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}
115 lines
4.2 KiB
C++
115 lines
4.2 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_PARSER_H_
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#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
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#include <string>
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#include <vector>
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#include "webrtc/call/rtc_event_log.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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// Files generated at build-time by the protobuf compiler.
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
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#else
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#include "webrtc/call/rtc_event_log.pb.h"
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#endif
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namespace webrtc {
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enum class MediaType;
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class ParsedRtcEventLog {
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friend class RtcEventLogTestHelper;
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public:
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enum EventType {
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UNKNOWN_EVENT = 0,
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LOG_START = 1,
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LOG_END = 2,
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RTP_EVENT = 3,
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RTCP_EVENT = 4,
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AUDIO_PLAYOUT_EVENT = 5,
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BWE_PACKET_LOSS_EVENT = 6,
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BWE_PACKET_DELAY_EVENT = 7,
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VIDEO_RECEIVER_CONFIG_EVENT = 8,
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VIDEO_SENDER_CONFIG_EVENT = 9,
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AUDIO_RECEIVER_CONFIG_EVENT = 10,
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AUDIO_SENDER_CONFIG_EVENT = 11
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};
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// Reads an RtcEventLog file and returns true if parsing was successful.
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bool ParseFile(const std::string& file_name);
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// Returns the number of events in an EventStream.
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size_t GetNumberOfEvents() const;
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// Reads the arrival timestamp (in microseconds) from a rtclog::Event.
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int64_t GetTimestamp(size_t index) const;
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// Reads the event type of the rtclog::Event at |index|.
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EventType GetEventType(size_t index) const;
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// Reads the header, direction, media type, header length and packet length
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// from the RTP event at |index|, and stores the values in the corresponding
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// output parameters. The output parameters can be set to nullptr if those
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// values aren't needed.
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// NB: The header must have space for at least IP_PACKET_SIZE bytes.
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void GetRtpHeader(size_t index,
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PacketDirection* incoming,
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MediaType* media_type,
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uint8_t* header,
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size_t* header_length,
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size_t* total_length) const;
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// Reads packet, direction, media type and packet length from the RTCP event
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// at |index|, and stores the values in the corresponding output parameters.
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// The output parameters can be set to nullptr if those values aren't needed.
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// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
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void GetRtcpPacket(size_t index,
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PacketDirection* incoming,
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MediaType* media_type,
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uint8_t* packet,
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size_t* length) const;
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// Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
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// Only the fields that are stored in the protobuf will be written.
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void GetVideoReceiveConfig(size_t index,
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VideoReceiveStream::Config* config) const;
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// Reads a config event to a (non-NULL) VideoSendStream::Config struct.
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// Only the fields that are stored in the protobuf will be written.
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void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) const;
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// Reads the SSRC from the audio playout event at |index|. The SSRC is stored
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// in the output parameter ssrc. The output parameter can be set to nullptr
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// and in that case the function only asserts that the event is well formed.
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void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
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// Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
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// expected packets from the BWE event at |index| and stores the values in
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// the corresponding output parameters. The output parameters can be set to
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// nullptr if those values aren't needed.
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// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
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void GetBwePacketLossEvent(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) const;
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private:
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std::vector<rtclog::Event> stream_;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
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