New parser for event log. Manually parse the outermost EventStream

webrtc/call/rtc_event_log_parser.cc

Index: webrtc/call/rtc_event_log_parser.cc
diff --git a/webrtc/call/rtc_event_log_parser.cc b/webrtc/call/rtc_event_log_parser.cc
new file mode 100644
index 0000000000000000000000000000000000000000..244f17c491e906050bb9dd400ac808d8483affcb
--- /dev/null
+++ b/webrtc/call/rtc_event_log_parser.cc
@@ -0,0 +1,395 @@
+/*
+ *  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/call/rtc_event_log_parser.h"
+
+#include <string.h>
+
+#include <fstream>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call.h"
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/file_wrapper.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;
+}
+
+ParsedRtcEventLog::EventType GetRuntimeEventType(
+    rtclog::Event::EventType event_type) {
+  switch (event_type) {
+    case rtclog::Event::UNKNOWN_EVENT:
+      return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
+    case rtclog::Event::LOG_START:
+      return ParsedRtcEventLog::EventType::LOG_START;
+    case rtclog::Event::LOG_END:
+      return ParsedRtcEventLog::EventType::LOG_END;
+    case rtclog::Event::RTP_EVENT:
+      return ParsedRtcEventLog::EventType::RTP_EVENT;
+    case rtclog::Event::RTCP_EVENT:
+      return ParsedRtcEventLog::EventType::RTCP_EVENT;
+    case rtclog::Event::AUDIO_PLAYOUT_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
+    case rtclog::Event::BWE_PACKET_LOSS_EVENT:
+      return ParsedRtcEventLog::EventType::BWE_PACKET_LOSS_EVENT;
+    case rtclog::Event::BWE_PACKET_DELAY_EVENT:
+      return ParsedRtcEventLog::EventType::BWE_PACKET_DELAY_EVENT;
+    case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
+    case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
+    case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
+    case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
+      return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
+  }
+  RTC_NOTREACHED();
+  return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
+}
+
+bool ParseVarInt(std::FILE* file, uint64_t* varint, size_t* bytes_read) {
+  uint8_t byte;
+  *varint = 0;
+  for (*bytes_read = 0; *bytes_read < 10 && fread(&byte, 1, 1, file) == 1;
+       ++(*bytes_read)) {
+    // The most significant bit of each byte is 0 if it is the last byte in
+    // the varint and 1 otherwise. Thus, we take the 7 least significant bits
+    // of each byte and shift them 7 bits for each byte read previously to get
+    // the (unsigned) integer.
+    *varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * *bytes_read);
+    if ((byte & 0x80) == 0) {
+      return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace
+
+bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
+  stream_.clear();
+  const size_t kMaxEventSize = (1u << 16) - 1;
+  char tmp_buffer[kMaxEventSize];
+
+  std::FILE* file = fopen(filename.c_str(), "rb");
+  if (!file) {
+    LOG(LS_WARNING) << "Could not open file for reading.";
+    return false;
+  }
+
+  while (1) {
+    // Peek at 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.
+    const uint64_t kExpectedTag = (1 << 3) | 2;
+    uint64_t tag;
+    size_t bytes_read;
+    if (!ParseVarInt(file, &tag, &bytes_read) || tag != kExpectedTag) {
+      fclose(file);
+      if (bytes_read == 0) {
+        return true;  // Reached end of file.
+      }
+      LOG(LS_WARNING) << "Missing field tag from beginning of protobuf event.";
+      return false;
+    }
+
+    // Peek at the length field.
+    uint64_t message_length;
+    if (!ParseVarInt(file, &message_length, &bytes_read)) {
+      LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
+      fclose(file);
+      return false;
+    } else if (message_length > kMaxEventSize) {
+      LOG(LS_WARNING) << "Protobuf message length is too large.";
+      fclose(file);
+      return false;
+    }
+
+    if (fread(tmp_buffer, 1, message_length, file) != message_length) {
+      LOG(LS_WARNING) << "Failed to read protobuf message from file.";
+      fclose(file);
+      return false;
+    }
+
+    rtclog::Event event;
+    if (!event.ParseFromArray(tmp_buffer, message_length)) {
+      LOG(LS_WARNING) << "Failed to parse protobuf message.";
+      fclose(file);
+      return false;
+    }
+    stream_.push_back(event);
+  }
+}
+
+size_t ParsedRtcEventLog::GetNumberOfEvents() const {
+  return stream_.size();
+}
+
+int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[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 = stream_[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 = stream_[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 = stream_[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,
+    VideoReceiveStream::Config* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  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->rtp.remote_ssrc = receiver_config.remote_ssrc();
+  RTC_CHECK(receiver_config.has_local_ssrc());
+  config->rtp.local_ssrc = receiver_config.local_ssrc();
+  // Get RTCP settings.
+  RTC_CHECK(receiver_config.has_rtcp_mode());
+  config->rtp.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
+  RTC_CHECK(receiver_config.has_remb());
+  config->rtp.remb = receiver_config.remb();
+  // Get RTX map.
+  config->rtp.rtx.clear();
+  for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
+    const rtclog::RtxMap& map = receiver_config.rtx_map(i);
+    RTC_CHECK(map.has_payload_type());
+    RTC_CHECK(map.has_config());
+    RTC_CHECK(map.config().has_rtx_ssrc());
+    RTC_CHECK(map.config().has_rtx_payload_type());
+    webrtc::VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
+    rtx_pair.ssrc = map.config().rtx_ssrc();
+    rtx_pair.payload_type = map.config().rtx_payload_type();
+    config->rtp.rtx.insert(std::make_pair(map.payload_type(), rtx_pair));
+  }
+  // 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));
+  }
+  // Get decoders.
+  config->decoders.clear();
+  for (int i = 0; i < receiver_config.decoders_size(); i++) {
+    RTC_CHECK(receiver_config.decoders(i).has_name());
+    RTC_CHECK(receiver_config.decoders(i).has_payload_type());
+    VideoReceiveStream::Decoder decoder;
+    decoder.payload_name = receiver_config.decoders(i).name();
+    decoder.payload_type = receiver_config.decoders(i).payload_type();
+    config->decoders.push_back(decoder);
+  }
+}
+
+void ParsedRtcEventLog::GetVideoSendConfig(
+    size_t index,
+    VideoSendStream::Config* config) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  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.
+  config->rtp.ssrcs.clear();
+  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
+    config->rtp.ssrcs.push_back(sender_config.ssrcs(i));
+  }
+  // Get header extensions.
+  config->rtp.extensions.clear();
+  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
+    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));
+  }
+  // Get RTX settings.
+  config->rtp.rtx.ssrcs.clear();
+  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
+    config->rtp.rtx.ssrcs.push_back(sender_config.rtx_ssrcs(i));
+  }
+  if (sender_config.rtx_ssrcs_size() > 0) {
+    RTC_CHECK(sender_config.has_rtx_payload_type());
+    config->rtp.rtx.payload_type = sender_config.rtx_payload_type();
+  } else {
+    // Reset RTX payload type default value if no RTX SSRCs are used.
+    config->rtp.rtx.payload_type = -1;
+  }
+  // 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::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[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();
+  }
+}
+
+void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index,
+                                              int32_t* bitrate,
+                                              uint8_t* fraction_loss,
+                                              int32_t* total_packets) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = stream_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PACKET_LOSS_EVENT);
+  RTC_CHECK(event.has_bwe_packet_loss_event());
+  const rtclog::BwePacketLossEvent& loss_event = event.bwe_packet_loss_event();
+  RTC_CHECK(loss_event.has_bitrate());
+  if (bitrate != nullptr) {
+    *bitrate = loss_event.bitrate();
+  }
+  RTC_CHECK(loss_event.has_fraction_loss());
+  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