Remove WebRtcVideoSendStream2::VideoSink inheritance. Remove sending black frame on source remova…

webrtc/media/engine/webrtcvideoengine2.h

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 213 matching lines @@
   bool ValidateSendSsrcAvailability(const StreamParams& sp) const
       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
   bool ValidateReceiveSsrcAvailability(const StreamParams& sp) const
       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
   void DeleteReceiveStream(WebRtcVideoReceiveStream* stream)
       EXCLUSIVE_LOCKS_REQUIRED(stream_crit_);
 
   static std::string CodecSettingsVectorToString(
       const std::vector<VideoCodecSettings>& codecs);
 
-  // Wrapper for the sender part, this is where the source is connected and
-  // frames are then converted from cricket frames to webrtc frames.
+  // Wrapper for the sender part.
   class WebRtcVideoSendStream
-      : public rtc::VideoSinkInterface<webrtc::VideoFrame>,
-        public rtc::VideoSourceInterface<webrtc::VideoFrame> {
+      : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
    public:
     WebRtcVideoSendStream(
         webrtc::Call* call,
         const StreamParams& sp,
         webrtc::VideoSendStream::Config config,
         const VideoOptions& options,
         WebRtcVideoEncoderFactory* external_encoder_factory,
         bool enable_cpu_overuse_detection,
         int max_bitrate_bps,
         const rtc::Optional<VideoCodecSettings>& codec_settings,
         const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
         const VideoSendParameters& send_params);
     virtual ~WebRtcVideoSendStream();
 
     void SetSendParameters(const ChangedSendParameters& send_params);
     bool SetRtpParameters(const webrtc::RtpParameters& parameters);
     webrtc::RtpParameters GetRtpParameters() const;
 
     // Implements rtc::VideoSourceInterface<webrtc::VideoFrame>.
     // WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream
-    // in |stream_|.
-    // TODO(perkj, nisse): Refactor WebRtcVideoSendStream to directly connect
-    // the camera input |source_|
-    void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
+    // in |stream_|. This is done to proxy VideoSinkWants from the encoder to
+    // the worker thread.
+    void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
                          const rtc::VideoSinkWants& wants) override;
-    void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+    void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
 
-    void OnFrame(const webrtc::VideoFrame& frame) override;
     bool SetVideoSend(bool mute,
                       const VideoOptions* options,
                       rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
 
     void SetSend(bool send);
 
     const std::vector<uint32_t>& GetSsrcs() const;
     VideoSenderInfo GetVideoSenderInfo(bool log_stats);
     void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
 
@@ skipping 22 matching lines @@
     struct AllocatedEncoder {
       AllocatedEncoder(webrtc::VideoEncoder* encoder,
                        const cricket::VideoCodec& codec,
                        bool external);
       webrtc::VideoEncoder* encoder;
       webrtc::VideoEncoder* external_encoder;
       cricket::VideoCodec codec;
       bool external;
     };
 
-    // TODO(perkj): VideoFrameInfo is currently used for sending a black frame
-    // when the video source is removed. Consider moving that logic to
-    // VieEncoder or remove it.
-    struct VideoFrameInfo {
-      VideoFrameInfo()
-          : width(0),
-            height(0),
-            rotation(webrtc::kVideoRotation_0),
-            is_texture(false) {}
-      int width;
-      int height;
-      webrtc::VideoRotation rotation;
-      bool is_texture;
-    };
-
     rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
     ConfigureVideoEncoderSettings(const VideoCodec& codec);
     AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec);
     void DestroyVideoEncoder(AllocatedEncoder* encoder);
     void SetCodec(const VideoCodecSettings& codec);
     void RecreateWebRtcStream();
     webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
         const VideoCodec& codec) const;
     void ReconfigureEncoder();
     bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
 
     // Calls Start or Stop according to whether or not |sending_| is true,
     // and whether or not the encoding in |rtp_parameters_| is active.
     void UpdateSendState();
 
     rtc::ThreadChecker thread_checker_;
     rtc::AsyncInvoker invoker_;
     rtc::Thread* worker_thread_;
     const std::vector<uint32_t> ssrcs_ ACCESS_ON(&thread_checker_);
     const std::vector<SsrcGroup> ssrc_groups_ ACCESS_ON(&thread_checker_);
     webrtc::Call* const call_;
     const bool enable_cpu_overuse_detection_;
     rtc::VideoSourceInterface<webrtc::VideoFrame>* source_
         ACCESS_ON(&thread_checker_);
     WebRtcVideoEncoderFactory* const external_encoder_factory_
         ACCESS_ON(&thread_checker_);
 
-    rtc::CriticalSection lock_;
     webrtc::VideoSendStream* stream_ ACCESS_ON(&thread_checker_);
     rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
-        GUARDED_BY(lock_);
+        ACCESS_ON(&thread_checker_);
     // Contains settings that are the same for all streams in the MediaChannel,
     // such as codecs, header extensions, and the global bitrate limit for the
     // entire channel.
     VideoSendStreamParameters parameters_ ACCESS_ON(&thread_checker_);
     // Contains settings that are unique for each stream, such as max_bitrate.
     // Does *not* contain codecs, however.
     // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
     // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
     // one stream per MediaChannel.
     webrtc::RtpParameters rtp_parameters_ ACCESS_ON(&thread_checker_);
     AllocatedEncoder allocated_encoder_ ACCESS_ON(&thread_checker_);
-    VideoFrameInfo last_frame_info_ GUARDED_BY(lock_);
 
     bool sending_ ACCESS_ON(&thread_checker_);
-
-    // The timestamp of the last frame received
-    // Used to generate timestamp for the black frame when source is removed
-    int64_t last_frame_timestamp_us_ GUARDED_BY(lock_);
   };
 
   // Wrapper for the receiver part, contains configs etc. that are needed to
   // reconstruct the underlying VideoReceiveStream.
   class WebRtcVideoReceiveStream
       : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
    public:
     WebRtcVideoReceiveStream(
         webrtc::Call* call,
         const StreamParams& sp,
@@ skipping 128 matching lines @@
   // send_params/recv_params, rtp_extensions, options, etc.
   VideoSendParameters send_params_;
   VideoOptions default_send_options_;
   VideoRecvParameters recv_params_;
   int64_t last_stats_log_ms_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_