Set ChannelReceive transport at construction time.
Followup to cl https://webrtc-review.googlesource.com/c/src/+/103640.
Set the rtcp_send_transport at construction time, delete
RegisterTransport, and the proxying of transport methods.
In addition, delete the unused RtcpRtpStats argument from the
constructor.
Bug: webrtc:9801
Change-Id: I80f25bc08dc2130386053568ddce4ef91654caeb
Reviewed-on: https://webrtc-review.googlesource.com/c/103803
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25010}diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 8e91665..8fcd566 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -39,6 +39,7 @@
#include "test/frame_generator.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
+#include "test/null_transport.h"
#include "test/rtp_rtcp_observer.h"
#include "test/single_threaded_task_queue.h"
#include "test/testsupport/fileutils.h"
@@ -163,6 +164,7 @@
std::unique_ptr<test::PacketTransport> audio_send_transport;
std::unique_ptr<test::PacketTransport> video_send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
+ test::NullTransport rtcp_send_transport;
AudioSendStream* audio_send_stream;
AudioReceiveStream* audio_receive_stream;
@@ -250,6 +252,7 @@
AudioReceiveStream::Config audio_recv_config;
audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
+ audio_recv_config.rtcp_send_transport = &rtcp_send_transport;
audio_recv_config.sync_group = kSyncGroup;
audio_recv_config.decoder_factory = audio_decoder_factory_;
audio_recv_config.decoder_map = {
