Index: webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc |
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc |
index 734433836ff0a5e4a9dfb7476c72ba8e421b7061..4f92e0670d20382304492d81fb7515db098d9f4e 100644 |
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc |
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc |
@@ -28,7 +28,7 @@ class ExtensionVerifyTransport : public webrtc::Transport { |
audio_level_id_(-1), |
absolute_sender_time_id_(-1) {} |
- int SendPacket(int channel, const void* data, size_t len) override { |
+ int SendPacket(const void* data, size_t len) override { |
webrtc::RTPHeader header; |
if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) { |
bool ok = true; |
@@ -51,7 +51,7 @@ class ExtensionVerifyTransport : public webrtc::Transport { |
return static_cast<int>(len); |
} |
- int SendRTCPPacket(int channel, const void* data, size_t len) override { |
+ int SendRTCPPacket(const void* data, size_t len) override { |
return static_cast<int>(len); |
} |