Index: webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h |
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h |
index 44fb0b2d76c423d60aac70aa896b90451d7e16ba..442f1966b0b890233c325aa68eb1c4ff94929c83 100644 |
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h |
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h |
@@ -48,7 +48,7 @@ class Sender { |
public: |
Sender(); |
void Setup(AudioCodingModule *acm, RTPStream *rtpStream, |
- std::string in_file_name, int sample_rate, int channels); |
+ std::string in_file_name, int sample_rate, size_t channels); |
void Teardown(); |
void Run(); |
bool Add10MsData(); |
@@ -71,7 +71,7 @@ class Receiver { |
Receiver(); |
virtual ~Receiver() {}; |
void Setup(AudioCodingModule *acm, RTPStream *rtpStream, |
- std::string out_file_name, int channels); |
+ std::string out_file_name, size_t channels); |
void Teardown(); |
void Run(); |
virtual bool IncomingPacket(); |