| 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);
|
| }
|
|
|
|
|