Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
index ea7931fadcd729c1ed149fe05a1f2f483443b00c..8b7bbb3d023cf66dfc3ac925d6cb560390e6481e 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc |
@@ -132,13 +132,11 @@ class RTCPSender::PacketBuiltCallback |
}; |
RTCPSender::RTCPSender( |
- int32_t id, |
bool audio, |
Clock* clock, |
ReceiveStatistics* receive_statistics, |
RtcpPacketTypeCounterObserver* packet_type_counter_observer) |
- : id_(id), |
- audio_(audio), |
+ : audio_(audio), |
clock_(clock), |
method_(kRtcpOff), |
critical_section_transport_( |
@@ -1119,7 +1117,7 @@ bool RTCPSender::PrepareReport(const FeedbackState& feedback_state, |
int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) { |
CriticalSectionScoped lock(critical_section_transport_.get()); |
if (cbTransport_) { |
- if (cbTransport_->SendRTCPPacket(id_, dataBuffer, length) > 0) |
+ if (cbTransport_->SendRTCPPacket(dataBuffer, length) > 0) |
return 0; |
} |
return -1; |
@@ -1218,18 +1216,17 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) { |
class Sender : public rtcp::RtcpPacket::PacketReadyCallback { |
public: |
- Sender(Transport* transport, int32_t id) |
- : transport_(transport), id_(id), send_failure_(false) {} |
+ Sender(Transport* transport) |
+ : transport_(transport), send_failure_(false) {} |
void OnPacketReady(uint8_t* data, size_t length) override { |
- if (transport_->SendRTCPPacket(id_, data, length) <= 0) |
+ if (transport_->SendRTCPPacket(data, length) <= 0) |
send_failure_ = true; |
} |
Transport* const transport_; |
- int32_t id_; |
bool send_failure_; |
- } sender(cbTransport_, id_); |
+ } sender(cbTransport_); |
uint8_t buffer[IP_PACKET_SIZE]; |
return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) && |