| 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_);
|
|
|