diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index d1f70fa..547f15f 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -608,6 +608,8 @@
 
 source_set("webrtc_opus") {
   sources = [
+    "codecs/opus/audio_encoder_opus.cc",
+    "codecs/opus/interface/audio_encoder_opus.h",
     "codecs/opus/interface/opus_interface.h",
     "codecs/opus/opus_inst.h",
     "codecs/opus/opus_interface.c",
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index f8142e2..f9cbe21 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -33,13 +33,13 @@
   // output.
   bool Encode(uint32_t timestamp,
               const int16_t* audio,
-              size_t num_samples,
+              size_t num_samples_per_channel,
               size_t max_encoded_bytes,
               uint8_t* encoded,
               size_t* encoded_bytes,
               uint32_t* encoded_timestamp) {
-    CHECK_EQ(num_samples,
-             static_cast<size_t>(sample_rate_hz() / 100 * num_channels()));
+    CHECK_EQ(num_samples_per_channel,
+             static_cast<size_t>(sample_rate_hz() / 100));
     bool ret = Encode(timestamp,
                       audio,
                       max_encoded_bytes,
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
new file mode 100644
index 0000000..0a3661f
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2014 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/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
+
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+namespace webrtc {
+
+namespace {
+
+// We always encode at 48 kHz.
+const int kSampleRateHz = 48000;
+
+int DivExact(int a, int b) {
+  CHECK_EQ(a % b, 0);
+  return a / b;
+}
+
+int16_t ClampInt16(size_t x) {
+  return static_cast<int16_t>(
+      std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())));
+}
+
+int16_t CastInt16(size_t x) {
+  DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()));
+  return static_cast<int16_t>(x);
+}
+
+}  // namespace
+
+AudioEncoderOpus::Config::Config() : frame_size_ms(20), num_channels(1) {}
+
+bool AudioEncoderOpus::Config::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (num_channels <= 0)
+    return false;
+  return true;
+}
+
+AudioEncoderOpus::AudioEncoderOpus(const Config& config)
+    : num_10ms_frames_per_packet_(DivExact(config.frame_size_ms, 10)),
+      num_channels_(config.num_channels),
+      samples_per_10ms_frame_(DivExact(kSampleRateHz, 100) * num_channels_) {
+  CHECK(config.IsOk());
+  input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
+  CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_));
+}
+
+AudioEncoderOpus::~AudioEncoderOpus() {
+  CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+int AudioEncoderOpus::sample_rate_hz() const {
+  return kSampleRateHz;
+}
+
+int AudioEncoderOpus::num_channels() const {
+  return num_channels_;
+}
+
+int AudioEncoderOpus::num_10ms_frames_per_packet() const {
+  return num_10ms_frames_per_packet_;
+}
+
+bool AudioEncoderOpus::Encode(uint32_t timestamp,
+                              const int16_t* audio,
+                              size_t max_encoded_bytes,
+                              uint8_t* encoded,
+                              size_t* encoded_bytes,
+                              uint32_t* encoded_timestamp) {
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = timestamp;
+  input_buffer_.insert(input_buffer_.end(), audio,
+                       audio + samples_per_10ms_frame_);
+  if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
+                              samples_per_10ms_frame_)) {
+    *encoded_bytes = 0;
+    return true;
+  }
+  CHECK_EQ(input_buffer_.size(),
+           static_cast<size_t>(num_10ms_frames_per_packet_) *
+           samples_per_10ms_frame_);
+  int16_t r = WebRtcOpus_Encode(
+      inst_, &input_buffer_[0],
+      DivExact(CastInt16(input_buffer_.size()), num_channels_),
+      ClampInt16(max_encoded_bytes), encoded);
+  input_buffer_.clear();
+  if (r < 0)
+    return false;
+  *encoded_bytes = r;
+  *encoded_timestamp = first_timestamp_in_buffer_;
+  return true;
+}
+
+}  // 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
new file mode 100644
index 0000000..7325b7e
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
+namespace webrtc {
+
+class AudioEncoderOpus : public AudioEncoder {
+ public:
+  struct Config {
+    Config();
+    bool IsOk() const;
+    int frame_size_ms;
+    int num_channels;
+  };
+
+  explicit AudioEncoderOpus(const Config& config);
+  virtual ~AudioEncoderOpus() OVERRIDE;
+
+  virtual int sample_rate_hz() const OVERRIDE;
+  virtual int num_channels() const OVERRIDE;
+  virtual int num_10ms_frames_per_packet() const OVERRIDE;
+
+ protected:
+  virtual bool Encode(uint32_t timestamp,
+                      const int16_t* audio,
+                      size_t max_encoded_bytes,
+                      uint8_t* encoded,
+                      size_t* encoded_bytes,
+                      uint32_t* encoded_timestamp) OVERRIDE;
+
+ private:
+  const int num_10ms_frames_per_packet_;
+  const int num_channels_;
+  const int samples_per_10ms_frame_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
index 89f0a54..b537285 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus.gypi
+++ b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
@@ -27,6 +27,8 @@
         '<(webrtc_root)',
       ],
       'sources': [
+        'audio_encoder_opus.cc',
+        'interface/audio_encoder_opus.h',
         'interface/opus_interface.h',
         'opus_inst.h',
         'opus_interface.c',
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 3a5a13f..b6c6ba1 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -26,7 +26,7 @@
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/system_wrappers/interface/data_log.h"
@@ -140,17 +140,20 @@
                           size_t input_len_samples,
                           uint8_t* output) {
     size_t enc_len_bytes = 0;
+    scoped_ptr<int16_t[]> interleaved_input(
+        new int16_t[channels_ * input_len_samples]);
     for (int i = 0; i < audio_encoder_->num_10ms_frames_per_packet(); ++i) {
       EXPECT_EQ(0u, enc_len_bytes);
-      EXPECT_TRUE(audio_encoder_->Encode(0,
-                                         input,
-                                         audio_encoder_->sample_rate_hz() / 100,
-                                         data_length_ * 2,
-                                         output,
-                                         &enc_len_bytes,
-                                         &output_timestamp_));
+
+      // Duplicate the mono input signal to however many channels the test
+      // wants.
+      test::InputAudioFile::DuplicateInterleaved(
+          input, input_len_samples, channels_, interleaved_input.get());
+
+      EXPECT_TRUE(audio_encoder_->Encode(
+          0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100,
+          data_length_ * 2, output, &enc_len_bytes, &output_timestamp_));
     }
-    EXPECT_EQ(input_len_samples, enc_len_bytes);
     return static_cast<int>(enc_len_bytes);
   }
 
@@ -636,56 +639,22 @@
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
     decoder_ = new AudioDecoderOpus(kDecoderOpus);
-    assert(decoder_);
-    WebRtcOpus_EncoderCreate(&encoder_, 1);
+    AudioEncoderOpus::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+    audio_encoder_.reset(new AudioEncoderOpus(config));
   }
-
-  ~AudioDecoderOpusTest() {
-    WebRtcOpus_EncoderFree(encoder_);
-  }
-
-  virtual void InitEncoder() {}
-
-  virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
-                          uint8_t* output) OVERRIDE {
-    int enc_len_bytes = WebRtcOpus_Encode(encoder_, const_cast<int16_t*>(input),
-        static_cast<int16_t>(input_len_samples),
-        static_cast<int16_t>(data_length_), output);
-    EXPECT_GT(enc_len_bytes, 0);
-    return enc_len_bytes;
-  }
-
-  OpusEncInst* encoder_;
 };
 
 class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
  protected:
   AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
     channels_ = 2;
-    WebRtcOpus_EncoderFree(encoder_);
     delete decoder_;
     decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch);
-    assert(decoder_);
-    WebRtcOpus_EncoderCreate(&encoder_, 2);
-  }
-
-  virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
-                          uint8_t* output) OVERRIDE {
-    // Create stereo by duplicating each sample in |input|.
-    const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
-    scoped_ptr<int16_t[]> input_stereo(new int16_t[input_stereo_samples]);
-    test::InputAudioFile::DuplicateInterleaved(
-        input, input_len_samples, 2, input_stereo.get());
-
-    // Note that the input length is given as samples per channel.
-    int enc_len_bytes =
-        WebRtcOpus_Encode(encoder_,
-                          input_stereo.get(),
-                          static_cast<int16_t>(input_len_samples),
-                          static_cast<int16_t>(data_length_),
-                          output);
-    EXPECT_GT(enc_len_bytes, 0);
-    return enc_len_bytes;
+    AudioEncoderOpus::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+    config.num_channels = 2;
+    audio_encoder_.reset(new AudioEncoderOpus(config));
   }
 };