8b60e8bc34
When a LossNotification RTCP message is received, the sequence numbers it refers to must be converted to timestamps before passing the message down to the encoder. This CL gives VideoSendStreamImpl access to that information via VideoSendStreamImpl::rtp_video_sender_. TBR=sprang@webrtc.org Bug: webrtc:10501 Change-Id: If207f0b6d2fb344da35b525cc104e8ba5cc614ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131323 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27489}
66 lines
2.6 KiB
C++
66 lines
2.6 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 <map>
|
|
#include <vector>
|
|
|
|
#include "absl/types/optional.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/utility/include/process_thread.h"
|
|
#include "modules/video_coding/include/video_codec_interface.h"
|
|
|
|
namespace webrtc {
|
|
class VideoBitrateAllocation;
|
|
struct FecProtectionParams;
|
|
|
|
class RtpVideoSenderInterface : public EncodedImageCallback {
|
|
public:
|
|
virtual void RegisterProcessThread(ProcessThread* module_process_thread) = 0;
|
|
virtual void DeRegisterProcessThread() = 0;
|
|
|
|
// RtpVideoSender will only route packets if being active, all
|
|
// packets will be dropped otherwise.
|
|
virtual void SetActive(bool active) = 0;
|
|
// Sets the sending status of the rtp modules and appropriately sets the
|
|
// RtpVideoSender to active if any rtp modules are active.
|
|
virtual void SetActiveModules(const std::vector<bool> active_modules) = 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(const uint8_t* packet, size_t length) = 0;
|
|
|
|
virtual void OnBitrateAllocationUpdated(
|
|
const VideoBitrateAllocation& bitrate) = 0;
|
|
virtual void OnBitrateUpdated(uint32_t bitrate_bps,
|
|
uint8_t fraction_loss,
|
|
int64_t rtt,
|
|
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 absl::optional<RtpSequenceNumberMap::Info> GetSentRtpPacketInfo(
|
|
uint32_t ssrc,
|
|
uint16_t seq_num) const = 0;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // CALL_RTP_VIDEO_SENDER_INTERFACE_H_
|