webrtc/api/rtp_headers.cc
Patrik Höglund 3e113438b1 Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:33:26 +00:00

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1.7 KiB
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/*
* Copyright (c) 2017 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.
*/
#include "api/rtp_headers.h"
#include <string.h>
#include <algorithm>
#include <limits>
#include <type_traits>
#include "rtc_base/checks.h"
#include "rtc_base/stringutils.h"
namespace webrtc {
RTPHeaderExtension::RTPHeaderExtension()
: hasTransmissionTimeOffset(false),
transmissionTimeOffset(0),
hasAbsoluteSendTime(false),
absoluteSendTime(0),
hasTransportSequenceNumber(false),
transportSequenceNumber(0),
hasAudioLevel(false),
voiceActivity(false),
audioLevel(0),
hasVideoRotation(false),
videoRotation(kVideoRotation_0),
hasVideoContentType(false),
videoContentType(VideoContentType::UNSPECIFIED),
has_video_timing(false) {}
RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
default;
RTPHeaderExtension& RTPHeaderExtension::operator=(
const RTPHeaderExtension& other) = default;
RTPHeader::RTPHeader()
: markerBit(false),
payloadType(0),
sequenceNumber(0),
timestamp(0),
ssrc(0),
numCSRCs(0),
arrOfCSRCs(),
paddingLength(0),
headerLength(0),
payload_type_frequency(0),
extension() {}
RTPHeader::RTPHeader(const RTPHeader& other) = default;
RTPHeader& RTPHeader::operator=(const RTPHeader& other) = default;
} // namespace webrtc