Reland of H.264 packetization mode 0 (try 3)

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index b82b66f5fe5a0e270a94b83041aaa18bc8150da8..9d71803f3b984887825d079e7a9897cd24ce2702 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -78,9 +78,10 @@ bool ParseStapAStartOffsets(const uint8_t* nalu_ptr,
 
 }  // namespace
 
-RtpPacketizerH264::RtpPacketizerH264(FrameType frame_type,
-                                     size_t max_payload_len)
-    : max_payload_len_(max_payload_len) {}
+RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len,
+                                     H264PacketizationMode packetization_mode)
+    : max_payload_len_(max_payload_len),
+      packetization_mode_(packetization_mode) {}
 
 RtpPacketizerH264::~RtpPacketizerH264() {
 }
@@ -163,11 +164,19 @@ void RtpPacketizerH264::SetPayloadData(
 
 void RtpPacketizerH264::GeneratePackets() {
   for (size_t i = 0; i < input_fragments_.size();) {
-    if (input_fragments_[i].length > max_payload_len_) {
-      PacketizeFuA(i);
-      ++i;
-    } else {
-      i = PacketizeStapA(i);
+    switch (packetization_mode_) {
+      case H264PacketizationMode::SingleNalUnit:
+        PacketizeSingleNalu(i);
+        ++i;
+        break;
+      case H264PacketizationMode::NonInterleaved:
+        if (input_fragments_[i].length > max_payload_len_) {
+          PacketizeFuA(i);
+          ++i;
+        } else {
+          i = PacketizeStapA(i);
+        }
+        break;
     }
   }
 }
@@ -230,6 +239,21 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
   return fragment_index;
 }
 
+void RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
+  // Add a single NALU to the queue, no aggregation.
+  size_t payload_size_left = max_payload_len_;
+  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");
+  RTC_CHECK_GT(fragment->length, 0u);
+  packets_.push(PacketUnit(*fragment, true /* first */, true /* last */,
+                           false /* aggregated */, fragment->buffer[0]));
+}
+
 bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet,
                                    bool* last_packet) {
   RTC_DCHECK(rtp_packet);
@@ -248,8 +272,10 @@ bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet,
     packets_.pop();
     input_fragments_.pop_front();
   } else if (packet.aggregated) {
+    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
     NextAggregatePacket(rtp_packet);
   } else {
+    RTC_CHECK(packetization_mode_ == H264PacketizationMode::NonInterleaved);
     NextFragmentPacket(rtp_packet);
   }
   RTC_DCHECK_LE(rtp_packet->payload_size(), max_payload_len_);