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The new API stores events gathered by event type. For example, it is possible to ask for a list of all incoming RTCP messages or all audio playout events. The new API is experimental and may change over next few weeks. Once it has stabilized and all unit tests and existing tools have been ported to the new API, the old one will be removed. This CL also updates the event_log_visualizer tool to use the new parser API. This is not a funcional change except for: - Incoming and outgoing audio level are now drawn in two separate plots. - Incoming and outgoing timstamps are now drawn in two separate plots. - RTCP count is no longer split into Video and Audio. It also counts all RTCP packets rather than only specific message types. - Slight timing difference in sendside BWE simulation due to only iterating over transport feedbacks and not over all RTCP packets. This timing changes are not visible in the plots. Media type for RTCP messages might not be identified correctly by rtc_event_log2text anymore. On the other hand, assigning a specific media type to an RTCP packet was a bit hacky to begin with. Bug: webrtc:8111 Change-Id: Ib244338c86a2c1a010c668a7aba440482023b512 Reviewed-on: https://webrtc-review.googlesource.com/73140 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23056}
93 lines
4 KiB
C++
93 lines
4 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_new.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 ParsedRtcEventLogNew& parsed_log,
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size_t index,
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const rtclog::StreamConfig& config);
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static void VerifyVideoSendStreamConfig(
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const ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& parsed_log,
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size_t index,
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const rtclog::StreamConfig& config);
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static void VerifyAudioSendStreamConfig(
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const ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& parsed_log,
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size_t index,
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uint32_t ssrc);
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static void VerifyBweLossEvent(const ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& parsed_log,
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size_t index,
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const AudioEncoderRuntimeConfig& config);
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static void VerifyLogStartEvent(const ParsedRtcEventLogNew& parsed_log,
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size_t index);
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static void VerifyLogEndEvent(const ParsedRtcEventLogNew& parsed_log,
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size_t index);
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static void VerifyBweProbeCluster(const ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& 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 ParsedRtcEventLogNew& 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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