This reverts commit 718d8631b0294a8bdc56366b68c51e2f04cd0c9e. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback"" > > This reverts commit 53d901332c2eb43cad0da5768c6f7a8c4aeb9590. > > Reason for revert: root cause has been found and will be addressed in the patch.The root cause was protection_bitrate_calculator_ is now destructed before worker_queue_, and worker_queue_ may contain tasks which involves protection_bitrate_calculator_, so they need to be destructed in the opposite order. > That was not an issue since before this cl we didn't allocate protection_bitrate_calculator_ on the heap. > > Original change's description: > > Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback" > > > > This reverts commit e58e91b6d143ef847f8df24b19de4ba98cdb6f72. > > > > Reason for revert: Breaks downstream project b/70848177 > > > > Original change's description: > > > Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback > > > > > > Bug: webrtc:8656 > > > Change-Id: Iab4f6ab8997cb082762218afc8580e9985ac2522 > > > Reviewed-on: https://webrtc-review.googlesource.com/33010 > > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21348} > > > > TBR=stefan@webrtc.org,philipel@webrtc.org,yinwa@webrtc.org > > > > Change-Id: Ic186ba78be429bd1046ceac15051a3382b6ffc4f > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:8656 > > Reviewed-on: https://webrtc-review.googlesource.com/35080 > > Commit-Queue: Lu Liu <lliuu@webrtc.org> > > Reviewed-by: Lu Liu <lliuu@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21374} > > TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org > > Change-Id: Ie2b5a2a2ead0f20ac67c1ea9b8d192af66bddf8d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8656 > Reviewed-on: https://webrtc-review.googlesource.com/35320 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Ying Wang <yinwa@webrtc.org> > Commit-Queue: Ying Wang <yinwa@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21409} TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org Change-Id: I9773aaa942054dcfbab6002a5d713ab3526b0534 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8656 Reviewed-on: https://webrtc-review.googlesource.com/35700 Reviewed-by: Ying Wang <yinwa@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21410}
116 lines
3.9 KiB
C++
116 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 VIDEO_VIDEO_SEND_STREAM_H_
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#define 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 "call/bitrate_allocator.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "common_video/libyuv/include/webrtc_libyuv.h"
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#include "modules/video_coding/protection_bitrate_calculator.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/event.h"
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#include "rtc_base/task_queue.h"
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#include "video/encoder_rtcp_feedback.h"
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#include "video/send_delay_stats.h"
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#include "video/send_statistics_proxy.h"
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#include "video/video_stream_encoder.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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// VideoStreamEncoder. VideoSendStreamInternal is created and deleted on
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// |worker_queue|.
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class VideoSendStream : public webrtc::VideoSendStream {
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public:
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VideoSendStream(
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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 std::map<uint32_t, RtpPayloadState>& suspended_payload_states);
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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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typedef std::map<uint32_t, RtpPayloadState> RtpPayloadStateMap;
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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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void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
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RtpPayloadStateMap* payload_state_map);
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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<VideoStreamEncoder> video_stream_encoder_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // VIDEO_VIDEO_SEND_STREAM_H_
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