(Auto)update libjingle 69359922-> 69365993

git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@6463 4adac7df-926f-26a2-2b94-8c16560cd09d

4 files changed, 188 insertions, 64 deletions

diff --git a/media/webrtc/fakewebrtcvoiceengine.h b/media/webrtc/fakewebrtcvoiceengine.h
index 7285908..25c952d 100644
--- a/media/webrtc/fakewebrtcvoiceengine.h
+++ b/media/webrtc/fakewebrtcvoiceengine.h
@@ -97,7 +97,8 @@ class FakeWebRtcVoiceEngine
           volume_pan_right(1.0),
           file(false),
           vad(false),
-          fec(false),
+          codec_fec(false),
+          red(false),
           nack(false),
           media_processor_registered(false),
           rx_agc_enabled(false),
@@ -105,7 +106,7 @@ class FakeWebRtcVoiceEngine
           cn8_type(13),
           cn16_type(105),
           dtmf_type(106),
-          fec_type(117),
+          red_type(117),
           nack_max_packets(0),
           vie_network(NULL),
           video_channel(-1),
@@ -125,7 +126,8 @@ class FakeWebRtcVoiceEngine
     float volume_pan_right;
     bool file;
     bool vad;
-    bool fec;
+    bool codec_fec;
+    bool red;
     bool nack;
     bool media_processor_registered;
     bool rx_agc_enabled;
@@ -134,7 +136,7 @@ class FakeWebRtcVoiceEngine
     int cn8_type;
     int cn16_type;
     int dtmf_type;
-    int fec_type;
+    int red_type;
     int nack_max_packets;
     webrtc::ViENetwork* vie_network;
     int video_channel;
@@ -215,8 +217,11 @@ class FakeWebRtcVoiceEngine
   bool GetVAD(int channel) {
     return channels_[channel]->vad;
   }
-  bool GetFEC(int channel) {
-    return channels_[channel]->fec;
+  bool GetRED(int channel) {
+    return channels_[channel]->red;
+  }
+  bool GetCodecFEC(int channel) {
+    return channels_[channel]->codec_fec;
   }
   bool GetNACK(int channel) {
     return channels_[channel]->nack;
@@ -244,8 +249,8 @@ class FakeWebRtcVoiceEngine
   int GetSendTelephoneEventPayloadType(int channel) {
     return channels_[channel]->dtmf_type;
   }
-  int GetSendFECPayloadType(int channel) {
-    return channels_[channel]->fec_type;
+  int GetSendREDPayloadType(int channel) {
+    return channels_[channel]->red_type;
   }
   bool CheckPacket(int channel, const void* data, size_t len) {
     bool result = !CheckNoPacket(channel);
@@ -531,6 +536,18 @@ 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);
+    channels_[channel]->codec_fec = enable;
+    return 0;
+  }
+  WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) {
+    WEBRTC_CHECK_CHANNEL(channel);
+    enable = channels_[channel]->codec_fec;
+    return 0;
+  }
+#endif  // USE_WEBRTC_DEV_BRANCH
 
   // webrtc::VoEDtmf
   WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code,
@@ -843,16 +860,24 @@ class FakeWebRtcVoiceEngine
     stats.packetsReceived = kIntStatValue;
     return 0;
   }
+#ifdef USE_WEBRTC_DEV_BRANCH
+  WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
+#else
   WEBRTC_FUNC(SetFECStatus, (int channel, bool enable, int redPayloadtype)) {
+#endif  // USE_WEBRTC_DEV_BRANCH
     WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->fec = enable;
-    channels_[channel]->fec_type = redPayloadtype;
+    channels_[channel]->red = enable;
+    channels_[channel]->red_type = redPayloadtype;
     return 0;
   }
+#ifdef USE_WEBRTC_DEV_BRANCH
+  WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
+#else
   WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) {
+#endif  // USE_WEBRTC_DEV_BRANCH
     WEBRTC_CHECK_CHANNEL(channel);
-    enable = channels_[channel]->fec;
-    redPayloadtype = channels_[channel]->fec_type;
+    enable = channels_[channel]->red;
+    redPayloadtype = channels_[channel]->red_type;
     return 0;
   }
   WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) {
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index 41518e8..dc9f4ab 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -558,6 +558,7 @@ class WebRtcOveruseObserver : public webrtc::CpuOveruseObserver {
   }
 
   void Enable(bool enable) {
+    LOG(LS_INFO) << "WebRtcOveruseObserver enable: " << enable;
     talk_base::CritScope cs(&crit_);
     enabled_ = enable;
   }
@@ -586,8 +587,7 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
         external_capture_(external_capture),
         capturer_updated_(false),
         interval_(0),
-        cpu_monitor_(cpu_monitor),
-        overuse_observer_enabled_(false) {
+        cpu_monitor_(cpu_monitor) {
   }
 
   int channel_id() const { return channel_id_; }
@@ -679,7 +679,8 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
     vie_wrapper->base()->RegisterCpuOveruseObserver(channel_id_,
                                                     overuse_observer_.get());
     // (Dis)connect the video adapter from the cpu monitor as appropriate.
-    SetCpuOveruseDetection(overuse_observer_enabled_);
+    SetCpuOveruseDetection(
+        video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false));
 
     SignalCpuAdaptationUnable.repeat(adapter->SignalCpuAdaptationUnable);
   }
@@ -698,10 +699,18 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
   }
 
   void ApplyCpuOptions(const VideoOptions& video_options) {
+    bool cpu_overuse_detection_changed =
+        video_options.cpu_overuse_detection.IsSet() &&
+        (video_options.cpu_overuse_detection.GetWithDefaultIfUnset(false) !=
+         video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false));
     // Use video_options_.SetAll() instead of assignment so that unset value in
     // video_options will not overwrite the previous option value.
     video_options_.SetAll(video_options);
     UpdateAdapterCpuOptions();
+    if (cpu_overuse_detection_changed) {
+      SetCpuOveruseDetection(
+          video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false));
+    }
   }
 
   void UpdateAdapterCpuOptions() {
@@ -709,15 +718,19 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
       return;
     }
 
-    bool cpu_adapt, cpu_smoothing, adapt_third;
+    bool cpu_smoothing, adapt_third;
     float low, med, high;
+    bool cpu_adapt =
+        video_options_.adapt_input_to_cpu_usage.GetWithDefaultIfUnset(false);
+    bool cpu_overuse_detection =
+        video_options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
 
     // TODO(thorcarpenter): Have VideoAdapter be responsible for setting
     // all these video options.
     CoordinatedVideoAdapter* video_adapter = video_capturer_->video_adapter();
-    if (video_options_.adapt_input_to_cpu_usage.Get(&cpu_adapt) ||
-        overuse_observer_enabled_) {
-      video_adapter->set_cpu_adaptation(cpu_adapt || overuse_observer_enabled_);
+    if (video_options_.adapt_input_to_cpu_usage.IsSet() ||
+        video_options_.cpu_overuse_detection.IsSet()) {
+      video_adapter->set_cpu_adaptation(cpu_adapt || cpu_overuse_detection);
     }
     if (video_options_.adapt_cpu_with_smoothing.Get(&cpu_smoothing)) {
       video_adapter->set_cpu_smoothing(cpu_smoothing);
@@ -737,8 +750,6 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
   }
 
   void SetCpuOveruseDetection(bool enable) {
-    overuse_observer_enabled_ = enable;
-
     if (overuse_observer_) {
       overuse_observer_->Enable(enable);
     }
@@ -747,10 +758,6 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
     // it will be signaled by cpu monitor.
     CoordinatedVideoAdapter* adapter = video_adapter();
     if (adapter) {
-      bool cpu_adapt = false;
-      video_options_.adapt_input_to_cpu_usage.Get(&cpu_adapt);
-      adapter->set_cpu_adaptation(
-          adapter->cpu_adaptation() || cpu_adapt || enable);
       if (cpu_monitor_) {
         if (enable) {
           cpu_monitor_->SignalUpdate.disconnect(adapter);
@@ -815,7 +822,6 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
 
   talk_base::CpuMonitor* cpu_monitor_;
   talk_base::scoped_ptr<WebRtcOveruseObserver> overuse_observer_;
-  bool overuse_observer_enabled_;
 
   VideoOptions video_options_;
 };
@@ -2967,9 +2973,6 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
   bool buffer_latency_changed = options.buffered_mode_latency.IsSet() &&
       (options_.buffered_mode_latency != options.buffered_mode_latency);
 
-  bool cpu_overuse_detection_changed = options.cpu_overuse_detection.IsSet() &&
-      (options_.cpu_overuse_detection != options.cpu_overuse_detection);
-
   bool dscp_option_changed = (options_.dscp != options.dscp);
 
   bool suspend_below_min_bitrate_changed =
@@ -3081,17 +3084,6 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
       }
     }
   }
-  if (cpu_overuse_detection_changed) {
-    bool cpu_overuse_detection =
-        options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
-    LOG(LS_INFO) << "CPU overuse detection is enabled? "
-                 << cpu_overuse_detection;
-    for (SendChannelMap::iterator iter = send_channels_.begin();
-         iter != send_channels_.end(); ++iter) {
-      WebRtcVideoChannelSendInfo* send_channel = iter->second;
-      send_channel->SetCpuOveruseDetection(cpu_overuse_detection);
-    }
-  }
   if (dscp_option_changed) {
     talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT;
     if (options_.dscp.GetWithDefaultIfUnset(false))
@@ -3576,10 +3568,6 @@ bool WebRtcVideoMediaChannel::ConfigureSending(int channel_id,
   send_channel->SignalCpuAdaptationUnable.connect(this,
       &WebRtcVideoMediaChannel::OnCpuAdaptationUnable);
 
-  if (options_.cpu_overuse_detection.GetWithDefaultIfUnset(false)) {
-    send_channel->SetCpuOveruseDetection(true);
-  }
-
   webrtc::CpuOveruseOptions overuse_options;
   if (GetCpuOveruseOptions(options_, &overuse_options)) {
     if (engine()->vie()->base()->SetCpuOveruseOptions(channel_id,
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index d7b3c4c..785cdf1 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -426,6 +426,16 @@ static int GetOpusBitrateFromParams(const AudioCodec& codec) {
   return bitrate;
 }
 
+// True if params["useinbandfec"] == "1"
+static bool IsOpusFecEnabled(const AudioCodec& codec) {
+  CodecParameterMap::const_iterator param =
+      codec.params.find(kCodecParamUseInbandFec);
+  if (param == codec.params.end())
+    return false;
+
+  return param->second == kParamValueTrue;
+}
+
 void WebRtcVoiceEngine::ConstructCodecs() {
   LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
@@ -1943,10 +1953,16 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
 
 bool WebRtcVoiceMediaChannel::SetSendCodecs(
     int channel, const std::vector<AudioCodec>& codecs) {
-  // Disable VAD, and FEC unless we know the other side wants them.
+  // Disable VAD, FEC, and RED unless we know the other side wants them.
   engine()->voe()->codec()->SetVADStatus(channel, false);
   engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
+#ifdef USE_WEBRTC_DEV_BRANCH
+  engine()->voe()->rtp()->SetREDStatus(channel, false);
+  engine()->voe()->codec()->SetFECStatus(channel, false);
+#else
+  // TODO(minyue): Remove code under #else case after new WebRTC roll.
   engine()->voe()->rtp()->SetFECStatus(channel, false);
+#endif  // USE_WEBRTC_DEV_BRANCH
 
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD and DTMF.
@@ -2005,11 +2021,24 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
       if (bitrate_from_params != 0) {
         voe_codec.rate = bitrate_from_params;
       }
+
+      // If FEC is enabled.
+      if (IsOpusFecEnabled(*it)) {
+        LOG(LS_INFO) << "Enabling Opus FEC on channel " << channel;
+#ifdef USE_WEBRTC_DEV_BRANCH
+        if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
+          // Enable in-band FEC of the Opus codec. Treat any failure as a fatal
+          // internal error.
+          LOG_RTCERR2(SetFECStatus, channel, true);
+          return false;
+        }
+#endif  // USE_WEBRTC_DEV_BRANCH
+      }
     }
 
     // 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.
-    // "red", for FEC audio, is a special case where the actual codec to be
+    // "red", for RED audio, is a special case where the actual codec to be
     // used is specified in params.
     if (IsRedCodec(it->name)) {
       // Parse out the RED parameters. If we fail, just ignore RED;
@@ -2020,9 +2049,16 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
 
       // Enable redundant encoding of the specified codec. Treat any
       // failure as a fatal internal error.
+#ifdef USE_WEBRTC_DEV_BRANCH
+      LOG(LS_INFO) << "Enabling RED on channel " << channel;
+      if (engine()->voe()->rtp()->SetREDStatus(channel, true, it->id) == -1) {
+        LOG_RTCERR3(SetREDStatus, channel, true, it->id);
+#else
+      // TODO(minyue): Remove code under #else case after new WebRTC roll.
       LOG(LS_INFO) << "Enabling FEC";
       if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) {
         LOG_RTCERR3(SetFECStatus, channel, true, it->id);
+#endif  // USE_WEBRTC_DEV_BRANCH
         return false;
       }
     } else {
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 5dab4ff..80a50c5 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -745,7 +745,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
   EXPECT_EQ(48000, gcodec.rate);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_FALSE(voe_.GetVAD(channel_num));
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(105, voe_.GetSendCNPayloadType(channel_num, true));
   EXPECT_EQ(106, voe_.GetSendTelephoneEventPayloadType(channel_num));
@@ -1144,6 +1144,81 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
   EXPECT_TRUE(voe_.GetNACK(channel_num));
 }
 
+#ifdef USE_WEBRTC_DEV_BRANCH
+// Test that without useinbandfec, Opus FEC is off.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFEC) {
+  EXPECT_TRUE(SetupEngine());
+  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));
+}
+
+// Test that with useinbandfec=0, Opus FEC is off.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusDisableFEC) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].params["useinbandfec"] = "0";
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  EXPECT_EQ(1, gcodec.channels);
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test that with useinbandfec=1, Opus FEC is on.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFEC) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].params["useinbandfec"] = "1";
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_TRUE(voe_.GetCodecFEC(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  EXPECT_EQ(1, gcodec.channels);
+  EXPECT_EQ(32000, gcodec.rate);
+}
+
+// Test that with useinbandfec=1, stereo=1, Opus FEC is on.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFECStereo) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].bitrate = 0;
+  codecs[0].params["stereo"] = "1";
+  codecs[0].params["useinbandfec"] = "1";
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_TRUE(voe_.GetCodecFEC(channel_num));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_STREQ("opus", gcodec.plname);
+  EXPECT_EQ(2, gcodec.channels);
+  EXPECT_EQ(64000, gcodec.rate);
+}
+
+// Test that with non-Opus, codec FEC is off.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacNoFEC) {
+  EXPECT_TRUE(SetupEngine());
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kIsacCodec);
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_FALSE(voe_.GetCodecFEC(channel_num));
+}
+#endif  // USE_WEBRTC_DEV_BRANCH
+
 // Test that we can apply CELT with stereo mode but fail with mono mode.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCelt) {
   EXPECT_TRUE(SetupEngine());
@@ -1315,7 +1390,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(voe_.GetVAD(channel_num));
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
   EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
@@ -1348,7 +1423,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(voe_.GetVAD(channel_num));
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
   EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
@@ -1412,13 +1487,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
   EXPECT_TRUE(voe_.GetVAD(channel_num));
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
   EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
   EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
   EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
 }
 
-// Test that we set up FEC correctly as caller.
+// Test that we set up RED correctly as caller.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCaller) {
   EXPECT_TRUE(SetupEngine());
   int channel_num = voe_.GetLastChannel();
@@ -1434,11 +1509,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCaller) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_TRUE(voe_.GetFEC(channel_num));
-  EXPECT_EQ(127, voe_.GetSendFECPayloadType(channel_num));
+  EXPECT_TRUE(voe_.GetRED(channel_num));
+  EXPECT_EQ(127, voe_.GetSendREDPayloadType(channel_num));
 }
 
-// Test that we set up FEC correctly as callee.
+// Test that we set up RED correctly as callee.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCallee) {
   EXPECT_TRUE(engine_.Init(talk_base::Thread::Current()));
   channel_ = engine_.CreateChannel();
@@ -1459,11 +1534,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCallee) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_TRUE(voe_.GetFEC(channel_num));
-  EXPECT_EQ(127, voe_.GetSendFECPayloadType(channel_num));
+  EXPECT_TRUE(voe_.GetRED(channel_num));
+  EXPECT_EQ(127, voe_.GetSendREDPayloadType(channel_num));
 }
 
-// Test that we set up FEC correctly if params are omitted.
+// Test that we set up RED correctly if params are omitted.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDNoParams) {
   EXPECT_TRUE(SetupEngine());
   int channel_num = voe_.GetLastChannel();
@@ -1478,8 +1553,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDNoParams) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_TRUE(voe_.GetFEC(channel_num));
-  EXPECT_EQ(127, voe_.GetSendFECPayloadType(channel_num));
+  EXPECT_TRUE(voe_.GetRED(channel_num));
+  EXPECT_EQ(127, voe_.GetSendREDPayloadType(channel_num));
 }
 
 // Test that we ignore RED if the parameters aren't named the way we expect.
@@ -1498,7 +1573,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED1) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
 }
 
 // Test that we ignore RED if it uses different primary/secondary encoding.
@@ -1517,7 +1592,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED2) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
 }
 
 // Test that we ignore RED if it uses more than 2 encodings.
@@ -1536,7 +1611,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED3) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
 }
 
 // Test that we ignore RED if it has bogus codec ids.
@@ -1555,7 +1630,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED4) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
 }
 
 // Test that we ignore RED if it refers to a codec that is not present.
@@ -1574,7 +1649,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBadRED5) {
   EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
   EXPECT_EQ(96, gcodec.pltype);
   EXPECT_STREQ("ISAC", gcodec.plname);
-  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_FALSE(voe_.GetRED(channel_num));
 }
 
 // Test support for audio level header extension.