| 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;
|
|
|