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); |