Removed RtcEventLog deps to call:call_interfaces.

webrtc/logging/rtc_event_log/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.
  */
 
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 
 #include <stdint.h>
 #include <string.h>
 
 #include <algorithm>
 #include <fstream>
 #include <istream>
 #include <map>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/protobuf_utils.h"
-#include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 namespace {
-MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
-  switch (media_type) {
-    case rtclog::MediaType::ANY:
-      return MediaType::ANY;
-    case rtclog::MediaType::AUDIO:
-      return MediaType::AUDIO;
-    case rtclog::MediaType::VIDEO:
-      return MediaType::VIDEO;
-    case rtclog::MediaType::DATA:
-      return MediaType::DATA;
-  }
-  RTC_NOTREACHED();
-  return MediaType::ANY;
-}
-
 RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
   switch (rtcp_mode) {
     case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
       return RtcpMode::kCompound;
     case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
       return RtcpMode::kReducedSize;
   }
   RTC_NOTREACHED();
   return RtcpMode::kOff;
 }
@@ skipping 113 matching lines @@
 
   while (1) {
     // Check whether we have reached end of file.
     stream.peek();
     if (stream.eof()) {
       return true;
     }
 
     // Read the next message tag. The tag number is defined as
     // (fieldnumber << 3) | wire_type. In our case, the field number is
-    // supposed to be 1 and the wire type for an length-delimited field is 2.
+    // supposed to be 1 and the wire type for an
+    // length-delimited field is 2.
     const uint64_t kExpectedTag = (1 << 3) | 2;
     std::tie(tag, success) = ParseVarInt(stream);
     if (!success) {
       LOG(LS_WARNING) << "Missing field tag from beginning of protobuf event.";
       return false;
     } else if (tag != kExpectedTag) {
       LOG(LS_WARNING) << "Unexpected field tag at beginning of protobuf event.";
       return false;
     }
 
@@ skipping 13 matching lines @@
       LOG(LS_WARNING) << "Failed to read protobuf message from file.";
       return false;
     }
 
     // Parse the protobuf event from the buffer.
     rtclog::Event event;
     if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
       LOG(LS_WARNING) << "Failed to parse protobuf message.";
       return false;
     }
+
+    EventType type = GetRuntimeEventType(event.type());
+    switch (type) {
+      case VIDEO_RECEIVER_CONFIG_EVENT: {
+        rtclog::StreamConfig config;
+        GetVideoReceiveConfig(event, &config);
+        streams_.emplace_back(config.remote_ssrc, MediaType::VIDEO,
+                              kIncomingPacket);
+        streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
+                              kOutgoingPacket);
+        break;
+      }
+      case VIDEO_SENDER_CONFIG_EVENT: {
+        rtclog::StreamConfig config;
+        GetVideoSendConfig(event, &config);
+        streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
+                              kOutgoingPacket);
+
+        streams_.emplace_back(config.rtx_ssrc, MediaType::VIDEO,
+                              kOutgoingPacket);
+        break;
+      }
+      case AUDIO_RECEIVER_CONFIG_EVENT: {
+        rtclog::StreamConfig config;
+        GetAudioReceiveConfig(event, &config);
+        streams_.emplace_back(config.remote_ssrc, MediaType::AUDIO,
+                              kIncomingPacket);
+        streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
+                              kOutgoingPacket);
+        break;
+      }
+      case AUDIO_SENDER_CONFIG_EVENT: {
+        rtclog::StreamConfig config;
+        GetAudioSendConfig(event, &config);
+        streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
+                              kOutgoingPacket);
+        break;
+      }
+      default:
+        break;
+    }
+
     events_.push_back(event);
   }
 }
 
 size_t ParsedRtcEventLog::GetNumberOfEvents() const {
   return events_.size();
 }
 
 int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_timestamp_us());
   return event.timestamp_us();
 }
 
 ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
     size_t index) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   return GetRuntimeEventType(event.type());
 }
 
 // The header must have space for at least IP_PACKET_SIZE bytes.
 void ParsedRtcEventLog::GetRtpHeader(size_t index,
                                      PacketDirection* incoming,
-                                     MediaType* media_type,
                                      uint8_t* header,
                                      size_t* header_length,
                                      size_t* total_length) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
   RTC_CHECK(event.has_rtp_packet());
   const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
   // Get direction of packet.
   RTC_CHECK(rtp_packet.has_incoming());
   if (incoming != nullptr) {
     *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
   }
-  // Get media type.
-  RTC_CHECK(rtp_packet.has_type());
-  if (media_type != nullptr) {
-    *media_type = GetRuntimeMediaType(rtp_packet.type());
-  }
   // Get packet length.
   RTC_CHECK(rtp_packet.has_packet_length());
   if (total_length != nullptr) {
     *total_length = rtp_packet.packet_length();
   }
   // Get header length.
   RTC_CHECK(rtp_packet.has_header());
   if (header_length != nullptr) {
     *header_length = rtp_packet.header().size();
   }
   // Get header contents.
   if (header != nullptr) {
     const size_t kMinRtpHeaderSize = 12;
     RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
     RTC_CHECK_LE(rtp_packet.header().size(),
                  static_cast<size_t>(IP_PACKET_SIZE));
     memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
   }
 }
 
 // The packet must have space for at least IP_PACKET_SIZE bytes.
 void ParsedRtcEventLog::GetRtcpPacket(size_t index,
                                       PacketDirection* incoming,
-                                      MediaType* media_type,
                                       uint8_t* packet,
                                       size_t* length) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
   RTC_CHECK(event.has_rtcp_packet());
   const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
   // Get direction of packet.
   RTC_CHECK(rtcp_packet.has_incoming());
   if (incoming != nullptr) {
     *incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
   }
-  // Get media type.
-  RTC_CHECK(rtcp_packet.has_type());
-  if (media_type != nullptr) {
-    *media_type = GetRuntimeMediaType(rtcp_packet.type());
-  }
   // Get packet length.
   RTC_CHECK(rtcp_packet.has_packet_data());
   if (length != nullptr) {
     *length = rtcp_packet.packet_data().size();
   }
   // Get packet contents.
   if (packet != nullptr) {
     RTC_CHECK_LE(rtcp_packet.packet_data().size(),
                  static_cast<unsigned>(IP_PACKET_SIZE));
     memcpy(packet, rtcp_packet.packet_data().data(),
            rtcp_packet.packet_data().size());
   }
 }
 
 void ParsedRtcEventLog::GetVideoReceiveConfig(
     size_t index,
     rtclog::StreamConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
-  const rtclog::Event& event = events_[index];
+  GetVideoReceiveConfig(events_[index], config);
+}
+
+void ParsedRtcEventLog::GetVideoReceiveConfig(
+    const rtclog::Event& event,
+    rtclog::StreamConfig* config) const {
   RTC_CHECK(config != nullptr);
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
   RTC_CHECK(event.has_video_receiver_config());
   const rtclog::VideoReceiveConfig& receiver_config =
       event.video_receiver_config();
   // Get SSRCs.
   RTC_CHECK(receiver_config.has_remote_ssrc());
   config->remote_ssrc = receiver_config.remote_ssrc();
   RTC_CHECK(receiver_config.has_local_ssrc());
@@ skipping 41 matching lines @@
     }
     config->codecs.emplace_back(receiver_config.decoders(i).name(),
                                 receiver_config.decoders(i).payload_type(),
                                 rtx_payload_type);
   }
 }
 
 void ParsedRtcEventLog::GetVideoSendConfig(size_t index,
                                            rtclog::StreamConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
-  const rtclog::Event& event = events_[index];
+  GetVideoSendConfig(events_[index], config);
+}
+void ParsedRtcEventLog::GetVideoSendConfig(const rtclog::Event& event,
+                                           rtclog::StreamConfig* config) const {
   RTC_CHECK(config != nullptr);
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
   RTC_CHECK(event.has_video_sender_config());
   const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
   // Get SSRCs.
   if (sender_config.ssrcs_size() > 0) {
     config->local_ssrc = sender_config.ssrcs(0);
     if (sender_config.ssrcs().size() > 1) {
       LOG(WARNING) << "VideoSendConfig contains multiple ssrcs.";
@@ skipping 17 matching lines @@
   config->codecs.emplace_back(
       sender_config.encoder().name(), sender_config.encoder().payload_type(),
       sender_config.has_rtx_payload_type() ? sender_config.rtx_payload_type()
                                            : 0);
 }
 
 void ParsedRtcEventLog::GetAudioReceiveConfig(
     size_t index,
     rtclog::StreamConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
-  const rtclog::Event& event = events_[index];
+  GetAudioReceiveConfig(events_[index], config);
+}
+
+void ParsedRtcEventLog::GetAudioReceiveConfig(
+    const rtclog::Event& event,
+    rtclog::StreamConfig* config) const {
   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->remote_ssrc = receiver_config.remote_ssrc();
   RTC_CHECK(receiver_config.has_local_ssrc());
   config->local_ssrc = receiver_config.local_ssrc();
   // Get header extensions.
   GetHeaderExtensions(&config->rtp_extensions,
                       receiver_config.header_extensions());
 }
 
 void ParsedRtcEventLog::GetAudioSendConfig(size_t index,
                                            rtclog::StreamConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
-  const rtclog::Event& event = events_[index];
+  GetAudioSendConfig(events_[index], config);
+}
+
+void ParsedRtcEventLog::GetAudioSendConfig(const rtclog::Event& event,
+                                           rtclog::StreamConfig* config) const {
   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.
   RTC_CHECK(sender_config.has_ssrc());
   config->local_ssrc = sender_config.ssrc();
   // Get header extensions.
   GetHeaderExtensions(&config->rtp_extensions,
@@ skipping 128 matching lines @@
     res.failure_reason =
         rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
   } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
     res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
   } else {
     RTC_NOTREACHED();
   }
 
   return res;
 }
+
+// Returns the MediaType for registered SSRCs. Search from the end to use last
+// registered types first.
+ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
+    uint32_t ssrc,
+    PacketDirection direction) const {
+  for (auto rit = streams_.rbegin(); rit != streams_.rend(); ++rit) {
+    if (rit->ssrc == ssrc && rit->direction == direction)
+      return rit->media_type;
+  }
+  return MediaType::ANY;
+}
 }  // namespace webrtc