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; |