1 files changed, 16 insertions, 11 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index d066bafb90..9c1dc4edb8 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -19,6 +19,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/algorithm/container.h"
 #include "absl/types/optional.h"
 #include "absl/types/variant.h"
 #include "common_video/h264/h264_common.h"
@@ -52,11 +53,14 @@ RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
     input_fragments_.push_back(
         payload.subview(nalu.payload_start_offset, nalu.payload_size));
   }
-
-  if (!GeneratePackets(packetization_mode)) {
-    // If failed to generate all the packets, discard already generated
-    // packets in case the caller would ignore return value and still try to
-    // call NextPacket().
+  bool has_empty_fragments = absl::c_any_of(
+      input_fragments_, [](const rtc::ArrayView<const uint8_t> fragment) {
+        return fragment.empty();
+      });
+  if (has_empty_fragments || !GeneratePackets(packetization_mode)) {
+    // If empty fragments were found or we failed to generate all the packets,
+    // discard already generated packets in case the caller would ignore the
+    // return value and still try to call NextPacket().
     num_packets_left_ = 0;
     while (!packets_.empty()) {
       packets_.pop();
@@ -73,6 +77,7 @@ size_t RtpPacketizerH264::NumPackets() const {
 bool RtpPacketizerH264::GeneratePackets(
     H264PacketizationMode packetization_mode) {
   for (size_t i = 0; i < input_fragments_.size();) {
+    RTC_DCHECK(!input_fragments_[i].empty());
     switch (packetization_mode) {
       case H264PacketizationMode::SingleNalUnit:
         if (!PacketizeSingleNalu(i))
@@ -173,11 +178,11 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
       return fragment_size;
     }
   };
-
   while (payload_size_left >= payload_size_needed()) {
     RTC_CHECK_GT(fragment.size(), 0);
-    packets_.push(PacketUnit(fragment, aggregated_fragments == 0, false, true,
-                             fragment[0]));
+
+    packets_.push(PacketUnit(fragment, /*first=*/aggregated_fragments == 0,
+                             /*last=*/false, /*aggregated=*/true, fragment[0]));
     payload_size_left -= fragment.size();
     payload_size_left -= fragment_headers_length;
 
@@ -218,9 +223,9 @@ bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
                       << limits_.max_payload_len;
     return false;
   }
-  RTC_CHECK_GT(fragment.size(), 0u);
-  packets_.push(PacketUnit(fragment, true /* first */, true /* last */,
-                           false /* aggregated */, fragment[0]));
+  RTC_CHECK(!fragment.empty());
+  packets_.push(PacketUnit(fragment, /*first=*/true, /*last=*/true,
+                           /*aggregated=*/false, fragment[0]));
   ++num_packets_left_;
   return true;
 }