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_; |