Index: webrtc/voice_engine/channel.h |
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
index 56f6ce4c057c10e15a8c9da39fcb5fbc2a67c65b..ac12b5bfc0da2b3499ce81ff1e35690e1211d7fe 100644 |
--- a/webrtc/voice_engine/channel.h |
+++ b/webrtc/voice_engine/channel.h |
@@ -364,24 +364,22 @@ public: |
size_t packet_length) override; |
// From RtpFeedback in the RTP/RTCP module |
- int32_t OnInitializeDecoder(int32_t id, |
- int8_t payloadType, |
+ int32_t OnInitializeDecoder(int8_t payloadType, |
const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
int frequency, |
uint8_t channels, |
uint32_t rate) override; |
- void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override; |
- void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override; |
+ void OnIncomingSSRCChanged(uint32_t ssrc) override; |
+ void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override; |
// From RtpAudioFeedback in the RTP/RTCP module |
- void OnPlayTelephoneEvent(int32_t id, |
- uint8_t event, |
+ void OnPlayTelephoneEvent(uint8_t event, |
uint16_t lengthMs, |
uint8_t volume) override; |
// From Transport (called by the RTP/RTCP module) |
- int SendPacket(int /*channel*/, const void* data, size_t len) override; |
- int SendRTCPPacket(int /*channel*/, const void* data, size_t len) override; |
+ int SendPacket(const void* data, size_t len) override; |
+ int SendRTCPPacket(const void* data, size_t len) override; |
// From MixerParticipant |
int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) override; |