Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
index 6b5a6c740a94d7c73d7e60358b972941fe593f32..f31dde9961cb162061bad4acb314955782efc186 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -211,7 +211,6 @@ class RTPSender { |
RtpState GetRtpState() const; |
void SetRtxRtpState(const RtpState& rtp_state); |
RtpState GetRtxRtpState() const; |
- bool ActivateCVORtpHeaderExtension(); |
protected: |
int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type); |
@@ -250,24 +249,9 @@ class RTPSender { |
int64_t capture_time_ms, |
uint32_t ssrc); |
- // Find the byte position of the RTP extension as indicated by |type| in |
- // |rtp_packet|. Return false if such extension doesn't exist. |
- bool FindHeaderExtensionPosition(RTPExtensionType type, |
- const uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const RTPHeader& rtp_header, |
- size_t* position) const |
- EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
- |
bool UpdateTransportSequenceNumber(RtpPacketToSend* packet, |
int* packet_id) const; |
- void UpdatePlayoutDelayLimits(uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const RTPHeader& rtp_header, |
- uint16_t min_playout_delay, |
- uint16_t max_playout_delay) const; |
- |
void UpdateRtpStats(const RtpPacketToSend& packet, |
bool is_rtx, |
bool is_retransmit); |
@@ -296,13 +280,11 @@ class RTPSender { |
std::map<int8_t, RtpUtility::Payload*> payload_type_map_; |
RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_); |
- bool video_rotation_active_; |
// Tracks the current request for playout delay limits from application |
// and decides whether the current RTP frame should include the playout |
// delay extension on header. |
PlayoutDelayOracle playout_delay_oracle_; |
- bool playout_delay_active_ GUARDED_BY(send_critsect_); |
RtpPacketHistory packet_history_; |