Revert of Add task queue to Call.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
@@ skipping 1263 matching lines @@
       encoder_config.streams = test::CreateVideoStreams(1);
       VideoStream* stream = &encoder_config.streams[0];
       stream->width = width;
       stream->height = height;
       stream->max_framerate = 5;
       stream->min_bitrate_bps = stream->target_bitrate_bps =
           stream->max_bitrate_bps = 100000;
 
       UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
 
-      send_streams[i] = sender_call->CreateVideoSendStream(
-          send_config.Copy(), encoder_config.Copy());
+      send_streams[i] =
+          sender_call->CreateVideoSendStream(send_config, encoder_config);
       send_streams[i]->Start();
 
       VideoReceiveStream::Config receive_config(receiver_transport.get());
       receive_config.rtp.remote_ssrc = ssrc;
       receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
       VideoReceiveStream::Decoder decoder =
           test::CreateMatchingDecoder(send_config.encoder_settings);
       allocated_decoders.push_back(
           std::unique_ptr<VideoDecoder>(decoder.decoder));
       receive_config.decoders.push_back(decoder);
@@ skipping 1183 matching lines @@
         VideoEncoderConfig* encoder_config) override {
       if (num_ssrcs_ > 1) {
         // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
         for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
           encoder_config->streams[i].min_bitrate_bps = 10000;
           encoder_config->streams[i].target_bitrate_bps = 15000;
           encoder_config->streams[i].max_bitrate_bps = 20000;
         }
       }
 
-      video_encoder_config_all_streams_ = encoder_config->Copy();
+      video_encoder_config_all_streams_ = *encoder_config;
       if (send_single_ssrc_first_)
         encoder_config->streams.resize(1);
     }
 
     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for "
                           << (send_single_ssrc_first_ ? "first SSRC."
                                                       : "SSRCs.");
 
       if (send_single_ssrc_first_) {
         // Set full simulcast and continue with the rest of the SSRCs.
         send_stream_->ReconfigureVideoEncoder(
-            std::move(video_encoder_config_all_streams_));
+            video_encoder_config_all_streams_);
         EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
       }
     }
 
    private:
     std::map<uint32_t, bool> valid_ssrcs_;
     std::map<uint32_t, bool> is_observed_;
 
     const size_t num_ssrcs_;
     const bool send_single_ssrc_first_;
@@ skipping 674 matching lines @@
 
   // Lower bitrates so that all streams send initially.
   for (size_t i = 0; i < video_encoder_config_.streams.size(); ++i) {
     video_encoder_config_.streams[i].min_bitrate_bps = 10000;
     video_encoder_config_.streams[i].target_bitrate_bps = 15000;
     video_encoder_config_.streams[i].max_bitrate_bps = 20000;
   }
 
   // Use the same total bitrates when sending a single stream to avoid lowering
   // the bitrate estimate and requiring a subsequent rampup.
-  VideoEncoderConfig one_stream = video_encoder_config_.Copy();
+  VideoEncoderConfig one_stream = video_encoder_config_;
   one_stream.streams.resize(1);
   for (size_t i = 1; i < video_encoder_config_.streams.size(); ++i) {
     one_stream.streams.front().min_bitrate_bps +=
         video_encoder_config_.streams[i].min_bitrate_bps;
     one_stream.streams.front().target_bitrate_bps +=
         video_encoder_config_.streams[i].target_bitrate_bps;
     one_stream.streams.front().max_bitrate_bps +=
         video_encoder_config_.streams[i].max_bitrate_bps;
   }
 
   CreateMatchingReceiveConfigs(&receive_transport);
 
   CreateVideoStreams();
   CreateFrameGeneratorCapturer();
 
   Start();
   EXPECT_TRUE(observer.Wait())
       << "Timed out waiting for all SSRCs to send packets.";
 
   // Test stream resetting more than once to make sure that the state doesn't
   // get set once (this could be due to using std::map::insert for instance).
   for (size_t i = 0; i < 3; ++i) {
     frame_generator_capturer_->Stop();
     sender_call_->DestroyVideoSendStream(video_send_stream_);
 
     // Re-create VideoSendStream with only one stream.
-    video_send_stream_ = sender_call_->CreateVideoSendStream(
-        video_send_config_.Copy(), one_stream.Copy());
+    video_send_stream_ =
+        sender_call_->CreateVideoSendStream(video_send_config_, one_stream);
     video_send_stream_->Start();
     if (provoke_rtcpsr_before_rtp) {
       // Rapid Resync Request forces sending RTCP Sender Report back.
       // Using this request speeds up this test because then there is no need
       // to wait for a second for periodic Sender Report.
       rtcp::RapidResyncRequest force_send_sr_back_request;
       rtc::Buffer packet = force_send_sr_back_request.Build();
       static_cast<webrtc::test::DirectTransport&>(receive_transport)
           .SendRtcp(packet.data(), packet.size());
     }
     CreateFrameGeneratorCapturer();
     frame_generator_capturer_->Start();
 
     observer.ResetExpectedSsrcs(1);
     EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
 
     // Reconfigure back to use all streams.
-    video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
+    video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
     observer.ResetExpectedSsrcs(kNumSsrcs);
     EXPECT_TRUE(observer.Wait())
         << "Timed out waiting for all SSRCs to send packets.";
 
     // Reconfigure down to one stream.
-    video_send_stream_->ReconfigureVideoEncoder(one_stream.Copy());
+    video_send_stream_->ReconfigureVideoEncoder(one_stream);
     observer.ResetExpectedSsrcs(1);
     EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
 
     // Reconfigure back to use all streams.
-    video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
+    video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_);
     observer.ResetExpectedSsrcs(kNumSsrcs);
     EXPECT_TRUE(observer.Wait())
         << "Timed out waiting for all SSRCs to send packets.";
   }
 
   send_transport.StopSending();
   receive_transport.StopSending();
 
   Stop();
   DestroyStreams();
@@ skipping 460 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc