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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/api/rtp_headers.cc
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/rtp_headers.cc')
-rw-r--r--third_party/libwebrtc/api/rtp_headers.cc54
1 files changed, 54 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtp_headers.cc b/third_party/libwebrtc/api/rtp_headers.cc
new file mode 100644
index 0000000000..40d4690397
--- /dev/null
+++ b/third_party/libwebrtc/api/rtp_headers.cc
@@ -0,0 +1,54 @@
+/*
+ * 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 {
+
+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),
+ csrcAudioLevels() {}
+
+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