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 9d71803f3b984887825d079e7a9897cd24ce2702..b82b66f5fe5a0e270a94b83041aaa18bc8150da8 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc |
@@ -78,10 +78,9 @@ |
} // namespace |
-RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len, |
- H264PacketizationMode packetization_mode) |
- : max_payload_len_(max_payload_len), |
- packetization_mode_(packetization_mode) {} |
+RtpPacketizerH264::RtpPacketizerH264(FrameType frame_type, |
+ size_t max_payload_len) |
+ : max_payload_len_(max_payload_len) {} |
RtpPacketizerH264::~RtpPacketizerH264() { |
} |
@@ -164,19 +163,11 @@ |
void RtpPacketizerH264::GeneratePackets() { |
for (size_t i = 0; i < input_fragments_.size();) { |
- 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; |
+ if (input_fragments_[i].length > max_payload_len_) { |
+ PacketizeFuA(i); |
+ ++i; |
+ } else { |
+ i = PacketizeStapA(i); |
} |
} |
} |
@@ -239,21 +230,6 @@ |
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); |
@@ -272,10 +248,8 @@ |
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_); |