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) { |