Replace SetCapturer and SetCaptureDevice by SetSource.

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 142 matching lines @@
                     bool mute,
                     const VideoOptions* options) override;
   bool AddSendStream(const StreamParams& sp) override;
   bool RemoveSendStream(uint32_t ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
   bool AddRecvStream(const StreamParams& sp, bool default_stream);
   bool RemoveRecvStream(uint32_t ssrc) override;
   bool SetSink(uint32_t ssrc,
                rtc::VideoSinkInterface<VideoFrame>* sink) override;
   bool GetStats(VideoMediaInfo* info) override;
-  bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override;
+  void SetSource(
+      uint32_t ssrc,
+      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
 
   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override;
   void SetInterface(NetworkInterface* iface) override;
 
   // Implemented for VideoMediaChannelTest.
   bool sending() const { return sending_; }
@@ skipping 25 matching lines @@
     // These optionals are unset if not changed.
     rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
     rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
   };
 
   bool GetChangedSendParameters(const VideoSendParameters& params,
                                 ChangedSendParameters* changed_params) const;
   bool GetChangedRecvParameters(const VideoRecvParameters& params,
                                 ChangedRecvParameters* changed_params) const;
 
-  bool MuteStream(uint32_t ssrc, bool mute);
-
   void SetMaxSendBandwidth(int bps);
   void SetOptions(uint32_t ssrc, const VideoOptions& options);
 
   void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
                             const StreamParams& sp) const;
   bool CodecIsExternallySupported(const std::string& name) const;
   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 capturer is connected and
+  // Wrapper for the sender part, this is where the source is connected and
   // frames are then converted from cricket frames to webrtc frames.
   class WebRtcVideoSendStream
       : public rtc::VideoSinkInterface<cricket::VideoFrame>,
         public webrtc::LoadObserver {
    public:
     WebRtcVideoSendStream(
         webrtc::Call* call,
         const StreamParams& sp,
         const 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 std::vector<webrtc::RtpExtension>& rtp_extensions,
         const VideoSendParameters& send_params);
     virtual ~WebRtcVideoSendStream();
 
     void SetOptions(const VideoOptions& options);
     // TODO(pbos): Move logic from SetOptions into this method.
     void SetSendParameters(const ChangedSendParameters& send_params);
     bool SetRtpParameters(const webrtc::RtpParameters& parameters);
     webrtc::RtpParameters GetRtpParameters() const;
 
     void OnFrame(const cricket::VideoFrame& frame) override;
-    bool SetCapturer(VideoCapturer* capturer);
-    void MuteStream(bool mute);
-    bool DisconnectCapturer();
+    void SetSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source);
+    void DisconnectSource();
 
     void SetSend(bool send);
 
     // Implements webrtc::LoadObserver.
     void OnLoadUpdate(Load load) override;
 
     const std::vector<uint32_t>& GetSsrcs() const;
     VideoSenderInfo GetVideoSenderInfo();
     void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
 
@@ skipping 81 matching lines @@
     void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_);
 
     rtc::ThreadChecker thread_checker_;
     rtc::AsyncInvoker invoker_;
     rtc::Thread* worker_thread_;
     const std::vector<uint32_t> ssrcs_;
     const std::vector<SsrcGroup> ssrc_groups_;
     webrtc::Call* const call_;
     rtc::VideoSinkWants sink_wants_;
     // Counter used for deciding if the video resolution is currently
-    // restricted by CPU usage. It is reset if |capturer_| is changed.
+    // restricted by CPU usage. It is reset if |source_| is changed.
     int cpu_restricted_counter_;
     // Total number of times resolution as been requested to be changed due to
     // CPU adaptation.
     int number_of_cpu_adapt_changes_;
-    VideoCapturer* capturer_;
+    rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
     WebRtcVideoEncoderFactory* const external_encoder_factory_
         GUARDED_BY(lock_);
 
     rtc::CriticalSection lock_;
     webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
     // 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_ GUARDED_BY(lock_);
     // Contains settings that are unique for each stream, such as max_bitrate.
     // 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_ GUARDED_BY(lock_);
     bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
     VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
     AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
+    // Original frame dimension, before adaptation due to cpu or
+    // network bottlenecks.
+    Dimensions input_dimensions_ GUARDED_BY(lock_);
     Dimensions last_dimensions_ GUARDED_BY(lock_);
     webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) =
         webrtc::kVideoRotation_0;
 
     bool sending_ GUARDED_BY(lock_);
-    bool muted_ GUARDED_BY(lock_);
 
     // The timestamp of the first frame received
     // Used to generate the timestamps of subsequent frames
     int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_);
 
     // The timestamp of the last frame received
-    // Used to generate timestamp for the black frame when capturer is removed
+    // Used to generate timestamp for the black frame when source is removed
     int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
   };
 
   // Wrapper for the receiver part, contains configs etc. that are needed to
   // reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
   // between rtc::VideoSinkInterface<webrtc::VideoFrame> and
   // rtc::VideoSinkInterface<cricket::VideoFrame>.
   class WebRtcVideoReceiveStream
       : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
    public:
@@ skipping 123 matching lines @@
   // TODO(deadbeef): Don't duplicate information between
   // send_params/recv_params, rtp_extensions, options, etc.
   VideoSendParameters send_params_;
   VideoOptions default_send_options_;
   VideoRecvParameters recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_