92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
112 lines
3.8 KiB
C++
112 lines
3.8 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 "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 "modules/video_coding/protection_bitrate_calculator.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/send_statistics_proxy.h"
|
|
#include "video/video_stream_encoder.h"
|
|
|
|
namespace webrtc {
|
|
|
|
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);
|
|
|
|
~VideoSendStream() override;
|
|
|
|
void SignalNetworkState(NetworkState state);
|
|
bool DeliverRtcp(const uint8_t* packet, size_t length);
|
|
|
|
// webrtc::VideoSendStream implementation.
|
|
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;
|
|
|
|
// 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;
|
|
|
|
RtpStateMap StopPermanentlyAndGetRtpStates();
|
|
|
|
void SetTransportOverhead(size_t transport_overhead_per_packet);
|
|
|
|
private:
|
|
class ConstructionTask;
|
|
class DestructAndGetRtpStateTask;
|
|
|
|
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_
|