| Index: webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
 | 
| diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
 | 
| index 014227584619f6a60fee934c57d45a77629114da..12ea300fe5760634b4698736abf2b15921f8a186 100644
 | 
| --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
 | 
| +++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
 | 
| @@ -46,7 +46,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
 | 
|        : timestamp_(0),
 | 
|          packet_sent_(false),
 | 
|          last_packet_send_timestamp_(timestamp_),
 | 
| -        last_frame_type_(kFrameEmpty) {
 | 
| +        last_frame_type_(kEmptyFrame) {
 | 
|      AudioCodingModule::Config config;
 | 
|      acm_.reset(new AudioCodingModuleImpl(config));
 | 
|      receiver_.reset(new AcmReceiver(config));
 | 
| @@ -120,7 +120,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
 | 
|                 const uint8_t* payload_data,
 | 
|                 size_t payload_len_bytes,
 | 
|                 const RTPFragmentationHeader* fragmentation) override {
 | 
| -    if (frame_type == kFrameEmpty)
 | 
| +    if (frame_type == kEmptyFrame)
 | 
|        return 0;
 | 
|  
 | 
|      rtp_header_.header.payloadType = payload_type;
 | 
| 
 |