diff --git a/webrtc/modules/include/module_common_types.h b/webrtc/modules/include/module_common_types.h
index 3df93b6c89..fa0826a551 100644
--- a/webrtc/modules/include/module_common_types.h
+++ b/webrtc/modules/include/module_common_types.h
@@ -271,6 +271,22 @@ enum class H264PacketizationMode {
   SingleNalUnit        // Mode 0 - only single NALU allowed
 };
 
+// This function is declared inline because it is not clear which
+// .cc file it should belong to.
+// TODO(hta): Refactor. https://bugs.webrtc.org/6842
+inline std::ostream& operator<<(std::ostream& stream,
+                                H264PacketizationMode mode) {
+  switch (mode) {
+    case H264PacketizationMode::NonInterleaved:
+      stream << "NonInterleaved";
+      break;
+    case H264PacketizationMode::SingleNalUnit:
+      stream << "SingleNalUnit";
+      break;
+  }
+  return stream;
+}
+
 struct NaluInfo {
   uint8_t type;
   int sps_id;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index 9d71803f3b..e896c65f97 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -245,10 +245,7 @@ void RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
   const Fragment* fragment = &input_fragments_[fragment_index];
   RTC_CHECK_GE(payload_size_left, fragment->length)
       << "Payload size left " << payload_size_left << ", fragment length "
-      << fragment->length << ", packetization mode "
-      << (packetization_mode_ == H264PacketizationMode::SingleNalUnit
-              ? "SingleNalUnit"
-              : "NonInterleaved");
+      << fragment->length << ", packetization mode " << packetization_mode_;
   RTC_CHECK_GT(fragment->length, 0u);
   packets_.push(PacketUnit(*fragment, true /* first */, true /* last */,
                            false /* aggregated */, fragment->buffer[0]));
@@ -272,10 +269,10 @@ bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet,
     packets_.pop();
     input_fragments_.pop_front();
   } else if (packet.aggregated) {
-    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
+    RTC_CHECK_EQ(H264PacketizationMode::NonInterleaved, packetization_mode_);
     NextAggregatePacket(rtp_packet);
   } else {
-    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
+    RTC_CHECK_EQ(H264PacketizationMode::NonInterleaved, packetization_mode_);
     NextFragmentPacket(rtp_packet);
   }
   RTC_DCHECK_LE(rtp_packet->payload_size(), max_payload_len_);