Index: webrtc/video/rtp_stream_receiver.cc |
diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc |
index 50c50d08ce111b76070c5aa321e01b38eb302397..af12b51556ce7a77341709d092ed46441b743f14 100644 |
--- a/webrtc/video/rtp_stream_receiver.cc |
+++ b/webrtc/video/rtp_stream_receiver.cc |
@@ -50,7 +50,6 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule( |
Transport* outgoing_transport, |
RtcpRttStats* rtt_stats, |
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
TransportSequenceNumberAllocator* transport_sequence_number_allocator) { |
RtpRtcp::Configuration configuration; |
configuration.audio = false; |
@@ -81,7 +80,6 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule( |
static const int kPacketLogIntervalMs = 10000; |
RtpStreamReceiver::RtpStreamReceiver( |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
Transport* transport, |
RtcpRttStats* rtt_stats, |
PacketRouter* packet_router, |
@@ -95,7 +93,6 @@ RtpStreamReceiver::RtpStreamReceiver( |
VCMTiming* timing) |
: clock_(Clock::GetRealTimeClock()), |
config_(*config), |
- remote_bitrate_estimator_(remote_bitrate_estimator), |
packet_router_(packet_router), |
remb_(remb), |
process_thread_(process_thread), |
@@ -114,7 +111,6 @@ RtpStreamReceiver::RtpStreamReceiver( |
transport, |
rtt_stats, |
receive_stats_proxy, |
- remote_bitrate_estimator_, |
packet_router)), |
complete_frame_callback_(complete_frame_callback), |
keyframe_request_sender_(keyframe_request_sender), |
@@ -309,7 +305,6 @@ void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) { |
bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet, |
size_t rtp_packet_length, |
const PacketTime& packet_time) { |
- RTC_DCHECK(remote_bitrate_estimator_); |
{ |
rtc::CritScope lock(&receive_cs_); |
if (!receiving_) { |