diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index ae092f3837..570b19b4ce 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -468,12 +468,7 @@ TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
   const int kRunTimeMs = 20000;
   TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
 }
-#if defined(WEBRTC_ANDROID)
-// This test is disabled on android as it does not update
-// sinkWants below 320x180, the starting resolution for these
-// tests.
-#define ReceivesCpuOveruseAndUnderuse DISABLED_ReceivesCpuOveruseAndUnderuse
-#endif
+
 TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) {
   class LoadObserver : public test::SendTest,
                        public test::FrameGeneratorCapturer::SinkWantsObserver {
@@ -486,6 +481,8 @@ TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) {
     void OnFrameGeneratorCapturerCreated(
         test::FrameGeneratorCapturer* frame_generator_capturer) override {
       frame_generator_capturer->SetSinkWantsObserver(this);
+      // Set a high initial resolution to be sure that we can scale down.
+      frame_generator_capturer->ChangeResolution(1920, 1080);
     }
 
     // OnSinkWantsChanged is called when FrameGeneratorCapturer::AddOrUpdateSink
diff --git a/webrtc/video/vie_encoder.cc b/webrtc/video/vie_encoder.cc
index 1458af266f..7880c4a275 100644
--- a/webrtc/video/vie_encoder.cc
+++ b/webrtc/video/vie_encoder.cc
@@ -35,14 +35,12 @@ using DegradationPreference = VideoSendStream::DegradationPreference;
 
 // Time interval for logging frame counts.
 const int64_t kFrameLogIntervalMs = 60000;
+
 // We will never ask for a resolution lower than this.
-#if defined(WEBRTC_ANDROID)
 // TODO(kthelgason): Lower this limit when better testing
 // on MediaCodec and fallback implementations are in place.
+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=7206
 const int kMinPixelsPerFrame = 320 * 180;
-#else
-const int kMinPixelsPerFrame = 120 * 90;
-#endif
 
 // The maximum number of frames to drop at beginning of stream
 // to try and achieve desired bitrate.