Adds logging of configuration information for VideoReceiveStream

webrtc/video/rtc_event_log_unittest.cc

Index: webrtc/video/rtc_event_log_unittest.cc
diff --git a/webrtc/video/rtc_event_log_unittest.cc b/webrtc/video/rtc_event_log_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..f0ccb70496cb9165b46339b4a3b39c5dfaa9a07a
--- /dev/null
+++ b/webrtc/video/rtc_event_log_unittest.cc
@@ -0,0 +1,425 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  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.
+ */
+
+#ifdef ENABLE_RTC_EVENT_LOG
+
+#include <stdio.h>
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/video/rtc_event_log.h"

[ivoc, 2015/07/17 12:14:29]

Could you sort these alphabetically?

[terelius, 2015/07/17 15:17:40]

Done.

+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/test/test_suite.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
+#else
+#include "webrtc/video/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+
+// Checks that the event has a timestamp, a type and exactly the data field
+// corresponding to the type.
+::testing::AssertionResult IsValidBasicEvent(const RelEvent& 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";
+  RelEvent_EventType type = event.type();
+  if ((type == RelEvent::RTP_EVENT) != event.has_rtp_packet())
+    return ::testing::AssertionFailure() << "Event of type" << type
+        << "has" << (event.has_rtp_packet()?"":"no") << "RTP packet";

[ivoc, 2015/07/17 12:14:29]

I wonder how this error message would look. It seems like there should be spaces
before & after each argument, e.g. "Event of type " << type. Also I'm guessing
the type itself will probably be printed as an int. We could just replace it
with the corresponding text in each of the ifs, e.g. "Event of type RTP_EVENT
has ". Alternatively we could make a function to convert enums to human-readable
strings.

[terelius, 2015/07/17 15:17:40]

You're right about the spaces. I'll fix that.

We can't replace 'type' with RTP_EVENT because the code will execute both if it
is an RTP_EVENT that doesn't have an rtp_packet, but also if it is another event
type that *does* have an rtp_packet even though it shouldn't.

+  if ((type == RelEvent::RTCP_EVENT) != event.has_rtcp_packet())
+    return ::testing::AssertionFailure() << "Event of type" << type
+        << "has" << (event.has_rtcp_packet()?"":"no") << "RTCP packet";
+  if ((type == RelEvent::DEBUG_EVENT) != event.has_debug_event())
+    return ::testing::AssertionFailure() << "Event of type" << type
+        << "has" << (event.has_debug_event()?"":"no") << "debug event";
+  if ((type == RelEvent::RECEIVER_CONFIG_EVENT) != event.has_receiver_config())
+    return ::testing::AssertionFailure() << "Event of type" << type
+        << "has" << (event.has_receiver_config()?"":"no") << "receiver config";
+  if ((type == RelEvent::SENDER_CONFIG_EVENT) != event.has_sender_config())
+    return ::testing::AssertionFailure() << "Event of type" << type
+        << "has" << (event.has_sender_config()?"":"no") << "sender config";
+  if ((type == RelEvent::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 == RelEvent::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 RelEvent& event,
+                               const VideoReceiveStream::Config& config) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(RelEvent::RECEIVER_CONFIG_EVENT, event.type());
+  const RelVideoReceiveConfig& receiver_config = event.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 == newapi::kRtcpCompound)
+    EXPECT_EQ(RelVideoReceiveConfig::RTCP_COMPOUND,
+              receiver_config.rtcp_mode());
+  else
+    EXPECT_EQ(RelVideoReceiveConfig::RTCP_REDUCEDSIZE,
+              receiver_config.rtcp_mode());
+  ASSERT_TRUE(receiver_config.has_receiver_reference_time_report());
+  EXPECT_EQ(config.rtp.rtcp_xr.receiver_reference_time_report,
+            receiver_config.receiver_reference_time_report());
+  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 (int i = 0; i < receiver_config.rtx_map_size(); i++) {
+    const RtxMap& mapping = receiver_config.rtx_map(i);
+    ASSERT_TRUE(mapping.has_payload_type());
+    ASSERT_TRUE(mapping.has_config());
+    EXPECT_EQ(1,
+              static_cast<int>(config.rtp.rtx.count(mapping.payload_type())));
+    const RtxConfig& rtx_config = mapping.config();
+    const VideoReceiveStream::Config::Rtp::Rtx& rtx =
+        config.rtp.rtx.at(mapping.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 RelEvent& event,
+                            const VideoSendStream::Config& config) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(RelEvent::SENDER_CONFIG_EVENT, event.type());
+  const RelVideoSendConfig& sender_config = event.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 CNAME.
+  ASSERT_TRUE(sender_config.has_c_name());
+  EXPECT_EQ(config.rtp.c_name, sender_config.c_name());
+  // 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 RelEvent& event,
+                    bool incoming,
+                    MediaType media_type,
+                    uint8_t* header,
+                    size_t header_size,
+                    size_t total_size) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(RelEvent::RTP_EVENT, event.type());
+  const RelRtpPacket& rtp_packet = event.rtp_packet();
+  ASSERT_TRUE(rtp_packet.has_direction());
+  EXPECT_EQ(incoming ? RelRtpPacket::INCOMING : RelRtpPacket::OUTGOING,
+            rtp_packet.direction());
+  ASSERT_TRUE(rtp_packet.has_type());
+  if (media_type == MediaType::VIDEO)
+    EXPECT_EQ(RelRtpPacket::VIDEO, rtp_packet.type());
+  else if (media_type == MediaType::AUDIO)
+    EXPECT_EQ(RelRtpPacket::AUDIO, rtp_packet.type());
+  else
+    EXPECT_EQ(RelRtpPacket::UNKNOWN_TYPE, 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 RelEvent& event,
+                     bool incoming,
+                     MediaType media_type,
+                     uint8_t* packet,
+                     size_t total_size) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(RelEvent::RTCP_EVENT, event.type());
+  const RelRtcpPacket& rtcp_packet = event.rtcp_packet();
+  ASSERT_TRUE(rtcp_packet.has_direction());
+  EXPECT_EQ(incoming ? RelRtcpPacket::INCOMING : RelRtcpPacket::OUTGOING,
+      rtcp_packet.direction());
+  ASSERT_TRUE(rtcp_packet.has_type());
+  if (media_type == MediaType::VIDEO)
+    EXPECT_EQ(RelRtcpPacket::VIDEO, rtcp_packet.type());
+  else if (media_type == MediaType::AUDIO)
+    EXPECT_EQ(RelRtcpPacket::AUDIO, rtcp_packet.type());
+  else
+    EXPECT_EQ(RelRtcpPacket::UNKNOWN_TYPE, rtcp_packet.type());
+  ASSERT_TRUE(rtcp_packet.has_data());
+  ASSERT_EQ(total_size, rtcp_packet.data().size());
+  for (size_t i = 0; i < total_size; i++) {
+    EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.data()[i]));
+  }
+}
+
+void VerifyLogStartEvent(const RelEvent& event) {
+  ASSERT_TRUE(IsValidBasicEvent(event));
+  ASSERT_EQ(RelEvent::DEBUG_EVENT, event.type());
+  const RelDebugEvent& debug_event = event.debug_event();
+  ASSERT_TRUE(debug_event.has_type());
+  EXPECT_EQ(RelDebugEvent::LOG_START, debug_event.type());
+  // TODO(terelius): Deliberately not verifying that there is a message field
+  // since our protobuf file says that the message is optional. Make a decision.
+}
+
+void GenerateVideoReceiveConfig(webrtc::VideoReceiveStream::Config* config) {
+  // Create a map from a payload type to an encoder name.
+  VideoReceiveStream::Decoder decoder;
+  decoder.payload_type = rand();
+  decoder.payload_name = (rand() % 2 ? "VP8" : "H264");
+  config->decoders.push_back(decoder);
+  // Add SSRCs for the stream.
+  config->rtp.remote_ssrc = rand();
+  config->rtp.local_ssrc = rand();
+  // Add extensions and settings for RTCP.
+  config->rtp.rtcp_mode = rand() % 2 ? newapi::kRtcpCompound
+                                     : newapi::kRtcpReducedSize;
+  config->rtp.rtcp_xr.receiver_reference_time_report =
+      static_cast<bool>(rand() % 2);
+  config->rtp.remb = static_cast<bool>(rand() % 2);
+  // Add a map from a payload type to a new ssrc and a new payload type for RTX.
+  webrtc::VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
+  rtx_pair.ssrc = rand();
+  rtx_pair.payload_type = rand();
+  config->rtp.rtx.insert(std::make_pair(rand(), rtx_pair));
+  // Add two random header extensions.
+  const char* extension_name = rand() % 2 ? RtpExtension::kTOffset
+                                          : RtpExtension::kVideoRotation;
+  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
+  extension_name = rand() % 2 ? RtpExtension::kAudioLevel
+                              : RtpExtension::kAbsSendTime;
+  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
+}
+
+void GenerateVideoSendConfig(webrtc::VideoSendStream::Config* config) {
+  // Create a map from a payload type to an encoder name.
+  config->encoder_settings.payload_type = rand();
+  config->encoder_settings.payload_name = (rand() % 2 ? "VP8" : "H264");
+  // Add SSRCs for the stream.
+  config->rtp.ssrcs.push_back(rand());
+  // Add a map from a payload type to new ssrcs and a new payload type for RTX.
+  config->rtp.rtx.ssrcs.push_back(rand());
+  config->rtp.rtx.payload_type = rand();
+  // Add a CNAME.
+  config->rtp.c_name = "some.user@some.host";
+  // Add two random header extensions.
+  const char* extension_name = rand() % 2 ? RtpExtension::kTOffset
+                                          : RtpExtension::kVideoRotation;
+  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
+  extension_name = rand() % 2 ? RtpExtension::kAudioLevel
+                              : RtpExtension::kAbsSendTime;
+  config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
+}
+
+// Test for the RtcEventLog class. Dumps some RTP packets to disk, then reads
+// them back to see if they match.
+void LogSessionAndReadBack(size_t rtp_count, unsigned random_seed) {
+  std::vector<std::vector<uint8_t>> rtp_packets;
+  std::vector<uint8_t> incoming_rtcp_packet;
+  std::vector<uint8_t> outgoing_rtcp_packet;
+
+  webrtc::VideoReceiveStream::Config receiver_config;
+  webrtc::VideoSendStream::Config sender_config;
+
+  srand(random_seed);
+
+  // Create rtp_count RTP packets containing random data.
+  const size_t rtp_header_size = 20;
+  for (size_t i = 0; i < rtp_count; i++) {
+    size_t packet_size = 1000 + rand() % 30;
+    rtp_packets.push_back(std::vector<uint8_t>());

[ivoc, 2015/07/17 12:14:29]

It would be good to call reserve here as well.

[terelius, 2015/07/17 15:17:40]

Done. Well spotted!

I initially used the constructor to allocate a vector containing all zeroes and
then changed the elements instead of appending them with push_back.

+    for (size_t j = 0; j < packet_size; j++) {
+      rtp_packets[i].push_back(rand());
+    }
+  }
+  // Create two RTCP packets containing random data.
+  size_t packet_size = 1000 + rand() % 30;
+  outgoing_rtcp_packet.reserve(packet_size);
+  for (size_t j = 0; j < packet_size; j++) {
+    outgoing_rtcp_packet.push_back(rand());
+  }
+  packet_size = 1000 + rand() % 30;
+  incoming_rtcp_packet.reserve(packet_size);
+  for (size_t j = 0; j < packet_size; j++) {
+    incoming_rtcp_packet.push_back(rand());
+  }
+  // Create configurations for the video streams.
+  GenerateVideoReceiveConfig(&receiver_config);
+  GenerateVideoSendConfig(&sender_config);
+
+  // Find the name of the current test, in order to use it as a temporary
+  // filename.
+  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
+  const std::string temp_filename =
+      test::OutputPath() + test_info->test_case_name() + test_info->name();
+
+  // When log_dumper goes out of scope, it causes the log file to be flushed
+  // to disk.
+  {
+    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    log_dumper->LogVideoReceiveStreamConfig(receiver_config);
+    log_dumper->LogVideoSendStreamConfig(sender_config);
+    size_t i = 0;
+    for (; i < rtp_count / 2; i++) {
+      log_dumper->LogRtpHeader(
+          (i % 2 == 0),  // Every second packet is incoming.
+          (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+          rtp_packets[i].data(),
+          rtp_header_size,
+          rtp_packets[i].size());
+    }
+    log_dumper->LogRtcpPacket(false,
+                              MediaType::AUDIO,
+                              outgoing_rtcp_packet.data(),
+                              outgoing_rtcp_packet.size());
+    log_dumper->StartLogging(temp_filename, 10000000);
+    for (; i < rtp_count; i++) {
+      log_dumper->LogRtpHeader(
+          (i % 2 == 0),  // Every second packet is incoming,
+          (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+          rtp_packets[i].data(),
+          rtp_header_size,
+          rtp_packets[i].size());
+    }
+    log_dumper->LogRtcpPacket(true,
+                              MediaType::VIDEO,
+                              incoming_rtcp_packet.data(),
+                              incoming_rtcp_packet.size());
+  }
+
+  const int config_count = 2;
+  const int rtcp_count = 2;
+  const int debug_count = 1;  // Only LogStart event,
+  const int event_count = config_count + debug_count + rtcp_count + rtp_count;
+
+  // Read the generated file from disk.
+  RelEventStream parsed_stream;
+
+  ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
+
+  // Verify the result.
+  EXPECT_EQ(event_count, parsed_stream.stream_size());
+  VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
+  VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
+  size_t i = 0;
+  for (; i < rtp_count / 2; i++) {
+    VerifyRtpEvent(parsed_stream.stream(config_count + i),
+                   (i % 2 == 0),  // Every second packet is incoming.
+                   (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+                   rtp_packets[i].data(),
+                   rtp_header_size,
+                   rtp_packets[i].size());
+  }
+  VerifyRtcpEvent(parsed_stream.stream(config_count + rtp_count / 2),
+                  false,
+                  MediaType::AUDIO,
+                  outgoing_rtcp_packet.data(),
+                  outgoing_rtcp_packet.size());
+
+  VerifyLogStartEvent(parsed_stream.stream(1 + config_count + rtp_count / 2));
+  for (; i < rtp_count; i++) {
+    VerifyRtpEvent(parsed_stream.stream(2 + config_count + i),
+                   (i % 2 == 0),  // Every second packet is incoming.
+                   (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
+                   rtp_packets[i].data(),
+                   rtp_header_size,
+                   rtp_packets[i].size());
+  }
+  VerifyRtcpEvent(parsed_stream.stream(2 + config_count + rtp_count),
+                  true,
+                  MediaType::VIDEO,
+                  incoming_rtcp_packet.data(),
+                  incoming_rtcp_packet.size());
+
+  // Clean up temporary file - can be pretty slow.
+  remove(temp_filename.c_str());
+}
+
+TEST(RtcEventLogTest, LogSessionAndReadBack) {
+  LogSessionAndReadBack(5, 321);
+  LogSessionAndReadBack(8, 3141592653U);
+  LogSessionAndReadBack(9, 2718281828U);
+}
+
+}  // namespace webrtc
+
+#endif  // ENABLE_RTC_EVENT_LOG