diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
new file mode 100644
index 0000000000..9b2f07fe3c
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class AudioEncoderOpusTest : public ::testing::Test {
+ protected:
+  // The constructor simply creates an Opus encoder with default configuration.
+  AudioEncoderOpusTest()
+      : opus_(new AudioEncoderOpus(AudioEncoderOpus::Config())) {}
+
+  // Repeatedly sets packet loss rates in the range [from, to], increasing by
+  // 0.01 in each step. The function verifies that the actual loss rate is
+  // |expected_return|.
+  void TestSetPacketLossRate(double from, double to, double expected_return) {
+    ASSERT_TRUE(opus_);
+    for (double loss = from; loss <= to;
+         (to >= from) ? loss += 0.01 : loss -= 0.01) {
+      opus_->SetProjectedPacketLossRate(loss);
+      EXPECT_DOUBLE_EQ(expected_return, opus_->packet_loss_rate());
+    }
+  }
+
+  scoped_ptr<AudioEncoderOpus> opus_;
+};
+
+namespace {
+// These constants correspond to those used in
+// AudioEncoderOpus::SetProjectedPacketLossRate.
+const double kPacketLossRate20 = 0.20;
+const double kPacketLossRate10 = 0.10;
+const double kPacketLossRate5 = 0.05;
+const double kPacketLossRate1 = 0.01;
+const double kLossRate20Margin = 0.02;
+const double kLossRate10Margin = 0.01;
+const double kLossRate5Margin = 0.01;
+}  // namespace
+
+TEST_F(AudioEncoderOpusTest, PacketLossRateOptimized) {
+  // Note that the order of the following calls is critical.
+  TestSetPacketLossRate(0.0, 0.0, 0.0);
+  TestSetPacketLossRate(kPacketLossRate1,
+                        kPacketLossRate5 + kLossRate5Margin - 0.01,
+                        kPacketLossRate1);
+  TestSetPacketLossRate(kPacketLossRate5 + kLossRate5Margin,
+                        kPacketLossRate10 + kLossRate10Margin - 0.01,
+                        kPacketLossRate5);
+  TestSetPacketLossRate(kPacketLossRate10 + kLossRate10Margin,
+                        kPacketLossRate20 + kLossRate20Margin - 0.01,
+                        kPacketLossRate10);
+  TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
+                        1.0,
+                        kPacketLossRate20);
+  TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
+                        kPacketLossRate20 - kLossRate20Margin,
+                        kPacketLossRate20);
+  TestSetPacketLossRate(kPacketLossRate20 - kLossRate20Margin - 0.01,
+                        kPacketLossRate10 - kLossRate10Margin,
+                        kPacketLossRate10);
+  TestSetPacketLossRate(kPacketLossRate10 - kLossRate10Margin - 0.01,
+                        kPacketLossRate5 - kLossRate5Margin,
+                        kPacketLossRate5);
+  TestSetPacketLossRate(kPacketLossRate5 - kLossRate5Margin - 0.01,
+                        kPacketLossRate1,
+                        kPacketLossRate1);
+  TestSetPacketLossRate(0.0, 0.0, 0.0);
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
index 477ee8fff1..c45388b8f7 100644
--- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
@@ -49,6 +49,7 @@ class AudioEncoderOpus final : public AudioEncoder {
   virtual int Max10MsFramesInAPacket() const OVERRIDE;
   void SetTargetBitrate(int bits_per_second) override;
   void SetProjectedPacketLossRate(double fraction) override;
+  double packet_loss_rate() const { return packet_loss_rate_; }
 
  protected:
   virtual bool EncodeInternal(uint32_t rtp_timestamp,
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 23ee29f431..376e6c5b45 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -105,6 +105,7 @@
             'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
             'audio_coding/codecs/isac/main/source/isac_unittest.cc',
             'audio_coding/codecs/isac/main/source/audio_encoder_isac_red_unittest.cc',
+            'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
             'audio_coding/codecs/opus/opus_unittest.cc',
             'audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc',
             'audio_coding/neteq/audio_classifier_unittest.cc',