blob: 15e3ce7eb8a15b42c7a700cdb15d2fca4e1f6cef [file] [log] [blame]
* Copyright (c) 2015 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 <memory>
#include <string>
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
class RtpHeaderParser;
namespace test {
class Packet;
class RtcEventLogSource : public PacketSource {
// Creates an RtcEventLogSource reading from |file_name|. If the file cannot
// be opened, or has the wrong format, NULL will be returned.
static RtcEventLogSource* Create(const std::string& file_name);
virtual ~RtcEventLogSource();
// Registers an RTP header extension and binds it to |id|.
virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
std::unique_ptr<Packet> NextPacket() override;
// Returns the timestamp of the next audio output event, in milliseconds. The
// maximum value of int64_t is returned if there are no more audio output
// events available.
int64_t NextAudioOutputEventMs();
bool OpenFile(const std::string& file_name);
size_t rtp_packet_index_ = 0;
size_t audio_output_index_ = 0;
ParsedRtcEventLog parsed_stream_;
std::unique_ptr<RtpHeaderParser> parser_;
} // namespace test
} // namespace webrtc