| 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 15d0bdeae671c4168c26f0c9ac9aa8c7c825e19c..5fbf738b4b7de4085de16e531a2e75a25ff847b1 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
|
| @@ -64,16 +64,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
|
| const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
| const PayloadUnion& specific_payload) const override;
|
|
|
| - // We do not allow codecs to have multiple payload types for audio, so we
|
| - // need to override the default behavior (which is to do nothing).
|
| - void PossiblyRemoveExistingPayloadType(
|
| - RtpUtility::PayloadTypeMap* payload_type_map,
|
| - const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
| - size_t payload_name_length,
|
| - uint32_t frequency,
|
| - uint8_t channels,
|
| - uint32_t rate) const;
|
| -
|
| // We need to look out for special payload types here and sometimes reset
|
| // statistics. In addition we sometimes need to tweak the frequency.
|
| void CheckPayloadChanged(int8_t payload_type,
|
| @@ -89,8 +79,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
|
| const AudioPayload& audio_specific,
|
| bool is_red);
|
|
|
| - uint32_t last_received_frequency_;
|
| -
|
| bool telephone_event_forward_to_decoder_;
|
| int8_t telephone_event_payload_type_;
|
| std::set<uint8_t> telephone_event_reported_;
|
|
|