mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
59ade0172f
This reverts commit75170be4ac. Reason for revert: Perf regression not affecting open source. Original change's description: > Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" > > This reverts commitd8c4de7172. > > Reason for revert: Tentative revert due to possible perf regression. b/260123362 > > Original change's description: > > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream > > > > 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} > > Bug: none > Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38725} Bug: b/260400659 Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900 Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38794}
69 lines
2.7 KiB
C++
69 lines
2.7 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 "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<bool>& 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<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 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<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
|
|
uint32_t ssrc,
|
|
rtc::ArrayView<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_
|