| 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;
|
|
|