chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.cc

/*
 *  Copyright (c) 2014 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 "modules/audio_coding/neteq/tools/packet.h"

#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"

namespace webrtc {
namespace test {

Packet::Packet(rtc::CopyOnWriteBuffer packet,
               size_t virtual_packet_length_bytes,
               double time_ms,
               const RtpHeaderExtensionMap* extension_map)
    : … { … }

Packet::Packet(const RTPHeader& header,
               size_t virtual_packet_length_bytes,
               size_t virtual_payload_length_bytes,
               double time_ms)
    : … { … }

Packet::~Packet() = default;

bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const { … }

void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) { … }

bool Packet::ParseHeader(const RtpHeaderExtensionMap* extension_map) { … }

void Packet::CopyToHeader(RTPHeader* destination) const { … }

}  // namespace test
}  // namespace webrtc