Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 67 insertions, 0 deletions

diff --git a/webrtc/modules/video_coding/utility/include/quality_scaler.h b/webrtc/modules/video_coding/utility/include/quality_scaler.h
new file mode 100644
index 0000000000..29a1496c05
--- /dev/null
+++ b/webrtc/modules/video_coding/utility/include/quality_scaler.h
@@ -0,0 +1,67 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
+
+#include "webrtc/common_video/libyuv/include/scaler.h"
+#include "webrtc/modules/video_coding/utility/include/moving_average.h"
+
+namespace webrtc {
+class QualityScaler {
+ public:
+  static const int kDefaultLowQpDenominator;
+  static const int kDefaultMinDownscaleDimension;
+  struct Resolution {
+    int width;
+    int height;
+  };
+
+  QualityScaler();
+  void Init(int low_qp_threshold,
+            int high_qp_threshold,
+            bool use_framerate_reduction);
+  void SetMinResolution(int min_width, int min_height);
+  void ReportFramerate(int framerate);
+  void ReportQP(int qp);
+  void ReportDroppedFrame();
+  void Reset(int framerate, int bitrate, int width, int height);
+  void OnEncodeFrame(const VideoFrame& frame);
+  Resolution GetScaledResolution() const;
+  const VideoFrame& GetScaledFrame(const VideoFrame& frame);
+  int GetTargetFramerate() const;
+  int downscale_shift() const { return downscale_shift_; }
+
+ private:
+  void AdjustScale(bool up);
+  void ClearSamples();
+
+  Scaler scaler_;
+  VideoFrame scaled_frame_;
+
+  size_t num_samples_;
+  int framerate_;
+  int target_framerate_;
+  int low_qp_threshold_;
+  int high_qp_threshold_;
+  MovingAverage<int> framedrop_percent_;
+  MovingAverage<int> average_qp_;
+  Resolution res_;
+
+  int downscale_shift_;
+  int framerate_down_;
+  bool use_framerate_reduction_;
+  int min_width_;
+  int min_height_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_