| Index: webrtc/call/rtx_receive_stream.cc
|
| diff --git a/webrtc/call/rtx_receive_stream.cc b/webrtc/call/rtx_receive_stream.cc
|
| index 6a5432fea6576965d50545b25974b7e3d27a6fb6..16463525c729de224d6a1dfb9befe76d4617a25f 100644
|
| --- a/webrtc/call/rtx_receive_stream.cc
|
| +++ b/webrtc/call/rtx_receive_stream.cc
|
| @@ -11,21 +11,17 @@
|
| #include <utility>
|
|
|
| #include "webrtc/call/rtx_receive_stream.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
|
| #include "webrtc/rtc_base/logging.h"
|
|
|
| namespace webrtc {
|
|
|
| -RtxReceiveStream::RtxReceiveStream(
|
| - RtpPacketSinkInterface* media_sink,
|
| - std::map<int, int> associated_payload_types,
|
| - uint32_t media_ssrc,
|
| - ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
|
| +RtxReceiveStream::RtxReceiveStream(RtpPacketSinkInterface* media_sink,
|
| + std::map<int, int> associated_payload_types,
|
| + uint32_t media_ssrc)
|
| : media_sink_(media_sink),
|
| associated_payload_types_(std::move(associated_payload_types)),
|
| - media_ssrc_(media_ssrc),
|
| - rtp_receive_statistics_(rtp_receive_statistics) {
|
| + media_ssrc_(media_ssrc) {
|
| if (associated_payload_types_.empty()) {
|
| LOG(LS_WARNING)
|
| << "RtxReceiveStream created with empty payload type mapping.";
|
| @@ -35,12 +31,6 @@
|
| RtxReceiveStream::~RtxReceiveStream() = default;
|
|
|
| void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
|
| - if (rtp_receive_statistics_) {
|
| - RTPHeader header;
|
| - rtx_packet.GetHeader(&header);
|
| - rtp_receive_statistics_->IncomingPacket(header, rtx_packet.size(),
|
| - false /* retransmitted */);
|
| - }
|
| rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
|
|
|
| if (payload.size() < kRtxHeaderSize) {
|
|
|