| 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_) {
|
|
|