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 b32e78ef9abee9f89e2f7a1ce742437af37468dc..5a1c176a6c6f2c5c9a2f6c2f3b79bb263f29d75b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -77,9 +77,10 @@ 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() {
 }
@@ -162,11 +163,17 @@ 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);
+    if (packetization_mode_ == kH264PacketizationMode0) {
+      PacketizeSingleNalu(i);
       ++i;
     } else {
-      i = PacketizeStapA(i);
+      RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);

[sprang_webrtc, 2016/12/01 16:58:57]

If you use enum class you can add switch statement here and get a compilation
error if the model is not covered by a case.

[hta-webrtc, 2016/12/01 18:26:27]

Done.

+      if (input_fragments_[i].length > max_payload_len_) {
+        PacketizeFuA(i);
+        ++i;
+      } else {
+        i = PacketizeStapA(i);
+      }
     }
   }
 }
@@ -229,6 +236,18 @@ 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_;
+  RTC_CHECK_GT(fragment->length, 0u);
+  packets_.push(PacketUnit(*fragment, true /* first */, true /* last */,
+                           false /* aggregated */, fragment->buffer[0]));
+}
+
 bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
                                    size_t* bytes_to_send,
                                    bool* last_packet) {
@@ -249,9 +268,11 @@ bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
     input_fragments_.pop_front();
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else if (packet.aggregated) {
+    RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);
     NextAggregatePacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else {
+    RTC_CHECK_EQ(packetization_mode_, kH264PacketizationMode1);
     NextFragmentPacket(buffer, bytes_to_send);
     RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   }