| 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 36d7eb573cbe43db51fe13a25183f41d0a974923..3fbca7b67d8714038067338f325b898e9239e3dc 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -133,7 +133,6 @@ RTPSender::RTPSender(
|
| transport_(transport),
|
| sending_media_(true), // Default to sending media.
|
| max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
|
| - packet_over_head_(28),
|
| payload_type_(-1),
|
| payload_type_map_(),
|
| rtp_header_extension_map_(),
|
| @@ -375,15 +374,12 @@ int RTPSender::SendPayloadFrequency() const {
|
| return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
|
| }
|
|
|
| -int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
|
| - uint16_t packet_over_head) {
|
| +void RTPSender::SetMaxPayloadLength(size_t max_payload_length) {
|
| // Sanity check.
|
| RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
|
| << "Invalid max payload length: " << max_payload_length;
|
| rtc::CritScope lock(&send_critsect_);
|
| max_payload_length_ = max_payload_length;
|
| - packet_over_head_ = packet_over_head;
|
| - return 0;
|
| }
|
|
|
| size_t RTPSender::MaxDataPayloadLength() const {
|
| @@ -405,8 +401,6 @@ size_t RTPSender::MaxPayloadLength() const {
|
| return max_payload_length_;
|
| }
|
|
|
| -uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; }
|
| -
|
| void RTPSender::SetRtxStatus(int mode) {
|
| rtc::CritScope lock(&send_critsect_);
|
| rtx_ = mode;
|
|
|