From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- .../libwebrtc/call/rtp_video_sender_interface.h | 69 ++++++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 third_party/libwebrtc/call/rtp_video_sender_interface.h (limited to 'third_party/libwebrtc/call/rtp_video_sender_interface.h') diff --git a/third_party/libwebrtc/call/rtp_video_sender_interface.h b/third_party/libwebrtc/call/rtp_video_sender_interface.h new file mode 100644 index 0000000000..3f2877155a --- /dev/null +++ b/third_party/libwebrtc/call/rtp_video_sender_interface.h @@ -0,0 +1,69 @@ +/* + * 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 +#include + +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "api/call/bitrate_allocation.h" +#include "api/fec_controller_override.h" +#include "api/video/video_layers_allocation.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" +#include "modules/video_coding/include/video_codec_interface.h" + +namespace webrtc { +class VideoBitrateAllocation; +struct FecProtectionParams; + +class RtpVideoSenderInterface : public EncodedImageCallback, + public FecControllerOverride { + public: + // Sets the sending status of the rtp modules and appropriately sets the + // RtpVideoSender to active if any rtp modules are active. + // A module will only send packet if beeing active. + virtual void SetActiveModules(const std::vector& active_modules) = 0; + // Set the sending status of all rtp modules to inactive. + virtual void Stop() = 0; + virtual bool IsActive() = 0; + + virtual void OnNetworkAvailability(bool network_available) = 0; + virtual std::map GetRtpStates() const = 0; + virtual std::map GetRtpPayloadStates() const = 0; + + virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 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 std::vector GetSentRtpPacketInfos( + uint32_t ssrc, + rtc::ArrayView sequence_numbers) const = 0; + + // Implements FecControllerOverride. + void SetFecAllowed(bool fec_allowed) override = 0; +}; +} // namespace webrtc +#endif // CALL_RTP_VIDEO_SENDER_INTERFACE_H_ -- cgit v1.2.3