Add task queue to Call.

webrtc/video_send_stream.h

 /*
  *  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 WEBRTC_VIDEO_SEND_STREAM_H_
 #define WEBRTC_VIDEO_SEND_STREAM_H_
 
 #include <map>
 #include <string>
+#include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/transport.h"
-#include "webrtc/media/base/videosinkinterface.h"
 
 namespace webrtc {
 
 class LoadObserver;
 class VideoEncoder;
 
 // Class to deliver captured frame to the video send stream.
 class VideoCaptureInput {
  public:
   // These methods do not lock internally and must be called sequentially.
@@ skipping 28 matching lines @@
     int avg_encode_time_ms = 0;
     int encode_usage_percent = 0;
     int target_media_bitrate_bps = 0;
     int media_bitrate_bps = 0;
     bool suspended = false;
     bool bw_limited_resolution = false;
     std::map<uint32_t, StreamStats> substreams;
   };
 
   struct Config {
+   private:
+    // Access to the copy constructor is private to force use of the Copy()
+    // method for those exceptional cases where we do use it.
+    Config(const Config&) = default;

[stefan-webrtc, 2016/07/08 15:56:43]

Perhaps put this last?

[perkj_webrtc, 2016/07/11 11:41:08]

This currently match with what has been done for VideoReceiveStream. I guess the
idea is to point it out but it is against the style guide so I moved it.
https://google.github.io/styleguide/cppguide.html#Declaration_Order

+
+   public:
     Config() = delete;
+    Config(Config&&) = default;
     explicit Config(Transport* send_transport)
         : send_transport(send_transport) {}
 
+    Config& operator=(Config&&) = default;
+    Config& operator=(const Config&) = delete;
+
+    // Mostly used by tests.  Avoid creating copies if you can.
+    Config Copy() const { return Config(*this); }
+
     std::string ToString() const;
 
     struct EncoderSettings {
       std::string ToString() const;
 
       std::string payload_name;
-      int payload_type = -1;
+      int payload_type = 0;

[stefan-webrtc, 2016/07/08 15:56:43]

Why was this changed?

[perkj_webrtc, 2016/07/11 11:41:08]

good question... 
VieKeyRequestTest.CreateAndTriggerRequest fails due to that
_codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec) dchecks... 

Ok, probably should fix that test to use a valid config instead.

 
       // TODO(sophiechang): Delete this field when no one is using internal
       // sources anymore.
       bool internal_source = false;
 
       // Allow 100% encoder utilization. Used for HW encoders where CPU isn't
       // expected to be the limiting factor, but a chip could be running at
       // 30fps (for example) exactly.
       bool full_overuse_time = false;
 
@@ skipping 48 matching lines @@
     // Called for each I420 frame before encoding the frame. Can be used for
     // effects, snapshots etc. 'nullptr' disables the callback.
     rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
 
     // Called for each encoded frame, e.g. used for file storage. 'nullptr'
     // disables the callback. Also measures timing and passes the time
     // spent on encoding. This timing will not fire if encoding takes longer
     // than the measuring window, since the sample data will have been dropped.
     EncodedFrameObserver* post_encode_callback = nullptr;
 
-    // Renderer for local preview. The local renderer will be called even if
-    // sending hasn't started. 'nullptr' disables local rendering.
-    rtc::VideoSinkInterface<VideoFrame>* local_renderer = nullptr;

[stefan-webrtc, 2016/07/08 15:56:43]

Wouldn't it make sense to do this separately?

[perkj_webrtc, 2016/07/11 11:41:08]

This is part of moving the input directly to ViEEncoder. How about I try to make
a separate cl of that? 

-
     // Expected delay needed by the renderer, i.e. the frame will be delivered
     // this many milliseconds, if possible, earlier than expected render time.
     // Only valid if |local_renderer| is set.
     int render_delay_ms = 0;
 
     // Target delay in milliseconds. A positive value indicates this stream is
     // used for streaming instead of a real-time call.
     int target_delay_ms = 0;
 
     // True if the stream should be suspended when the available bitrate fall
     // below the minimum configured bitrate. If this variable is false, the
     // stream may send at a rate higher than the estimated available bitrate.
     bool suspend_below_min_bitrate = false;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // Gets interface used to insert captured frames. Valid as long as the
   // VideoSendStream is valid.
   virtual VideoCaptureInput* Input() = 0;
 
   // Set which streams to send. Must have at least as many SSRCs as configured
   // in the config. Encoder settings are passed on to the encoder instance along
   // with the VideoStream settings.
-  virtual void ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
+  virtual void ReconfigureVideoEncoder(VideoEncoderConfig config) = 0;
 
   virtual Stats GetStats() = 0;
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_