webrtc/api/rtp_headers.cc
Joachim Reiersen a341fe31d4 Remove deprecated accessors for audio_level in RTPHeaderExtension
Bug: webrtc:15788
Change-Id: I0247e19edf89ed2212b93227c05136b87d56d8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Joachim Reiersen <joachimr@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42101}
2024-04-17 15:41:59 +00:00

58 lines
1.7 KiB
C++

/*
* 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"
namespace webrtc {
AudioLevel::AudioLevel() : voice_activity_(false), audio_level_(0) {}
AudioLevel::AudioLevel(bool voice_activity, int audio_level)
: voice_activity_(voice_activity), audio_level_(audio_level) {
RTC_CHECK_GE(audio_level, 0);
RTC_CHECK_LE(audio_level, 127);
}
RTPHeaderExtension::RTPHeaderExtension()
: hasTransmissionTimeOffset(false),
transmissionTimeOffset(0),
hasAbsoluteSendTime(false),
absoluteSendTime(0),
hasTransportSequenceNumber(false),
transportSequenceNumber(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),
extension() {}
RTPHeader::RTPHeader(const RTPHeader& other) = default;
RTPHeader& RTPHeader::operator=(const RTPHeader& other) = default;
} // namespace webrtc