webrtc/call/rtp_video_sender_interface.h
Danil Chapovalov c9752917bd Replace ArrayView with std::span everywhere except api
ArrayView is an alias to std::span. This change switch to use
std::span directly instead of through the alias.

Search&Replace MakeArrayView and ArrayView with std::span
Search&Replace include "api/array_view.h" with include <span>
Remove <span> include where std::span is not mentioned in the file
Remove build dependencies on array_view target

Bug: webrtc:439801349
Change-Id: I64bbdcf1300126286158ccbeda62c49e0c3ec83f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460501
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47294}
2026-03-30 02:05:06 -07:00

68 lines
2.5 KiB
C++

/*
* Copyright (c) 2018 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 CALL_RTP_VIDEO_SENDER_INTERFACE_H_
#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
#include <cstddef>
#include <cstdint>
#include <map>
#include <span>
#include <vector>
#include "api/call/bitrate_allocation.h"
#include "api/fec_controller_override.h"
#include "api/video/video_layers_allocation.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
namespace webrtc {
class VideoBitrateAllocation;
struct FecProtectionParams;
class RtpVideoSenderInterface : public EncodedImageCallback,
public FecControllerOverride {
public:
// Sets weather or not RTP packets is allowed to be sent on this sender.
virtual void SetSending(bool enabled) = 0;
virtual bool IsActive() = 0;
virtual void OnNetworkAvailability(bool network_available) = 0;
virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
virtual void DeliverRtcp(std::span<const uint8_t> packet) = 0;
virtual void OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) = 0;
virtual void OnVideoLayersAllocationUpdated(
const VideoLayersAllocation& allocation) = 0;
virtual void OnBitrateUpdated(BitrateAllocationUpdate update,
int framerate) = 0;
virtual void OnTransportOverheadChanged(
size_t transport_overhead_bytes_per_packet) = 0;
virtual uint32_t GetPayloadBitrateBps() const = 0;
virtual uint32_t GetProtectionBitrateBps() const = 0;
virtual void SetEncodingData(size_t width,
size_t height,
size_t num_temporal_layers) = 0;
virtual void SetCsrcs(std::span<const uint32_t> csrcs) = 0;
virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
uint32_t ssrc,
std::span<const uint16_t> sequence_numbers) const = 0;
// Implements FecControllerOverride.
void SetFecAllowed(bool fec_allowed) override = 0;
};
} // namespace webrtc
#endif // CALL_RTP_VIDEO_SENDER_INTERFACE_H_