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https://github.com/mollyim/webrtc.git
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440216fcf3
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>. Split LogSessionAndReadBack into three functions and create class to share state between them. Split VerifyRtpEvent into one incoming and one outgoing version. Originally uploaded as https://codereview.webrtc.org/2997973002/ Bug: webrtc:8111 Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3 Reviewed-on: https://webrtc-review.googlesource.com/5020 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20063}
91 lines
4.1 KiB
C++
91 lines
4.1 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
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#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
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#include "call/call.h"
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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namespace webrtc {
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class RtcEventLogTestHelper {
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public:
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static void VerifyVideoReceiveStreamConfig(
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const ParsedRtcEventLog& parsed_log,
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size_t index,
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const rtclog::StreamConfig& config);
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static void VerifyVideoSendStreamConfig(const ParsedRtcEventLog& parsed_log,
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size_t index,
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const rtclog::StreamConfig& config);
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static void VerifyAudioReceiveStreamConfig(
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const ParsedRtcEventLog& parsed_log,
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size_t index,
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const rtclog::StreamConfig& config);
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static void VerifyAudioSendStreamConfig(const ParsedRtcEventLog& parsed_log,
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size_t index,
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const rtclog::StreamConfig& config);
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static void VerifyIncomingRtpEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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const RtpPacketReceived& expected_packet);
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static void VerifyOutgoingRtpEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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const RtpPacketToSend& expected_packet);
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static void VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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PacketDirection direction,
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const uint8_t* packet,
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size_t total_size);
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static void VerifyPlayoutEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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uint32_t ssrc);
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static void VerifyBweLossEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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int32_t bitrate,
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uint8_t fraction_loss,
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int32_t total_packets);
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static void VerifyBweDelayEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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int32_t bitrate,
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BandwidthUsage detector_state);
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static void VerifyAudioNetworkAdaptation(
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const ParsedRtcEventLog& parsed_log,
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size_t index,
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const AudioEncoderRuntimeConfig& config);
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static void VerifyLogStartEvent(const ParsedRtcEventLog& parsed_log,
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size_t index);
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static void VerifyLogEndEvent(const ParsedRtcEventLog& parsed_log,
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size_t index);
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static void VerifyBweProbeCluster(const ParsedRtcEventLog& parsed_log,
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size_t index,
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uint32_t id,
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uint32_t bitrate_bps,
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uint32_t min_probes,
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uint32_t min_bytes);
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static void VerifyProbeResultSuccess(const ParsedRtcEventLog& parsed_log,
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size_t index,
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uint32_t id,
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uint32_t bitrate_bps);
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static void VerifyProbeResultFailure(const ParsedRtcEventLog& parsed_log,
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size_t index,
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uint32_t id,
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ProbeFailureReason failure_reason);
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};
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} // namespace webrtc
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#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
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