| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| index 9d134afbccc81a934c0146af9ffb116395dc1ac1..252ffb2a3e59737e125e10c605b137ec9d67aeca 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -695,7 +695,6 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
|
| size_t length = IP_PACKET_SIZE;
|
| uint8_t data_buffer[IP_PACKET_SIZE];
|
| int64_t capture_time_ms;
|
| -
|
| if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
|
| data_buffer, &length,
|
| &capture_time_ms)) {
|
| @@ -923,8 +922,8 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
|
| // TODO(sprang): Potentially too much overhead in IsRegistered()?
|
| bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
|
| kRtpExtensionTransportSequenceNumber) &&
|
| - transport_sequence_number_allocator_;
|
| -
|
| + transport_sequence_number_allocator_ &&
|
| + !is_retransmit;
|
| PacketOptions options;
|
| if (using_transport_seq) {
|
| options.packet_id =
|
|
|