Adds logging of configuration information for VideoReceiveStream

webrtc/modules/audio_coding/main/acm2/acm_dump_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.
  */
 
@@ skipping 13 matching lines @@
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h"
 #else
 #include "webrtc/audio_coding/dump.pb.h"
 #endif
 
 namespace webrtc {
 
-// Test for the acm dump class. Dumps some RTP packets to disk, then reads them
+void VerifyReceiveStreamConfig(const ACMDumpEvent& event,
+                               const VideoReceiveStream::Config& config) {
+  EXPECT_TRUE(event.has_timestamp_us());
+  ASSERT_TRUE(event.has_type());
+  EXPECT_EQ(ACMDumpEvent::RECEIVER_CONFIG_EVENT, event.type());
+  EXPECT_FALSE(event.has_rtp_packet());
+  EXPECT_FALSE(event.has_rtcp_packet());
+  EXPECT_FALSE(event.has_debug_event());
+  ASSERT_TRUE(event.has_receiver_config());
+  EXPECT_FALSE(event.has_sender_config());
+  EXPECT_FALSE(event.has_audio_receiver_config());
+  EXPECT_FALSE(event.has_audio_sender_config());
+  const ACMDumpVideoReceiveConfig& 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(ACMDumpVideoReceiveConfig::RTCP_COMPOUND,
+              receiver_config.rtcp_mode());
+  else
+    EXPECT_EQ(ACMDumpVideoReceiveConfig::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 =

[ivoc, 2015/07/14 12:13:14]

We could consider making this an auto, since it's such a long typename.

[terelius, 2015/07/16 12:47:03]

While I like using auto to hide obvious boiler-plate code (e.g. iterators,
yuck!), I prefer not to use it to hide actual type information. Doing so makes
it hard to edit the code without an IDE, imo. 

[ivoc, 2015/07/17 12:14:28]

Okay, that makes sense.

+        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 ACMDumpEvent& event,
+                            const VideoSendStream::Config& config) {
+  EXPECT_TRUE(event.has_timestamp_us());

[ivoc, 2015/07/14 12:13:14]

I think it would be a good idea to refactor the common verification checks
(timestamp, event type, etc) into a function that is used in multiple tests.

[stefan-webrtc, 2015/07/14 13:28:56]

Agree, these functions are very long. If possible it would be good to refactor,
assuming there's code duplication.

[terelius, 2015/07/16 12:47:03]

The first 10 lines of each function are similar. I've refactored to eliminate 9
of the lines at the cost of a rather ugly looking helper function. The
difficulty is that the gtest ASSERT_TRUE only returns from the current function,
so an assertion failure in a subroutine allows the surrounding test to continue.
I've instead used custom AssertionFailure()'s to pass information back to the
caller, so that the caller can use the ASSERT_TRUE, but still get the detailed
error message.

Apart from the first 10 lines, the functions are too different to benefit from
refactoring imo.

+  ASSERT_TRUE(event.has_type());
+  EXPECT_EQ(ACMDumpEvent::SENDER_CONFIG_EVENT, event.type());
+  EXPECT_FALSE(event.has_rtp_packet());
+  EXPECT_FALSE(event.has_rtcp_packet());
+  EXPECT_FALSE(event.has_debug_event());
+  EXPECT_FALSE(event.has_receiver_config());
+  ASSERT_TRUE(event.has_sender_config());
+  EXPECT_FALSE(event.has_audio_receiver_config());
+  EXPECT_FALSE(event.has_audio_sender_config());
+  const ACMDumpVideoSendConfig& 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 ACMDumpEvent& event,
+                    bool incoming,
+                    MediaType media_type,
+                    uint8_t* header,
+                    size_t header_size,
+                    size_t total_size) {
+  EXPECT_TRUE(event.has_timestamp_us());
+  ASSERT_TRUE(event.has_type());
+  EXPECT_EQ(ACMDumpEvent::RTP_EVENT, event.type());
+  ASSERT_TRUE(event.has_rtp_packet());
+  EXPECT_FALSE(event.has_rtcp_packet());
+  EXPECT_FALSE(event.has_debug_event());
+  EXPECT_FALSE(event.has_receiver_config());
+  EXPECT_FALSE(event.has_sender_config());
+  EXPECT_FALSE(event.has_audio_receiver_config());
+  EXPECT_FALSE(event.has_audio_sender_config());
+  const ACMDumpRtpPacket& rtp_packet = event.rtp_packet();
+  ASSERT_TRUE(rtp_packet.has_direction());
+  EXPECT_EQ(incoming ? ACMDumpRtpPacket::INCOMING : ACMDumpRtpPacket::OUTGOING,
+            rtp_packet.direction());
+  ASSERT_TRUE(rtp_packet.has_type());
+  if (media_type == MediaType::VIDEO)
+    EXPECT_EQ(ACMDumpRtpPacket::VIDEO, rtp_packet.type());
+  else if (media_type == MediaType::AUDIO)
+    EXPECT_EQ(ACMDumpRtpPacket::AUDIO, rtp_packet.type());
+  else
+    EXPECT_EQ(ACMDumpRtpPacket::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 ACMDumpEvent& event,
+                     bool incoming,
+                     MediaType media_type,
+                     uint8_t* packet,
+                     size_t total_size) {
+  EXPECT_TRUE(event.has_timestamp_us());
+  ASSERT_TRUE(event.has_type());
+  EXPECT_EQ(ACMDumpEvent::RTCP_EVENT, event.type());
+  EXPECT_FALSE(event.has_rtp_packet());
+  ASSERT_TRUE(event.has_rtcp_packet());
+  EXPECT_FALSE(event.has_debug_event());
+  EXPECT_FALSE(event.has_receiver_config());
+  EXPECT_FALSE(event.has_sender_config());
+  EXPECT_FALSE(event.has_audio_receiver_config());
+  EXPECT_FALSE(event.has_audio_sender_config());
+  const ACMDumpRtcpPacket& rtcp_packet = event.rtcp_packet();
+  ASSERT_TRUE(rtcp_packet.has_direction());
+  EXPECT_EQ(
+      incoming ? ACMDumpRtcpPacket::INCOMING : ACMDumpRtcpPacket::OUTGOING,
+      rtcp_packet.direction());
+  ASSERT_TRUE(rtcp_packet.has_type());
+  if (media_type == MediaType::VIDEO)
+    EXPECT_EQ(ACMDumpRtcpPacket::VIDEO, rtcp_packet.type());
+  else if (media_type == MediaType::AUDIO)
+    EXPECT_EQ(ACMDumpRtcpPacket::AUDIO, rtcp_packet.type());
+  else
+    EXPECT_EQ(ACMDumpRtcpPacket::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 ACMDumpEvent& event) {
+  EXPECT_TRUE(event.has_timestamp_us());
+  ASSERT_TRUE(event.has_type());
+  EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, event.type());
+  EXPECT_FALSE(event.has_rtp_packet());
+  EXPECT_FALSE(event.has_rtcp_packet());
+  ASSERT_TRUE(event.has_debug_event());
+  EXPECT_FALSE(event.has_receiver_config());
+  EXPECT_FALSE(event.has_sender_config());
+  EXPECT_FALSE(event.has_audio_receiver_config());
+  EXPECT_FALSE(event.has_audio_sender_config());
+  const ACMDumpDebugEvent& debug_event = event.debug_event();
+  ASSERT_TRUE(debug_event.has_type());
+  EXPECT_EQ(ACMDumpDebugEvent::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.

[ivoc, 2015/07/14 12:13:14]

I think that's fine, especially since the message field is not really used for
anything in the LOG_START event. Right now it contains an empty string, but I
guess that might change at some point.

[terelius, 2015/07/16 12:47:03]

We might want to make LogStart its own event type rather than trying to shoehorn
it into DebugEvent. If we don't want to store any data apart from wall clock
with the LogStart event we could even skip the event field and just make
LOG_START a value of the RelEvent::EventType enum.

[ivoc, 2015/07/17 12:14:28]

Good idea.

+}
+
+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 ACMDump class. Dumps some RTP packets to disk, then reads them
 // back to see if they match.
-class AcmDumpTest : public ::testing::Test {
- public:
-  void VerifyResults(const ACMDumpEventStream& parsed_stream,
-                     size_t packet_size) {
-    // Verify the result.
-    EXPECT_EQ(5, parsed_stream.stream_size());
-    const ACMDumpEvent& start_event = parsed_stream.stream(2);
-    ASSERT_TRUE(start_event.has_type());
-    EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, start_event.type());
-    EXPECT_TRUE(start_event.has_timestamp_us());
-    EXPECT_FALSE(start_event.has_packet());
-    ASSERT_TRUE(start_event.has_debug_event());
-    auto start_debug_event = start_event.debug_event();
-    ASSERT_TRUE(start_debug_event.has_type());
-    EXPECT_EQ(ACMDumpDebugEvent::LOG_START, start_debug_event.type());
-    ASSERT_TRUE(start_debug_event.has_message());
-
-    for (int i = 0; i < parsed_stream.stream_size(); i++) {
-      if (i == 2) {
-        // This is the LOG_START packet that was already verified.
-        continue;
-      }
-      const ACMDumpEvent& test_event = parsed_stream.stream(i);
-      ASSERT_TRUE(test_event.has_type());
-      EXPECT_EQ(ACMDumpEvent::RTP_EVENT, test_event.type());
-      EXPECT_TRUE(test_event.has_timestamp_us());
-      EXPECT_FALSE(test_event.has_debug_event());
-      ASSERT_TRUE(test_event.has_packet());
-      const ACMDumpRTPPacket& test_packet = test_event.packet();
-      ASSERT_TRUE(test_packet.has_direction());
-      if (i <= 1) {
-        EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction());
-      } else if (i >= 3) {
-        EXPECT_EQ(ACMDumpRTPPacket::OUTGOING, test_packet.direction());
-      }
-      ASSERT_TRUE(test_packet.has_rtp_data());
-      ASSERT_EQ(packet_size, test_packet.rtp_data().size());
-      for (size_t i = 0; i < packet_size; i++) {
-        EXPECT_EQ(rtp_packet_[i],
-                  static_cast<uint8_t>(test_packet.rtp_data()[i]));
-      }
+void LogSessionAndReadBack(size_t rtp_count, unsigned random_seed) {
+  std::vector<uint8_t*> rtp_packets;
+  std::vector<size_t> rtp_packet_sizes;

[ivoc, 2015/07/14 12:13:14]

I would prefer a vector<vector<uint8_t>> here, the sizes don't have to be stored
seperately then (each vector has a size).

[stefan-webrtc, 2015/07/14 13:28:56]

And you can also avoid the delete[]

[terelius, 2015/07/16 12:47:03]

Done.

+  uint8_t* incoming_rtcp_packet;
+  uint8_t* outgoing_rtcp_packet;

[ivoc, 2015/07/14 12:13:14]

These should be vector<uint8_t>.

[terelius, 2015/07/16 12:47:03]

Done.

+  size_t incoming_rtcp_packet_size;
+  size_t outgoing_rtcp_packet_size;
+  webrtc::VideoReceiveStream::Config receiver_config;
+  webrtc::VideoSendStream::Config sender_config;
+
+  srand(random_seed);
+
+  // Create rtp_count RTP packets containing random data.
+  size_t rtp_header_size = 20;

[ivoc, 2015/07/14 12:13:14]

Should be const.

[terelius, 2015/07/16 12:47:03]

Done.

+  for (size_t i = 0; i < rtp_count; i++) {
+    size_t packet_size = 1000 + rand() % 30;
+    rtp_packets.push_back(new uint8_t[packet_size]);
+    for (size_t j = 0; j < packet_size; j++) {
+      rtp_packets[i][j] = rand();
     }
-  }
-
-  void Run(int packet_size, int random_seed) {
-    rtp_packet_.clear();
-    rtp_packet_.reserve(packet_size);
-    srand(random_seed);
-    // Fill the packet vector with random data.
-    for (int i = 0; i < packet_size; i++) {
-      rtp_packet_.push_back(rand());
+    rtp_packet_sizes.push_back(packet_size);
+  }
+  // Create two RTCP packets containing random data.
+  outgoing_rtcp_packet_size = 1000 + rand() % 30;
+  outgoing_rtcp_packet = new uint8_t[outgoing_rtcp_packet_size];
+  for (size_t j = 0; j < outgoing_rtcp_packet_size; j++) {
+    outgoing_rtcp_packet[j] = rand();
+  }
+  incoming_rtcp_packet_size = 1000 + rand() % 30;
+  incoming_rtcp_packet = new uint8_t[incoming_rtcp_packet_size];
+  for (size_t j = 0; j < incoming_rtcp_packet_size; j++) {
+    incoming_rtcp_packet[j] = 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<AcmDump> log_dumper(AcmDump::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],
+          rtp_header_size,
+          rtp_packet_sizes[i]);
     }
-    // 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<AcmDump> log_dumper(AcmDump::Create());
-      log_dumper->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
-      log_dumper->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
-      log_dumper->StartLogging(temp_filename, 10000000);
-      log_dumper->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
-      log_dumper->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
+    log_dumper->LogRtcpPacket(false,
+                              MediaType::AUDIO,
+                              outgoing_rtcp_packet,
+                              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],
+          rtp_header_size,
+          rtp_packet_sizes[i]);
     }
-
-    // Read the generated file from disk.
-    ACMDumpEventStream parsed_stream;
-
-    ASSERT_EQ(true, AcmDump::ParseAcmDump(temp_filename, &parsed_stream));
-
-    VerifyResults(parsed_stream, packet_size);
-
-    // Clean up temporary file - can be pretty slow.
-    remove(temp_filename.c_str());
-  }
-  std::vector<uint8_t> rtp_packet_;
-};
-
-TEST_F(AcmDumpTest, DumpAndRead) {
-  Run(256, 321);
+    log_dumper->LogRtcpPacket(true,
+                              MediaType::VIDEO,
+                              incoming_rtcp_packet,
+                              incoming_rtcp_packet_size);
+  }
+
+  int config_count = 2;
+  int rtcp_count = 2;
+  int debug_count = 1;  // Only LogStart event,

[ivoc, 2015/07/14 12:13:14]

These should be const.

[terelius, 2015/07/16 12:47:03]

Done.

+  int event_count = config_count + debug_count + rtcp_count + rtp_count;

[ivoc, 2015/07/14 12:13:14]

This one as well.

[terelius, 2015/07/16 12:47:03]

Done.

+
+  // Read the generated file from disk.
+  ACMDumpEventStream parsed_stream;
+
+  ASSERT_TRUE(AcmDump::ParseAcmDump(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],
+                   rtp_header_size,
+                   rtp_packet_sizes[i]);
+  }
+  VerifyRtcpEvent(parsed_stream.stream(config_count + rtp_count / 2),
+                  false,
+                  MediaType::AUDIO,
+                  outgoing_rtcp_packet,
+                  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],
+                   rtp_header_size,
+                   rtp_packet_sizes[i]);
+  }
+  VerifyRtcpEvent(parsed_stream.stream(2 + config_count + rtp_count),
+                  true,
+                  MediaType::VIDEO,
+                  incoming_rtcp_packet,
+                  incoming_rtcp_packet_size);
+
+  // Clean up temporary file - can be pretty slow.
+  remove(temp_filename.c_str());
+
+  // Free allocated memory,
+  for (size_t i = 0; i < rtp_count; i++) {
+    delete[] rtp_packets[i];
+  }
+  delete[] incoming_rtcp_packet;
+  delete[] outgoing_rtcp_packet;
+}
+
+TEST(AcmDumpTest, LogSessionAndReadBack) {
+  LogSessionAndReadBack(5, 321);
+  LogSessionAndReadBack(8, 3141592653U);
+  LogSessionAndReadBack(9, 2718281828U);
 }
 
 }  // namespace webrtc
 
 #endif  // RTC_AUDIOCODING_DEBUG_DUMP