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; |