| Index: webrtc/call/rtp_stream_receiver_controller.cc
|
| diff --git a/webrtc/call/rtp_stream_receiver_controller.cc b/webrtc/call/rtp_stream_receiver_controller.cc
|
| index 94fa83b60e6699ae3c0cb1373a4c1143af43371e..123866535299f3ce8d5104241bcb1c3b3164edea 100644
|
| --- a/webrtc/call/rtp_stream_receiver_controller.cc
|
| +++ b/webrtc/call/rtp_stream_receiver_controller.cc
|
| @@ -9,8 +9,6 @@
|
| */
|
|
|
| #include "webrtc/call/rtp_stream_receiver_controller.h"
|
| -
|
| -#include "webrtc/rtc_base/logging.h"
|
| #include "webrtc/rtc_base/ptr_util.h"
|
|
|
| namespace webrtc {
|
| @@ -20,11 +18,7 @@
|
| uint32_t ssrc,
|
| RtpPacketSinkInterface* sink)
|
| : controller_(controller), sink_(sink) {
|
| - const bool sink_added = controller_->AddSink(ssrc, sink_);
|
| - if (!sink_added) {
|
| - LOG(LS_ERROR) << "RtpStreamReceiverController::Receiver::Receiver: Sink "
|
| - << "could not be added for SSRC=" << ssrc << ".";
|
| - }
|
| + controller_->AddSink(ssrc, sink_);
|
| }
|
|
|
| RtpStreamReceiverController::Receiver::~Receiver() {
|
| @@ -49,7 +43,7 @@
|
| return demuxer_.OnRtpPacket(packet);
|
| }
|
|
|
| -bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
|
| +void RtpStreamReceiverController::AddSink(uint32_t ssrc,
|
| RtpPacketSinkInterface* sink) {
|
| rtc::CritScope cs(&lock_);
|
| return demuxer_.AddSink(ssrc, sink);
|
|
|