Files
platform-external-webrtc/video/video_send_stream.h
Per Kjellander 59ade0172f Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880.

Reason for revert: Perf regression not affecting open source.

Original change's description:
> Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
>
> This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc.
>
> 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}
2022-12-02 12:03:25 +00:00

127 lines
4.3 KiB
C++

/*
* Copyright (c) 2013 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 VIDEO_VIDEO_SEND_STREAM_H_
#define VIDEO_VIDEO_SEND_STREAM_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "call/bitrate_allocator.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "modules/utility/maybe_worker_thread.h"
#include "rtc_base/event.h"
#include "rtc_base/system/no_unique_address.h"
#include "video/encoder_rtcp_feedback.h"
#include "video/send_delay_stats.h"
#include "video/send_statistics_proxy.h"
#include "video/video_send_stream_impl.h"
#include "video/video_stream_encoder_interface.h"
namespace webrtc {
namespace test {
class VideoSendStreamPeer;
} // namespace test
class CallStats;
class IvfFileWriter;
class RateLimiter;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class RtcEventLog;
namespace internal {
class VideoSendStreamImpl;
// VideoSendStream implements webrtc::VideoSendStream.
// Internally, it delegates all public methods to VideoSendStreamImpl and / or
// VideoStreamEncoder.
class VideoSendStream : public webrtc::VideoSendStream {
public:
using RtpStateMap = std::map<uint32_t, RtpState>;
using RtpPayloadStateMap = std::map<uint32_t, RtpPayloadState>;
VideoSendStream(
Clock* clock,
int num_cpu_cores,
TaskQueueFactory* task_queue_factory,
TaskQueueBase* network_queue,
RtcpRttStats* call_stats,
RtpTransportControllerSendInterface* transport,
BitrateAllocatorInterface* bitrate_allocator,
SendDelayStats* send_delay_stats,
RtcEventLog* event_log,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
std::unique_ptr<FecController> fec_controller,
const FieldTrialsView& field_trials);
~VideoSendStream() override;
void DeliverRtcp(const uint8_t* packet, size_t length);
// webrtc::VideoSendStream implementation.
void Start() override;
void StartPerRtpStream(std::vector<bool> active_layers) override;
void Stop() override;
bool started() override;
void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override;
std::vector<rtc::scoped_refptr<Resource>> GetAdaptationResources() override;
void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const DegradationPreference& degradation_preference) override;
void ReconfigureVideoEncoder(VideoEncoderConfig config) override;
void ReconfigureVideoEncoder(VideoEncoderConfig config,
SetParametersCallback callback) override;
Stats GetStats() override;
void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
RtpPayloadStateMap* payload_state_map);
void GenerateKeyFrame(const std::vector<std::string>& rids) override;
private:
friend class test::VideoSendStreamPeer;
absl::optional<float> GetPacingFactorOverride() const;
RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
MaybeWorkerThread* const rtp_transport_queue_;
RtpTransportControllerSendInterface* const transport_;
rtc::Event thread_sync_event_;
rtc::scoped_refptr<PendingTaskSafetyFlag> transport_queue_safety_ =
PendingTaskSafetyFlag::CreateDetached();
SendStatisticsProxy stats_proxy_;
const VideoSendStream::Config config_;
const VideoEncoderConfig::ContentType content_type_;
std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_;
EncoderRtcpFeedback encoder_feedback_;
RtpVideoSenderInterface* const rtp_video_sender_;
VideoSendStreamImpl send_stream_;
bool running_ RTC_GUARDED_BY(thread_checker_) = false;
};
} // namespace internal
} // namespace webrtc
#endif // VIDEO_VIDEO_SEND_STREAM_H_