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 "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/test/testsupport/perf_test.h"
+#include "webrtc/test/encoder_settings.h"

[mflodman, 2016/09/27 11:28:00]

Alphabetic order.

[perkj_webrtc, 2016/09/27 13:45:17]

Done.

 
 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)
     ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
@@ skipping 62 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 {
+    RTC_DCHECK_EQ(width, test::CallTest::kDefaultWidth);
+    RTC_DCHECK_EQ(height, test::CallTest::kDefaultHeight);
+    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->encoder_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 birate to send all the target streams plus the min bitrate of
+    // enough bitrate to send all the target streams plus the min bitrate of
     // the last one.
-    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;
+    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;
     }
   }
 
   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