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/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
#include <string>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class RtpHeaderParser;
namespace rtclog {
class EventStream;
} // namespace rtclog
namespace test {
class Packet;
class RtcEventLogSource : public PacketSource {
public:
// 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);
// Returns a pointer to the next packet. Returns NULL if end of file was
// reached.
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();
private:
RtcEventLogSource();
bool OpenFile(const std::string& file_name);
int rtp_packet_index_ = 0;
int audio_output_index_ = 0;
rtc::scoped_ptr<rtclog::EventStream> event_log_;
rtc::scoped_ptr<RtpHeaderParser> parser_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
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