
Bug: webrtc:9517 Change-Id: I18397a28067dbe5029fc80fe2eef360869abb339 Reviewed-on: https://webrtc-review.googlesource.com/89380 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24039}
61 lines
2.3 KiB
C++
61 lines
2.3 KiB
C++
/*
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* Copyright (c) 2018 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 CALL_RTP_VIDEO_SENDER_INTERFACE_H_
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#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
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#include <map>
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#include <vector>
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#include "call/rtp_config.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/include/video_codec_interface.h"
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namespace webrtc {
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class VideoBitrateAllocation;
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struct FecProtectionParams;
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class RtpVideoSenderInterface : public EncodedImageCallback {
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public:
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virtual void RegisterProcessThread(ProcessThread* module_process_thread) = 0;
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virtual void DeRegisterProcessThread() = 0;
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// RtpVideoSender will only route packets if being active, all
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// packets will be dropped otherwise.
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virtual void SetActive(bool active) = 0;
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// Sets the sending status of the rtp modules and appropriately sets the
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// RtpVideoSender to active if any rtp modules are active.
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virtual void SetActiveModules(const std::vector<bool> active_modules) = 0;
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virtual bool IsActive() = 0;
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virtual void OnNetworkAvailability(bool network_available) = 0;
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virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
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virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
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virtual bool FecEnabled() const = 0;
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virtual bool NackEnabled() const = 0;
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virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 0;
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virtual void ProtectionRequest(const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params,
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uint32_t* sent_video_rate_bps,
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uint32_t* sent_nack_rate_bps,
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uint32_t* sent_fec_rate_bps) = 0;
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virtual void SetMaxRtpPacketSize(size_t max_rtp_packet_size) = 0;
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virtual void OnBitrateAllocationUpdated(
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const VideoBitrateAllocation& bitrate) = 0;
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};
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} // namespace webrtc
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#endif // CALL_RTP_VIDEO_SENDER_INTERFACE_H_
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