webrtc_m130/video/video_send_stream.h
Ying Wang 38a31b0af6 Revert "Revert "Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback"""
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}
2017-12-21 12:26:31 +00:00

116 lines
3.9 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,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states);
~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;
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:
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_