Adds logging of configuration information for VideoReceiveStream

webrtc/video/rtc_event_log_unittest.cc

+/*
+ *  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/call.h"  // For MediaType definition
+#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"
+#include "webrtc/video/rtc_event_log.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";
+  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>());
+    rtp_packets[i].reserve(packet_size);
+    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