blob: d06a7d007f5c1cd46de1a3c866bff7c5494ccdcb [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 LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
namespace webrtc {
class RtcEventLogTestHelper {
public:
static void VerifyVideoReceiveStreamConfig(
const ParsedRtcEventLogNew& parsed_log,
size_t index,
const rtclog::StreamConfig& config);
static void VerifyVideoSendStreamConfig(
const ParsedRtcEventLogNew& parsed_log,
size_t index,
const rtclog::StreamConfig& config);
static void VerifyAudioReceiveStreamConfig(
const ParsedRtcEventLogNew& parsed_log,
size_t index,
const rtclog::StreamConfig& config);
static void VerifyAudioSendStreamConfig(
const ParsedRtcEventLogNew& parsed_log,
size_t index,
const rtclog::StreamConfig& config);
static void VerifyIncomingRtpEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index,
const RtpPacketReceived& expected_packet);
static void VerifyOutgoingRtpEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index,
const RtpPacketToSend& expected_packet);
static void VerifyRtcpEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index,
PacketDirection direction,
const uint8_t* packet,
size_t total_size);
static void VerifyPlayoutEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index,
uint32_t ssrc);
static void VerifyBweLossEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index,
int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets);
static void VerifyBweDelayEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index,
int32_t bitrate,
BandwidthUsage detector_state);
static void VerifyAudioNetworkAdaptation(
const ParsedRtcEventLogNew& parsed_log,
size_t index,
const AudioEncoderRuntimeConfig& config);
static void VerifyLogStartEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index);
static void VerifyLogEndEvent(const ParsedRtcEventLogNew& parsed_log,
size_t index);
static void VerifyBweProbeCluster(const ParsedRtcEventLogNew& parsed_log,
size_t index,
int32_t id,
int32_t bitrate_bps,
uint32_t min_probes,
uint32_t min_bytes);
static void VerifyProbeResultSuccess(const ParsedRtcEventLogNew& parsed_log,
size_t index,
int32_t id,
int32_t bitrate_bps);
static void VerifyProbeResultFailure(const ParsedRtcEventLogNew& parsed_log,
size_t index,
int32_t id,
ProbeFailureReason failure_reason);
};
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_