e5c4a810e2
This makes more sense since logically it's a transport level feature, not a media stream feature. Even if the implementation details forces it to be an rtp stream detail, for the moment. BUG=webrtc:7907 Review-Url: https://codereview.webrtc.org/2978503002 Cr-Commit-Position: refs/heads/master@{#18963}
112 lines
3.9 KiB
C++
112 lines
3.9 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
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#define WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "webrtc/call/bitrate_allocator.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/modules/video_coding/protection_bitrate_calculator.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/event.h"
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#include "webrtc/rtc_base/task_queue.h"
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#include "webrtc/video/encoder_rtcp_feedback.h"
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#include "webrtc/video/send_delay_stats.h"
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#include "webrtc/video/send_statistics_proxy.h"
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#include "webrtc/video/vie_encoder.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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namespace webrtc {
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class CallStats;
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class SendSideCongestionController;
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class IvfFileWriter;
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class ProcessThread;
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class RtpRtcp;
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class RtpTransportControllerSendInterface;
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class RtcEventLog;
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namespace internal {
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class VideoSendStreamImpl;
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// VideoSendStream implements webrtc::VideoSendStream.
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// Internally, it delegates all public methods to VideoSendStreamImpl and / or
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// VieEncoder. VideoSendStreamInternal is created and deleted on |worker_queue|.
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class VideoSendStream : public webrtc::VideoSendStream {
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public:
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VideoSendStream(int num_cpu_cores,
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ProcessThread* module_process_thread,
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rtc::TaskQueue* worker_queue,
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CallStats* call_stats,
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RtpTransportControllerSendInterface* transport,
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BitrateAllocator* bitrate_allocator,
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SendDelayStats* send_delay_stats,
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RtcEventLog* event_log,
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config,
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const std::map<uint32_t, RtpState>& suspended_ssrcs,
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const RtpKeepAliveConfig& keepalive_config);
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~VideoSendStream() override;
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void SignalNetworkState(NetworkState state);
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bool DeliverRtcp(const uint8_t* packet, size_t length);
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// webrtc::VideoSendStream implementation.
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void Start() override;
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void Stop() override;
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void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
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const DegradationPreference& degradation_preference) override;
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void ReconfigureVideoEncoder(VideoEncoderConfig) override;
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Stats GetStats() override;
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typedef std::map<uint32_t, RtpState> RtpStateMap;
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// Takes ownership of each file, is responsible for closing them later.
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// Calling this method will close and finalize any current logs.
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// Giving rtc::kInvalidPlatformFileValue in any position disables logging
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// for the corresponding stream.
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// If a frame to be written would make the log too large the write fails and
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// the log is closed and finalized. A |byte_limit| of 0 means no limit.
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void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
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size_t byte_limit) override;
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RtpStateMap StopPermanentlyAndGetRtpStates();
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void SetTransportOverhead(size_t transport_overhead_per_packet);
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private:
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class ConstructionTask;
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class DestructAndGetRtpStateTask;
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rtc::ThreadChecker thread_checker_;
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rtc::TaskQueue* const worker_queue_;
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rtc::Event thread_sync_event_;
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SendStatisticsProxy stats_proxy_;
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const VideoSendStream::Config config_;
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const VideoEncoderConfig::ContentType content_type_;
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std::unique_ptr<VideoSendStreamImpl> send_stream_;
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std::unique_ptr<ViEEncoder> vie_encoder_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
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