Added logging for audio send/receive stream configs.

webrtc/call/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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 358 matching lines @@
   }
   // Get encoder.
   RTC_CHECK(sender_config.has_encoder());
   RTC_CHECK(sender_config.encoder().has_name());
   RTC_CHECK(sender_config.encoder().has_payload_type());
   config->encoder_settings.payload_name = sender_config.encoder().name();
   config->encoder_settings.payload_type =
       sender_config.encoder().payload_type();
 }
 
+void ParsedRtcEventLog::GetAudioReceiveConfig(
+    size_t index,
+    AudioReceiveStream::Config* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(config != nullptr);
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
+  RTC_CHECK(event.has_audio_receiver_config());
+  const rtclog::AudioReceiveConfig& receiver_config =
+      event.audio_receiver_config();
+  // Get SSRCs.
+  RTC_CHECK(receiver_config.has_remote_ssrc());
+  config->rtp.remote_ssrc = receiver_config.remote_ssrc();
+  RTC_CHECK(receiver_config.has_local_ssrc());
+  config->rtp.local_ssrc = receiver_config.local_ssrc();
+  // Get header extensions.
+  config->rtp.extensions.clear();
+  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
+    RTC_CHECK(receiver_config.header_extensions(i).has_name());
+    RTC_CHECK(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();
+    config->rtp.extensions.push_back(RtpExtension(name, id));
+  }
+}
+
+void ParsedRtcEventLog::GetAudioSendConfig(
+    size_t index,
+    AudioSendStream::Config* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(config != nullptr);
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
+  RTC_CHECK(event.has_audio_sender_config());
+  const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
+  // Get SSRCs.
+  config->rtp.ssrc = sender_config.ssrc();

[terelius, 2016/09/21 10:11:26]

RTC_CHECK(sender_config.has_ssrc())

[ivoc, 2016/09/22 09:20:55]

Good catch, added.

+  // Get header extensions.
+  config->rtp.extensions.clear();
+  for (int i = 0; i < sender_config.header_extensions_size(); i++) {

[the sun, 2016/09/22 06:51:02]

Can we make a utility function out of this loop, or abstract it some other way?

[ivoc, 2016/09/22 09:20:55]

It's not as simple as it seems, because the type of both |sender_config| and
|config| are different in each of the functions where this loop appears. I guess
a function with two template parameters could work, but I'm not sure if it's
worth the added complexity, WDYT?

[the sun, 2016/09/22 19:22:05]

All the stream config objects store the extensions in a
std::vector<RtpExtension> so that type should need no templating. The protobuf
side would, unless you can do something with the result from e.g. this method in
rtc_event_log.pb.h:

  const ::google::protobuf::RepeatedPtrField<
::webrtc::rtclog::RtpHeaderExtension >&
      header_extensions() const;

?
(I haven't used the C++ proto API in a while so I'm not sure it's useful.)

[ivoc, 2016/09/23 12:40:27]

Thanks for the suggestion, that's a really good idea. Done.

+    RTC_CHECK(sender_config.header_extensions(i).has_name());
+    RTC_CHECK(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();
+    config->rtp.extensions.push_back(RtpExtension(name, id));
+  }
+}
+
 void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
   RTC_CHECK(event.has_audio_playout_event());
   const rtclog::AudioPlayoutEvent& loss_event = event.audio_playout_event();
   RTC_CHECK(loss_event.has_local_ssrc());
   if (ssrc != nullptr) {
     *ssrc = loss_event.local_ssrc();
@@ skipping 18 matching lines @@
   if (fraction_loss != nullptr) {
     *fraction_loss = loss_event.fraction_loss();
   }
   RTC_CHECK(loss_event.has_total_packets());
   if (total_packets != nullptr) {
     *total_packets = loss_event.total_packets();
   }
 }
 
 }  // namespace webrtc