webrtc/api/rtp_headers.cc
Joachim Reiersen 5075cb4a60 Expose AudioLevel as an absl::optional struct in api/rtp_headers.h
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.

The old fields are preserved for compatibility with downstream projects, but will be removed in the future.

Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
2024-03-22 10:07:47 +00:00

79 lines
2.2 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),
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;
absl::optional<AudioLevel> RTPHeaderExtension::audio_level() const {
if (!hasAudioLevel) {
return absl::nullopt;
}
return AudioLevel(voiceActivity, audioLevel);
}
void RTPHeaderExtension::set_audio_level(
absl::optional<AudioLevel> audio_level) {
if (audio_level) {
hasAudioLevel = true;
voiceActivity = audio_level->voice_activity();
audioLevel = audio_level->level();
} else {
hasAudioLevel = false;
}
}
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