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_; |