| Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
|
| index a7efcbba344e19bfbc86a307ec6493159597a804..176852e01ef597ea620c4b9cf550653976075a24 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
|
| @@ -28,8 +28,7 @@ class CriticalSectionWrapper;
|
| class RTPReceiverAudio : public RTPReceiverStrategy,
|
| public TelephoneEventHandler {
|
| public:
|
| - RTPReceiverAudio(const int32_t id,
|
| - RtpData* data_callback,
|
| + RTPReceiverAudio(RtpData* data_callback,
|
| RtpAudioFeedback* incoming_messages_callback);
|
| virtual ~RTPReceiverAudio() {}
|
|
|
| @@ -74,7 +73,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
|
|
|
| int32_t InvokeOnInitializeDecoder(
|
| RtpFeedback* callback,
|
| - int32_t id,
|
| int8_t payload_type,
|
| const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
| const PayloadUnion& specific_payload) const override;
|
| @@ -106,8 +104,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
|
| const AudioPayload& audio_specific,
|
| bool is_red);
|
|
|
| - int32_t id_;
|
| -
|
| uint32_t last_received_frequency_;
|
|
|
| bool telephone_event_forward_to_decoder_;
|
|
|