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Unified Diff: webrtc/call/rtc_event_log_parser.h

Issue 1768773002: New parser for event log. Manually parse the outermost EventStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Move parser to its own build target. Fix comments from Henrik. Created 4 years, 9 months ago
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Index: webrtc/call/rtc_event_log_parser.h
diff --git a/webrtc/call/rtc_event_log_parser.h b/webrtc/call/rtc_event_log_parser.h
new file mode 100644
index 0000000000000000000000000000000000000000..d02fa73b7dee1d0ff5e2d2bc344e5c6700dd0782
--- /dev/null
+++ b/webrtc/call/rtc_event_log_parser.h
@@ -0,0 +1,113 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
ivoc 2016/04/01 08:27:59 Should be 2016 I guess?
terelius 2016/04/19 17:01:45 I created the file in 2015 when I started on the p
ivoc 2016/04/25 08:08:12 Ok, I'll leave it up to you :)
+ *
+ * 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_CALL_RTC_EVENT_LOG_PARSER_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/video_receive_stream.h"
+#include "webrtc/video_send_stream.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+
+enum class MediaType;
+
+class ParsedRtcEventLog {
+ friend class RtcEventLogTestHelper;
+
+ public:
+ enum EventType {
ivoc 2016/04/01 08:27:59 Would it be possible to reuse the enum from the ge
terelius 2016/04/19 17:01:45 It is non-trivial. The enum is actually named Even
ivoc 2016/04/25 08:08:12 Acknowledged.
+ UNKNOWN_EVENT = 0,
+ LOG_START = 1,
+ LOG_END = 2,
+ RTP_EVENT = 3,
+ RTCP_EVENT = 4,
+ AUDIO_PLAYOUT_EVENT = 5,
+ BWE_PACKET_LOSS_EVENT = 6,
+ BWE_PACKET_DELAY_EVENT = 7,
+ VIDEO_RECEIVER_CONFIG_EVENT = 8,
+ VIDEO_SENDER_CONFIG_EVENT = 9,
+ AUDIO_RECEIVER_CONFIG_EVENT = 10,
+ AUDIO_SENDER_CONFIG_EVENT = 11
+ };
+
+ // Reads an RtcEventLog file and returns true if parsing was successful.
+ bool ParseFile(const std::string& file_name);
ivoc 2016/04/01 08:27:59 I guess it doesn't make a big difference either wa
terelius 2016/04/19 17:01:45 Yeah, I though about that. I don't have any strong
ivoc 2016/04/25 08:08:12 Ok, sounds reasonable.
+
+ // Returns the number of events in an EventStream.
+ size_t GetNumberOfEvents() const;
+
+ // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
+ int64_t GetTimestamp(size_t i) const;
ivoc 2016/04/01 08:27:59 I would prefer a more descriptive name instead of
terelius 2016/04/19 17:01:45 Done.
+
+ // Reads the event type of the rtclog::Event at index i.
+ EventType GetEventType(size_t i) const;
+
+ // Reads the header, direction, media type, header length and packet length
+ // from the RTP event at index i. The values are stored in the corresponding
+ // output parameters. If some value is irrelevant, then that output parameter
ivoc 2016/04/01 08:27:59 I think this could be rephrased a bit more clearly
terelius 2016/04/19 17:01:45 Done.
+ // can be set to NULL.
+ // NB: The header must have space for at least IP_PACKET_SIZE bytes.
+ void GetRtpHeader(size_t i,
+ PacketDirection* incoming,
+ MediaType* media_type,
+ uint8_t* header,
+ size_t* header_length,
+ size_t* total_length) const;
+
+ // Reads packet, direction, media type and packet length from the RTCP event
+ // at index i. The values are stored in the corresponding output parameters.
+ // If some value is irrelevant, then that output parameter can be set to NULL.
+ // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+ void GetRtcpPacket(size_t i,
+ PacketDirection* incoming,
+ MediaType* media_type,
+ uint8_t* packet,
+ size_t* length) const;
+
+ // Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
+ // Only the fields that are stored in the protobuf will be written.
+ void GetVideoReceiveConfig(size_t i,
+ VideoReceiveStream::Config* config) const;
+
+ // Reads a config event to a (non-NULL) VideoSendStream::Config struct.
+ // Only the fields that are stored in the protobuf will be written.
+ void GetVideoSendConfig(size_t i, VideoSendStream::Config* config) const;
+
+ // Reads the SSRC from the audio playout event at index i. The SSRC is stored
+ // in the output parameter ssrc. To only check that the event is well formed,
+ // the output parameter can be set to NULL.
+ void GetAudioPlayout(size_t i, uint32_t* ssrc) const;
+
+ // Reads packet, direction, media type and packet length from the RTCP event
+ // at index i. The values are stored in the corresponding output parameters.
+ // If some value is irrelevant, then that output parameter can be set to NULL.
+ // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+ void GetBwePacketLossEvent(size_t i,
+ int32_t* bitrate,
+ uint8_t* fraction_loss,
+ int32_t* total_packets) const;
+
+ private:
+ std::vector<rtclog::Event> stream_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_

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