| 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 17bcc7c625009abfa7b1f9be7afa6d0a6a11b65f..d1de668e69d5ed992e6e42cab1c299b017f5610a 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -464,10 +464,10 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
|
| size_t RTPSender::SendPadData(size_t bytes,
|
| const PacedPacketInfo& pacing_info) {
|
| size_t padding_bytes_in_packet;
|
| + size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
|
|
|
| if (audio_configured_) {
|
| // Allow smaller padding packets for audio.
|
| - size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
|
| padding_bytes_in_packet =
|
| std::min(std::max(bytes, kMinAudioPaddingLength), max_payload_size);
|
| if (padding_bytes_in_packet > kMaxPaddingLength)
|
| @@ -477,10 +477,6 @@ size_t RTPSender::SendPadData(size_t bytes,
|
| // RtpPacketSender, which will make sure we don't send too much padding even
|
| // if a single packet is larger than requested.
|
| // We do this to avoid frequently sending small packets on higher bitrates.
|
| - size_t max_payload_size =
|
| - max_packet_size_ - RtpHeaderLength() // RTP overhead.
|
| - - video_->FecPacketOverhead() // FEC/ULP/RED overhead.
|
| - - (RtxStatus() ? kRtxHeaderSize : 0); // RTX overhead.
|
| padding_bytes_in_packet = std::min(max_payload_size, kMaxPaddingLength);
|
| }
|
| size_t bytes_sent = 0;
|
|
|