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, 82 insertions, 0 deletions

diff --git a/webrtc/modules/video_coding/main/test/video_source.h b/webrtc/modules/video_coding/main/test/video_source.h
new file mode 100644
index 0000000000..05deb4a39b
--- /dev/null
+++ b/webrtc/modules/video_coding/main/test/video_source.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2012 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_TEST_VIDEO_SOURCE_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
+
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/typedefs.h"
+
+#include <string>
+
+enum VideoSize
+    {
+        kUndefined,
+        kSQCIF,     // 128*96       = 12 288
+        kQQVGA,     // 160*120      = 19 200
+        kQCIF,      // 176*144      = 25 344
+        kCGA,       // 320*200      = 64 000
+        kQVGA,      // 320*240      = 76 800
+        kSIF,       // 352*240      = 84 480
+        kWQVGA,     // 400*240      = 96 000
+        kCIF,       // 352*288      = 101 376
+        kW288p,     // 512*288      = 147 456 (WCIF)
+        k448p,      // 576*448      = 281 088
+        kVGA,       // 640*480      = 307 200
+        k432p,      // 720*432      = 311 040
+        kW432p,     // 768*432      = 331 776
+        k4SIF,      // 704*480      = 337 920
+        kW448p,     // 768*448      = 344 064
+        kNTSC,		// 720*480      = 345 600
+        kFW448p,    // 800*448      = 358 400
+        kWVGA,      // 800*480      = 384 000
+        k4CIF,      // 704*576      = 405 504
+        kSVGA,      // 800*600      = 480 000
+        kW544p,     // 960*544      = 522 240
+        kW576p,     // 1024*576     = 589 824 (W4CIF)
+        kHD,        // 960*720      = 691 200
+        kXGA,       // 1024*768     = 786 432
+        kWHD,       // 1280*720     = 921 600
+        kFullHD,   // 1440*1080    = 1 555 200
+        kWFullHD,  // 1920*1080    = 2 073 600
+
+        kNumberOfVideoSizes
+    };
+
+
+class VideoSource
+{
+public:
+  VideoSource();
+  VideoSource(std::string fileName, VideoSize size, float frameRate, webrtc::VideoType type = webrtc::kI420);
+  VideoSource(std::string fileName, uint16_t width, uint16_t height,
+      float frameRate = 30, webrtc::VideoType type = webrtc::kI420);
+
+    std::string GetFileName() const { return _fileName; }
+    uint16_t  GetWidth() const { return _width; }
+    uint16_t GetHeight() const { return _height; }
+    webrtc::VideoType GetType() const { return _type; }
+    float GetFrameRate() const { return _frameRate; }
+    int GetWidthHeight( VideoSize size);
+
+    // Returns the filename with the path (including the leading slash) removed.
+    std::string GetName() const;
+
+    size_t GetFrameLength() const;
+
+private:
+    std::string         _fileName;
+    uint16_t      _width;
+    uint16_t      _height;
+    webrtc::VideoType   _type;
+    float               _frameRate;
+};
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_