diff --git a/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index 0d47191ee..58874093a 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/talk/app/webrtc/webrtcsdp.cc
@@ -71,6 +71,7 @@ using cricket::kCodecParamUseInbandFec;
 using cricket::kCodecParamSctpProtocol;
 using cricket::kCodecParamSctpStreams;
 using cricket::kCodecParamMaxAverageBitrate;
+using cricket::kCodecParamMaxPlaybackRate;
 using cricket::kCodecParamAssociatedPayloadType;
 using cricket::kWildcardPayloadType;
 using cricket::MediaContentDescription;
@@ -1534,7 +1535,8 @@ bool IsFmtpParam(const std::string& name) {
     kCodecParamStereo, kCodecParamUseInbandFec, kCodecParamStartBitrate,
     kCodecParamMaxBitrate, kCodecParamMinBitrate, kCodecParamMaxQuantization,
     kCodecParamSctpProtocol, kCodecParamSctpStreams,
-    kCodecParamMaxAverageBitrate, kCodecParamAssociatedPayloadType
+    kCodecParamMaxAverageBitrate, kCodecParamMaxPlaybackRate,
+    kCodecParamAssociatedPayloadType
   };
   for (size_t i = 0; i < ARRAY_SIZE(kFmtpParams); ++i) {
     if (_stricmp(name.c_str(), kFmtpParams[i]) == 0) {
diff --git a/talk/media/base/constants.cc b/talk/media/base/constants.cc
index cd10ef75f..19a960f35 100644
--- a/talk/media/base/constants.cc
+++ b/talk/media/base/constants.cc
@@ -59,6 +59,7 @@ const char kCodecParamSPropStereo[] = "sprop-stereo";
 const char kCodecParamStereo[] = "stereo";
 const char kCodecParamUseInbandFec[] = "useinbandfec";
 const char kCodecParamMaxAverageBitrate[] = "maxaveragebitrate";
+const char kCodecParamMaxPlaybackRate[] = "maxplaybackrate";
 
 const char kCodecParamSctpProtocol[] = "protocol";
 const char kCodecParamSctpStreams[] = "streams";
@@ -72,6 +73,7 @@ const int kOpusDefaultMinPTime = 3;
 const int kOpusDefaultSPropStereo = 0;
 const int kOpusDefaultStereo = 0;
 const int kOpusDefaultUseInbandFec = 0;
+const int kOpusDefaultMaxPlaybackRate = 48000;
 
 const int kPreferredMaxPTime = 60;
 const int kPreferredMinPTime = 10;
diff --git a/talk/media/base/constants.h b/talk/media/base/constants.h
index 5ac1be2b6..5168acb89 100644
--- a/talk/media/base/constants.h
+++ b/talk/media/base/constants.h
@@ -62,6 +62,7 @@ extern const char kCodecParamSPropStereo[];
 extern const char kCodecParamStereo[];
 extern const char kCodecParamUseInbandFec[];
 extern const char kCodecParamMaxAverageBitrate[];
+extern const char kCodecParamMaxPlaybackRate[];
 extern const char kCodecParamSctpProtocol[];
 extern const char kCodecParamSctpStreams[];
 
@@ -79,6 +80,8 @@ extern const int kOpusDefaultMinPTime;
 extern const int kOpusDefaultSPropStereo;
 extern const int kOpusDefaultStereo;
 extern const int kOpusDefaultUseInbandFec;
+extern const int kOpusDefaultMaxPlaybackRate;
+
 // Prefered values in this code base. Note that they may differ from the default
 // values in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
 // Only frames larger or equal to 10 ms are currently supported in this code
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index fe443904d..c984a01da 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -61,6 +61,12 @@ static const int kFakeDeviceId = 0;
 static const int kFakeDeviceId = 1;
 #endif
 
+static const int kOpusBandwidthNb = 4000;
+static const int kOpusBandwidthMb = 6000;
+static const int kOpusBandwidthWb = 8000;
+static const int kOpusBandwidthSwb = 12000;
+static const int kOpusBandwidthFb = 20000;
+
 // Verify the header extension ID, if enabled, is within the bounds specified in
 // [RFC5285]: 1-14 inclusive.
 #define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \
@@ -180,6 +186,7 @@ class FakeWebRtcVoiceEngine
           file(false),
           vad(false),
           codec_fec(false),
+          max_encoding_bandwidth(0),
           red(false),
           nack(false),
           media_processor_registered(false),
@@ -209,6 +216,7 @@ class FakeWebRtcVoiceEngine
     bool file;
     bool vad;
     bool codec_fec;
+    int max_encoding_bandwidth;
     bool red;
     bool nack;
     bool media_processor_registered;
@@ -305,6 +313,9 @@ class FakeWebRtcVoiceEngine
   bool GetCodecFEC(int channel) {
     return channels_[channel]->codec_fec;
   }
+  int GetMaxEncodingBandwidth(int channel) {
+    return channels_[channel]->max_encoding_bandwidth;
+  }
   bool GetNACK(int channel) {
     return channels_[channel]->nack;
   }
@@ -624,10 +635,11 @@ class FakeWebRtcVoiceEngine
   }
   WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
                              webrtc::VadModes& mode, bool& disabledDTX));
+
 #ifdef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
     WEBRTC_CHECK_CHANNEL(channel);
-    if (strcmp(channels_[channel]->send_codec.plname, "opus")) {
+    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
       // Return -1 if current send codec is not Opus.
       // TODO(minyue): Excludes other codecs if they support inband FEC.
       return -1;
@@ -640,6 +652,25 @@ class FakeWebRtcVoiceEngine
     enable = channels_[channel]->codec_fec;
     return 0;
   }
+
+  WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) {
+    WEBRTC_CHECK_CHANNEL(channel);
+    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
+      // Return -1 if current send codec is not Opus.
+      return -1;
+    }
+    if (frequency_hz <= 8000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb;
+    else if (frequency_hz <= 12000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb;
+    else if (frequency_hz <= 16000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb;
+    else if (frequency_hz <= 24000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb;
+    else
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
+    return 0;
+  }
 #endif  // USE_WEBRTC_DEV_BRANCH
 
   // webrtc::VoEDtmf
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 90ca8cbff..3d5d9ee66 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -120,6 +120,7 @@ static const int kOpusStereoBitrate = 64000;
 // Opus bitrate should be in the range between 6000 and 510000.
 static const int kOpusMinBitrate = 6000;
 static const int kOpusMaxBitrate = 510000;
+
 // Default audio dscp value.
 // See http://tools.ietf.org/html/rfc2474 for details.
 // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
@@ -404,6 +405,7 @@ static bool IsOpusStereoEnabled(const AudioCodec& codec) {
   return codec.GetParam(kCodecParamStereo, &value) && value == 1;
 }
 
+// TODO(minyue): Clamp bitrate when invalid.
 static bool IsValidOpusBitrate(int bitrate) {
   return (bitrate >= kOpusMinBitrate && bitrate <= kOpusMaxBitrate);
 }
@@ -430,6 +432,59 @@ static bool IsOpusFecEnabled(const AudioCodec& codec) {
   return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
 }
 
+// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
+// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
+static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
+  int value;
+  if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
+    return value;
+  }
+  return kOpusDefaultMaxPlaybackRate;
+}
+
+static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
+                          bool* enable_codec_fec, int* max_playback_rate) {
+  *enable_codec_fec = IsOpusFecEnabled(codec);
+  *max_playback_rate = GetOpusMaxPlaybackRate(codec);
+
+  // If OPUS, change what we send according to the "stereo" codec
+  // parameter, and not the "channels" parameter.  We set
+  // voe_codec.channels to 2 if "stereo=1" and 1 otherwise.  If
+  // the bitrate is not specified, i.e. is zero, we set it to the
+  // appropriate default value for mono or stereo Opus.
+
+  // TODO(minyue): The determination of bit rate might take the maximum playback
+  // rate into account.
+
+  if (IsOpusStereoEnabled(codec)) {
+    voe_codec->channels = 2;
+    if (!IsValidOpusBitrate(codec.bitrate)) {
+      if (codec.bitrate != 0) {
+        LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
+                        << codec.bitrate
+                        << ") with default opus stereo bitrate: "
+                        << kOpusStereoBitrate;
+      }
+      voe_codec->rate = kOpusStereoBitrate;
+    }
+  } else {
+    voe_codec->channels = 1;
+    if (!IsValidOpusBitrate(codec.bitrate)) {
+      if (codec.bitrate != 0) {
+        LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
+                        << codec.bitrate
+                        << ") with default opus mono bitrate: "
+                        << kOpusMonoBitrate;
+      }
+      voe_codec->rate = kOpusMonoBitrate;
+    }
+  }
+  int bitrate_from_params = GetOpusBitrateFromParams(codec);
+  if (bitrate_from_params != 0) {
+    voe_codec->rate = bitrate_from_params;
+  }
+}
+
 void WebRtcVoiceEngine::ConstructCodecs() {
   LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
@@ -1993,6 +2048,10 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
   bool nack_enabled = nack_enabled_;
   bool enable_codec_fec = false;
 
+  // max_playback_rate <= 0 will not trigger setting of maximum encoding
+  // bandwidth.
+  int max_playback_rate = 0;
+
   // Set send codec (the first non-telephone-event/CN codec)
   for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
        it != codecs.end(); ++it) {
@@ -2009,40 +2068,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
       continue;
     }
 
-    // If OPUS, change what we send according to the "stereo" codec
-    // parameter, and not the "channels" parameter.  We set
-    // voe_codec.channels to 2 if "stereo=1" and 1 otherwise.  If
-    // the bitrate is not specified, i.e. is zero, we set it to the
-    // appropriate default value for mono or stereo Opus.
-    if (IsOpus(*it)) {
-      if (IsOpusStereoEnabled(*it)) {
-        voe_codec.channels = 2;
-        if (!IsValidOpusBitrate(it->bitrate)) {
-          if (it->bitrate != 0) {
-            LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
-                            << it->bitrate
-                            << ") with default opus stereo bitrate: "
-                            << kOpusStereoBitrate;
-          }
-          voe_codec.rate = kOpusStereoBitrate;
-        }
-      } else {
-        voe_codec.channels = 1;
-        if (!IsValidOpusBitrate(it->bitrate)) {
-          if (it->bitrate != 0) {
-            LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
-                            << it->bitrate
-                            << ") with default opus mono bitrate: "
-                            << kOpusMonoBitrate;
-          }
-          voe_codec.rate = kOpusMonoBitrate;
-        }
-      }
-      int bitrate_from_params = GetOpusBitrateFromParams(*it);
-      if (bitrate_from_params != 0) {
-        voe_codec.rate = bitrate_from_params;
-      }
-    }
 
     // We'll use the first codec in the list to actually send audio data.
     // Be sure to use the payload type requested by the remote side.
@@ -2072,8 +2097,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
     } else {
       send_codec = voe_codec;
       nack_enabled = IsNackEnabled(*it);
-      // For Opus as the send codec, we enable inband FEC if requested.
-      enable_codec_fec = IsOpus(*it) && IsOpusFecEnabled(*it);
+      // For Opus as the send codec, we are to enable inband FEC if requested
+      // and set maximum playback rate.
+      if (IsOpus(*it)) {
+        GetOpusConfig(*it, &send_codec, &enable_codec_fec, &max_playback_rate);
+      }
     }
     found_send_codec = true;
     break;
@@ -2107,6 +2135,21 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
 #endif  // USE_WEBRTC_DEV_BRANCH
   }
 
+  // maxplaybackrate should be set after SetSendCodec.
+  if (max_playback_rate > 0) {
+    LOG(LS_INFO) << "Attempt to set maximum playback rate to "
+                 << max_playback_rate
+                 << " Hz on channel "
+                 << channel;
+#ifdef USE_WEBRTC_DEV_BRANCH
+    // (max_playback_rate + 1) >> 1 is to obtain ceil(max_playback_rate / 2.0).
+    if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
+        channel, max_playback_rate) == -1) {
+      LOG(LS_WARNING) << "Could not set maximum playback rate.";
+    }
+#endif
+  }
+
   // Always update the |send_codec_| to the currently set send codec.
   send_codec_.reset(new webrtc::CodecInst(send_codec));
 
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index f5b8260f0..b044e924c 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -1152,7 +1152,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFec) {
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
   codecs.push_back(kOpusCodec);
-  codecs[0].bitrate = 0;
   EXPECT_TRUE(channel_->SetSendCodecs(codecs));
   EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
 }
@@ -1229,6 +1228,159 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacWithParamNoFec) {
   EXPECT_TRUE(channel_->SetSendCodecs(codecs));
   EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
 }
+
+// Test that Opus FEC status can be changed.
+TEST_F(WebRtcVoiceEngineTestFake, ChangeOpusFecStatus) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
+  codecs[0].params["useinbandfec"] = "1";
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_TRUE(voe_.GetCodecFEC(channel_num));
+}
+
+// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateNb) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthNb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  // TODO(minyue): Default bit rate is not but can in future be affected by
+  // kCodecParamMaxPlaybackRate.
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateMb) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8001);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthMb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  // TODO(minyue): Default bit rate is not but can in future be affected by
+  // kCodecParamMaxPlaybackRate.
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateWb) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 12001);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthWb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  // TODO(minyue): Default bit rate is not but can in future be affected by
+  // kCodecParamMaxPlaybackRate.
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateSwb) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 16001);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthSwb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  // TODO(minyue): Default bit rate is not but can in future be affected by
+  // kCodecParamMaxPlaybackRate.
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test 24000 < maxplaybackrate triggers Opus full band mode.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateFb) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 24001);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthFb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  // TODO(minyue): Default bit rate is not but can in future be affected by
+  // kCodecParamMaxPlaybackRate.
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test Opus that without maxplaybackrate, default playback rate is used.
+TEST_F(WebRtcVoiceEngineTestFake, DefaultOpusMaxPlaybackRate) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthFb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+}
+
+// Test the with non-Opus, maxplaybackrate has no effect.
+TEST_F(WebRtcVoiceEngineTestFake, SetNonOpusMaxPlaybackRate) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kIsacCodec);
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 32000);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(0, voe_.GetMaxEncodingBandwidth(channel_num));
+}
+
+// Test maxplaybackrate can be set on two streams.
+TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  // Default bandwidth is 24000.
+  EXPECT_EQ(cricket::kOpusBandwidthFb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+
+  codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000);
+
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_EQ(cricket::kOpusBandwidthNb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+
+  channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc2));
+  channel_num = voe_.GetLastChannel();
+  EXPECT_EQ(cricket::kOpusBandwidthNb,
+            voe_.GetMaxEncodingBandwidth(channel_num));
+}
 #endif  // USE_WEBRTC_DEV_BRANCH
 
 // Test that we can apply CELT with stereo mode but fail with mono mode.