| /* | 
 |  *  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_ |