Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estima…

webrtc/call/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.
  */
 
@@ skipping 257 matching lines @@
 }
 
 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
  * presence of extension number i from kExtensionTypes / kExtensionNames.
  * The least significant bit extension_bitvector has number 0.
  */
@@ skipping 102 matching lines @@
           RtpExtension(kExtensionNames[i], 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,
                            size_t rtcp_count,
                            size_t playout_count,
+                           size_t bwe_loss_count,
                            uint32_t extensions_bitvector,
                            uint32_t csrcs_count,
                            unsigned random_seed) {
   ASSERT_LE(rtcp_count, rtp_count);
   ASSERT_LE(playout_count, rtp_count);
+  ASSERT_LE(bwe_loss_count, rtp_count);
   std::vector<rtc::Buffer> rtp_packets;
   std::vector<rtc::Buffer> rtcp_packets;
   std::vector<size_t> rtp_header_sizes;
   std::vector<uint32_t> playout_ssrcs;
+  std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
 
   VideoReceiveStream::Config receiver_config(nullptr);
   VideoSendStream::Config sender_config(nullptr);
 
   srand(random_seed);
 
   // Create rtp_count RTP packets containing random data.
   for (size_t i = 0; i < rtp_count; i++) {
     size_t packet_size = 1000 + rand() % 64;
     rtp_packets.push_back(rtc::Buffer(packet_size));
     size_t header_size = GenerateRtpPacket(extensions_bitvector, csrcs_count,
                                            rtp_packets[i].data(), packet_size);
     rtp_header_sizes.push_back(header_size);
   }
   // Create rtcp_count RTCP packets containing random data.
   for (size_t i = 0; i < rtcp_count; i++) {
     size_t packet_size = 1000 + rand() % 64;
     rtcp_packets.push_back(rtc::Buffer(packet_size));
     GenerateRtcpPacket(rtcp_packets[i].data(), packet_size);
   }
   // Create playout_count random SSRCs to use when logging AudioPlayout events.
   for (size_t i = 0; i < playout_count; i++) {
     playout_ssrcs.push_back(static_cast<uint32_t>(rand()));
   }
+  // Create bwe_loss_count random bitrate updates for BwePacketLoss.
+  for (size_t i = 0; i < bwe_loss_count; i++) {

[ivoc, 2015/10/19 16:03:38]

why not just make i an int? I don't see any compelling reason to make it a
size_t (nor in the previous loops for that matter).

[terelius, 2015/10/26 17:40:18]

No, there is no strong reason for doing it either way in this case. In general,
I prefer to always use size_t for the length of an array and by extension also
for indices into an array.

The google standard is ambiguous on this issue:
"We use int very often, for integers we know are not going to be too big, e.g.,
loop counters."
But also:
"When appropriate, you are welcome to use standard types like size_t"

Both are used about equally in WebRTC; 977 occurrences of "for (int i" vs 927
occurrences of "for (size_t i".

[stefan-webrtc, 2015/10/28 16:03:18]

Chromium style is more explicit when it comes to size_t:
https://www.chromium.org/developers/coding-style

[terelius, 2015/10/30 10:51:40]

Acknowledged.

+    bwe_loss_updates.push_back(std::pair<int32_t, uint8_t>(rand(), rand()));
+  }
   // Create configurations for the video streams.
   GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config);
   GenerateVideoSendConfig(extensions_bitvector, &sender_config);
   const int config_count = 2;
 
   // 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 rtcp_index = 1, playout_index = 1;
+    size_t rtcp_index = 1, playout_index = 1, bwe_loss_index = 1;

[stefan-webrtc, 2015/10/19 08:43:22]

Personally, I prefer to have one variable per line.

[terelius, 2015/10/26 17:40:18]

Done.

     for (size_t i = 1; 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 - 1].data(), rtp_packets[i - 1].size());
       if (i * rtcp_count >= rtcp_index * rtp_count) {
         log_dumper->LogRtcpPacket(
             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());
         rtcp_index++;
       }
       if (i * playout_count >= playout_index * rtp_count) {
         log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
         playout_index++;
       }
+      if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
+        log_dumper->LogBwePacketLossEvent(
+            bwe_loss_updates[bwe_loss_index - 1].first,
+            bwe_loss_updates[bwe_loss_index - 1].second, i);
+        bwe_loss_index++;
+      }
       if (i == rtp_count / 2) {
         log_dumper->StartLogging(temp_filename, 10000000);
       }
     }
   }
 
   // Read the generated file from disk.
   rtclog::EventStream parsed_stream;
 
   ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream));
 
   // Verify the result.
-  const int event_count =
-      config_count + playout_count + rtcp_count + rtp_count + 1;
+  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);
-  size_t event_index = config_count, rtcp_index = 1, playout_index = 1;
+  size_t event_index = config_count, rtcp_index = 1, playout_index = 1,
+         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());
     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());
       event_index++;
       rtcp_index++;
     }
     if (i * playout_count >= playout_index * rtp_count) {
       VerifyPlayoutEvent(parsed_stream.stream(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);
+      event_index++;
+      bwe_loss_index++;
+    }
     if (i == rtp_count / 2) {
       VerifyLogStartEvent(parsed_stream.stream(event_index));
       event_index++;
     }
   }
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
 
 TEST(RtcEventLogTest, LogSessionAndReadBack) {
-  // Log 5 RTP, 2 RTCP, and 0 playout events with no header extensions or CSRCS.
-  LogSessionAndReadBack(5, 2, 0, 0, 0, 321);
+  // Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events
+  // with no header extensions or CSRCS.
+  LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321);
 
-  // Enable AbsSendTime and TransportSequenceNumbers
+  // Enable AbsSendTime and TransportSequenceNumbers.
   uint32_t extensions = 0;
   for (uint32_t i = 0; i < kNumExtensions; i++) {
     if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime ||
         kExtensionTypes[i] ==
             RTPExtensionType::kRtpExtensionTransportSequenceNumber) {
       extensions |= 1u << i;
     }
   }
-  LogSessionAndReadBack(8, 2, 0, extensions, 0, 3141592653u);
+  LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u);
 
-  extensions = (1u << kNumExtensions) - 1;  // Enable all header extensions
-  LogSessionAndReadBack(9, 2, 3, extensions, 2, 2718281828u);
+  extensions = (1u << kNumExtensions) - 1;  // Enable all header extensions.
+  LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u);
 
   // Try all combinations of header extensions and up to 2 CSRCS.
   for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) {
     for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) {
       LogSessionAndReadBack(5 + extensions,   // Number of RTP packets.
                             2 + csrcs_count,  // Number of RTCP packets.
-                            3 + csrcs_count,  // Number of playout events
-                            extensions,       // Bit vector choosing extensions
-                            csrcs_count,      // Number of contributing sources
+                            3 + csrcs_count,  // Number of playout events.
+                            1 + csrcs_count,  // Number of BWE loss events.
+                            extensions,       // Bit vector choosing extensions.
+                            csrcs_count,      // Number of contributing sources.
                             rand());
     }
   }
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG