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_); |