| /* |
| * 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_ |