d8c4de7172
VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream. Therefore this cl: - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience. - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing. - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used. Bug: none Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38698}
70 lines
2.7 KiB
C++
70 lines
2.7 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 "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/call/bitrate_allocation.h"
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#include "api/fec_controller_override.h"
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#include "api/video/video_layers_allocation.h"
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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/rtp_rtcp/source/rtp_sequence_number_map.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 FecControllerOverride {
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public:
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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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// A module will only send packet if beeing active.
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virtual void SetActiveModules(const std::vector<bool>& active_modules) = 0;
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// Set the sending status of all rtp modules to inactive.
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virtual void Stop() = 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 void DeliverRtcp(const uint8_t* packet, size_t length) = 0;
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virtual void OnBitrateAllocationUpdated(
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const VideoBitrateAllocation& bitrate) = 0;
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virtual void OnVideoLayersAllocationUpdated(
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const VideoLayersAllocation& allocation) = 0;
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virtual void OnBitrateUpdated(BitrateAllocationUpdate update,
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int framerate) = 0;
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virtual void OnTransportOverheadChanged(
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size_t transport_overhead_bytes_per_packet) = 0;
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virtual uint32_t GetPayloadBitrateBps() const = 0;
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virtual uint32_t GetProtectionBitrateBps() const = 0;
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virtual void SetEncodingData(size_t width,
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size_t height,
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size_t num_temporal_layers) = 0;
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virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
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uint32_t ssrc,
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rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
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// Implements FecControllerOverride.
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void SetFecAllowed(bool fec_allowed) override = 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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