chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.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/rtp_file_source.h"

#include <string.h>

#include "absl/strings/string_view.h"
#ifndef WIN32
#include <netinet/in.h>
#endif

#include <memory>

#include "modules/audio_coding/neteq/tools/packet.h"
#include "rtc_base/checks.h"
#include "test/rtp_file_reader.h"

namespace webrtc {
namespace test {

RtpFileSource* RtpFileSource::Create(absl::string_view file_name,
                                     absl::optional<uint32_t> ssrc_filter) { … }

bool RtpFileSource::ValidRtpDump(absl::string_view file_name) { … }

bool RtpFileSource::ValidPcap(absl::string_view file_name) { … }

RtpFileSource::~RtpFileSource() { … }

bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
                                               uint8_t id) { … }

std::unique_ptr<Packet> RtpFileSource::NextPacket() { … }

RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
    : … { … }

bool RtpFileSource::OpenFile(absl::string_view file_name) { … }

}  // namespace test
}  // namespace webrtc