Index: webrtc/modules/video_coding/main/source/jitter_buffer_common.h |
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_common.h b/webrtc/modules/video_coding/main/source/jitter_buffer_common.h |
deleted file mode 100644 |
index 97af78087afff02ff32af7f9436c28aaffbeb970..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_common.h |
+++ /dev/null |
@@ -1,72 +0,0 @@ |
-/* |
- * 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_ |