| /* |
| * 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_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_ |
| #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_ |
| |
| #include "webrtc/call/call.h" |
| #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h" |
| |
| namespace webrtc { |
| |
| class RtcEventLogTestHelper { |
| public: |
| static void VerifyVideoReceiveStreamConfig( |
| const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| const rtclog::StreamConfig& config); |
| static void VerifyVideoSendStreamConfig(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| const rtclog::StreamConfig& config); |
| static void VerifyAudioReceiveStreamConfig( |
| const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| const rtclog::StreamConfig& config); |
| static void VerifyAudioSendStreamConfig(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| const rtclog::StreamConfig& config); |
| static void VerifyRtpEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| PacketDirection direction, |
| const uint8_t* header, |
| size_t header_size, |
| size_t total_size); |
| static void VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| PacketDirection direction, |
| const uint8_t* packet, |
| size_t total_size); |
| static void VerifyPlayoutEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| uint32_t ssrc); |
| static void VerifyBweLossEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| int32_t bitrate, |
| uint8_t fraction_loss, |
| int32_t total_packets); |
| static void VerifyBweDelayEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| int32_t bitrate, |
| BandwidthUsage detector_state); |
| |
| static void VerifyAudioNetworkAdaptation( |
| const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| const AudioEncoderRuntimeConfig& config); |
| |
| static void VerifyLogStartEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index); |
| static void VerifyLogEndEvent(const ParsedRtcEventLog& parsed_log, |
| size_t index); |
| |
| static void VerifyBweProbeCluster(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| uint32_t id, |
| uint32_t bitrate_bps, |
| uint32_t min_probes, |
| uint32_t min_bytes); |
| |
| static void VerifyProbeResultSuccess(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| uint32_t id, |
| uint32_t bitrate_bps); |
| |
| static void VerifyProbeResultFailure(const ParsedRtcEventLog& parsed_log, |
| size_t index, |
| uint32_t id, |
| ProbeFailureReason failure_reason); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_ |