Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/video_coding/jitter_buffer_common.h

@@ -0,0 +1,72 @@
+/*
+ *  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_CODING_JITTER_BUFFER_COMMON_H_
+#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Used to estimate rolling average of packets per frame.
+static const float kFastConvergeMultiplier = 0.4f;
+static const float kNormalConvergeMultiplier = 0.2f;
+
+enum { kMaxNumberOfFrames = 300 };
+enum { kStartNumberOfFrames = 6 };
+enum { kMaxVideoDelayMs = 10000 };
+enum { kPacketsPerFrameMultiplier = 5 };
+enum { kFastConvergeThreshold = 5 };
+
+enum VCMJitterBufferEnum {
+  kMaxConsecutiveOldFrames = 60,
+  kMaxConsecutiveOldPackets = 300,
+  // TODO(sprang): Reduce this limit once codecs don't sometimes wildly
+  // overshoot bitrate target.
+  kMaxPacketsInSession = 1400,      // Allows ~2MB frames.
+  kBufferIncStepSizeBytes = 30000,  // >20 packets.
+  kMaxJBFrameSizeBytes = 4000000    // sanity don't go above 4Mbyte.
+};
+
+enum VCMFrameBufferEnum {
+  kOutOfBoundsPacket = -7,
+  kNotInitialized = -6,
+  kOldPacket = -5,
+  kGeneralError = -4,
+  kFlushIndicator = -3,  // Indicator that a flush has occurred.
+  kTimeStampError = -2,
+  kSizeError = -1,
+  kNoError = 0,
+  kIncomplete = 1,        // Frame incomplete.
+  kCompleteSession = 3,   // at least one layer in the frame complete.
+  kDecodableSession = 4,  // Frame incomplete, but ready to be decoded
+  kDuplicatePacket = 5    // We're receiving a duplicate packet.
+};
+
+enum VCMFrameBufferStateEnum {
+  kStateEmpty,       // frame popped by the RTP receiver
+  kStateIncomplete,  // frame that have one or more packet(s) stored
+  kStateComplete,    // frame that have all packets
+  kStateDecodable    // Hybrid mode - frame can be decoded
+};
+
+enum { kH264StartCodeLengthBytes = 4 };
+
+// Used to indicate if a received packet contain a complete NALU (or equivalent)
+enum VCMNaluCompleteness {
+  kNaluUnset = 0,     // Packet has not been filled.
+  kNaluComplete = 1,  // Packet can be decoded as is.
+  kNaluStart,         // Packet contain beginning of NALU
+  kNaluIncomplete,    // Packet is not beginning or end of NALU
+  kNaluEnd,           // Packet is the end of a NALU
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_