d5c1a0bd5d
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt. Review-Url: https://codereview.webrtc.org/1768773002 Cr-Commit-Position: refs/heads/master@{#12714}
59 lines
2.4 KiB
C++
59 lines
2.4 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 WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
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#define WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
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#include "webrtc/call.h"
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#include "webrtc/call/rtc_event_log_parser.h"
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namespace webrtc {
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class RtcEventLogTestHelper {
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public:
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static void VerifyReceiveStreamConfig(
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const ParsedRtcEventLog& parsed_log,
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size_t index,
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const VideoReceiveStream::Config& config);
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static void VerifySendStreamConfig(const ParsedRtcEventLog& parsed_log,
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size_t index,
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const VideoSendStream::Config& config);
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static void VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
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size_t index,
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PacketDirection direction,
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MediaType media_type,
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const uint8_t* header,
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size_t header_size,
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size_t total_size);
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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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MediaType media_type,
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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 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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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
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