diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 4275e59..06b250d 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -27,21 +27,26 @@
 
 using MediaType = webrtc::ParsedRtcEventLog::MediaType;
 
-DEFINE_bool(noaudio,
-            false,
-            "Excludes audio packets from the converted RTPdump file.");
-DEFINE_bool(novideo,
-            false,
-            "Excludes video packets from the converted RTPdump file.");
-DEFINE_bool(nodata,
-            false,
-            "Excludes data packets from the converted RTPdump file.");
-DEFINE_bool(nortp,
-            false,
-            "Excludes RTP packets from the converted RTPdump file.");
-DEFINE_bool(nortcp,
-            false,
-            "Excludes RTCP packets from the converted RTPdump file.");
+DEFINE_bool(
+    audio,
+    true,
+    "Use --noaudio to exclude audio packets from the converted RTPdump file.");
+DEFINE_bool(
+    video,
+    true,
+    "Use --novideo to exclude video packets from the converted RTPdump file.");
+DEFINE_bool(
+    data,
+    true,
+    "Use --nodata to exclude data packets from the converted RTPdump file.");
+DEFINE_bool(
+    rtp,
+    true,
+    "Use --nortp to exclude RTP packets from the converted RTPdump file.");
+DEFINE_bool(
+    rtcp,
+    true,
+    "Use --nortcp to exclude RTCP packets from the converted RTPdump file.");
 DEFINE_string(ssrc,
               "",
               "Store only packets with this SSRC (decimal or hex, the latter "
@@ -122,7 +127,7 @@
     // some required fields and we attempt to access them. We could consider
     // a softer failure option, but it does not seem useful to generate
     // RTP dumps based on broken event logs.
-    if (!FLAG_nortp &&
+    if (FLAG_rtp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
@@ -143,11 +148,11 @@
       rtp_parser.Parse(&parsed_header);
       MediaType media_type =
           parsed_stream.GetMediaType(parsed_header.ssrc, direction);
-      if (FLAG_noaudio && media_type == MediaType::AUDIO)
+      if (!FLAG_audio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAG_novideo && media_type == MediaType::VIDEO)
+      if (!FLAG_video && media_type == MediaType::VIDEO)
         continue;
-      if (FLAG_nodata && media_type == MediaType::DATA)
+      if (!FLAG_data && media_type == MediaType::DATA)
         continue;
       if (strlen(FLAG_ssrc) > 0) {
         const uint32_t packet_ssrc =
@@ -160,9 +165,8 @@
       rtp_writer->WritePacket(&packet);
       rtp_counter++;
     }
-    if (!FLAG_nortcp &&
-        parsed_stream.GetEventType(i) ==
-            webrtc::ParsedRtcEventLog::RTCP_EVENT) {
+    if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
+                         webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
       parsed_stream.GetRtcpPacket(i, &direction, packet.data, &packet.length);
@@ -181,11 +185,11 @@
       const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
           reinterpret_cast<const uint8_t*>(packet.data + 4));
       MediaType media_type = parsed_stream.GetMediaType(packet_ssrc, direction);
-      if (FLAG_noaudio && media_type == MediaType::AUDIO)
+      if (!FLAG_audio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAG_novideo && media_type == MediaType::VIDEO)
+      if (!FLAG_video && media_type == MediaType::VIDEO)
         continue;
-      if (FLAG_nodata && media_type == MediaType::DATA)
+      if (!FLAG_data && media_type == MediaType::DATA)
         continue;
       if (strlen(FLAG_ssrc) > 0) {
         if (packet_ssrc != ssrc_filter)
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index bba6ace..0f9d674 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -40,17 +40,17 @@
 
 namespace {
 
-DEFINE_bool(noconfig, false, "Excludes stream configurations.");
-DEFINE_bool(noincoming, false, "Excludes incoming packets.");
-DEFINE_bool(nooutgoing, false, "Excludes outgoing packets.");
+DEFINE_bool(config, true, "Use --noconfig to exclude stream configurations.");
+DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets.");
+DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
-DEFINE_bool(noaudio, false, "Excludes audio packets.");
+DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
-DEFINE_bool(novideo, false, "Excludes video packets.");
+DEFINE_bool(video, true, "Use --novideo to exclude video packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
-DEFINE_bool(nodata, false, "Excludes data packets.");
-DEFINE_bool(nortp, false, "Excludes RTP packets.");
-DEFINE_bool(nortcp, false, "Excludes RTCP packets.");
+DEFINE_bool(data, true, "Use --nodata to exclude data packets.");
+DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets.");
+DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
 // TODO(terelius): Allow a list of SSRCs.
 DEFINE_string(ssrc,
               "",
@@ -84,15 +84,15 @@
 bool ExcludePacket(webrtc::PacketDirection direction,
                    MediaType media_type,
                    uint32_t packet_ssrc) {
-  if (FLAG_nooutgoing && direction == webrtc::kOutgoingPacket)
+  if (!FLAG_outgoing && direction == webrtc::kOutgoingPacket)
     return true;
-  if (FLAG_noincoming && direction == webrtc::kIncomingPacket)
+  if (!FLAG_incoming && direction == webrtc::kIncomingPacket)
     return true;
-  if (FLAG_noaudio && media_type == MediaType::AUDIO)
+  if (!FLAG_audio && media_type == MediaType::AUDIO)
     return true;
-  if (FLAG_novideo && media_type == MediaType::VIDEO)
+  if (!FLAG_video && media_type == MediaType::VIDEO)
     return true;
-  if (FLAG_nodata && media_type == MediaType::DATA)
+  if (!FLAG_data && media_type == MediaType::DATA)
     return true;
   if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc)
     return true;
@@ -386,7 +386,7 @@
   }
 
   for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
-    if (!FLAG_noconfig && !FLAG_novideo && !FLAG_noincoming &&
+    if (FLAG_config && FLAG_video && FLAG_incoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
@@ -407,7 +407,7 @@
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAG_noconfig && !FLAG_novideo && !FLAG_nooutgoing &&
+    if (FLAG_config && FLAG_video && FLAG_outgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
       std::vector<webrtc::rtclog::StreamConfig> configs =
@@ -430,7 +430,7 @@
         std::cout << "}" << std::endl;
       }
     }
-    if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_noincoming &&
+    if (FLAG_config && FLAG_audio && FLAG_incoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
@@ -451,7 +451,7 @@
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_nooutgoing &&
+    if (FLAG_config && FLAG_audio && FLAG_outgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
@@ -470,7 +470,7 @@
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAG_nortp &&
+    if (FLAG_rtp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       size_t header_length;
       size_t total_length;
@@ -521,9 +521,8 @@
       }
       std::cout << std::endl;
     }
-    if (!FLAG_nortcp &&
-        parsed_stream.GetEventType(i) ==
-            webrtc::ParsedRtcEventLog::RTCP_EVENT) {
+    if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
+                         webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       size_t length;
       uint8_t packet[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;