Revert of Let ViEEncoder handle resolution changes.

webrtc/call/rampup_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 "webrtc/call/rampup_tests.h"
 
 #include "webrtc/test/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/testsupport/perf_test.h"
 
 namespace webrtc {
 namespace {
 
 static const int64_t kPollIntervalMs = 20;
 
 std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
   std::vector<uint32_t> ssrcs;
   for (size_t i = 0; i != num_streams; ++i)
@@ skipping 63 matching lines @@
 }
 
 size_t RampUpTester::GetNumVideoStreams() const {
   return num_video_streams_;
 }
 
 size_t RampUpTester::GetNumAudioStreams() const {
   return num_audio_streams_;
 }
 
-class RampUpTester::VideoStreamFactory
-    : public VideoEncoderConfig::VideoStreamFactoryInterface {
- public:
-  VideoStreamFactory() {}
-
- private:
-  std::vector<VideoStream> CreateEncoderStreams(
-      int width,
-      int height,
-      const VideoEncoderConfig& encoder_config) override {
-    std::vector<VideoStream> streams =
-        test::CreateVideoStreams(width, height, encoder_config);
-    if (encoder_config.number_of_streams == 1) {
-      streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000;
-    }
-    return streams;
-  }
-};
-
 void RampUpTester::ModifyVideoConfigs(
     VideoSendStream::Config* send_config,
     std::vector<VideoReceiveStream::Config>* receive_configs,
     VideoEncoderConfig* encoder_config) {
   send_config->suspend_below_min_bitrate = true;
-  encoder_config->number_of_streams = num_video_streams_;
-  encoder_config->max_bitrate_bps = 2000000;
-  encoder_config->video_stream_factory =
-      new rtc::RefCountedObject<RampUpTester::VideoStreamFactory>();
+
   if (num_video_streams_ == 1) {
+    encoder_config->streams[0].target_bitrate_bps =
+        encoder_config->streams[0].max_bitrate_bps = 2000000;
     // For single stream rampup until 1mbps
     expected_bitrate_bps_ = kSingleStreamTargetBps;
   } else {
     // For multi stream rampup until all streams are being sent. That means
-    // enough bitrate to send all the target streams plus the min bitrate of
+    // enough birate to send all the target streams plus the min bitrate of
     // the last one.
-    std::vector<VideoStream> streams = test::CreateVideoStreams(
-        test::CallTest::kDefaultWidth, test::CallTest::kDefaultHeight,
-        *encoder_config);
-    expected_bitrate_bps_ = streams.back().min_bitrate_bps;
-    for (size_t i = 0; i < streams.size() - 1; ++i) {
-      expected_bitrate_bps_ += streams[i].target_bitrate_bps;
+    expected_bitrate_bps_ = encoder_config->streams.back().min_bitrate_bps;
+    for (size_t i = 0; i < encoder_config->streams.size() - 1; ++i) {
+      expected_bitrate_bps_ += encoder_config->streams[i].target_bitrate_bps;
     }
   }
 
   send_config->rtp.extensions.clear();
 
   bool remb;
   bool transport_cc;
   if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
     remb = true;
     transport_cc = false;
@@ skipping 469 matching lines @@
                     true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumberUri, false, false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc