Let ViEEncoder express resolution requests as Sinkwants

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 <utility>
 #include <vector>
 #include <utility>
 
 #include "webrtc/base/platform_file.h"
 #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/media/base/videosourceinterface.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
-class LoadObserver;
 class VideoEncoder;
 
 class VideoSendStream {
  public:
   struct StreamStats {
     std::string ToString() const;
 
     FrameCounts frame_counts;
     bool is_rtx = false;
     int width = 0;
@@ skipping 19 matching lines @@
     // Bitrate the encoder is currently configured to use due to bandwidth
     // limitations.
     int target_media_bitrate_bps = 0;
     // Bitrate the encoder is actually producing.
     int media_bitrate_bps = 0;
     // Media bitrate this VideoSendStream is configured to prefer if there are
     // no bandwidth limitations.
     int preferred_media_bitrate_bps = 0;
     bool suspended = false;
     bool bw_limited_resolution = false;
+    bool cpu_limited_resolution = false;
+    // Total number of times resolution as been requested to be changed due to
+    // CPU adaptation.
+    int number_of_cpu_adapt_changes = 0;
     std::map<uint32_t, StreamStats> substreams;
   };
 
   struct Config {
    public:
     Config() = delete;
     Config(Config&&) = default;
     explicit Config(Transport* send_transport)
         : send_transport(send_transport) {}
 
@@ skipping 64 matching lines @@
         int payload_type = -1;
       } rtx;
 
       // RTCP CNAME, see RFC 3550.
       std::string c_name;
     } rtp;
 
     // Transport for outgoing packets.
     Transport* send_transport = nullptr;
 
-    // Callback for overuse and normal usage based on the jitter of incoming
-    // captured frames. 'nullptr' disables the callback.
-    LoadObserver* overuse_callback = nullptr;
-
     // 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;
 
@@ skipping 17 matching lines @@
     Config(const Config&) = default;
   };
 
   // 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;
 
+  // Based on the spec in
+  // https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
+  enum class DegradationPreference {
+    kMaintainResolution,
+    // TODO(perkj): Implement kMaintainFrameRate. kBalanced will drop frames
+    // if the encoder overshootor or the encoder can not encode fast enough.

[nisse-webrtc, 2016/10/28 10:53:44]

"s " missing in "overshoots or"

[perkj_webrtc, 2016/10/28 14:05:34]

Done.

+    kBalanced,
+  };
   virtual void SetSource(
-      rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
+      const DegradationPreference& degradation_preference) = 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(VideoEncoderConfig config) = 0;
 
   virtual Stats GetStats() = 0;
 
   // Takes ownership of each file, is responsible for closing them later.
   // Calling this method will close and finalize any current logs.
@@ skipping 11 matching lines @@
     EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
   }
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_