want this state change to be reflected properly in the RtpRtcp modules. Each video send stream can contain multiple rtp_rtcp_modules pertaining to different simulcast streams. These modules are currently all turned on/off when the send stream is started and stopped. This change allows for individual modules to be turned on/off. This means if a module stops sending it will send a bye message, so the receiving side will not expect more frames to be sent when the stream is inactive and the encoder is no longer encoding/sending images. Bug: webrtc:8653 Change-Id: Ib6d00240f627b4ff1714646e847026f24c7c3aa4 Reviewed-on: https://webrtc-review.googlesource.com/42841 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21880}
80 lines
2.6 KiB
C++
80 lines
2.6 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef VIDEO_PAYLOAD_ROUTER_H_
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#define VIDEO_PAYLOAD_ROUTER_H_
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#include <map>
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#include <vector>
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#include "api/video_codecs/video_encoder.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class RTPFragmentationHeader;
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class RtpRtcp;
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struct RTPVideoHeader;
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// PayloadRouter routes outgoing data to the correct sending RTP module, based
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// on the simulcast layer in RTPVideoHeader.
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class PayloadRouter : public EncodedImageCallback {
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public:
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// Rtp modules are assumed to be sorted in simulcast index order.
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PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
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const std::vector<uint32_t>& ssrcs,
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int payload_type,
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const std::map<uint32_t, RtpPayloadState>& states);
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~PayloadRouter();
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// PayloadRouter will only route packets if being active, all packets will be
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// dropped otherwise.
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void SetActive(bool active);
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// Sets the sending status of the rtp modules and appropriately sets the
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// payload router to active if any rtp modules are active.
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void SetActiveModules(const std::vector<bool> active_modules);
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bool IsActive();
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std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
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// Implements EncodedImageCallback.
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// Returns 0 if the packet was routed / sent, -1 otherwise.
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EncodedImageCallback::Result OnEncodedImage(
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const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info,
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const RTPFragmentationHeader* fragmentation) override;
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void OnBitrateAllocationUpdated(const BitrateAllocation& bitrate);
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private:
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class RtpPayloadParams;
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void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
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rtc::CriticalSection crit_;
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bool active_ RTC_GUARDED_BY(crit_);
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// Rtp modules are assumed to be sorted in simulcast index order. Not owned.
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const std::vector<RtpRtcp*> rtp_modules_;
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const int payload_type_;
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const bool forced_fallback_enabled_;
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std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_);
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RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
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};
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} // namespace webrtc
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#endif // VIDEO_PAYLOAD_ROUTER_H_
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