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Unified Diff: webrtc/call/rtc_event_log_unittest.cc

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: Comments from ivoc Created 4 years, 8 months ago
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Index: webrtc/call/rtc_event_log_unittest.cc
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index e3104591d9ab55847f424d97c8dae5451dc20f7f..0ebb59496d7a363b5997065cf70e4179495a6125 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -22,6 +22,8 @@
#include "webrtc/base/thread.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/call/rtc_event_log_unittest_helper.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
@@ -55,242 +57,6 @@ const size_t kNumExtensions = 5;
} // namespace
-// TODO(terelius): Place this definition with other parsing functions?
-MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
- switch (media_type) {
- case rtclog::MediaType::ANY:
- return MediaType::ANY;
- case rtclog::MediaType::AUDIO:
- return MediaType::AUDIO;
- case rtclog::MediaType::VIDEO:
- return MediaType::VIDEO;
- case rtclog::MediaType::DATA:
- return MediaType::DATA;
- }
- RTC_NOTREACHED();
- return MediaType::ANY;
-}
-
-// Checks that the event has a timestamp, a type and exactly the data field
-// corresponding to the type.
-::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
- if (!event.has_timestamp_us())
- return ::testing::AssertionFailure() << "Event has no timestamp";
- if (!event.has_type())
- return ::testing::AssertionFailure() << "Event has no event type";
- rtclog::Event_EventType type = event.type();
- if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet())
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
- if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet())
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
- if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
- event.has_audio_playout_event())
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_audio_playout_event() ? "" : "no ")
- << "audio_playout event";
- if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
- event.has_video_receiver_config())
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_video_receiver_config() ? "" : "no ")
- << "receiver config";
- if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
- event.has_video_sender_config())
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_video_sender_config() ? "" : "no ") << "sender config";
- if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
- event.has_audio_receiver_config()) {
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_audio_receiver_config() ? "" : "no ")
- << "audio receiver config";
- }
- if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
- event.has_audio_sender_config()) {
- return ::testing::AssertionFailure()
- << "Event of type " << type << " has "
- << (event.has_audio_sender_config() ? "" : "no ")
- << "audio sender config";
- }
- return ::testing::AssertionSuccess();
-}
-
-void VerifyReceiveStreamConfig(const rtclog::Event& event,
- const VideoReceiveStream::Config& config) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
- const rtclog::VideoReceiveConfig& receiver_config =
- event.video_receiver_config();
- // Check SSRCs.
- ASSERT_TRUE(receiver_config.has_remote_ssrc());
- EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
- ASSERT_TRUE(receiver_config.has_local_ssrc());
- EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
- // Check RTCP settings.
- ASSERT_TRUE(receiver_config.has_rtcp_mode());
- if (config.rtp.rtcp_mode == RtcpMode::kCompound)
- EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
- receiver_config.rtcp_mode());
- else
- EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
- receiver_config.rtcp_mode());
- ASSERT_TRUE(receiver_config.has_remb());
- EXPECT_EQ(config.rtp.remb, receiver_config.remb());
- // Check RTX map.
- ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()),
- receiver_config.rtx_map_size());
- for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
- ASSERT_TRUE(rtx_map.has_payload_type());
- ASSERT_TRUE(rtx_map.has_config());
- EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type()));
- const rtclog::RtxConfig& rtx_config = rtx_map.config();
- const VideoReceiveStream::Config::Rtp::Rtx& rtx =
- config.rtp.rtx.at(rtx_map.payload_type());
- ASSERT_TRUE(rtx_config.has_rtx_ssrc());
- ASSERT_TRUE(rtx_config.has_rtx_payload_type());
- EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc());
- EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type());
- }
- // Check header extensions.
- ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
- receiver_config.header_extensions_size());
- for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
- ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
- ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
- const std::string& name = receiver_config.header_extensions(i).name();
- int id = receiver_config.header_extensions(i).id();
- EXPECT_EQ(config.rtp.extensions[i].id, id);
- EXPECT_EQ(config.rtp.extensions[i].name, name);
- }
- // Check decoders.
- ASSERT_EQ(static_cast<int>(config.decoders.size()),
- receiver_config.decoders_size());
- for (int i = 0; i < receiver_config.decoders_size(); i++) {
- ASSERT_TRUE(receiver_config.decoders(i).has_name());
- ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
- const std::string& decoder_name = receiver_config.decoders(i).name();
- int decoder_type = receiver_config.decoders(i).payload_type();
- EXPECT_EQ(config.decoders[i].payload_name, decoder_name);
- EXPECT_EQ(config.decoders[i].payload_type, decoder_type);
- }
-}
-
-void VerifySendStreamConfig(const rtclog::Event& event,
- const VideoSendStream::Config& config) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
- const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
- // Check SSRCs.
- ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()),
- sender_config.ssrcs_size());
- for (int i = 0; i < sender_config.ssrcs_size(); i++) {
- EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
- }
- // Check header extensions.
- ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
- sender_config.header_extensions_size());
- for (int i = 0; i < sender_config.header_extensions_size(); i++) {
- ASSERT_TRUE(sender_config.header_extensions(i).has_name());
- ASSERT_TRUE(sender_config.header_extensions(i).has_id());
- const std::string& name = sender_config.header_extensions(i).name();
- int id = sender_config.header_extensions(i).id();
- EXPECT_EQ(config.rtp.extensions[i].id, id);
- EXPECT_EQ(config.rtp.extensions[i].name, name);
- }
- // Check RTX settings.
- ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()),
- sender_config.rtx_ssrcs_size());
- for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
- EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i));
- }
- if (sender_config.rtx_ssrcs_size() > 0) {
- ASSERT_TRUE(sender_config.has_rtx_payload_type());
- EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
- }
- // Check encoder.
- ASSERT_TRUE(sender_config.has_encoder());
- ASSERT_TRUE(sender_config.encoder().has_name());
- ASSERT_TRUE(sender_config.encoder().has_payload_type());
- EXPECT_EQ(config.encoder_settings.payload_name,
- sender_config.encoder().name());
- EXPECT_EQ(config.encoder_settings.payload_type,
- sender_config.encoder().payload_type());
-}
-
-void VerifyRtpEvent(const rtclog::Event& event,
- bool incoming,
- MediaType media_type,
- const uint8_t* header,
- size_t header_size,
- size_t total_size) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
- const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
- ASSERT_TRUE(rtp_packet.has_incoming());
- EXPECT_EQ(incoming, rtp_packet.incoming());
- ASSERT_TRUE(rtp_packet.has_type());
- EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
- ASSERT_TRUE(rtp_packet.has_packet_length());
- EXPECT_EQ(total_size, rtp_packet.packet_length());
- ASSERT_TRUE(rtp_packet.has_header());
- ASSERT_EQ(header_size, rtp_packet.header().size());
- for (size_t i = 0; i < header_size; i++) {
- EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
- }
-}
-
-void VerifyRtcpEvent(const rtclog::Event& event,
- bool incoming,
- MediaType media_type,
- const uint8_t* packet,
- size_t total_size) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
- const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
- ASSERT_TRUE(rtcp_packet.has_incoming());
- EXPECT_EQ(incoming, rtcp_packet.incoming());
- ASSERT_TRUE(rtcp_packet.has_type());
- EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
- ASSERT_TRUE(rtcp_packet.has_packet_data());
- ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
- for (size_t i = 0; i < total_size; i++) {
- EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
- }
-}
-
-void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
- const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
- ASSERT_TRUE(playout_event.has_local_ssrc());
- EXPECT_EQ(ssrc, playout_event.local_ssrc());
-}
-
-void VerifyBweLossEvent(const rtclog::Event& event,
- int32_t bitrate,
- uint8_t fraction_loss,
- int32_t total_packets) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
- const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
- ASSERT_TRUE(bwe_event.has_bitrate());
- EXPECT_EQ(bitrate, bwe_event.bitrate());
- ASSERT_TRUE(bwe_event.has_fraction_loss());
- EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
- ASSERT_TRUE(bwe_event.has_total_packets());
- EXPECT_EQ(total_packets, bwe_event.total_packets());
-}
-
-void VerifyLogStartEvent(const rtclog::Event& event) {
- ASSERT_TRUE(IsValidBasicEvent(event));
- EXPECT_EQ(rtclog::Event::LOG_START, event.type());
-}
/*
* Bit number i of extension_bitvector is set to indicate the
@@ -508,53 +274,56 @@ void LogSessionAndReadBack(size_t rtp_count,
}
// Read the generated file from disk.
- rtclog::EventStream parsed_stream;
+ ParsedRtcEventLog parsed_log;
- ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+ ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
// Verify that what we read back from the event log is the same as
// what we wrote down. For RTCP we log the full packets, but for
// RTP we should only log the header.
- const int event_count = config_count + playout_count + bwe_loss_count +
- rtcp_count + rtp_count + 1;
- EXPECT_EQ(event_count, parsed_stream.stream_size());
- VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
- VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
+ const size_t event_count = config_count + playout_count + bwe_loss_count +
+ rtcp_count + rtp_count + 1;
+ EXPECT_EQ(event_count, parsed_log.GetNumberOfEvents());
+ RtcEventLogTestHelper::VerifyReceiveStreamConfig(parsed_log, 0,
+ receiver_config);
+ RtcEventLogTestHelper::VerifySendStreamConfig(parsed_log, 1, sender_config);
size_t event_index = config_count;
size_t rtcp_index = 1;
size_t playout_index = 1;
size_t bwe_loss_index = 1;
for (size_t i = 1; i <= rtp_count; i++) {
- VerifyRtpEvent(parsed_stream.stream(event_index),
- (i % 2 == 0), // Every second packet is incoming.
- (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
- rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
- rtp_packets[i - 1].size());
+ RtcEventLogTestHelper::VerifyRtpEvent(
+ parsed_log, event_index,
+ (i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
+ (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+ rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
+ rtp_packets[i - 1].size());
event_index++;
if (i * rtcp_count >= rtcp_index * rtp_count) {
- VerifyRtcpEvent(parsed_stream.stream(event_index),
- rtcp_index % 2 == 0, // Every second packet is incoming.
- rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
- rtcp_packets[rtcp_index - 1].data(),
- rtcp_packets[rtcp_index - 1].size());
+ RtcEventLogTestHelper::VerifyRtcpEvent(
+ parsed_log, event_index,
+ rtcp_index % 2 == 0 ? kIncomingPacket : kOutgoingPacket,
+ rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
+ rtcp_packets[rtcp_index - 1].data(),
+ rtcp_packets[rtcp_index - 1].size());
event_index++;
rtcp_index++;
}
if (i * playout_count >= playout_index * rtp_count) {
- VerifyPlayoutEvent(parsed_stream.stream(event_index),
- playout_ssrcs[playout_index - 1]);
+ RtcEventLogTestHelper::VerifyPlayoutEvent(
+ parsed_log, event_index, playout_ssrcs[playout_index - 1]);
event_index++;
playout_index++;
}
if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
- VerifyBweLossEvent(parsed_stream.stream(event_index),
- bwe_loss_updates[bwe_loss_index - 1].first,
- bwe_loss_updates[bwe_loss_index - 1].second, i);
+ RtcEventLogTestHelper::VerifyBweLossEvent(
+ parsed_log, event_index, bwe_loss_updates[bwe_loss_index - 1].first,
+ bwe_loss_updates[bwe_loss_index - 1].second, i);
event_index++;
bwe_loss_index++;
}
if (i == rtp_count / 2) {
- VerifyLogStartEvent(parsed_stream.stream(event_index));
+ RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, event_index);
event_index++;
}
}
@@ -660,21 +429,23 @@ void DropOldEvents(uint32_t extensions_bitvector,
}
// Read the generated file from disk.
- rtclog::EventStream parsed_stream;
- ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+ ParsedRtcEventLog parsed_log;
+ ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
// Verify that what we read back from the event log is the same as
// what we wrote. Old RTP and RTCP events should have been discarded,
// but old configuration events should still be available.
- EXPECT_EQ(5, parsed_stream.stream_size());
- VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
- VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
- VerifyLogStartEvent(parsed_stream.stream(2));
- VerifyRtpEvent(parsed_stream.stream(3), true, MediaType::VIDEO,
- recent_rtp_packet.data(), recent_header_size,
- recent_rtp_packet.size());
- VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO,
- recent_rtcp_packet.data(), recent_rtcp_packet.size());
+ EXPECT_EQ(5u, parsed_log.GetNumberOfEvents());
+ RtcEventLogTestHelper::VerifyReceiveStreamConfig(parsed_log, 0,
+ receiver_config);
+ RtcEventLogTestHelper::VerifySendStreamConfig(parsed_log, 1, sender_config);
+ RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 2);
+ RtcEventLogTestHelper::VerifyRtpEvent(
+ parsed_log, 3, kIncomingPacket, MediaType::VIDEO,
+ recent_rtp_packet.data(), recent_header_size, recent_rtp_packet.size());
+ RtcEventLogTestHelper::VerifyRtcpEvent(
+ parsed_log, 4, kOutgoingPacket, MediaType::VIDEO,
+ recent_rtcp_packet.data(), recent_rtcp_packet.size());
// Clean up temporary file - can be pretty slow.
remove(temp_filename.c_str());

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