25e5110ab0
Injecting the retransmission rate limiter used in video send stream directly rather than using the transport controller reference. This prepares for removing ownership of the retransmission rate limiter from the congestion controller. Bug: webrtc:8415 Change-Id: Iee8af53e62f407ee430625008f2d2b0cabb1f369 Reviewed-on: https://webrtc-review.googlesource.com/58800 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22251}
128 lines
4.2 KiB
C++
128 lines
4.2 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 <vector>
|
|
|
|
#include "api/fec_controller.h"
|
|
#include "call/bitrate_allocator.h"
|
|
#include "call/video_receive_stream.h"
|
|
#include "call/video_send_stream.h"
|
|
#include "common_video/libyuv/include/webrtc_libyuv.h"
|
|
#include "rtc_base/criticalsection.h"
|
|
#include "rtc_base/event.h"
|
|
#include "rtc_base/task_queue.h"
|
|
#include "video/encoder_rtcp_feedback.h"
|
|
#include "video/send_delay_stats.h"
|
|
#include "video/payload_router.h"
|
|
#include "video/send_statistics_proxy.h"
|
|
#include "video/video_stream_encoder.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
class VideoSendStreamPeer;
|
|
} // namespace test
|
|
|
|
class CallStats;
|
|
class SendSideCongestionController;
|
|
class IvfFileWriter;
|
|
class ProcessThread;
|
|
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. VideoSendStreamInternal is created and deleted on
|
|
// |worker_queue|.
|
|
class VideoSendStream : public webrtc::VideoSendStream {
|
|
public:
|
|
VideoSendStream(
|
|
int num_cpu_cores,
|
|
ProcessThread* module_process_thread,
|
|
rtc::TaskQueue* worker_queue,
|
|
CallStats* call_stats,
|
|
RtpTransportControllerSendInterface* transport,
|
|
BitrateAllocator* 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,
|
|
RateLimiter* retransmission_limiter);
|
|
|
|
~VideoSendStream() override;
|
|
|
|
void SignalNetworkState(NetworkState state);
|
|
bool DeliverRtcp(const uint8_t* packet, size_t length);
|
|
|
|
// webrtc::VideoSendStream implementation.
|
|
void UpdateActiveSimulcastLayers(
|
|
const std::vector<bool> active_layers) override;
|
|
void Start() override;
|
|
void Stop() override;
|
|
|
|
void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
|
|
const DegradationPreference& degradation_preference) override;
|
|
|
|
void ReconfigureVideoEncoder(VideoEncoderConfig) override;
|
|
Stats GetStats() override;
|
|
|
|
typedef std::map<uint32_t, RtpState> RtpStateMap;
|
|
typedef std::map<uint32_t, RtpPayloadState> RtpPayloadStateMap;
|
|
|
|
// Takes ownership of each file, is responsible for closing them later.
|
|
// Calling this method will close and finalize any current logs.
|
|
// Giving rtc::kInvalidPlatformFileValue in any position disables logging
|
|
// for the corresponding stream.
|
|
// If a frame to be written would make the log too large the write fails and
|
|
// the log is closed and finalized. A |byte_limit| of 0 means no limit.
|
|
void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
|
|
size_t byte_limit) override;
|
|
|
|
void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
|
|
RtpPayloadStateMap* payload_state_map);
|
|
|
|
void SetTransportOverhead(size_t transport_overhead_per_packet);
|
|
|
|
private:
|
|
friend class test::VideoSendStreamPeer;
|
|
|
|
class ConstructionTask;
|
|
class DestructAndGetRtpStateTask;
|
|
|
|
rtc::Optional<float> GetPacingFactorOverride() const;
|
|
|
|
rtc::ThreadChecker thread_checker_;
|
|
rtc::TaskQueue* const worker_queue_;
|
|
rtc::Event thread_sync_event_;
|
|
|
|
SendStatisticsProxy stats_proxy_;
|
|
const VideoSendStream::Config config_;
|
|
const VideoEncoderConfig::ContentType content_type_;
|
|
std::unique_ptr<VideoSendStreamImpl> send_stream_;
|
|
std::unique_ptr<VideoStreamEncoder> video_stream_encoder_;
|
|
};
|
|
|
|
} // namespace internal
|
|
} // namespace webrtc
|
|
|
|
#endif // VIDEO_VIDEO_SEND_STREAM_H_
|