Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 102 insertions, 0 deletions

diff --git a/webrtc/modules/video_processing/include/video_processing.h b/webrtc/modules/video_processing/include/video_processing.h
new file mode 100644
index 0000000000..a8d6358887
--- /dev/null
+++ b/webrtc/modules/video_processing/include/video_processing.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2011 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_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
+#define WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_processing/include/video_processing_defines.h"
+#include "webrtc/video_frame.h"
+
+// The module is largely intended to process video streams, except functionality
+// provided by static functions which operate independent of previous frames. It
+// is recommended, but not required that a unique instance be used for each
+// concurrently processed stream. Similarly, it is recommended to call Reset()
+// before switching to a new stream, but this is not absolutely required.
+//
+// The module provides basic thread safety by permitting only a single function
+// to execute concurrently.
+
+namespace webrtc {
+
+class VideoProcessing {
+ public:
+  struct FrameStats {
+    uint32_t hist[256];  // Frame histogram.
+    uint32_t mean;
+    uint32_t sum;
+    uint32_t num_pixels;
+    uint32_t sub_sampling_factor;  // Sub-sampling factor, in powers of 2.
+  };
+
+  enum BrightnessWarning { kNoWarning, kDarkWarning, kBrightWarning };
+
+  static VideoProcessing* Create();
+  virtual ~VideoProcessing() {}
+
+  // Retrieves statistics for the input frame. This function must be used to
+  // prepare a FrameStats struct for use in certain VPM functions.
+  static void GetFrameStats(const VideoFrame& frame, FrameStats* stats);
+
+  // Checks the validity of a FrameStats struct. Currently, valid implies only
+  // that is had changed from its initialized state.
+  static bool ValidFrameStats(const FrameStats& stats);
+
+  static void ClearFrameStats(FrameStats* stats);
+
+  // Increases/decreases the luminance value. 'delta' can be in the range {}
+  static void Brighten(int delta, VideoFrame* frame);
+
+  // Detects and removes camera flicker from a video stream. Every frame from
+  // the stream must be passed in. A frame will only be altered if flicker has
+  // been detected. Has a fixed-point implementation.
+  // Frame statistics provided by GetFrameStats(). On return the stats will
+  // be reset to zero if the frame was altered. Call GetFrameStats() again
+  // if the statistics for the altered frame are required.
+  virtual int32_t Deflickering(VideoFrame* frame, FrameStats* stats) = 0;
+
+  // Detects if a video frame is excessively bright or dark. Returns a
+  // warning if this is the case. Multiple frames should be passed in before
+  // expecting a warning. Has a floating-point implementation.
+  virtual int32_t BrightnessDetection(const VideoFrame& frame,
+                                      const FrameStats& stats) = 0;
+
+  // The following functions refer to the pre-processor unit within VPM. The
+  // pre-processor perfoms spatial/temporal decimation and content analysis on
+  // the frames prior to encoding.
+
+  // Enable/disable temporal decimation
+  virtual void EnableTemporalDecimation(bool enable) = 0;
+
+  virtual int32_t SetTargetResolution(uint32_t width,
+                                      uint32_t height,
+                                      uint32_t frame_rate) = 0;
+
+  virtual void SetTargetFramerate(int frame_rate) = 0;
+
+  virtual uint32_t GetDecimatedFrameRate() = 0;
+  virtual uint32_t GetDecimatedWidth() const = 0;
+  virtual uint32_t GetDecimatedHeight() const = 0;
+
+  // Set the spatial resampling settings of the VPM according to
+  // VideoFrameResampling.
+  virtual void SetInputFrameResampleMode(
+      VideoFrameResampling resampling_mode) = 0;
+
+  virtual void EnableDenosing(bool enable) = 0;
+  virtual const VideoFrame* PreprocessFrame(const VideoFrame& frame) = 0;
+
+  virtual VideoContentMetrics* GetContentMetrics() const = 0;
+  virtual void EnableContentAnalysis(bool enable) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_PROCESSING_INCLUDE_VIDEO_PROCESSING_H_