blob: 02ab897566190b335215cc683f9609b9bf720e8b [file] [log] [blame]
* 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 <stdio.h>
#include <memory>
#include <string>
#include "absl/types/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
class RtpHeaderParser;
namespace test {
class RtpFileReader;
class RtpFileSource : public PacketSource {
// Creates an RtpFileSource reading from |file_name|. If the file cannot be
// opened, or has the wrong format, NULL will be returned.
static RtpFileSource* Create(
const std::string& file_name,
absl::optional<uint32_t> ssrc_filter = absl::nullopt);
// Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
static bool ValidRtpDump(const std::string& file_name);
static bool ValidPcap(const std::string& file_name);
~RtpFileSource() override;
// Registers an RTP header extension and binds it to |id|.
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
std::unique_ptr<Packet> NextPacket() override;
static const int kFirstLineLength = 40;
static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
static const size_t kPacketHeaderSize = 8;
explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
bool OpenFile(const std::string& file_name);
std::unique_ptr<RtpFileReader> rtp_reader_;
std::unique_ptr<RtpHeaderParser> parser_;
const absl::optional<uint32_t> ssrc_filter_;
} // namespace test
} // namespace webrtc