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Side by Side Diff: webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

Issue 1979443004: Add H264 bitstream rewriting to limit frame reordering marker in header (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rewriting on the receiver side as well Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "webrtc/modules/audio_coding/neteq/packet.h" 11 #include "webrtc/modules/rtp_rtcp/source/h264/pps_parser.h"
12 12
13 namespace webrtc { 13 namespace webrtc {} // namespace webrtc
14
15 Packet::Packet() = default;
16
17 Packet::~Packet() = default;
18
19 } // namespace webrtc
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