mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
59 lines
2 KiB
C++
59 lines
2 KiB
C++
/*
|
|
* 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 MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
|
|
#define MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_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.
|
|
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
|
|
};
|
|
|
|
enum { kH264StartCodeLengthBytes = 4 };
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
|