blob: acdfa77da177e605ddfa94e80dccea6abef37847 [file] [log] [blame]
/*
* Copyright (c) 2016 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_CALL_RTC_EVENT_LOG_PARSER_H_
#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
#include <string>
#include <vector>
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
#else
#include "webrtc/call/rtc_event_log.pb.h"
#endif
namespace webrtc {
enum class MediaType;
class ParsedRtcEventLog {
friend class RtcEventLogTestHelper;
public:
enum EventType {
UNKNOWN_EVENT = 0,
LOG_START = 1,
LOG_END = 2,
RTP_EVENT = 3,
RTCP_EVENT = 4,
AUDIO_PLAYOUT_EVENT = 5,
BWE_PACKET_LOSS_EVENT = 6,
BWE_PACKET_DELAY_EVENT = 7,
VIDEO_RECEIVER_CONFIG_EVENT = 8,
VIDEO_SENDER_CONFIG_EVENT = 9,
AUDIO_RECEIVER_CONFIG_EVENT = 10,
AUDIO_SENDER_CONFIG_EVENT = 11
};
// Reads an RtcEventLog file and returns true if parsing was successful.
bool ParseFile(const std::string& file_name);
// Returns the number of events in an EventStream.
size_t GetNumberOfEvents() const;
// Reads the arrival timestamp (in microseconds) from a rtclog::Event.
int64_t GetTimestamp(size_t index) const;
// Reads the event type of the rtclog::Event at |index|.
EventType GetEventType(size_t index) const;
// Reads the header, direction, media type, header length and packet length
// from the RTP event at |index|, and stores the values in the corresponding
// output parameters. The output parameters can be set to nullptr if those
// values aren't needed.
// NB: The header must have space for at least IP_PACKET_SIZE bytes.
void GetRtpHeader(size_t index,
PacketDirection* incoming,
MediaType* media_type,
uint8_t* header,
size_t* header_length,
size_t* total_length) const;
// Reads packet, direction, media type and packet length from the RTCP event
// at |index|, and stores the values in the corresponding output parameters.
// The output parameters can be set to nullptr if those values aren't needed.
// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
void GetRtcpPacket(size_t index,
PacketDirection* incoming,
MediaType* media_type,
uint8_t* packet,
size_t* length) const;
// Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
// Only the fields that are stored in the protobuf will be written.
void GetVideoReceiveConfig(size_t index,
VideoReceiveStream::Config* config) const;
// Reads a config event to a (non-NULL) VideoSendStream::Config struct.
// Only the fields that are stored in the protobuf will be written.
void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) const;
// Reads the SSRC from the audio playout event at |index|. The SSRC is stored
// in the output parameter ssrc. The output parameter can be set to nullptr
// and in that case the function only asserts that the event is well formed.
void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
// Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
// expected packets from the BWE event at |index| and stores the values in
// the corresponding output parameters. The output parameters can be set to
// nullptr if those values aren't needed.
// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
void GetBwePacketLossEvent(size_t index,
int32_t* bitrate,
uint8_t* fraction_loss,
int32_t* total_packets) const;
private:
std::vector<rtclog::Event> stream_;
};
} // namespace webrtc
#endif // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_