commit 7a9a092708f1f3abc45f9aabda2db205132cc4ac
author Bjorn A Mellem <mellem@webrtc.org> Tue Nov 26 17:19:40 2019
committer Commit Bot <commit-bot@chromium.org> Tue Nov 26 19:19:36 2019
tree 571b3fa38dc3967749eeced63f545b3e9760c31a
parent 2b4bd975171eecdc460787b23456326e12eecdb3

Delete media transport integration.

MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}

test/call_test.cc

diff --git a/test/call_test.cc b/test/call_test.cc
index 9f26cc6..10b631a 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -36,7 +36,7 @@
       task_queue_factory_(CreateDefaultTaskQueueFactory()),
       send_event_log_(std::make_unique<RtcEventLogNull>()),
       recv_event_log_(std::make_unique<RtcEventLogNull>()),
-      audio_send_config_(/*send_transport=*/nullptr, MediaTransportConfig()),
+      audio_send_config_(/*send_transport=*/nullptr),
       audio_send_stream_(nullptr),
       frame_generator_capturer_(nullptr),
       fake_encoder_factory_([this]() {
@@ -275,8 +275,7 @@
   RTC_DCHECK_LE(num_audio_streams, 1);
   RTC_DCHECK_LE(num_flexfec_streams, 1);
   if (num_audio_streams > 0) {
-    AudioSendStream::Config audio_send_config(send_transport,
-                                              MediaTransportConfig());
+    AudioSendStream::Config audio_send_config(send_transport);
     audio_send_config.rtp.ssrc = kAudioSendSsrc;
     audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
         kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});

test/peer_scenario/peer_scenario_client.cc

diff --git a/test/peer_scenario/peer_scenario_client.cc b/test/peer_scenario/peer_scenario_client.cc
index 28cbb6e..d8f2b65 100644
--- a/test/peer_scenario/peer_scenario_client.cc
+++ b/test/peer_scenario/peer_scenario_client.cc
@@ -185,7 +185,6 @@
   pcf_deps.fec_controller_factory = nullptr;
   pcf_deps.network_controller_factory = nullptr;
   pcf_deps.network_state_predictor_factory = nullptr;
-  pcf_deps.media_transport_factory = nullptr;
 
   pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
 

test/scenario/audio_stream.cc

diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index f5d2116..2738f69 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -73,8 +73,7 @@
     rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
     Transport* send_transport)
     : sender_(sender), config_(config) {
-  AudioSendStream::Config send_config(send_transport,
-                                      webrtc::MediaTransportConfig());
+  AudioSendStream::Config send_config(send_transport);
   ssrc_ = sender->GetNextAudioSsrc();
   send_config.rtp.ssrc = ssrc_;
   SdpAudioFormat::Parameters sdp_params;

test/scenario/stats_collection.h

diff --git a/test/scenario/stats_collection.h b/test/scenario/stats_collection.h
index 64cb58c..908385e 100644
--- a/test/scenario/stats_collection.h
+++ b/test/scenario/stats_collection.h
@@ -15,6 +15,7 @@
 
 #include "absl/types/optional.h"
 #include "call/call.h"
+#include "rtc_base/thread.h"
 #include "test/logging/log_writer.h"
 #include "test/scenario/performance_stats.h"